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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-rtp-usage-08.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Real-Time Communication in WEB-browsers W=
orking Group of the IETF.

	Title           : Web Real-Time Communication (WebRTC): Media Transport an=
d Use of RTP
	Author(s)       : Colin Perkins
                          Magnus Westerlund
                          Joerg Ott
	Filename        : draft-ietf-rtcweb-rtp-usage-08.txt
	Pages           : 40
	Date            : 2013-09-01

Abstract:
   The Web Real-Time Communication (WebRTC) framework provides support
   for direct interactive rich communication using audio, video, text,
   collaboration, games, etc.  between two peers' web-browsers.  This
   memo describes the media transport aspects of the WebRTC framework.
   It specifies how the Real-time Transport Protocol (RTP) is used in
   the WebRTC context, and gives requirements for which RTP features,
   profiles, and extensions need to be supported.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-rtp-usage

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-08

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-rtp-usage-08


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


From csp@csperkins.org  Sun Sep  1 10:15:41 2013
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From: Colin Perkins <csp@csperkins.org>
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The main changes in this version are to reorganise and rewrite Section =
12, and remove most of the Appendix, with the exception of sections A.5 =
and A.6, which have been merged into Section 12. We've also tried to =
address the comments made on the list recently.=20

Colin


On 1 Sep 2013, at 18:02, Internet-Drafts@ietf.org wrote:
> A New Internet-Draft is available from the on-line Internet-Drafts =
directories.
> This draft is a work item of the Real-Time Communication in =
WEB-browsers Working Group of the IETF.
>=20
> 	Title           : Web Real-Time Communication (WebRTC): Media =
Transport and Use of RTP
> 	Author(s)       : Colin Perkins
>                          Magnus Westerlund
>                          Joerg Ott
> 	Filename        : draft-ietf-rtcweb-rtp-usage-08.txt
> 	Pages           : 40
> 	Date            : 2013-09-01
>=20
> Abstract:
>   The Web Real-Time Communication (WebRTC) framework provides support
>   for direct interactive rich communication using audio, video, text,
>   collaboration, games, etc.  between two peers' web-browsers.  This
>   memo describes the media transport aspects of the WebRTC framework.
>   It specifies how the Real-time Transport Protocol (RTP) is used in
>   the WebRTC context, and gives requirements for which RTP features,
>   profiles, and extensions need to be supported.
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-rtp-usage
>=20
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-08
>=20
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-rtp-usage-08

...

--=20
Colin Perkins
http://csperkins.org/




From cowwoc@bbs.darktech.org  Sun Sep  1 12:44:39 2013
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Date: Sun, 01 Sep 2013 15:44:31 -0400
From: cowwoc <cowwoc@bbs.darktech.org>
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Subject: [rtcweb] Virnetx IPR
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Hi,

     Is WebRTC affected by this?

http://apple.slashdot.org/story/13/09/01/1233230/apple-now-relaying-all-facetime-calls-due-to-lost-patent-dispute 
and
http://arstechnica.com/tech-policy/2010/08/virnetx-files-vpn-patent-suit-against-apple-cisco-nec/

Thanks,
Gili

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From: Silvia Pfeiffer <silviapfeiffer1@gmail.com>
Date: Mon, 2 Sep 2013 08:38:59 +1000
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Subject: Re: [rtcweb] Virnetx IPR
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FAIK it's unrelated: WebRTC is not using a vpn to establish
connections. But IANAL. :-)
Silvia.

On Mon, Sep 2, 2013 at 5:44 AM, cowwoc <cowwoc@bbs.darktech.org> wrote:
> Hi,
>
>     Is WebRTC affected by this?
>
> http://apple.slashdot.org/story/13/09/01/1233230/apple-now-relaying-all-facetime-calls-due-to-lost-patent-dispute
> and
> http://arstechnica.com/tech-policy/2010/08/virnetx-files-vpn-patent-suit-against-apple-cisco-nec/
>
> Thanks,
> Gili
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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Silvia,

     Something doesn't make sense. It's my understanding that Facetime 
involves direct video chat connections (no VPN) and yet Apple was found 
to be infringing Virnetx patents.

     Quoting 
http://arstechnica.com/tech-policy/2013/08/report-after-patent-loss-apple-tweaks-facetime-and-logs-500000-complaints/:

Both sides in the litigation admit that if Apple routes its FaceTime 
calls through relay servers, it will avoid infringing the VirnetX 
patents. Once Apple was found to be infringing---and realized it could 
end up paying an ongoing royalty for using FaceTime---*the company 
redesigned the system so that all FaceTime calls would rely on relay 
servers. Lease believes the switch happened in April*.

     This makes me believe the patent affect direct p2p connections.

Gili

On 01/09/2013 6:38 PM, Silvia Pfeiffer wrote:
> FAIK it's unrelated: WebRTC is not using a vpn to establish
> connections. But IANAL. :-)
> Silvia.
>
> On Mon, Sep 2, 2013 at 5:44 AM, cowwoc <cowwoc@bbs.darktech.org> wrote:
>> Hi,
>>
>>      Is WebRTC affected by this?
>>
>> http://apple.slashdot.org/story/13/09/01/1233230/apple-now-relaying-all-facetime-calls-due-to-lost-patent-dispute
>> and
>> http://arstechnica.com/tech-policy/2010/08/virnetx-files-vpn-patent-suit-against-apple-cisco-nec/
>>
>> Thanks,
>> Gili
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb


--------------090102090504040902040108
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: 7bit

<html>
  <head>
    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
  </head>
  <body text="#000000" bgcolor="#FFFFFF">
    <div class="moz-cite-prefix">Silvia,<br>
      <br>
      &nbsp;&nbsp;&nbsp; Something doesn't make sense. It's my understanding that
      Facetime involves direct video chat connections (no VPN) and yet
      Apple was found to be infringing Virnetx patents.<br>
      <br>
      &nbsp;&nbsp;&nbsp; Quoting <a
href="http://arstechnica.com/tech-policy/2013/08/report-after-patent-loss-apple-tweaks-facetime-and-logs-500000-complaints/">http://arstechnica.com/tech-policy/2013/08/report-after-patent-loss-apple-tweaks-facetime-and-logs-500000-complaints/</a>:<br>
      <br>
      <span style="color: rgb(0, 0, 0); font-family: Arial, sans-serif;
        font-size: 13px; font-style: normal; font-variant: normal;
        font-weight: normal; letter-spacing: normal; line-height: 19px;
        orphans: auto; text-align: left; text-indent: 0px;
        text-transform: none; white-space: normal; widows: auto;
        word-spacing: 0px; -webkit-text-stroke-width: 0px;
        background-color: rgb(255, 255, 255); display: inline
        !important; float: none;">Both sides in the litigation admit
        that if Apple routes its FaceTime calls through relay servers,
        it will avoid infringing the VirnetX patents. Once Apple was
        found to be infringing&#8212;and realized it could end up paying an
        ongoing royalty for using FaceTime&#8212;</span><b style="outline:
        none; vertical-align: baseline; font-family: Arial, sans-serif;
        font-style: normal; font-size: 13px; padding: 0px; margin: 0px;
        color: rgb(0, 0, 0); font-variant: normal; letter-spacing:
        normal; line-height: 19px; orphans: auto; text-align: left;
        text-indent: 0px; text-transform: none; white-space: normal;
        widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;
        background-color: rgb(255, 255, 255);">the company redesigned
        the system so that all FaceTime calls would rely on relay
        servers. Lease believes the switch happened in April</b><span
        style="color: rgb(0, 0, 0); font-family: Arial, sans-serif;
        font-size: 13px; font-style: normal; font-variant: normal;
        font-weight: normal; letter-spacing: normal; line-height: 19px;
        orphans: auto; text-align: left; text-indent: 0px;
        text-transform: none; white-space: normal; widows: auto;
        word-spacing: 0px; -webkit-text-stroke-width: 0px;
        background-color: rgb(255, 255, 255); display: inline
        !important; float: none;">.<br>
        <br>
        &nbsp;&nbsp;&nbsp; This makes me believe the patent affect direct p2p
        connections.<br>
      </span><br>
      Gili<br>
      <br>
      On 01/09/2013 6:38 PM, Silvia Pfeiffer wrote:<br>
    </div>
    <blockquote
cite="mid:CAHp8n2=ywG8j1rKNAbY-VLbeSk_G6vL999s55-Nf--7s1cybCA@mail.gmail.com"
      type="cite">
      <pre wrap="">FAIK it's unrelated: WebRTC is not using a vpn to establish
connections. But IANAL. :-)
Silvia.

On Mon, Sep 2, 2013 at 5:44 AM, cowwoc <a class="moz-txt-link-rfc2396E" href="mailto:cowwoc@bbs.darktech.org">&lt;cowwoc@bbs.darktech.org&gt;</a> wrote:
</pre>
      <blockquote type="cite">
        <pre wrap="">Hi,

    Is WebRTC affected by this?

<a class="moz-txt-link-freetext" href="http://apple.slashdot.org/story/13/09/01/1233230/apple-now-relaying-all-facetime-calls-due-to-lost-patent-dispute">http://apple.slashdot.org/story/13/09/01/1233230/apple-now-relaying-all-facetime-calls-due-to-lost-patent-dispute</a>
and
<a class="moz-txt-link-freetext" href="http://arstechnica.com/tech-policy/2010/08/virnetx-files-vpn-patent-suit-against-apple-cisco-nec/">http://arstechnica.com/tech-policy/2010/08/virnetx-files-vpn-patent-suit-against-apple-cisco-nec/</a>

Thanks,
Gili
_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
      </blockquote>
    </blockquote>
    <br>
  </body>
</html>

--------------090102090504040902040108--

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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Real-Time Communication in WEB-browsers W=
orking Group of the IETF.

	Title           : Transports for RTCWEB
	Author(s)       : Harald Alvestrand
	Filename        : draft-ietf-rtcweb-transports-01.txt
	Pages           : 8
	Date            : 2013-09-03

Abstract:
   This document describes the data transport protocols used by RTCWEB,
   including the protocols used for interaction with intermediate boxes
   such as firewalls, relays and NAT boxes.



The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-transports

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-transports-01

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-transports-01


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

Internet-Drafts are also available by anonymous FTP at:
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Since I had so much good feedback on the previous version, I've created 
an update.

There's still a couple of open questions, in particular around support 
of TURN/TLS (TLS for connecting from the client to the TURN server).

I have also said nothing about HTTP proxies - after reading through that 
text in -firewall-, I concluded that we need a protocol specification to 
refer to before we can require clients to connect through HTTP proxies.

The protocol specification MIGHT fit on one page - but then again, it 
might not.


-------- Original Message --------
Subject: 	New Version Notification for draft-ietf-rtcweb-transports-01.txt
Date: 	Tue, 03 Sep 2013 02:40:45 -0700
From: 	internet-drafts@ietf.org
To: 	Harald T. Alvestrand <harald@alvestrand.no>, Harald Alvestrand 
<harald@alvestrand.no>



A new version of I-D, draft-ietf-rtcweb-transports-01.txt
has been successfully submitted by Harald Alvestrand and posted to the
IETF repository.

Filename:	 draft-ietf-rtcweb-transports
Revision:	 01
Title:		 Transports for RTCWEB
Creation date:	 2013-09-03
Group:		 rtcweb
Number of pages: 8
URL:             http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-transports-01.txt
Status:          http://datatracker.ietf.org/doc/draft-ietf-rtcweb-transports
Htmlized:        http://tools.ietf.org/html/draft-ietf-rtcweb-transports-01
Diff:            http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-transports-01

Abstract:
    This document describes the data transport protocols used by RTCWEB,
    including the protocols used for interaction with intermediate boxes
    such as firewalls, relays and NAT boxes.


                                                                                   


Please note that it may take a couple of minutes from the time of submission
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The IETF Secretariat




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<html>
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    Since I had so much good feedback on the previous version, I've
    created an update.<br>
    <br>
    There's still a couple of open questions, in particular around
    support of TURN/TLS (TLS for connecting from the client to the TURN
    server).<br>
    <br>
    I have also said nothing about HTTP proxies - after reading through
    that text in -firewall-, I concluded that we need a protocol
    specification to refer to before we can require clients to connect
    through HTTP proxies.<br>
    <br>
    The protocol specification MIGHT fit on one page - but then again,
    it might not.<br>
    <div class="moz-forward-container"><br>
      <br>
      -------- Original Message --------
      <table class="moz-email-headers-table" border="0" cellpadding="0"
        cellspacing="0">
        <tbody>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject:
            </th>
            <td>New Version Notification for
              draft-ietf-rtcweb-transports-01.txt</td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date: </th>
            <td>Tue, 03 Sep 2013 02:40:45 -0700</td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From: </th>
            <td><a class="moz-txt-link-abbreviated" href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a></td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To: </th>
            <td>Harald T. Alvestrand <a class="moz-txt-link-rfc2396E" href="mailto:harald@alvestrand.no">&lt;harald@alvestrand.no&gt;</a>,
              Harald Alvestrand <a class="moz-txt-link-rfc2396E" href="mailto:harald@alvestrand.no">&lt;harald@alvestrand.no&gt;</a></td>
          </tr>
        </tbody>
      </table>
      <br>
      <br>
      <pre>A new version of I-D, draft-ietf-rtcweb-transports-01.txt
has been successfully submitted by Harald Alvestrand and posted to the
IETF repository.

Filename:	 draft-ietf-rtcweb-transports
Revision:	 01
Title:		 Transports for RTCWEB
Creation date:	 2013-09-03
Group:		 rtcweb
Number of pages: 8
URL:             <a class="moz-txt-link-freetext" href="http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-transports-01.txt">http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-transports-01.txt</a>
Status:          <a class="moz-txt-link-freetext" href="http://datatracker.ietf.org/doc/draft-ietf-rtcweb-transports">http://datatracker.ietf.org/doc/draft-ietf-rtcweb-transports</a>
Htmlized:        <a class="moz-txt-link-freetext" href="http://tools.ietf.org/html/draft-ietf-rtcweb-transports-01">http://tools.ietf.org/html/draft-ietf-rtcweb-transports-01</a>
Diff:            <a class="moz-txt-link-freetext" href="http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-transports-01">http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-transports-01</a>

Abstract:
   This document describes the data transport protocols used by RTCWEB,
   including the protocols used for interaction with intermediate boxes
   such as firewalls, relays and NAT boxes.


                                                                                  


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</pre>
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Subject: Re: [rtcweb] References to ICE in -overview
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Sorry for not replying to this earlier; I'm preparing the new version of 
-overview now.

On 08/15/2013 07:04 PM, Martin Thomson wrote:
> I just noticed that the only concrete RFC 5245 reference in -overview
> is in the appendicies.  The term "ICE agent" is defined by not used in
> the main body of the text.  ICE is never expanded on first use.
I expanded the reference in the terminology section. The main purpose of 
that definition was to get people conscious that when they use the term 
"Agent", they need to qualify what kind of agent it is.

I also deleted the ICE reference in section 4 (Data transport), since 
the -transports- draft goes into much more detail about ICE than the 
-overview- draft should.
>
> Should there be a more explicit reference to ICE in Section 7?

I don't think section 7 is the right place for it; section 7 is where we 
say "we don't want to do SIP, but we require the need to do what SIP is 
able to do". (Unless my numbering's gone haywire).


> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-overview-08.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Real-Time Communication in WEB-browsers W=
orking Group of the IETF.

	Title           : Overview: Real Time Protocols for Brower-based Applicati=
ons
	Author(s)       : Harald T. Alvestrand
	Filename        : draft-ietf-rtcweb-overview-08.txt
	Pages           : 20
	Date            : 2013-09-03

Abstract:
   This document gives an overview and context of a protocol suite
   intended for use with real-time applications that can be deployed in
   browsers - "real time communication on the Web".

   It intends to serve as a starting and coordination point to make sure
   all the parts that are needed to achieve this goal are findable, and
   that the parts that belong in the Internet protocol suite are fully
   specified and on the right publication track.

   This document is a work item of the RTCWEB working group.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-overview

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-overview-08

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-overview-08


Please note that it may take a couple of minutes from the time of submission
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Subject: Re: [rtcweb] References to ICE in -overview
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On 3 September 2013 05:21, Harald Alvestrand <harald@alvestrand.no> wrote:
> I also deleted the ICE reference in section 4 (Data transport), since the
> -transports- draft goes into much more detail about ICE than the -overview-
> draft should.

Makes sense.  I made the comment before -transports- draft was published.

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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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We would like to start a working group last call of draft-ietf-rtcweb-use-c=
ases-and-requirements-11.

Please send comments by the end of the day on September 21.=20

Thank you,=20

The chairs =85.





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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: Bo Burman <bo.burman@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: I-D Action: draft-ietf-rtcweb-audio-02.txt
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
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I support the proposed text addition. But rather than adding at the end of =
section 3, replace the beginning of section 3 as below.

OLD:
   To ensure a baseline level of interoperability between WebRTC
   clients, a minimum set of required codecs are specified below.
   While this section specifies the codecs that will be mandated for all
   WebRTC client implementations, it leaves the question of supporting
   additional codecs to the will of the implementer.

NEW:
   To ensure a baseline level of interoperability between WebRTC
   clients, a minimum set of required codecs are specified below.
   If other suitable audio codecs are available to the browser to use,=20
   it is RECOMMENDED that they are also included in the offer in order=20
   to maximize the possibility to establish the session without the need=20
   for audio transcoding.

Mo

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Bo Burman
Sent: Tuesday, August 27, 2013 10:53 AM
To: rtcweb@ietf.org
Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt

Based on the previous discussion thread http://www.ietf.org/mail-archive/we=
b/rtcweb/current/msg08501.html, I believe there is support to add some text=
 into this document.

I thus propose to add the following text at the end of section 3:

If other suitable audio codecs are available to the browser to use,=20
it is RECOMMENDED that they are also included in the offer in order=20
to maximize the possibility to establish the session without the need=20
for audio transcoding

Cheers,
Bo

> -----Original Message-----
> From: i-d-announce-bounces@ietf.org [mailto:i-d-announce-bounces@ietf.org=
] On Behalf Of internet-drafts@ietf.org
> Sent: den 2 augusti 2013 18:30
> To: i-d-announce@ietf.org
> Cc: rtcweb@ietf.org
> Subject: I-D Action: draft-ietf-rtcweb-audio-02.txt
>=20
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts direct=
ories.
>  This draft is a work item of the Real-Time Communication in WEB-browsers=
 Working Group of the IETF.
>=20
> 	Title           : WebRTC Audio Codec and Processing Requirements
> 	Author(s)       : Jean-Marc Valin
>                           Cary Bran
> 	Filename        : draft-ietf-rtcweb-audio-02.txt
> 	Pages           : 6
> 	Date            : 2013-08-02
>=20
> Abstract:
>    This document outlines the audio codec and processing requirements
>    for WebRTC client application and endpoint devices.
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-audio
>=20
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-rtcweb-audio-02
>=20
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-audio-02
>=20
>=20
> Please note that it may take a couple of minutes from the time of submiss=
ion until the htmlized version and diff are
> available at tools.ietf.org.
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> _______________________________________________
> I-D-Announce mailing list
> I-D-Announce@ietf.org
> https://www.ietf.org/mailman/listinfo/i-d-announce
> Internet-Draft directories: http://www.ietf.org/shadow.html or ftp://ftp.=
ietf.org/ietf/1shadow-sites.txt
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

From stefan.lk.hakansson@ericsson.com  Thu Sep  5 01:45:51 2013
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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On 2013-09-04 18:23, Cullen Jennings (fluffy) wrote:=0A=
>=0A=
> We would like to start a working group last call of draft-ietf-rtcweb-use=
-cases-and-requirements-11.=0A=
>=0A=
> Please send comments by the end of the day on September 21.=0A=
=0A=
In the current version (-11) we've not addressed Andrew Hutton's input =0A=
on handling the case with a FW that only allows traffic via a HTTP Proxy =
=0A=
[1].=0A=
=0A=
This should be fixed for the next version.=0A=
=0A=
[11] http://www.ietf.org/mail-archive/web/rtcweb/current/msg08264.html=0A=
=0A=
>=0A=
> Thank you,=0A=
>=0A=
> The chairs =85.=0A=
>=0A=
>=0A=
>=0A=
>=0A=
>=0A=
=0A=

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From: "Rauschenbach, Uwe (NSN - DE/Munich)" <uwe.rauschenbach@nsn.com>
To: "ext Mo Zanaty (mzanaty)" <mzanaty@cisco.com>, Bo Burman <bo.burman@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: I-D Action: draft-ietf-rtcweb-audio-02.txt
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
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I support this proposal.

Kind regards,
Uwe=20


> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of ext Mo Zanaty (mzanaty)
> Sent: Wednesday, September 04, 2013 10:03 PM
> To: Bo Burman; rtcweb@ietf.org
> Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
>=20
> I support the proposed text addition. But rather than adding at the end
> of section 3, replace the beginning of section 3 as below.
>=20
> OLD:
>    To ensure a baseline level of interoperability between WebRTC
>    clients, a minimum set of required codecs are specified below.
>    While this section specifies the codecs that will be mandated for
> all
>    WebRTC client implementations, it leaves the question of supporting
>    additional codecs to the will of the implementer.
>=20
> NEW:
>    To ensure a baseline level of interoperability between WebRTC
>    clients, a minimum set of required codecs are specified below.
>    If other suitable audio codecs are available to the browser to use,
>    it is RECOMMENDED that they are also included in the offer in order
>    to maximize the possibility to establish the session without the
> need
>    for audio transcoding.
>=20
> Mo
>=20
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Bo Burman
> Sent: Tuesday, August 27, 2013 10:53 AM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
>=20
> Based on the previous discussion thread http://www.ietf.org/mail-
> archive/web/rtcweb/current/msg08501.html, I believe there is support to
> add some text into this document.
>=20
> I thus propose to add the following text at the end of section 3:
>=20
> If other suitable audio codecs are available to the browser to use,
> it is RECOMMENDED that they are also included in the offer in order
> to maximize the possibility to establish the session without the need
> for audio transcoding
>=20
> Cheers,
> Bo
>=20
> > -----Original Message-----
> > From: i-d-announce-bounces@ietf.org [mailto:i-d-announce-
> bounces@ietf.org] On Behalf Of internet-drafts@ietf.org
> > Sent: den 2 augusti 2013 18:30
> > To: i-d-announce@ietf.org
> > Cc: rtcweb@ietf.org
> > Subject: I-D Action: draft-ietf-rtcweb-audio-02.txt
> >
> >
> > A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> >  This draft is a work item of the Real-Time Communication in WEB-
> browsers Working Group of the IETF.
> >
> > 	Title           : WebRTC Audio Codec and Processing Requirements
> > 	Author(s)       : Jean-Marc Valin
> >                           Cary Bran
> > 	Filename        : draft-ietf-rtcweb-audio-02.txt
> > 	Pages           : 6
> > 	Date            : 2013-08-02
> >
> > Abstract:
> >    This document outlines the audio codec and processing requirements
> >    for WebRTC client application and endpoint devices.
> >
> >
> > The IETF datatracker status page for this draft is:
> > https://datatracker.ietf.org/doc/draft-ietf-rtcweb-audio
> >
> > There's also a htmlized version available at:
> > http://tools.ietf.org/html/draft-ietf-rtcweb-audio-02
> >
> > A diff from the previous version is available at:
> > http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-audio-02
> >
> >
> > Please note that it may take a couple of minutes from the time of
> submission until the htmlized version and diff are
> > available at tools.ietf.org.
> >
> > Internet-Drafts are also available by anonymous FTP at:
> > ftp://ftp.ietf.org/internet-drafts/
> >
> > _______________________________________________
> > I-D-Announce mailing list
> > I-D-Announce@ietf.org
> > https://www.ietf.org/mailman/listinfo/i-d-announce
> > Internet-Draft directories: http://www.ietf.org/shadow.html or
> ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From internet-drafts@ietf.org  Thu Sep  5 03:16:45 2013
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Date: Thu, 05 Sep 2013 03:16:09 -0700
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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-rtp-usage-09.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Real-Time Communication in WEB-browsers W=
orking Group of the IETF.

	Title           : Web Real-Time Communication (WebRTC): Media Transport an=
d Use of RTP
	Author(s)       : Colin Perkins
                          Magnus Westerlund
                          Joerg Ott
	Filename        : draft-ietf-rtcweb-rtp-usage-09.txt
	Pages           : 40
	Date            : 2013-09-05

Abstract:
   The Web Real-Time Communication (WebRTC) framework provides support
   for direct interactive rich communication using audio, video, text,
   collaboration, games, etc.  between two peers' web-browsers.  This
   memo describes the media transport aspects of the WebRTC framework.
   It specifies how the Real-time Transport Protocol (RTP) is used in
   the WebRTC context, and gives requirements for which RTP features,
   profiles, and extensions need to be supported.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-rtp-usage

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-09

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-rtp-usage-09


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


From csp@csperkins.org  Thu Sep  5 03:18:19 2013
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From: Colin Perkins <csp@csperkins.org>
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-rtp-usage-09.txt
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This just updates the references, since a few drafts have been published =
as RFCs recently. No other content changes.

Colin



On 5 Sep 2013, at 11:16, Internet-Drafts@ietf.org wrote:
> A New Internet-Draft is available from the on-line Internet-Drafts =
directories.
> This draft is a work item of the Real-Time Communication in =
WEB-browsers Working Group of the IETF.
>=20
> 	Title           : Web Real-Time Communication (WebRTC): Media =
Transport and Use of RTP
> 	Author(s)       : Colin Perkins
>                          Magnus Westerlund
>                          Joerg Ott
> 	Filename        : draft-ietf-rtcweb-rtp-usage-09.txt
> 	Pages           : 40
> 	Date            : 2013-09-05
>=20
> Abstract:
>   The Web Real-Time Communication (WebRTC) framework provides support
>   for direct interactive rich communication using audio, video, text,
>   collaboration, games, etc.  between two peers' web-browsers.  This
>   memo describes the media transport aspects of the WebRTC framework.
>   It specifies how the Real-time Transport Protocol (RTP) is used in
>   the WebRTC context, and gives requirements for which RTP features,
>   profiles, and extensions need to be supported.
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-rtp-usage
>=20
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-09
>=20
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-rtp-usage-09
>=20
>=20
> Please note that it may take a couple of minutes from the time of =
submission
> until the htmlized version and diff are available at tools.ietf.org.
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



--=20
Colin Perkins
http://csperkins.org/




From bo.burman@ericsson.com  Thu Sep  5 06:06:22 2013
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From: Bo Burman <bo.burman@ericsson.com>
To: "Rauschenbach, Uwe (NSN - DE/Munich)" <uwe.rauschenbach@nsn.com>, "ext Mo Zanaty (mzanaty)" <mzanaty@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: I-D Action: draft-ietf-rtcweb-audio-02.txt
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Date: Thu, 5 Sep 2013 13:05:46 +0000
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References: <20130802162957.17108.79281.idtracker@ietfa.amsl.com> <BBE9739C2C302046BD34B42713A1E2A22DF83C31@ESESSMB105.ericsson.se> <3879D71E758A7E4AA99A35DD8D41D3D91D5260A2@xmb-rcd-x14.cisco.com> <56C2F665D49E0341B9DF5938005ACDF80E8A65@DEMUMBX005.nsn-intra.net>
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
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I also support it.
/Bo

> -----Original Message-----
> From: Rauschenbach, Uwe (NSN - DE/Munich) [mailto:uwe.rauschenbach@nsn.co=
m]
> Sent: den 5 september 2013 12:14
> To: ext Mo Zanaty (mzanaty); Bo Burman; rtcweb@ietf.org
> Subject: RE: I-D Action: draft-ietf-rtcweb-audio-02.txt
>=20
> I support this proposal.
>=20
> Kind regards,
> Uwe
>=20
>=20
> > -----Original Message-----
> > From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> > Behalf Of ext Mo Zanaty (mzanaty)
> > Sent: Wednesday, September 04, 2013 10:03 PM
> > To: Bo Burman; rtcweb@ietf.org
> > Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
> >
> > I support the proposed text addition. But rather than adding at the
> > end of section 3, replace the beginning of section 3 as below.
> >
> > OLD:
> >    To ensure a baseline level of interoperability between WebRTC
> >    clients, a minimum set of required codecs are specified below.
> >    While this section specifies the codecs that will be mandated for
> > all
> >    WebRTC client implementations, it leaves the question of supporting
> >    additional codecs to the will of the implementer.
> >
> > NEW:
> >    To ensure a baseline level of interoperability between WebRTC
> >    clients, a minimum set of required codecs are specified below.
> >    If other suitable audio codecs are available to the browser to use,
> >    it is RECOMMENDED that they are also included in the offer in order
> >    to maximize the possibility to establish the session without the
> > need
> >    for audio transcoding.
> >
> > Mo
> >
> > -----Original Message-----
> > From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> > Behalf Of Bo Burman
> > Sent: Tuesday, August 27, 2013 10:53 AM
> > To: rtcweb@ietf.org
> > Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
> >
> > Based on the previous discussion thread http://www.ietf.org/mail-
> > archive/web/rtcweb/current/msg08501.html, I believe there is support
> > to add some text into this document.
> >
> > I thus propose to add the following text at the end of section 3:
> >
> > If other suitable audio codecs are available to the browser to use, it
> > is RECOMMENDED that they are also included in the offer in order to
> > maximize the possibility to establish the session without the need for
> > audio transcoding
> >
> > Cheers,
> > Bo
> >
> > > -----Original Message-----
> > > From: i-d-announce-bounces@ietf.org [mailto:i-d-announce-
> > bounces@ietf.org] On Behalf Of internet-drafts@ietf.org
> > > Sent: den 2 augusti 2013 18:30
> > > To: i-d-announce@ietf.org
> > > Cc: rtcweb@ietf.org
> > > Subject: I-D Action: draft-ietf-rtcweb-audio-02.txt
> > >
> > >
> > > A New Internet-Draft is available from the on-line Internet-Drafts
> > directories.
> > >  This draft is a work item of the Real-Time Communication in WEB-
> > browsers Working Group of the IETF.
> > >
> > > 	Title           : WebRTC Audio Codec and Processing Requirements
> > > 	Author(s)       : Jean-Marc Valin
> > >                           Cary Bran
> > > 	Filename        : draft-ietf-rtcweb-audio-02.txt
> > > 	Pages           : 6
> > > 	Date            : 2013-08-02
> > >
> > > Abstract:
> > >    This document outlines the audio codec and processing requirements
> > >    for WebRTC client application and endpoint devices.
> > >
> > >
> > > The IETF datatracker status page for this draft is:
> > > https://datatracker.ietf.org/doc/draft-ietf-rtcweb-audio
> > >
> > > There's also a htmlized version available at:
> > > http://tools.ietf.org/html/draft-ietf-rtcweb-audio-02
> > >
> > > A diff from the previous version is available at:
> > > http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-audio-02
> > >
> > >
> > > Please note that it may take a couple of minutes from the time of
> > submission until the htmlized version and diff are
> > > available at tools.ietf.org.
> > >
> > > Internet-Drafts are also available by anonymous FTP at:
> > > ftp://ftp.ietf.org/internet-drafts/
> > >
> > > _______________________________________________
> > > I-D-Announce mailing list
> > > I-D-Announce@ietf.org
> > > https://www.ietf.org/mailman/listinfo/i-d-announce
> > > Internet-Draft directories: http://www.ietf.org/shadow.html or
> > ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb

From lgeyser@gmail.com  Thu Sep  5 07:06:07 2013
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Date: Thu, 5 Sep 2013 16:06:04 +0200
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From: Leon Geyser <lgeyser@gmail.com>
To: Bo Burman <bo.burman@ericsson.com>
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
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It would be nice if a word like mandated/mandatory is present in the new
text like it was in the old text. Maybe REQUIRED in uppercase?


On 5 September 2013 15:05, Bo Burman <bo.burman@ericsson.com> wrote:

> I also support it.
> /Bo
>
> > -----Original Message-----
> > From: Rauschenbach, Uwe (NSN - DE/Munich) [mailto:
> uwe.rauschenbach@nsn.com]
> > Sent: den 5 september 2013 12:14
> > To: ext Mo Zanaty (mzanaty); Bo Burman; rtcweb@ietf.org
> > Subject: RE: I-D Action: draft-ietf-rtcweb-audio-02.txt
> >
> > I support this proposal.
> >
> > Kind regards,
> > Uwe
> >
> >
> > > -----Original Message-----
> > > From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> > > Behalf Of ext Mo Zanaty (mzanaty)
> > > Sent: Wednesday, September 04, 2013 10:03 PM
> > > To: Bo Burman; rtcweb@ietf.org
> > > Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
> > >
> > > I support the proposed text addition. But rather than adding at the
> > > end of section 3, replace the beginning of section 3 as below.
> > >
> > > OLD:
> > >    To ensure a baseline level of interoperability between WebRTC
> > >    clients, a minimum set of required codecs are specified below.
> > >    While this section specifies the codecs that will be mandated for
> > > all
> > >    WebRTC client implementations, it leaves the question of supporting
> > >    additional codecs to the will of the implementer.
> > >
> > > NEW:
> > >    To ensure a baseline level of interoperability between WebRTC
> > >    clients, a minimum set of required codecs are specified below.
> > >    If other suitable audio codecs are available to the browser to use,
> > >    it is RECOMMENDED that they are also included in the offer in order
> > >    to maximize the possibility to establish the session without the
> > > need
> > >    for audio transcoding.
> > >
> > > Mo
> > >
> > > -----Original Message-----
> > > From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> > > Behalf Of Bo Burman
> > > Sent: Tuesday, August 27, 2013 10:53 AM
> > > To: rtcweb@ietf.org
> > > Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
> > >
> > > Based on the previous discussion thread http://www.ietf.org/mail-
> > > archive/web/rtcweb/current/msg08501.html, I believe there is support
> > > to add some text into this document.
> > >
> > > I thus propose to add the following text at the end of section 3:
> > >
> > > If other suitable audio codecs are available to the browser to use, it
> > > is RECOMMENDED that they are also included in the offer in order to
> > > maximize the possibility to establish the session without the need for
> > > audio transcoding
> > >
> > > Cheers,
> > > Bo
> > >
> > > > -----Original Message-----
> > > > From: i-d-announce-bounces@ietf.org [mailto:i-d-announce-
> > > bounces@ietf.org] On Behalf Of internet-drafts@ietf.org
> > > > Sent: den 2 augusti 2013 18:30
> > > > To: i-d-announce@ietf.org
> > > > Cc: rtcweb@ietf.org
> > > > Subject: I-D Action: draft-ietf-rtcweb-audio-02.txt
> > > >
> > > >
> > > > A New Internet-Draft is available from the on-line Internet-Drafts
> > > directories.
> > > >  This draft is a work item of the Real-Time Communication in WEB-
> > > browsers Working Group of the IETF.
> > > >
> > > >   Title           : WebRTC Audio Codec and Processing Requirements
> > > >   Author(s)       : Jean-Marc Valin
> > > >                           Cary Bran
> > > >   Filename        : draft-ietf-rtcweb-audio-02.txt
> > > >   Pages           : 6
> > > >   Date            : 2013-08-02
> > > >
> > > > Abstract:
> > > >    This document outlines the audio codec and processing requirements
> > > >    for WebRTC client application and endpoint devices.
> > > >
> > > >
> > > > The IETF datatracker status page for this draft is:
> > > > https://datatracker.ietf.org/doc/draft-ietf-rtcweb-audio
> > > >
> > > > There's also a htmlized version available at:
> > > > http://tools.ietf.org/html/draft-ietf-rtcweb-audio-02
> > > >
> > > > A diff from the previous version is available at:
> > > > http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-audio-02
> > > >
> > > >
> > > > Please note that it may take a couple of minutes from the time of
> > > submission until the htmlized version and diff are
> > > > available at tools.ietf.org.
> > > >
> > > > Internet-Drafts are also available by anonymous FTP at:
> > > > ftp://ftp.ietf.org/internet-drafts/
> > > >
> > > > _______________________________________________
> > > > I-D-Announce mailing list
> > > > I-D-Announce@ietf.org
> > > > https://www.ietf.org/mailman/listinfo/i-d-announce
> > > > Internet-Draft directories: http://www.ietf.org/shadow.html or
> > > ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> > > _______________________________________________
> > > rtcweb mailing list
> > > rtcweb@ietf.org
> > > https://www.ietf.org/mailman/listinfo/rtcweb
> > > _______________________________________________
> > > rtcweb mailing list
> > > rtcweb@ietf.org
> > > https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--001a11c34374348a5804e5a36ca8
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<div dir=3D"ltr">It would be nice if a word like mandated/mandatory is pres=
ent in the new text like it was in the old text. Maybe REQUIRED in uppercas=
e?<br></div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">O=
n 5 September 2013 15:05, Bo Burman <span dir=3D"ltr">&lt;<a href=3D"mailto=
:bo.burman@ericsson.com" target=3D"_blank">bo.burman@ericsson.com</a>&gt;</=
span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">I also support it.<br>
<span class=3D"HOEnZb"><font color=3D"#888888">/Bo<br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
&gt; -----Original Message-----<br>
&gt; From: Rauschenbach, Uwe (NSN - DE/Munich) [mailto:<a href=3D"mailto:uw=
e.rauschenbach@nsn.com">uwe.rauschenbach@nsn.com</a>]<br>
&gt; Sent: den 5 september 2013 12:14<br>
&gt; To: ext Mo Zanaty (mzanaty); Bo Burman; <a href=3D"mailto:rtcweb@ietf.=
org">rtcweb@ietf.org</a><br>
&gt; Subject: RE: I-D Action: draft-ietf-rtcweb-audio-02.txt<br>
&gt;<br>
&gt; I support this proposal.<br>
&gt;<br>
&gt; Kind regards,<br>
&gt; Uwe<br>
&gt;<br>
&gt;<br>
&gt; &gt; -----Original Message-----<br>
&gt; &gt; From: <a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@i=
etf.org</a> [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounc=
es@ietf.org</a>] On<br>
&gt; &gt; Behalf Of ext Mo Zanaty (mzanaty)<br>
&gt; &gt; Sent: Wednesday, September 04, 2013 10:03 PM<br>
&gt; &gt; To: Bo Burman; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org=
</a><br>
&gt; &gt; Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt<=
br>
&gt; &gt;<br>
&gt; &gt; I support the proposed text addition. But rather than adding at t=
he<br>
&gt; &gt; end of section 3, replace the beginning of section 3 as below.<br=
>
&gt; &gt;<br>
&gt; &gt; OLD:<br>
&gt; &gt; =A0 =A0To ensure a baseline level of interoperability between Web=
RTC<br>
&gt; &gt; =A0 =A0clients, a minimum set of required codecs are specified be=
low.<br>
&gt; &gt; =A0 =A0While this section specifies the codecs that will be manda=
ted for<br>
&gt; &gt; all<br>
&gt; &gt; =A0 =A0WebRTC client implementations, it leaves the question of s=
upporting<br>
&gt; &gt; =A0 =A0additional codecs to the will of the implementer.<br>
&gt; &gt;<br>
&gt; &gt; NEW:<br>
&gt; &gt; =A0 =A0To ensure a baseline level of interoperability between Web=
RTC<br>
&gt; &gt; =A0 =A0clients, a minimum set of required codecs are specified be=
low.<br>
&gt; &gt; =A0 =A0If other suitable audio codecs are available to the browse=
r to use,<br>
&gt; &gt; =A0 =A0it is RECOMMENDED that they are also included in the offer=
 in order<br>
&gt; &gt; =A0 =A0to maximize the possibility to establish the session witho=
ut the<br>
&gt; &gt; need<br>
&gt; &gt; =A0 =A0for audio transcoding.<br>
&gt; &gt;<br>
&gt; &gt; Mo<br>
&gt; &gt;<br>
&gt; &gt; -----Original Message-----<br>
&gt; &gt; From: <a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@i=
etf.org</a> [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounc=
es@ietf.org</a>] On<br>
&gt; &gt; Behalf Of Bo Burman<br>
&gt; &gt; Sent: Tuesday, August 27, 2013 10:53 AM<br>
&gt; &gt; To: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; &gt; Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt<=
br>
&gt; &gt;<br>
&gt; &gt; Based on the previous discussion thread <a href=3D"http://www.iet=
f.org/mail-" target=3D"_blank">http://www.ietf.org/mail-</a><br>
&gt; &gt; archive/web/rtcweb/current/msg08501.html, I believe there is supp=
ort<br>
&gt; &gt; to add some text into this document.<br>
&gt; &gt;<br>
&gt; &gt; I thus propose to add the following text at the end of section 3:=
<br>
&gt; &gt;<br>
&gt; &gt; If other suitable audio codecs are available to the browser to us=
e, it<br>
&gt; &gt; is RECOMMENDED that they are also included in the offer in order =
to<br>
&gt; &gt; maximize the possibility to establish the session without the nee=
d for<br>
&gt; &gt; audio transcoding<br>
&gt; &gt;<br>
&gt; &gt; Cheers,<br>
&gt; &gt; Bo<br>
&gt; &gt;<br>
&gt; &gt; &gt; -----Original Message-----<br>
&gt; &gt; &gt; From: <a href=3D"mailto:i-d-announce-bounces@ietf.org">i-d-a=
nnounce-bounces@ietf.org</a> [mailto:<a href=3D"mailto:i-d-announce-">i-d-a=
nnounce-</a><br>
&gt; &gt; <a href=3D"mailto:bounces@ietf.org">bounces@ietf.org</a>] On Beha=
lf Of <a href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org<=
/a><br>
&gt; &gt; &gt; Sent: den 2 augusti 2013 18:30<br>
&gt; &gt; &gt; To: <a href=3D"mailto:i-d-announce@ietf.org">i-d-announce@ie=
tf.org</a><br>
&gt; &gt; &gt; Cc: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><b=
r>
&gt; &gt; &gt; Subject: I-D Action: draft-ietf-rtcweb-audio-02.txt<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; A New Internet-Draft is available from the on-line Internet-=
Drafts<br>
&gt; &gt; directories.<br>
&gt; &gt; &gt; =A0This draft is a work item of the Real-Time Communication =
in WEB-<br>
&gt; &gt; browsers Working Group of the IETF.<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; =A0 Title =A0 =A0 =A0 =A0 =A0 : WebRTC Audio Codec and Proce=
ssing Requirements<br>
&gt; &gt; &gt; =A0 Author(s) =A0 =A0 =A0 : Jean-Marc Valin<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Cary Bra=
n<br>
&gt; &gt; &gt; =A0 Filename =A0 =A0 =A0 =A0: draft-ietf-rtcweb-audio-02.txt=
<br>
&gt; &gt; &gt; =A0 Pages =A0 =A0 =A0 =A0 =A0 : 6<br>
&gt; &gt; &gt; =A0 Date =A0 =A0 =A0 =A0 =A0 =A0: 2013-08-02<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; Abstract:<br>
&gt; &gt; &gt; =A0 =A0This document outlines the audio codec and processing=
 requirements<br>
&gt; &gt; &gt; =A0 =A0for WebRTC client application and endpoint devices.<b=
r>
&gt; &gt; &gt;<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; The IETF datatracker status page for this draft is:<br>
&gt; &gt; &gt; <a href=3D"https://datatracker.ietf.org/doc/draft-ietf-rtcwe=
b-audio" target=3D"_blank">https://datatracker.ietf.org/doc/draft-ietf-rtcw=
eb-audio</a><br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; There&#39;s also a htmlized version available at:<br>
&gt; &gt; &gt; <a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-audi=
o-02" target=3D"_blank">http://tools.ietf.org/html/draft-ietf-rtcweb-audio-=
02</a><br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; A diff from the previous version is available at:<br>
&gt; &gt; &gt; <a href=3D"http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtc=
web-audio-02" target=3D"_blank">http://www.ietf.org/rfcdiff?url2=3Ddraft-ie=
tf-rtcweb-audio-02</a><br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; Please note that it may take a couple of minutes from the ti=
me of<br>
&gt; &gt; submission until the htmlized version and diff are<br>
&gt; &gt; &gt; available at <a href=3D"http://tools.ietf.org" target=3D"_bl=
ank">tools.ietf.org</a>.<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; Internet-Drafts are also available by anonymous FTP at:<br>
&gt; &gt; &gt; <a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=3D"_b=
lank">ftp://ftp.ietf.org/internet-drafts/</a><br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; _______________________________________________<br>
&gt; &gt; &gt; I-D-Announce mailing list<br>
&gt; &gt; &gt; <a href=3D"mailto:I-D-Announce@ietf.org">I-D-Announce@ietf.o=
rg</a><br>
&gt; &gt; &gt; <a href=3D"https://www.ietf.org/mailman/listinfo/i-d-announc=
e" target=3D"_blank">https://www.ietf.org/mailman/listinfo/i-d-announce</a>=
<br>
&gt; &gt; &gt; Internet-Draft directories: <a href=3D"http://www.ietf.org/s=
hadow.html" target=3D"_blank">http://www.ietf.org/shadow.html</a> or<br>
&gt; &gt; <a href=3D"ftp://ftp.ietf.org/ietf/1shadow-sites.txt" target=3D"_=
blank">ftp://ftp.ietf.org/ietf/1shadow-sites.txt</a><br>
&gt; &gt; _______________________________________________<br>
&gt; &gt; rtcweb mailing list<br>
&gt; &gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; &gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=
=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt; &gt; _______________________________________________<br>
&gt; &gt; rtcweb mailing list<br>
&gt; &gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; &gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=
=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>

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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
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On 09/05/2013 04:06 PM, Leon Geyser wrote:
> It would be nice if a word like mandated/mandatory is present in the 
> new text like it was in the old text. Maybe REQUIRED in uppercase?

REQUIRED is an RFC 2119 reserved word, and means the same thing as MUST.
I would strenously object to REQUIRED.

RECOMMENDED is also an RFC 2119 reserved word, and means the same thing 
as SHOULD.
I have misgivings about giving such a strongly worded recommendation, 
but I can live with it.


>
>
> On 5 September 2013 15:05, Bo Burman <bo.burman@ericsson.com 
> <mailto:bo.burman@ericsson.com>> wrote:
>
>     I also support it.
>     /Bo
>
>     > -----Original Message-----
>     > From: Rauschenbach, Uwe (NSN - DE/Munich)
>     [mailto:uwe.rauschenbach@nsn.com <mailto:uwe.rauschenbach@nsn.com>]
>     > Sent: den 5 september 2013 12:14
>     > To: ext Mo Zanaty (mzanaty); Bo Burman; rtcweb@ietf.org
>     <mailto:rtcweb@ietf.org>
>     > Subject: RE: I-D Action: draft-ietf-rtcweb-audio-02.txt
>     >
>     > I support this proposal.
>     >
>     > Kind regards,
>     > Uwe
>     >
>     >
>     > > -----Original Message-----
>     > > From: rtcweb-bounces@ietf.org <mailto:rtcweb-bounces@ietf.org>
>     [mailto:rtcweb-bounces@ietf.org <mailto:rtcweb-bounces@ietf.org>] On
>     > > Behalf Of ext Mo Zanaty (mzanaty)
>     > > Sent: Wednesday, September 04, 2013 10:03 PM
>     > > To: Bo Burman; rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     > > Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
>     > >
>     > > I support the proposed text addition. But rather than adding
>     at the
>     > > end of section 3, replace the beginning of section 3 as below.
>     > >
>     > > OLD:
>     > >    To ensure a baseline level of interoperability between WebRTC
>     > >    clients, a minimum set of required codecs are specified below.
>     > >    While this section specifies the codecs that will be
>     mandated for
>     > > all
>     > >    WebRTC client implementations, it leaves the question of
>     supporting
>     > >    additional codecs to the will of the implementer.
>     > >
>     > > NEW:
>     > >    To ensure a baseline level of interoperability between WebRTC
>     > >    clients, a minimum set of required codecs are specified below.
>     > >    If other suitable audio codecs are available to the browser
>     to use,
>     > >    it is RECOMMENDED that they are also included in the offer
>     in order
>     > >    to maximize the possibility to establish the session
>     without the
>     > > need
>     > >    for audio transcoding.
>     > >
>     > > Mo
>     > >
>     > > -----Original Message-----
>     > > From: rtcweb-bounces@ietf.org <mailto:rtcweb-bounces@ietf.org>
>     [mailto:rtcweb-bounces@ietf.org <mailto:rtcweb-bounces@ietf.org>] On
>     > > Behalf Of Bo Burman
>     > > Sent: Tuesday, August 27, 2013 10:53 AM
>     > > To: rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     > > Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
>     > >
>     > > Based on the previous discussion thread http://www.ietf.org/mail-
>     > > archive/web/rtcweb/current/msg08501.html, I believe there is
>     support
>     > > to add some text into this document.
>     > >
>     > > I thus propose to add the following text at the end of section 3:
>     > >
>     > > If other suitable audio codecs are available to the browser to
>     use, it
>     > > is RECOMMENDED that they are also included in the offer in
>     order to
>     > > maximize the possibility to establish the session without the
>     need for
>     > > audio transcoding
>     > >
>     > > Cheers,
>     > > Bo
>     > >
>     > > > -----Original Message-----
>     > > > From: i-d-announce-bounces@ietf.org
>     <mailto:i-d-announce-bounces@ietf.org> [mailto:i-d-announce-
>     <mailto:i-d-announce->
>     > > bounces@ietf.org <mailto:bounces@ietf.org>] On Behalf Of
>     internet-drafts@ietf.org <mailto:internet-drafts@ietf.org>
>     > > > Sent: den 2 augusti 2013 18:30
>     > > > To: i-d-announce@ietf.org <mailto:i-d-announce@ietf.org>
>     > > > Cc: rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     > > > Subject: I-D Action: draft-ietf-rtcweb-audio-02.txt
>     > > >
>     > > >
>     > > > A New Internet-Draft is available from the on-line
>     Internet-Drafts
>     > > directories.
>     > > >  This draft is a work item of the Real-Time Communication in
>     WEB-
>     > > browsers Working Group of the IETF.
>     > > >
>     > > >   Title           : WebRTC Audio Codec and Processing
>     Requirements
>     > > >   Author(s)       : Jean-Marc Valin
>     > > >                           Cary Bran
>     > > >   Filename        : draft-ietf-rtcweb-audio-02.txt
>     > > >   Pages           : 6
>     > > >   Date            : 2013-08-02
>     > > >
>     > > > Abstract:
>     > > >    This document outlines the audio codec and processing
>     requirements
>     > > >    for WebRTC client application and endpoint devices.
>     > > >
>     > > >
>     > > > The IETF datatracker status page for this draft is:
>     > > > https://datatracker.ietf.org/doc/draft-ietf-rtcweb-audio
>     > > >
>     > > > There's also a htmlized version available at:
>     > > > http://tools.ietf.org/html/draft-ietf-rtcweb-audio-02
>     > > >
>     > > > A diff from the previous version is available at:
>     > > > http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-audio-02
>     > > >
>     > > >
>     > > > Please note that it may take a couple of minutes from the
>     time of
>     > > submission until the htmlized version and diff are
>     > > > available at tools.ietf.org <http://tools.ietf.org>.
>     > > >
>     > > > Internet-Drafts are also available by anonymous FTP at:
>     > > > ftp://ftp.ietf.org/internet-drafts/
>     > > >
>     > > > _______________________________________________
>     > > > I-D-Announce mailing list
>     > > > I-D-Announce@ietf.org <mailto:I-D-Announce@ietf.org>
>     > > > https://www.ietf.org/mailman/listinfo/i-d-announce
>     > > > Internet-Draft directories: http://www.ietf.org/shadow.html or
>     > > ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>     > > _______________________________________________
>     > > rtcweb mailing list
>     > > rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     > > https://www.ietf.org/mailman/listinfo/rtcweb
>     > > _______________________________________________
>     > > rtcweb mailing list
>     > > rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     > > https://www.ietf.org/mailman/listinfo/rtcweb
>     _______________________________________________
>     rtcweb mailing list
>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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    <div class="moz-cite-prefix">On 09/05/2013 04:06 PM, Leon Geyser
      wrote:<br>
    </div>
    <blockquote
cite="mid:CAGgHUiSK-bZrdXtxf-An8NM+pw-iqoWCrsG_bRUpxcD2DOCQrQ@mail.gmail.com"
      type="cite">
      <div dir="ltr">It would be nice if a word like mandated/mandatory
        is present in the new text like it was in the old text. Maybe
        REQUIRED in uppercase?<br>
      </div>
    </blockquote>
    <br>
    REQUIRED is an RFC 2119 reserved word, and means the same thing as
    MUST.<br>
    I would strenously object to REQUIRED.<br>
    <br>
    RECOMMENDED is also an RFC 2119 reserved word, and means the same
    thing as SHOULD.<br>
    I have misgivings about giving such a strongly worded
    recommendation, but I can live with it.<br>
    <br>
    <br>
    <blockquote
cite="mid:CAGgHUiSK-bZrdXtxf-An8NM+pw-iqoWCrsG_bRUpxcD2DOCQrQ@mail.gmail.com"
      type="cite">
      <div class="gmail_extra"><br>
        <br>
        <div class="gmail_quote">On 5 September 2013 15:05, Bo Burman <span
            dir="ltr">&lt;<a moz-do-not-send="true"
              href="mailto:bo.burman@ericsson.com" target="_blank">bo.burman@ericsson.com</a>&gt;</span>
          wrote:<br>
          <blockquote class="gmail_quote" style="margin:0 0 0
            .8ex;border-left:1px #ccc solid;padding-left:1ex">I also
            support it.<br>
            <span class="HOEnZb"><font color="#888888">/Bo<br>
              </font></span>
            <div class="HOEnZb">
              <div class="h5"><br>
                &gt; -----Original Message-----<br>
                &gt; From: Rauschenbach, Uwe (NSN - DE/Munich) [mailto:<a
                  moz-do-not-send="true"
                  href="mailto:uwe.rauschenbach@nsn.com">uwe.rauschenbach@nsn.com</a>]<br>
                &gt; Sent: den 5 september 2013 12:14<br>
                &gt; To: ext Mo Zanaty (mzanaty); Bo Burman; <a
                  moz-do-not-send="true" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                &gt; Subject: RE: I-D Action:
                draft-ietf-rtcweb-audio-02.txt<br>
                &gt;<br>
                &gt; I support this proposal.<br>
                &gt;<br>
                &gt; Kind regards,<br>
                &gt; Uwe<br>
                &gt;<br>
                &gt;<br>
                &gt; &gt; -----Original Message-----<br>
                &gt; &gt; From: <a moz-do-not-send="true"
                  href="mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a>
                [mailto:<a moz-do-not-send="true"
                  href="mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a>]
                On<br>
                &gt; &gt; Behalf Of ext Mo Zanaty (mzanaty)<br>
                &gt; &gt; Sent: Wednesday, September 04, 2013 10:03 PM<br>
                &gt; &gt; To: Bo Burman; <a moz-do-not-send="true"
                  href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                &gt; &gt; Subject: Re: [rtcweb] I-D Action:
                draft-ietf-rtcweb-audio-02.txt<br>
                &gt; &gt;<br>
                &gt; &gt; I support the proposed text addition. But
                rather than adding at the<br>
                &gt; &gt; end of section 3, replace the beginning of
                section 3 as below.<br>
                &gt; &gt;<br>
                &gt; &gt; OLD:<br>
                &gt; &gt; &nbsp; &nbsp;To ensure a baseline level of
                interoperability between WebRTC<br>
                &gt; &gt; &nbsp; &nbsp;clients, a minimum set of required codecs
                are specified below.<br>
                &gt; &gt; &nbsp; &nbsp;While this section specifies the codecs
                that will be mandated for<br>
                &gt; &gt; all<br>
                &gt; &gt; &nbsp; &nbsp;WebRTC client implementations, it leaves
                the question of supporting<br>
                &gt; &gt; &nbsp; &nbsp;additional codecs to the will of the
                implementer.<br>
                &gt; &gt;<br>
                &gt; &gt; NEW:<br>
                &gt; &gt; &nbsp; &nbsp;To ensure a baseline level of
                interoperability between WebRTC<br>
                &gt; &gt; &nbsp; &nbsp;clients, a minimum set of required codecs
                are specified below.<br>
                &gt; &gt; &nbsp; &nbsp;If other suitable audio codecs are
                available to the browser to use,<br>
                &gt; &gt; &nbsp; &nbsp;it is RECOMMENDED that they are also
                included in the offer in order<br>
                &gt; &gt; &nbsp; &nbsp;to maximize the possibility to establish
                the session without the<br>
                &gt; &gt; need<br>
                &gt; &gt; &nbsp; &nbsp;for audio transcoding.<br>
                &gt; &gt;<br>
                &gt; &gt; Mo<br>
                &gt; &gt;<br>
                &gt; &gt; -----Original Message-----<br>
                &gt; &gt; From: <a moz-do-not-send="true"
                  href="mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a>
                [mailto:<a moz-do-not-send="true"
                  href="mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a>]
                On<br>
                &gt; &gt; Behalf Of Bo Burman<br>
                &gt; &gt; Sent: Tuesday, August 27, 2013 10:53 AM<br>
                &gt; &gt; To: <a moz-do-not-send="true"
                  href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                &gt; &gt; Subject: Re: [rtcweb] I-D Action:
                draft-ietf-rtcweb-audio-02.txt<br>
                &gt; &gt;<br>
                &gt; &gt; Based on the previous discussion thread <a
                  moz-do-not-send="true"
                  href="http://www.ietf.org/mail-" target="_blank">http://www.ietf.org/mail-</a><br>
                &gt; &gt; archive/web/rtcweb/current/msg08501.html, I
                believe there is support<br>
                &gt; &gt; to add some text into this document.<br>
                &gt; &gt;<br>
                &gt; &gt; I thus propose to add the following text at
                the end of section 3:<br>
                &gt; &gt;<br>
                &gt; &gt; If other suitable audio codecs are available
                to the browser to use, it<br>
                &gt; &gt; is RECOMMENDED that they are also included in
                the offer in order to<br>
                &gt; &gt; maximize the possibility to establish the
                session without the need for<br>
                &gt; &gt; audio transcoding<br>
                &gt; &gt;<br>
                &gt; &gt; Cheers,<br>
                &gt; &gt; Bo<br>
                &gt; &gt;<br>
                &gt; &gt; &gt; -----Original Message-----<br>
                &gt; &gt; &gt; From: <a moz-do-not-send="true"
                  href="mailto:i-d-announce-bounces@ietf.org">i-d-announce-bounces@ietf.org</a>
                [mailto:<a moz-do-not-send="true"
                  href="mailto:i-d-announce-">i-d-announce-</a><br>
                &gt; &gt; <a moz-do-not-send="true"
                  href="mailto:bounces@ietf.org">bounces@ietf.org</a>]
                On Behalf Of <a moz-do-not-send="true"
                  href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a><br>
                &gt; &gt; &gt; Sent: den 2 augusti 2013 18:30<br>
                &gt; &gt; &gt; To: <a moz-do-not-send="true"
                  href="mailto:i-d-announce@ietf.org">i-d-announce@ietf.org</a><br>
                &gt; &gt; &gt; Cc: <a moz-do-not-send="true"
                  href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                &gt; &gt; &gt; Subject: I-D Action:
                draft-ietf-rtcweb-audio-02.txt<br>
                &gt; &gt; &gt;<br>
                &gt; &gt; &gt;<br>
                &gt; &gt; &gt; A New Internet-Draft is available from
                the on-line Internet-Drafts<br>
                &gt; &gt; directories.<br>
                &gt; &gt; &gt; &nbsp;This draft is a work item of the
                Real-Time Communication in WEB-<br>
                &gt; &gt; browsers Working Group of the IETF.<br>
                &gt; &gt; &gt;<br>
                &gt; &gt; &gt; &nbsp; Title &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; : WebRTC Audio Codec
                and Processing Requirements<br>
                &gt; &gt; &gt; &nbsp; Author(s) &nbsp; &nbsp; &nbsp; : Jean-Marc Valin<br>
                &gt; &gt; &gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Cary Bran<br>
                &gt; &gt; &gt; &nbsp; Filename &nbsp; &nbsp; &nbsp; &nbsp;:
                draft-ietf-rtcweb-audio-02.txt<br>
                &gt; &gt; &gt; &nbsp; Pages &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; : 6<br>
                &gt; &gt; &gt; &nbsp; Date &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;: 2013-08-02<br>
                &gt; &gt; &gt;<br>
                &gt; &gt; &gt; Abstract:<br>
                &gt; &gt; &gt; &nbsp; &nbsp;This document outlines the audio codec
                and processing requirements<br>
                &gt; &gt; &gt; &nbsp; &nbsp;for WebRTC client application and
                endpoint devices.<br>
                &gt; &gt; &gt;<br>
                &gt; &gt; &gt;<br>
                &gt; &gt; &gt; The IETF datatracker status page for this
                draft is:<br>
                &gt; &gt; &gt; <a moz-do-not-send="true"
                  href="https://datatracker.ietf.org/doc/draft-ietf-rtcweb-audio"
                  target="_blank">https://datatracker.ietf.org/doc/draft-ietf-rtcweb-audio</a><br>
                &gt; &gt; &gt;<br>
                &gt; &gt; &gt; There's also a htmlized version available
                at:<br>
                &gt; &gt; &gt; <a moz-do-not-send="true"
                  href="http://tools.ietf.org/html/draft-ietf-rtcweb-audio-02"
                  target="_blank">http://tools.ietf.org/html/draft-ietf-rtcweb-audio-02</a><br>
                &gt; &gt; &gt;<br>
                &gt; &gt; &gt; A diff from the previous version is
                available at:<br>
                &gt; &gt; &gt; <a moz-do-not-send="true"
                  href="http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-audio-02"
                  target="_blank">http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-audio-02</a><br>
                &gt; &gt; &gt;<br>
                &gt; &gt; &gt;<br>
                &gt; &gt; &gt; Please note that it may take a couple of
                minutes from the time of<br>
                &gt; &gt; submission until the htmlized version and diff
                are<br>
                &gt; &gt; &gt; available at <a moz-do-not-send="true"
                  href="http://tools.ietf.org" target="_blank">tools.ietf.org</a>.<br>
                &gt; &gt; &gt;<br>
                &gt; &gt; &gt; Internet-Drafts are also available by
                anonymous FTP at:<br>
                &gt; &gt; &gt; <a moz-do-not-send="true"
                  href="ftp://ftp.ietf.org/internet-drafts/"
                  target="_blank">ftp://ftp.ietf.org/internet-drafts/</a><br>
                &gt; &gt; &gt;<br>
                &gt; &gt; &gt;
                _______________________________________________<br>
                &gt; &gt; &gt; I-D-Announce mailing list<br>
                &gt; &gt; &gt; <a moz-do-not-send="true"
                  href="mailto:I-D-Announce@ietf.org">I-D-Announce@ietf.org</a><br>
                &gt; &gt; &gt; <a moz-do-not-send="true"
                  href="https://www.ietf.org/mailman/listinfo/i-d-announce"
                  target="_blank">https://www.ietf.org/mailman/listinfo/i-d-announce</a><br>
                &gt; &gt; &gt; Internet-Draft directories: <a
                  moz-do-not-send="true"
                  href="http://www.ietf.org/shadow.html" target="_blank">http://www.ietf.org/shadow.html</a>
                or<br>
                &gt; &gt; <a moz-do-not-send="true"
                  href="ftp://ftp.ietf.org/ietf/1shadow-sites.txt"
                  target="_blank">ftp://ftp.ietf.org/ietf/1shadow-sites.txt</a><br>
                &gt; &gt;
                _______________________________________________<br>
                &gt; &gt; rtcweb mailing list<br>
                &gt; &gt; <a moz-do-not-send="true"
                  href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                &gt; &gt; <a moz-do-not-send="true"
                  href="https://www.ietf.org/mailman/listinfo/rtcweb"
                  target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
                &gt; &gt;
                _______________________________________________<br>
                &gt; &gt; rtcweb mailing list<br>
                &gt; &gt; <a moz-do-not-send="true"
                  href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                &gt; &gt; <a moz-do-not-send="true"
                  href="https://www.ietf.org/mailman/listinfo/rtcweb"
                  target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
                _______________________________________________<br>
                rtcweb mailing list<br>
                <a moz-do-not-send="true" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                <a moz-do-not-send="true"
                  href="https://www.ietf.org/mailman/listinfo/rtcweb"
                  target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
              </div>
            </div>
          </blockquote>
        </div>
        <br>
      </div>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
  </body>
</html>

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From mzanaty@cisco.com  Thu Sep  5 07:41:59 2013
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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: Leon Geyser <lgeyser@gmail.com>, Bo Burman <bo.burman@ericsson.com>
Thread-Topic: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
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Date: Thu, 5 Sep 2013 14:41:47 +0000
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References: <20130802162957.17108.79281.idtracker@ietfa.amsl.com> <BBE9739C2C302046BD34B42713A1E2A22DF83C31@ESESSMB105.ericsson.se> <3879D71E758A7E4AA99A35DD8D41D3D91D5260A2@xmb-rcd-x14.cisco.com> <56C2F665D49E0341B9DF5938005ACDF80E8A65@DEMUMBX005.nsn-intra.net> <BBE9739C2C302046BD34B42713A1E2A22DF88232@ESESSMB105.ericsson.se> <CAGgHUiSK-bZrdXtxf-An8NM+pw-iqoWCrsG_bRUpxcD2DOCQrQ@mail.gmail.com>
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If you mean for the required codecs, that is already in the next sentence. =
Full context:

OLD:

To ensure a baseline level of interoperability between WebRTC
clients, a minimum set of required codecs are specified below.
While this section specifies the codecs that will be mandated for all
WebRTC client implementations, it leaves the question of supporting
additional codecs to the will of the implementer.

WebRTC clients are REQUIRED to implement the following audio codecs.

NEW:

To ensure a baseline level of interoperability between WebRTC
clients, a minimum set of required codecs are specified below.
If other suitable audio codecs are available to the browser to use,
it is RECOMMENDED that they are also included in the offer in order
to maximize the possibility to establish the session without the need
for audio transcoding.

WebRTC clients are REQUIRED to implement the following audio codecs.


From: Leon Geyser [mailto:lgeyser@gmail.com]
Sent: Thursday, September 05, 2013 10:06 AM
To: Bo Burman
Cc: Rauschenbach, Uwe (NSN - DE/Munich); Mo Zanaty (mzanaty); rtcweb@ietf.o=
rg
Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt

It would be nice if a word like mandated/mandatory is present in the new te=
xt like it was in the old text. Maybe REQUIRED in uppercase?

On 5 September 2013 15:05, Bo Burman <bo.burman@ericsson.com<mailto:bo.burm=
an@ericsson.com>> wrote:
I also support it.
/Bo

> -----Original Message-----
> From: Rauschenbach, Uwe (NSN - DE/Munich) [mailto:uwe.rauschenbach@nsn.co=
m<mailto:uwe.rauschenbach@nsn.com>]
> Sent: den 5 september 2013 12:14
> To: ext Mo Zanaty (mzanaty); Bo Burman; rtcweb@ietf.org<mailto:rtcweb@iet=
f.org>
> Subject: RE: I-D Action: draft-ietf-rtcweb-audio-02.txt
>
> I support this proposal.
>
> Kind regards,
> Uwe
>
>
> > -----Original Message-----
> > From: rtcweb-bounces@ietf.org<mailto:rtcweb-bounces@ietf.org> [mailto:r=
tcweb-bounces@ietf.org<mailto:rtcweb-bounces@ietf.org>] On
> > Behalf Of ext Mo Zanaty (mzanaty)
> > Sent: Wednesday, September 04, 2013 10:03 PM
> > To: Bo Burman; rtcweb@ietf.org<mailto:rtcweb@ietf.org>
> > Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
> >
> > I support the proposed text addition. But rather than adding at the
> > end of section 3, replace the beginning of section 3 as below.
> >
> > OLD:
> >    To ensure a baseline level of interoperability between WebRTC
> >    clients, a minimum set of required codecs are specified below.
> >    While this section specifies the codecs that will be mandated for
> > all
> >    WebRTC client implementations, it leaves the question of supporting
> >    additional codecs to the will of the implementer.
> >
> > NEW:
> >    To ensure a baseline level of interoperability between WebRTC
> >    clients, a minimum set of required codecs are specified below.
> >    If other suitable audio codecs are available to the browser to use,
> >    it is RECOMMENDED that they are also included in the offer in order
> >    to maximize the possibility to establish the session without the
> > need
> >    for audio transcoding.
> >
> > Mo
> >
> > -----Original Message-----
> > From: rtcweb-bounces@ietf.org<mailto:rtcweb-bounces@ietf.org> [mailto:r=
tcweb-bounces@ietf.org<mailto:rtcweb-bounces@ietf.org>] On
> > Behalf Of Bo Burman
> > Sent: Tuesday, August 27, 2013 10:53 AM
> > To: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
> > Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
> >
> > Based on the previous discussion thread http://www.ietf.org/mail-
> > archive/web/rtcweb/current/msg08501.html, I believe there is support
> > to add some text into this document.
> >
> > I thus propose to add the following text at the end of section 3:
> >
> > If other suitable audio codecs are available to the browser to use, it
> > is RECOMMENDED that they are also included in the offer in order to
> > maximize the possibility to establish the session without the need for
> > audio transcoding
> >
> > Cheers,
> > Bo
> >
> > > -----Original Message-----
> > > From: i-d-announce-bounces@ietf.org<mailto:i-d-announce-bounces@ietf.=
org> [mailto:i-d-announce-<mailto:i-d-announce->
> > bounces@ietf.org<mailto:bounces@ietf.org>] On Behalf Of internet-drafts=
@ietf.org<mailto:internet-drafts@ietf.org>
> > > Sent: den 2 augusti 2013 18:30
> > > To: i-d-announce@ietf.org<mailto:i-d-announce@ietf.org>
> > > Cc: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
> > > Subject: I-D Action: draft-ietf-rtcweb-audio-02.txt
> > >
> > >
> > > A New Internet-Draft is available from the on-line Internet-Drafts
> > directories.
> > >  This draft is a work item of the Real-Time Communication in WEB-
> > browsers Working Group of the IETF.
> > >
> > >   Title           : WebRTC Audio Codec and Processing Requirements
> > >   Author(s)       : Jean-Marc Valin
> > >                           Cary Bran
> > >   Filename        : draft-ietf-rtcweb-audio-02.txt
> > >   Pages           : 6
> > >   Date            : 2013-08-02
> > >
> > > Abstract:
> > >    This document outlines the audio codec and processing requirements
> > >    for WebRTC client application and endpoint devices.
> > >
> > >
> > > The IETF datatracker status page for this draft is:
> > > https://datatracker.ietf.org/doc/draft-ietf-rtcweb-audio
> > >
> > > There's also a htmlized version available at:
> > > http://tools.ietf.org/html/draft-ietf-rtcweb-audio-02
> > >
> > > A diff from the previous version is available at:
> > > http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-audio-02
> > >
> > >
> > > Please note that it may take a couple of minutes from the time of
> > submission until the htmlized version and diff are
> > > available at tools.ietf.org<http://tools.ietf.org>.
> > >
> > > Internet-Drafts are also available by anonymous FTP at:
> > > ftp://ftp.ietf.org/internet-drafts/
> > >
> > > _______________________________________________
> > > I-D-Announce mailing list
> > > I-D-Announce@ietf.org<mailto:I-D-Announce@ietf.org>
> > > https://www.ietf.org/mailman/listinfo/i-d-announce
> > > Internet-Draft directories: http://www.ietf.org/shadow.html or
> > ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org<mailto:rtcweb@ietf.org>
> > https://www.ietf.org/mailman/listinfo/rtcweb
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org<mailto:rtcweb@ietf.org>
> > https://www.ietf.org/mailman/listinfo/rtcweb
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org>
https://www.ietf.org/mailman/listinfo/rtcweb


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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">If you mean for the required codecs, th=
at is already in the next sentence. Full context:<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">OLD:<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">To ensure a baseline level of interoper=
ability between WebRTC<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">clients, a minimum set of required code=
cs are specified below.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">While this section specifies the codecs=
 that will be mandated for all<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">WebRTC client implementations, it leave=
s the question of supporting<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">additional codecs to the will of the im=
plementer.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">WebRTC clients are REQUIRED to implemen=
t the following audio codecs.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">NEW:<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">To ensure a baseline level of interoper=
ability between WebRTC<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">clients, a minimum set of required code=
cs are specified below.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">If other suitable audio codecs are avai=
lable to the browser to use,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">it is RECOMMENDED that they are also in=
cluded in the offer in order<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">to maximize the possibility to establis=
h the session without the need<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">for audio transcoding.<o:p></o:p></span=
></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">WebRTC clients are REQUIRED to implemen=
t the following audio codecs.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Leon Gey=
ser [mailto:lgeyser@gmail.com]
<br>
<b>Sent:</b> Thursday, September 05, 2013 10:06 AM<br>
<b>To:</b> Bo Burman<br>
<b>Cc:</b> Rauschenbach, Uwe (NSN - DE/Munich); Mo Zanaty (mzanaty); rtcweb=
@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt<o:p=
></o:p></span></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">It would be nice if a word like mandated/mandatory i=
s present in the new text like it was in the old text. Maybe REQUIRED in up=
percase?<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On 5 September 2013 15:05, Bo Burman &lt;<a href=3D"=
mailto:bo.burman@ericsson.com" target=3D"_blank">bo.burman@ericsson.com</a>=
&gt; wrote:<o:p></o:p></p>
<p class=3D"MsoNormal">I also support it.<br>
<span class=3D"hoenzb"><span style=3D"color:#888888">/Bo</span></span><o:p>=
</o:p></p>
<div>
<div>
<p class=3D"MsoNormal"><br>
&gt; -----Original Message-----<br>
&gt; From: Rauschenbach, Uwe (NSN - DE/Munich) [mailto:<a href=3D"mailto:uw=
e.rauschenbach@nsn.com">uwe.rauschenbach@nsn.com</a>]<br>
&gt; Sent: den 5 september 2013 12:14<br>
&gt; To: ext Mo Zanaty (mzanaty); Bo Burman; <a href=3D"mailto:rtcweb@ietf.=
org">rtcweb@ietf.org</a><br>
&gt; Subject: RE: I-D Action: draft-ietf-rtcweb-audio-02.txt<br>
&gt;<br>
&gt; I support this proposal.<br>
&gt;<br>
&gt; Kind regards,<br>
&gt; Uwe<br>
&gt;<br>
&gt;<br>
&gt; &gt; -----Original Message-----<br>
&gt; &gt; From: <a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@i=
etf.org</a> [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounc=
es@ietf.org</a>] On<br>
&gt; &gt; Behalf Of ext Mo Zanaty (mzanaty)<br>
&gt; &gt; Sent: Wednesday, September 04, 2013 10:03 PM<br>
&gt; &gt; To: Bo Burman; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org=
</a><br>
&gt; &gt; Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt<=
br>
&gt; &gt;<br>
&gt; &gt; I support the proposed text addition. But rather than adding at t=
he<br>
&gt; &gt; end of section 3, replace the beginning of section 3 as below.<br=
>
&gt; &gt;<br>
&gt; &gt; OLD:<br>
&gt; &gt; &nbsp; &nbsp;To ensure a baseline level of interoperability betwe=
en WebRTC<br>
&gt; &gt; &nbsp; &nbsp;clients, a minimum set of required codecs are specif=
ied below.<br>
&gt; &gt; &nbsp; &nbsp;While this section specifies the codecs that will be=
 mandated for<br>
&gt; &gt; all<br>
&gt; &gt; &nbsp; &nbsp;WebRTC client implementations, it leaves the questio=
n of supporting<br>
&gt; &gt; &nbsp; &nbsp;additional codecs to the will of the implementer.<br=
>
&gt; &gt;<br>
&gt; &gt; NEW:<br>
&gt; &gt; &nbsp; &nbsp;To ensure a baseline level of interoperability betwe=
en WebRTC<br>
&gt; &gt; &nbsp; &nbsp;clients, a minimum set of required codecs are specif=
ied below.<br>
&gt; &gt; &nbsp; &nbsp;If other suitable audio codecs are available to the =
browser to use,<br>
&gt; &gt; &nbsp; &nbsp;it is RECOMMENDED that they are also included in the=
 offer in order<br>
&gt; &gt; &nbsp; &nbsp;to maximize the possibility to establish the session=
 without the<br>
&gt; &gt; need<br>
&gt; &gt; &nbsp; &nbsp;for audio transcoding.<br>
&gt; &gt;<br>
&gt; &gt; Mo<br>
&gt; &gt;<br>
&gt; &gt; -----Original Message-----<br>
&gt; &gt; From: <a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@i=
etf.org</a> [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounc=
es@ietf.org</a>] On<br>
&gt; &gt; Behalf Of Bo Burman<br>
&gt; &gt; Sent: Tuesday, August 27, 2013 10:53 AM<br>
&gt; &gt; To: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; &gt; Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt<=
br>
&gt; &gt;<br>
&gt; &gt; Based on the previous discussion thread <a href=3D"http://www.iet=
f.org/mail-" target=3D"_blank">
http://www.ietf.org/mail-</a><br>
&gt; &gt; archive/web/rtcweb/current/msg08501.html, I believe there is supp=
ort<br>
&gt; &gt; to add some text into this document.<br>
&gt; &gt;<br>
&gt; &gt; I thus propose to add the following text at the end of section 3:=
<br>
&gt; &gt;<br>
&gt; &gt; If other suitable audio codecs are available to the browser to us=
e, it<br>
&gt; &gt; is RECOMMENDED that they are also included in the offer in order =
to<br>
&gt; &gt; maximize the possibility to establish the session without the nee=
d for<br>
&gt; &gt; audio transcoding<br>
&gt; &gt;<br>
&gt; &gt; Cheers,<br>
&gt; &gt; Bo<br>
&gt; &gt;<br>
&gt; &gt; &gt; -----Original Message-----<br>
&gt; &gt; &gt; From: <a href=3D"mailto:i-d-announce-bounces@ietf.org">i-d-a=
nnounce-bounces@ietf.org</a> [mailto:<a href=3D"mailto:i-d-announce-">i-d-a=
nnounce-</a><br>
&gt; &gt; <a href=3D"mailto:bounces@ietf.org">bounces@ietf.org</a>] On Beha=
lf Of <a href=3D"mailto:internet-drafts@ietf.org">
internet-drafts@ietf.org</a><br>
&gt; &gt; &gt; Sent: den 2 augusti 2013 18:30<br>
&gt; &gt; &gt; To: <a href=3D"mailto:i-d-announce@ietf.org">i-d-announce@ie=
tf.org</a><br>
&gt; &gt; &gt; Cc: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><b=
r>
&gt; &gt; &gt; Subject: I-D Action: draft-ietf-rtcweb-audio-02.txt<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; A New Internet-Draft is available from the on-line Internet-=
Drafts<br>
&gt; &gt; directories.<br>
&gt; &gt; &gt; &nbsp;This draft is a work item of the Real-Time Communicati=
on in WEB-<br>
&gt; &gt; browsers Working Group of the IETF.<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; &nbsp; Title &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; : WebRTC Aud=
io Codec and Processing Requirements<br>
&gt; &gt; &gt; &nbsp; Author(s) &nbsp; &nbsp; &nbsp; : Jean-Marc Valin<br>
&gt; &gt; &gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; Cary Bran<br>
&gt; &gt; &gt; &nbsp; Filename &nbsp; &nbsp; &nbsp; &nbsp;: draft-ietf-rtcw=
eb-audio-02.txt<br>
&gt; &gt; &gt; &nbsp; Pages &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; : 6<br>
&gt; &gt; &gt; &nbsp; Date &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;: 2013-=
08-02<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; Abstract:<br>
&gt; &gt; &gt; &nbsp; &nbsp;This document outlines the audio codec and proc=
essing requirements<br>
&gt; &gt; &gt; &nbsp; &nbsp;for WebRTC client application and endpoint devi=
ces.<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; The IETF datatracker status page for this draft is:<br>
&gt; &gt; &gt; <a href=3D"https://datatracker.ietf.org/doc/draft-ietf-rtcwe=
b-audio" target=3D"_blank">
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-audio</a><br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; There's also a htmlized version available at:<br>
&gt; &gt; &gt; <a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-audi=
o-02" target=3D"_blank">
http://tools.ietf.org/html/draft-ietf-rtcweb-audio-02</a><br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; A diff from the previous version is available at:<br>
&gt; &gt; &gt; <a href=3D"http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtc=
web-audio-02" target=3D"_blank">
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-audio-02</a><br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; Please note that it may take a couple of minutes from the ti=
me of<br>
&gt; &gt; submission until the htmlized version and diff are<br>
&gt; &gt; &gt; available at <a href=3D"http://tools.ietf.org" target=3D"_bl=
ank">tools.ietf.org</a>.<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; Internet-Drafts are also available by anonymous FTP at:<br>
&gt; &gt; &gt; <a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=3D"_b=
lank">ftp://ftp.ietf.org/internet-drafts/</a><br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; _______________________________________________<br>
&gt; &gt; &gt; I-D-Announce mailing list<br>
&gt; &gt; &gt; <a href=3D"mailto:I-D-Announce@ietf.org">I-D-Announce@ietf.o=
rg</a><br>
&gt; &gt; &gt; <a href=3D"https://www.ietf.org/mailman/listinfo/i-d-announc=
e" target=3D"_blank">
https://www.ietf.org/mailman/listinfo/i-d-announce</a><br>
&gt; &gt; &gt; Internet-Draft directories: <a href=3D"http://www.ietf.org/s=
hadow.html" target=3D"_blank">
http://www.ietf.org/shadow.html</a> or<br>
&gt; &gt; <a href=3D"ftp://ftp.ietf.org/ietf/1shadow-sites.txt" target=3D"_=
blank">ftp://ftp.ietf.org/ietf/1shadow-sites.txt</a><br>
&gt; &gt; _______________________________________________<br>
&gt; &gt; rtcweb mailing list<br>
&gt; &gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; &gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=
=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt; &gt; _______________________________________________<br>
&gt; &gt; rtcweb mailing list<br>
&gt; &gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; &gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=
=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></o:p></p>
</div>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
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From: "Lijing (Jessie, Huawei)" <lijing80@huawei.com>
To: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>, Bo Burman <bo.burman@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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+1

I also support this proposal.

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Mo Zanaty (mzanaty)
Sent: Thursday, September 05, 2013 4:03 AM
To: Bo Burman; rtcweb@ietf.org
Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt

I support the proposed text addition. But rather than adding at the end of =
section 3, replace the beginning of section 3 as below.

OLD:
   To ensure a baseline level of interoperability between WebRTC
   clients, a minimum set of required codecs are specified below.
   While this section specifies the codecs that will be mandated for all
   WebRTC client implementations, it leaves the question of supporting
   additional codecs to the will of the implementer.

NEW:
   To ensure a baseline level of interoperability between WebRTC
   clients, a minimum set of required codecs are specified below.
   If other suitable audio codecs are available to the browser to use,=20
   it is RECOMMENDED that they are also included in the offer in order=20
   to maximize the possibility to establish the session without the need=20
   for audio transcoding.

Mo

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Bo Burman
Sent: Tuesday, August 27, 2013 10:53 AM
To: rtcweb@ietf.org
Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt

Based on the previous discussion thread http://www.ietf.org/mail-archive/we=
b/rtcweb/current/msg08501.html, I believe there is support to add some text=
 into this document.

I thus propose to add the following text at the end of section 3:

If other suitable audio codecs are available to the browser to use,=20
it is RECOMMENDED that they are also included in the offer in order=20
to maximize the possibility to establish the session without the need=20
for audio transcoding

Cheers,
Bo

> -----Original Message-----
> From: i-d-announce-bounces@ietf.org [mailto:i-d-announce-bounces@ietf.org=
] On Behalf Of internet-drafts@ietf.org
> Sent: den 2 augusti 2013 18:30
> To: i-d-announce@ietf.org
> Cc: rtcweb@ietf.org
> Subject: I-D Action: draft-ietf-rtcweb-audio-02.txt
>=20
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts direct=
ories.
>  This draft is a work item of the Real-Time Communication in WEB-browsers=
 Working Group of the IETF.
>=20
> 	Title           : WebRTC Audio Codec and Processing Requirements
> 	Author(s)       : Jean-Marc Valin
>                           Cary Bran
> 	Filename        : draft-ietf-rtcweb-audio-02.txt
> 	Pages           : 6
> 	Date            : 2013-08-02
>=20
> Abstract:
>    This document outlines the audio codec and processing requirements
>    for WebRTC client application and endpoint devices.
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-audio
>=20
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-rtcweb-audio-02
>=20
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-audio-02
>=20
>=20
> Please note that it may take a couple of minutes from the time of submiss=
ion until the htmlized version and diff are
> available at tools.ietf.org.
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> _______________________________________________
> I-D-Announce mailing list
> I-D-Announce@ietf.org
> https://www.ietf.org/mailman/listinfo/i-d-announce
> Internet-Draft directories: http://www.ietf.org/shadow.html or ftp://ftp.=
ietf.org/ietf/1shadow-sites.txt
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

From andrew.hutton@siemens-enterprise.com  Fri Sep  6 09:59:24 2013
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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: "Lijing (Jessie, Huawei)" <lijing80@huawei.com>, "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>, Bo Burman <bo.burman@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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There was a discussion on this back in January and I then I thought we had =
converged on the following text which made a lower case type recommendation=
:

"If other suitable audio codecs are available to the browser to use it is r=
ecommended that they are also included in the offer in order to maximize th=
e possibility to establish the session without the need for audio transcodi=
ng"

See http://www.ietf.org/mail-archive/web/rtcweb/current/msg06121.html

Andy



> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Lijing (Jessie, Huawei)
> Sent: 06 September 2013 05:06
> To: Mo Zanaty (mzanaty); Bo Burman; rtcweb@ietf.org
> Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
>=20
> +1
>=20
> I also support this proposal.
>=20
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Mo Zanaty (mzanaty)
> Sent: Thursday, September 05, 2013 4:03 AM
> To: Bo Burman; rtcweb@ietf.org
> Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
>=20
> I support the proposed text addition. But rather than adding at the end
> of section 3, replace the beginning of section 3 as below.
>=20
> OLD:
>    To ensure a baseline level of interoperability between WebRTC
>    clients, a minimum set of required codecs are specified below.
>    While this section specifies the codecs that will be mandated for
> all
>    WebRTC client implementations, it leaves the question of supporting
>    additional codecs to the will of the implementer.
>=20
> NEW:
>    To ensure a baseline level of interoperability between WebRTC
>    clients, a minimum set of required codecs are specified below.
>    If other suitable audio codecs are available to the browser to use,
>    it is RECOMMENDED that they are also included in the offer in order
>    to maximize the possibility to establish the session without the
> need
>    for audio transcoding.
>=20
> Mo
>=20
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Bo Burman
> Sent: Tuesday, August 27, 2013 10:53 AM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
>=20
> Based on the previous discussion thread http://www.ietf.org/mail-
> archive/web/rtcweb/current/msg08501.html, I believe there is support to
> add some text into this document.
>=20
> I thus propose to add the following text at the end of section 3:
>=20
> If other suitable audio codecs are available to the browser to use,
> it is RECOMMENDED that they are also included in the offer in order
> to maximize the possibility to establish the session without the need
> for audio transcoding
>=20
> Cheers,
> Bo
>=20
> > -----Original Message-----
> > From: i-d-announce-bounces@ietf.org [mailto:i-d-announce-
> bounces@ietf.org] On Behalf Of internet-drafts@ietf.org
> > Sent: den 2 augusti 2013 18:30
> > To: i-d-announce@ietf.org
> > Cc: rtcweb@ietf.org
> > Subject: I-D Action: draft-ietf-rtcweb-audio-02.txt
> >
> >
> > A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> >  This draft is a work item of the Real-Time Communication in WEB-
> browsers Working Group of the IETF.
> >
> > 	Title           : WebRTC Audio Codec and Processing Requirements
> > 	Author(s)       : Jean-Marc Valin
> >                           Cary Bran
> > 	Filename        : draft-ietf-rtcweb-audio-02.txt
> > 	Pages           : 6
> > 	Date            : 2013-08-02
> >
> > Abstract:
> >    This document outlines the audio codec and processing requirements
> >    for WebRTC client application and endpoint devices.
> >
> >
> > The IETF datatracker status page for this draft is:
> > https://datatracker.ietf.org/doc/draft-ietf-rtcweb-audio
> >
> > There's also a htmlized version available at:
> > http://tools.ietf.org/html/draft-ietf-rtcweb-audio-02
> >
> > A diff from the previous version is available at:
> > http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-audio-02
> >
> >
> > Please note that it may take a couple of minutes from the time of
> submission until the htmlized version and diff are
> > available at tools.ietf.org.
> >
> > Internet-Drafts are also available by anonymous FTP at:
> > ftp://ftp.ietf.org/internet-drafts/
> >
> > _______________________________________________
> > I-D-Announce mailing list
> > I-D-Announce@ietf.org
> > https://www.ietf.org/mailman/listinfo/i-d-announce
> > Internet-Draft directories: http://www.ietf.org/shadow.html or
> ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From jmvalin@mozilla.com  Fri Sep  6 11:49:44 2013
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

There was a consensus call made a while ago and as far as I remember,
there was no WG consensus for any SHOULD/RECOMMENDED codecs.

Cheers,

	Jean-Marc

On 08/27/2013 10:53 AM, Bo Burman wrote:
> Based on the previous discussion thread 
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg08501.html,
> I believe there is support to add some text into this document.
> 
> I thus propose to add the following text at the end of section 3:
> 
> If other suitable audio codecs are available to the browser to use,
>  it is RECOMMENDED that they are also included in the offer in
> order to maximize the possibility to establish the session without
> the need for audio transcoding
> 
> Cheers, Bo
> 
>> -----Original Message----- From: i-d-announce-bounces@ietf.org 
>> [mailto:i-d-announce-bounces@ietf.org] On Behalf Of 
>> internet-drafts@ietf.org Sent: den 2 augusti 2013 18:30 To: 
>> i-d-announce@ietf.org Cc: rtcweb@ietf.org Subject: I-D Action: 
>> draft-ietf-rtcweb-audio-02.txt
>> 
>> 
>> A New Internet-Draft is available from the on-line
>> Internet-Drafts directories. This draft is a work item of the
>> Real-Time Communication in WEB-browsers Working Group of the
>> IETF.
>> 
>> Title           : WebRTC Audio Codec and Processing Requirements
>>  Author(s)       : Jean-Marc Valin Cary Bran Filename        : 
>> draft-ietf-rtcweb-audio-02.txt Pages           : 6 Date :
>> 2013-08-02
>> 
>> Abstract: This document outlines the audio codec and processing 
>> requirements for WebRTC client application and endpoint devices.
>> 
>> 
>> The IETF datatracker status page for this draft is: 
>> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-audio
>> 
>> There's also a htmlized version available at: 
>> http://tools.ietf.org/html/draft-ietf-rtcweb-audio-02
>> 
>> A diff from the previous version is available at: 
>> http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-audio-02
>> 
>> 
>> Please note that it may take a couple of minutes from the time
>> of submission until the htmlized version and diff are available
>> at tools.ietf.org.
>> 
>> Internet-Drafts are also available by anonymous FTP at: 
>> ftp://ftp.ietf.org/internet-drafts/
>> 
>> _______________________________________________ I-D-Announce 
>> mailing list I-D-Announce@ietf.org 
>> https://www.ietf.org/mailman/listinfo/i-d-announce
>> Internet-Draft directories: http://www.ietf.org/shadow.html or 
>> ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> _______________________________________________ rtcweb mailing list
>  rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
> 

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From mzanaty@cisco.com  Fri Sep  6 13:28:42 2013
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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: Jean-Marc Valin <jmvalin@mozilla.com>, Bo Burman <bo.burman@ericsson.com>
Thread-Topic: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
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Date: Fri, 6 Sep 2013 20:28:36 +0000
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
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Clearly no consensus for any more specific codecs. But the proposed text ju=
st says if you have more codecs available (not specific ones), you should o=
ffer them all, rather than suppress them and only offer the required codecs=
. So if the "will of the implementer" (current text) is neutral, this may s=
way them to offer more than required. If the will is hostile to other codec=
s, this text won't matter.

Mo

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Jean-Marc Valin
Sent: Friday, September 06, 2013 2:50 PM
To: Bo Burman
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

There was a consensus call made a while ago and as far as I remember,
there was no WG consensus for any SHOULD/RECOMMENDED codecs.

Cheers,

	Jean-Marc

On 08/27/2013 10:53 AM, Bo Burman wrote:
> Based on the previous discussion thread=20
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg08501.html,
> I believe there is support to add some text into this document.
>=20
> I thus propose to add the following text at the end of section 3:
>=20
> If other suitable audio codecs are available to the browser to use,
>  it is RECOMMENDED that they are also included in the offer in
> order to maximize the possibility to establish the session without
> the need for audio transcoding
>=20
> Cheers, Bo
>=20
>> -----Original Message----- From: i-d-announce-bounces@ietf.org=20
>> [mailto:i-d-announce-bounces@ietf.org] On Behalf Of=20
>> internet-drafts@ietf.org Sent: den 2 augusti 2013 18:30 To:=20
>> i-d-announce@ietf.org Cc: rtcweb@ietf.org Subject: I-D Action:=20
>> draft-ietf-rtcweb-audio-02.txt
>>=20
>>=20
>> A New Internet-Draft is available from the on-line
>> Internet-Drafts directories. This draft is a work item of the
>> Real-Time Communication in WEB-browsers Working Group of the
>> IETF.
>>=20
>> Title           : WebRTC Audio Codec and Processing Requirements
>>  Author(s)       : Jean-Marc Valin Cary Bran Filename        :=20
>> draft-ietf-rtcweb-audio-02.txt Pages           : 6 Date :
>> 2013-08-02
>>=20
>> Abstract: This document outlines the audio codec and processing=20
>> requirements for WebRTC client application and endpoint devices.
>>=20
>>=20
>> The IETF datatracker status page for this draft is:=20
>> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-audio
>>=20
>> There's also a htmlized version available at:=20
>> http://tools.ietf.org/html/draft-ietf-rtcweb-audio-02
>>=20
>> A diff from the previous version is available at:=20
>> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-audio-02
>>=20
>>=20
>> Please note that it may take a couple of minutes from the time
>> of submission until the htmlized version and diff are available
>> at tools.ietf.org.
>>=20
>> Internet-Drafts are also available by anonymous FTP at:=20
>> ftp://ftp.ietf.org/internet-drafts/
>>=20
>> _______________________________________________ I-D-Announce=20
>> mailing list I-D-Announce@ietf.org=20
>> https://www.ietf.org/mailman/listinfo/i-d-announce
>> Internet-Draft directories: http://www.ietf.org/shadow.html or=20
>> ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> _______________________________________________ rtcweb mailing list
>  rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
>=20

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_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

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On 6 September 2013 13:28, Mo Zanaty (mzanaty) <mzanaty@cisco.com> wrote:
> If the will is hostile to other codecs, this text won't matter.

The natural disposition of the implementer is hostility toward more
work, so I don't see the text making much difference either way.

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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
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On 6 September 2013 14:42, Mo Zanaty (mzanaty) <mzanaty@cisco.com> wrote:
> True. The same can be said about every SHOULD, but we still bother to specify it.

Most SHOULDs I've written are more like "MUST unless you have some
some good reasons that we haven't bothered to enumerate".  These serve
as an escape valve for people who want to break the rules in a
controlled environment.

That's interoperability requirements though, not implementation.  I
haven't ever done anything equivalent to MUST implement/SHOULD
implement, which is what we are talking about here.

From harald@alvestrand.no  Fri Sep  6 15:27:22 2013
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
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On 09/06/2013 11:42 PM, Mo Zanaty (mzanaty) wrote:
> True. The same can be said about every SHOULD, but we still bother to specify it.
> Maybe we should update RFC 2119 to state:
>
> Obey: "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT"
> Ignore: "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", "OPTIONAL"

At the time 2119 was written (I remember discussing this with Scott 
Bradner several times), the intent of the text for "SHOULD" was "you'd 
better do this if you don't have a real good reason why it's not a 
reasonable thing to do in your particular case".

Quoth:

3. SHOULD   This word, or the adjective "RECOMMENDED", mean that there
    may exist valid reasons in particular circumstances to ignore a
    particular item, but the full implications must be understood and
    carefully weighed before choosing a different course.

Of course, people who don't want to cover all the SHOULDs have pushed 
towards interpreting it as "weak recommendation that we can ignore if we 
feel like it", but that wasn't the original intent of 2119, and some of 
us still want to have that word available for use in the stronger meaning.

At one time there was push towards saying that "if you write SHOULD in 
an RFC, you need to spell out the circumstances where it'll be 
reasonable to ignore it - otherwise, write MUST or MAY". That push has 
petered out at this time, but I felt sympathetic to the idea.

That's why I'm so reluctant to use the word in cases where I think a 
large part of the implementors are going to ignore it, and where (in my 
opinion) no great harm comes to interoperability when they do.

But this instance not something I storm barricades over; if the 
consensus of the WG is to use "RECOMMENDED" rather than "recommended", 
I'll note that I'm the rough part of the consensus, and live with it.


>
> -----Original Message-----
> From: Martin Thomson [mailto:martin.thomson@gmail.com]
> Sent: Friday, September 06, 2013 4:38 PM
> To: Mo Zanaty (mzanaty)
> Cc: Jean-Marc Valin; Bo Burman; rtcweb@ietf.org
> Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
>
> On 6 September 2013 13:28, Mo Zanaty (mzanaty) <mzanaty@cisco.com> wrote:
>> If the will is hostile to other codecs, this text won't matter.
> The natural disposition of the implementer is hostility toward more
> work, so I don't see the text making much difference either way.
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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On 6 September 2013 15:27, Harald Alvestrand <harald@alvestrand.no> wrote:
> That's why I'm so reluctant to use the word in cases where I think a large
> part of the implementors are going to ignore it, and where (in my opinion)
> no great harm comes to interoperability when they do.

I tend to agree, though we have to be careful not to end up with the
RFC 6919, Section 1 problem at the same time.

I still try to do the "MUST, unless ..." form rather than SHOULD.
SHOULD is a bit wishy-washy.

In this case, I don't see the point of saying anything at all.  I'm
sure that browser implementers will add what they believe will be best
for their users.  And avoiding transcoding is good, which should be
sufficient.  Ultimately, we're all grown-ups, and having the IETF
tells us to eat our vegetables is insulting.

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Let's avoid reopening the whole discussion about this normative wording!=20

I would like to recall again that the proposal from Bo comes from a comprom=
ise statement that was almost reached in e-mail discussions last January (f=
rom an initial proposal from Andrew Allen) and almost reached now again
http://www.ietf.org/mail-archive/web/rtcweb/current/msg08501.html

So it confirms clearly that the consensus of the group is on the proposed t=
ext and especially because it does not include any formal normative languag=
e.=20
So I would strongly suggest to not reinvent anything new and close now this=
 long discussion on the last proposal from Bo on which several supports hav=
e been expressed and no objections, possibly with the slight modification s=
uggested by Mo Zanaty on the place where the text can be added (beginning o=
f section 3) which is a good idea that I support as well
http://www.ietf.org/mail-archive/web/rtcweb/current/msg08750.html

St=E9phane






-----Message d'origine-----
De=A0: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] De la part =
de Martin Thomson
Envoy=E9=A0: samedi 7 septembre 2013 00:40
=C0=A0: Harald Alvestrand
Cc=A0: rtcweb@ietf.org
Objet=A0: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt

On 6 September 2013 15:27, Harald Alvestrand <harald@alvestrand.no> wrote:
> That's why I'm so reluctant to use the word in cases where I think a=20
> large part of the implementors are going to ignore it, and where (in=20
> my opinion) no great harm comes to interoperability when they do.

I tend to agree, though we have to be careful not to end up with the RFC 69=
19, Section 1 problem at the same time.

I still try to do the "MUST, unless ..." form rather than SHOULD.
SHOULD is a bit wishy-washy.

In this case, I don't see the point of saying anything at all.  I'm sure th=
at browser implementers will add what they believe will be best for their u=
sers.  And avoiding transcoding is good, which should be sufficient.  Ultim=
ately, we're all grown-ups, and having the IETF tells us to eat our vegetab=
les is insulting.
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

___________________________________________________________________________=
______________________________________________

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Thank you.


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Subject: [rtcweb] Adopting draft-muthu-behave-consent-freshness?
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WG,

This is a call for WG adoption of STUN Usage for Consent Freshness
(draft-muthu-behave-consent-freshness-04). This document defines a STUN
usage for consent freshness. As this requires no protocol extensions we
as intended users can define this usage in our WG. Such work also
matches our charter. The draft-ietf-rtcweb-security-arch-07 is
normatively dependent on this STUN usage.

Document:
https://datatracker.ietf.org/doc/draft-muthu-behave-consent-freshness/

WG, please indicate your support or issues with adopting this document
as WG item with a proposed milestone:

Mar 2014 Send STUN Usage for Consent Freshness to IESG for publication
as proposed standard.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From harald@alvestrand.no  Mon Sep  9 01:47:39 2013
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From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
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This last round was triggered by Leon Geyser suggesting that we put 
REQUIRED in there.

I'm happy with the proposal made in January (lowercase "recommended"), 
weakly opposed to SHOULD / RECOMMENDED and strongly opposed to MUST / 
REQUIRED. I'm perfectly willing to go along with a chairs' declaration 
that the WG has consensus for SHOULD / RECOMMENDED, but it's up to the 
chairs to make the call.

I'll try to remain silent on the issue until the chairs have spoken.


On 09/09/2013 08:47 AM, stephane.proust@orange.com wrote:
> Let's avoid reopening the whole discussion about this normative wording!
>
> I would like to recall again that the proposal from Bo comes from a compromise statement that was almost reached in e-mail discussions last January (from an initial proposal from Andrew Allen) and almost reached now again
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg08501.html
>
> So it confirms clearly that the consensus of the group is on the proposed text and especially because it does not include any formal normative language.
> So I would strongly suggest to not reinvent anything new and close now this long discussion on the last proposal from Bo on which several supports have been expressed and no objections, possibly with the slight modification suggested by Mo Zanaty on the place where the text can be added (beginning of section 3) which is a good idea that I support as well
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg08750.html
>
> Stéphane
>
>
>
>
>
>
> -----Message d'origine-----
> De : rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] De la part de Martin Thomson
> Envoyé : samedi 7 septembre 2013 00:40
> À : Harald Alvestrand
> Cc : rtcweb@ietf.org
> Objet : Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
>
> On 6 September 2013 15:27, Harald Alvestrand <harald@alvestrand.no> wrote:
>> That's why I'm so reluctant to use the word in cases where I think a
>> large part of the implementors are going to ignore it, and where (in
>> my opinion) no great harm comes to interoperability when they do.
> I tend to agree, though we have to be careful not to end up with the RFC 6919, Section 1 problem at the same time.
>
> I still try to do the "MUST, unless ..." form rather than SHOULD.
> SHOULD is a bit wishy-washy.
>
> In this case, I don't see the point of saying anything at all.  I'm sure that browser implementers will add what they believe will be best for their users.  And avoiding transcoding is good, which should be sufficient.  Ultimately, we're all grown-ups, and having the IETF tells us to eat our vegetables is insulting.
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
> _________________________________________________________________________________________________________________________
>
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From simon.perreault@viagenie.ca  Mon Sep  9 02:24:45 2013
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Subject: Re: [rtcweb] Adopting draft-muthu-behave-consent-freshness?
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Le 2013-09-09 10:36, Magnus Westerlund a écrit :
> This is a call for WG adoption of STUN Usage for Consent Freshness
> (draft-muthu-behave-consent-freshness-04). This document defines a STUN
> usage for consent freshness. As this requires no protocol extensions we
> as intended users can define this usage in our WG. Such work also
> matches our charter. The draft-ietf-rtcweb-security-arch-07 is
> normatively dependent on this STUN usage.
> 
> Document:
> https://datatracker.ietf.org/doc/draft-muthu-behave-consent-freshness/
> 
> WG, please indicate your support or issues with adopting this document
> as WG item with a proposed milestone:
> 
> Mar 2014 Send STUN Usage for Consent Freshness to IESG for publication
> as proposed standard.

I have commented multiple times on this doc. I support its adoption now.

Simon

From harald@alvestrand.no  Mon Sep  9 03:00:47 2013
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Subject: Re: [rtcweb] Adopting draft-muthu-behave-consent-freshness?
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I support adoption as a working group document.

On 09/09/2013 10:36 AM, Magnus Westerlund wrote:
> WG,
>
> This is a call for WG adoption of STUN Usage for Consent Freshness
> (draft-muthu-behave-consent-freshness-04). This document defines a STUN
> usage for consent freshness. As this requires no protocol extensions we
> as intended users can define this usage in our WG. Such work also
> matches our charter. The draft-ietf-rtcweb-security-arch-07 is
> normatively dependent on this STUN usage.
>
> Document:
> https://datatracker.ietf.org/doc/draft-muthu-behave-consent-freshness/
>
> WG, please indicate your support or issues with adopting this document
> as WG item with a proposed milestone:
>
> Mar 2014 Send STUN Usage for Consent Freshness to IESG for publication
> as proposed standard.
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> Färögatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From christer.holmberg@ericsson.com  Mon Sep  9 05:22:29 2013
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To: Magnus Westerlund <magnus.westerlund@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: Re: [rtcweb] Adopting draft-muthu-behave-consent-freshness?
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Hi,

I don't object adoption, but a question for clarification.

The draft says:

   "While a WebRTC browser could verify whether the peer continues to
   send SRTCP reports before sending traffic to the peer, the usage of
   SRTCP together with Security Descriptions [RFC4568] requires exposing
   the media keys to the JavaScript and renders SRTCP unsuitable for
   consent freshness."

Now, as we have decided to not use SDES, I guess that can be removed.

But, based on that, I'd just like to verify whether there is still a need f=
or the draft :)

Regards,

Christer



-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Magnus Westerlund
Sent: 9. syyskuuta 2013 11:37
To: rtcweb@ietf.org
Subject: [rtcweb] Adopting draft-muthu-behave-consent-freshness?

WG,

This is a call for WG adoption of STUN Usage for Consent Freshness (draft-m=
uthu-behave-consent-freshness-04). This document defines a STUN usage for c=
onsent freshness. As this requires no protocol extensions we as intended us=
ers can define this usage in our WG. Such work also matches our charter. Th=
e draft-ietf-rtcweb-security-arch-07 is normatively dependent on this STUN =
usage.

Document:
https://datatracker.ietf.org/doc/draft-muthu-behave-consent-freshness/

WG, please indicate your support or issues with adopting this document as W=
G item with a proposed milestone:

Mar 2014 Send STUN Usage for Consent Freshness to IESG for publication as p=
roposed standard.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
F=E4r=F6gatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------

_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

From harald@alvestrand.no  Mon Sep  9 05:29:51 2013
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Subject: [rtcweb] Consent freshness in the light of no-SDES (Re: Adopting draft-muthu-behave-consent-freshness?)
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Changing the subject line to make life simpler for those who count the 
respondents :-)

On 09/09/2013 02:20 PM, Christer Holmberg wrote:
> Hi,
>
> I don't object adoption, but a question for clarification.
>
> The draft says:
>
>     "While a WebRTC browser could verify whether the peer continues to
>     send SRTCP reports before sending traffic to the peer, the usage of
>     SRTCP together with Security Descriptions [RFC4568] requires exposing
>     the media keys to the JavaScript and renders SRTCP unsuitable for
>     consent freshness."
>
> Now, as we have decided to not use SDES, I guess that can be removed.
>
> But, based on that, I'd just like to verify whether there is still a need for the draft :)
>

Personal opinion: I think SRTCP is still unsuitable for consent 
freshness. One reason (apart from any aspect that deals with security, 
on which I don't want to comment) is that if one sets up a connection 
with only a data channel, there will be no SSRCs to send SRTCP on.




From christer.holmberg@ericsson.com  Mon Sep  9 05:46:55 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Consent freshness in the light of no-SDES (Re: Adopting draft-muthu-behave-consent-freshness?)
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Subject: Re: [rtcweb] Consent freshness in the light of no-SDES (Re: Adopting draft-muthu-behave-consent-freshness?)
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Hi,

And, from a security perspective (the backward compatibility is a separate =
issue) the requirement to integrity protect the STUN binding requests are n=
ot affected by the SDES decision?

Regards,

Christer

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Harald Alvestrand
Sent: 9. syyskuuta 2013 15:28
To: rtcweb@ietf.org
Subject: [rtcweb] Consent freshness in the light of no-SDES (Re: Adopting d=
raft-muthu-behave-consent-freshness?)

Changing the subject line to make life simpler for those who count the resp=
ondents :-)

On 09/09/2013 02:20 PM, Christer Holmberg wrote:
> Hi,
>
> I don't object adoption, but a question for clarification.
>
> The draft says:
>
>     "While a WebRTC browser could verify whether the peer continues to
>     send SRTCP reports before sending traffic to the peer, the usage of
>     SRTCP together with Security Descriptions [RFC4568] requires exposing
>     the media keys to the JavaScript and renders SRTCP unsuitable for
>     consent freshness."
>
> Now, as we have decided to not use SDES, I guess that can be removed.
>
> But, based on that, I'd just like to verify whether there is still a=20
> need for the draft :)
>

Personal opinion: I think SRTCP is still unsuitable for consent freshness. =
One reason (apart from any aspect that deals with security, on which I don'=
t want to comment) is that if one sets up a connection with only a data cha=
nnel, there will be no SSRCs to send SRTCP on.



_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

From ekr@rtfm.com  Mon Sep  9 06:08:40 2013
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
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On Mon, Sep 9, 2013 at 1:47 AM, Harald Alvestrand <harald@alvestrand.no>wro=
te:

> This last round was triggered by Leon Geyser suggesting that we put
> REQUIRED in there.
>
> I'm happy with the proposal made in January (lowercase "recommended"),
> weakly opposed to SHOULD / RECOMMENDED and strongly opposed to MUST /
> REQUIRED. I'm perfectly willing to go along with a chairs' declaration th=
at
> the WG has consensus for SHOULD / RECOMMENDED, but it's up to the chairs =
to
> make the call.
>
> I'll try to remain silent on the issue until the chairs have spoken.


I would like to hear from the chairs as well. It was not my impression that
we had consensus on any kind of recommendation here but rather that
those against a recommendation had merely grown tired of arguing.

I would suggest that we schedule a small amount of discussion time
at the next IETF to resolve this (20 min should be sufficient) and verify
consensus. However, I will defer to the chairs' judgement.

-Ekr


>
> On 09/09/2013 08:47 AM, stephane.proust@orange.com wrote:
>
>> Let's avoid reopening the whole discussion about this normative wording!
>>
>> I would like to recall again that the proposal from Bo comes from a
>> compromise statement that was almost reached in e-mail discussions last
>> January (from an initial proposal from Andrew Allen) and almost reached =
now
>> again
>> http://www.ietf.org/mail-**archive/web/rtcweb/current/**msg08501.html<ht=
tp://www.ietf.org/mail-archive/web/rtcweb/current/msg08501.html>
>>
>> So it confirms clearly that the consensus of the group is on the propose=
d
>> text and especially because it does not include any formal normative
>> language.
>> So I would strongly suggest to not reinvent anything new and close now
>> this long discussion on the last proposal from Bo on which several suppo=
rts
>> have been expressed and no objections, possibly with the slight
>> modification suggested by Mo Zanaty on the place where the text can be
>> added (beginning of section 3) which is a good idea that I support as we=
ll
>> http://www.ietf.org/mail-**archive/web/rtcweb/current/**msg08750.html<ht=
tp://www.ietf.org/mail-archive/web/rtcweb/current/msg08750.html>
>>
>> St=E9phane
>>
>>
>>
>>
>>
>>
>> -----Message d'origine-----
>> De : rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.**org<rtcweb-bo=
unces@ietf.org>]
>> De la part de Martin Thomson
>> Envoy=E9 : samedi 7 septembre 2013 00:40
>> =C0 : Harald Alvestrand
>> Cc : rtcweb@ietf.org
>> Objet : Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
>>
>> On 6 September 2013 15:27, Harald Alvestrand <harald@alvestrand.no>
>> wrote:
>>
>>> That's why I'm so reluctant to use the word in cases where I think a
>>> large part of the implementors are going to ignore it, and where (in
>>> my opinion) no great harm comes to interoperability when they do.
>>>
>> I tend to agree, though we have to be careful not to end up with the RFC
>> 6919, Section 1 problem at the same time.
>>
>> I still try to do the "MUST, unless ..." form rather than SHOULD.
>> SHOULD is a bit wishy-washy.
>>
>> In this case, I don't see the point of saying anything at all.  I'm sure
>> that browser implementers will add what they believe will be best for th=
eir
>> users.  And avoiding transcoding is good, which should be sufficient.
>>  Ultimately, we're all grown-ups, and having the IETF tells us to eat ou=
r
>> vegetables is insulting.
>> ______________________________**_________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/mail=
man/listinfo/rtcweb>
>>
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>>
> ______________________________**_________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/mailm=
an/listinfo/rtcweb>
>

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<div dir=3D"ltr">On Mon, Sep 9, 2013 at 1:47 AM, Harald Alvestrand <span di=
r=3D"ltr">&lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">har=
ald@alvestrand.no</a>&gt;</span> wrote:<br><div class=3D"gmail_extra"><div =
class=3D"gmail_quote">

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">This last round was triggered by Leon Geyser=
 suggesting that we put REQUIRED in there.<br>
<br>
I&#39;m happy with the proposal made in January (lowercase &quot;recommende=
d&quot;), weakly opposed to SHOULD / RECOMMENDED and strongly opposed to MU=
ST / REQUIRED. I&#39;m perfectly willing to go along with a chairs&#39; dec=
laration that the WG has consensus for SHOULD / RECOMMENDED, but it&#39;s u=
p to the chairs to make the call.<br>


<br>
I&#39;ll try to remain silent on the issue until the chairs have spoken.</b=
lockquote><div><br></div><div>I would like to hear from the chairs as well.=
 It was not my impression that</div><div>we had consensus on any kind of re=
commendation here but rather that</div>

<div>those against a recommendation had merely grown tired of arguing.</div=
><div><br></div><div>I would suggest that we schedule a small amount of dis=
cussion time</div><div>at the next IETF to resolve this (20 min should be s=
ufficient) and verify</div>

<div>consensus. However, I will defer to the chairs&#39; judgement.</div><d=
iv><br></div><div>-Ekr</div><div><br></div><blockquote class=3D"gmail_quote=
" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div class=3D"HOEnZb">
<div class=3D"h5">
<br>
<br>
On 09/09/2013 08:47 AM, <a href=3D"mailto:stephane.proust@orange.com" targe=
t=3D"_blank">stephane.proust@orange.com</a> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Let&#39;s avoid reopening the whole discussion about this normative wording=
!<br>
<br>
I would like to recall again that the proposal from Bo comes from a comprom=
ise statement that was almost reached in e-mail discussions last January (f=
rom an initial proposal from Andrew Allen) and almost reached now again<br>


<a href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg08501.htm=
l" target=3D"_blank">http://www.ietf.org/mail-<u></u>archive/web/rtcweb/cur=
rent/<u></u>msg08501.html</a><br>
<br>
So it confirms clearly that the consensus of the group is on the proposed t=
ext and especially because it does not include any formal normative languag=
e.<br>
So I would strongly suggest to not reinvent anything new and close now this=
 long discussion on the last proposal from Bo on which several supports hav=
e been expressed and no objections, possibly with the slight modification s=
uggested by Mo Zanaty on the place where the text can be added (beginning o=
f section 3) which is a good idea that I support as well<br>


<a href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg08750.htm=
l" target=3D"_blank">http://www.ietf.org/mail-<u></u>archive/web/rtcweb/cur=
rent/<u></u>msg08750.html</a><br>
<br>
St=E9phane<br>
<br>
<br>
<br>
<br>
<br>
<br>
-----Message d&#39;origine-----<br>
De : <a href=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_blank">rtcweb-bo=
unces@ietf.org</a> [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org" targe=
t=3D"_blank">rtcweb-bounces@ietf.<u></u>org</a>] De la part de Martin Thoms=
on<br>


Envoy=E9 : samedi 7 septembre 2013 00:40<br>
=C0 : Harald Alvestrand<br>
Cc : <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</=
a><br>
Objet : Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt<br>
<br>
On 6 September 2013 15:27, Harald Alvestrand &lt;<a href=3D"mailto:harald@a=
lvestrand.no" target=3D"_blank">harald@alvestrand.no</a>&gt; wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
That&#39;s why I&#39;m so reluctant to use the word in cases where I think =
a<br>
large part of the implementors are going to ignore it, and where (in<br>
my opinion) no great harm comes to interoperability when they do.<br>
</blockquote>
I tend to agree, though we have to be careful not to end up with the RFC 69=
19, Section 1 problem at the same time.<br>
<br>
I still try to do the &quot;MUST, unless ...&quot; form rather than SHOULD.=
<br>
SHOULD is a bit wishy-washy.<br>
<br>
In this case, I don&#39;t see the point of saying anything at all. =A0I&#39=
;m sure that browser implementers will add what they believe will be best f=
or their users. =A0And avoiding transcoding is good, which should be suffic=
ient. =A0Ultimately, we&#39;re all grown-ups, and having the IETF tells us =
to eat our vegetables is insulting.<br>


______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
<br>
______________________________<u></u>______________________________<u></u>_=
_____________________________<u></u>______________________________<u></u>_<=
br>
<br>
Ce message et ses pieces jointes peuvent contenir des informations confiden=
tielles ou privilegiees et ne doivent donc<br>
pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu=
 ce message par erreur, veuillez le signaler<br>
a l&#39;expediteur et le detruire ainsi que les pieces jointes. Les message=
s electroniques etant susceptibles d&#39;alteration,<br>
Orange decline toute responsabilite si ce message a ete altere, deforme ou =
falsifie. Merci.<br>
<br>
This message and its attachments may contain confidential or privileged inf=
ormation that may be protected by law;<br>
they should not be distributed, used or copied without authorisation.<br>
If you have received this email in error, please notify the sender and dele=
te this message and its attachments.<br>
As emails may be altered, Orange is not liable for messages that have been =
modified, changed or falsified.<br>
Thank you.<br>
<br>
</blockquote>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div></div>

--001a1133dd4cef56f304e5f30914--

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Date: Mon, 09 Sep 2013 15:38:36 +0200
From: "Alfred E. Heggestad" <aeh@db.org>
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Subject: Re: [rtcweb] Adopting draft-muthu-behave-consent-freshness?
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On 09/09/13 10:36, Magnus Westerlund wrote:
> WG,
>
> This is a call for WG adoption of STUN Usage for Consent Freshness
> (draft-muthu-behave-consent-freshness-04). This document defines a STUN
> usage for consent freshness. As this requires no protocol extensions we
> as intended users can define this usage in our WG. Such work also
> matches our charter. The draft-ietf-rtcweb-security-arch-07 is
> normatively dependent on this STUN usage.
>
> Document:
> https://datatracker.ietf.org/doc/draft-muthu-behave-consent-freshness/
>
> WG, please indicate your support or issues with adopting this document
> as WG item with a proposed milestone:
>

I support adopting this document.



/alfred

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+1
Harald said:=20
> I support adoption as a working group document.

 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'><div>+1</div><div><br></div><div=
>Harald said:&nbsp=3B</div><div><br>&gt=3B I support adoption as a working =
group document.<br><br></div> 		 	   		  </div></body>
</html>=

--_c54d25a5-1f1c-41c6-be7b-3bb8e7145466_--

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=0A=
=0A=
=0A=

Christer said:=20
"Now=2C as we have decided to not use SDES=2C I guess that can be removed."
[BA] Actually=2C I believe that a sentence or two describing why the needs =
of consent freshness are not met by mechanisms such as SRTCP is worth keepi=
ng=2C and possibly should even be expanded upon.  One reason is that we als=
o have the "circuit breakers" mechanism which *does* rely (in part) on SRTC=
P.  =20
Christer said:=20
"But=2C based on that=2C I'd just like to verify whether there is still a n=
eed for the draft :)"
[BA] I do think that the draft is still necessary=2C even without SDES/SRTP=
.   The usage of ICE for consent seems to me to be less susceptible to a va=
riety of issues than a dependency on SRTCP might be.=20
However=2C I am concerned about the interaction between circuit breakers an=
d consent freshness.  At IETF 87=2C we heard that the circuit breakers mech=
anism could fire even when the overall loss rate was quite low=2C due to bu=
rst losses.  That shouldn't be true of consent=2C since loss of consent req=
uires up to 30 consecutive losses over a 15 second period.  However=2C it w=
ould appear to me that the current circuit breakers algorithms will cause m=
edia sending to be curtailed prior to loss of consent in many cases. =20

=0A=
 		 	   		  =

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</head>
<body class=3D'hmmessage'><div dir=3D'ltr'>=0A=
=0A=
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--></style>=0A=
<div dir=3D"ltr"><br><span style=3D"font-size: 12pt=3B ">Christer said:&nbs=
p=3B</span><div><div><br></div><div>"Now=2C as we have decided to not use S=
DES=2C I guess that can be removed."</div><div><br></div><div>[BA] Actually=
=2C I believe that a sentence or two describing why the needs of consent fr=
eshness are not met by mechanisms such as SRTCP is worth keeping=2C and pos=
sibly should even be expanded upon. &nbsp=3BOne reason is that we also have=
 the "circuit breakers" mechanism which *does* rely (in part) on SRTCP. &nb=
sp=3B&nbsp=3B</div><div><br>Christer said:&nbsp=3B</div><div><br></div><div=
>"But=2C based on that=2C I'd just like to verify whether there is still a =
need for the draft :)"</div><div><br></div><div>[BA] I do think that the dr=
aft is still necessary=2C even without SDES/SRTP. &nbsp=3B The usage of ICE=
 for consent seems to me to be less susceptible to a variety of issues than=
 a dependency on SRTCP might be.&nbsp=3B</div><div><br></div><div>However=
=2C I am concerned about the interaction between circuit breakers and conse=
nt freshness. &nbsp=3BAt IETF 87=2C we heard that the circuit breakers mech=
anism could fire even when the overall loss rate was quite low=2C due to bu=
rst losses. &nbsp=3BThat shouldn't be true of consent=2C since loss of cons=
ent requires up to 30 consecutive losses over a 15 second period. &nbsp=3BH=
owever=2C it would appear to me that the current circuit breakers algorithm=
s will cause media sending to be curtailed prior to loss of consent in many=
 cases. &nbsp=3B<br><br></div></div></div>=0A=
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--_d5d4e94b-4032-4052-9dc2-5ecf8cb4c1c5_--

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Subject: Re: [rtcweb] draft-ietf-rtcweb-transports-01 TURN/IPV6 RFC 6156.
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Le 2013-09-09 18:36, Hutton, Andrew a écrit :
> Currently draft-ietf-rtcweb-transports-01 does not say anything about IPV6 which I assume it should. Specifically I am thinking that it needs to state a requirement to support RFC6156 support "Traversal Using Relays around NAT (TURN) Extension for IPv6".

+1

> I am not sure how much we need to say about webrtc client procedures around RFC6156 and whether they should be included in the draft-ietf-rtcweb-transport or whether it is something we should add to our nat/firewall draft (draft-hutton-rtcweb-nat-firewall-considerations).  Any opinions on this?

Something along the lines of "IPv6 allocations MUST be requested even
when the client only has IPv4 connectivity." ?

Simon

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Subject: Re: [rtcweb] draft-ietf-rtcweb-transports-01 TURN/IPV6 RFC 6156.
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On Mon, Sep 9, 2013 at 9:36 AM, Hutton, Andrew
<andrew.hutton@siemens-enterprise.com> wrote:
> Currently draft-ietf-rtcweb-transports-01 does not say anything about IPV=
6 which I assume it should. Specifically I am thinking that it needs to sta=
te a requirement to support RFC6156 support "Traversal Using Relays around =
NAT (TURN) Extension for IPv6".
>
> I am not sure how much we need to say about webrtc client procedures arou=
nd RFC6156 and whether they should be included in the draft-ietf-rtcweb-tra=
nsport or whether it is something we should add to our nat/firewall draft (=
draft-hutton-rtcweb-nat-firewall-considerations).  Any opinions on this?
>

+1 on covering IPv6 in the transport draft.

I would specifically state that much of the challenges with NATs are
resolved by preferring IPv6.

I would also cover the use case of using TURN to bridge IPv6-only
clients to communicate with IPv4 clients.

It would also be wise to reference RFC6540 to underscore that IPv6 use
is a BCP of the IETF

CB


> Regards
> Andy
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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--_72251298-e3a3-42af-90dd-d4acfa7eeea6_
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> I would specifically state that much of the challenges with NATs are
> resolved by preferring IPv6.
 [BA] That isn't what we've seen in practice=2C since IPv6 addresses (parti=
cularly tunnel addresses) may not be routable. So in practice=2C things see=
m to work better if IPv4 addresses are preferred and in fact=2C government =
profiles such as DISA UCR 2008 require the ability to configure an IPv4 pre=
ference. =20
> I would also cover the use case of using TURN to bridge IPv6-only
> clients to communicate with IPv4 clients.
[BA] Yes=2C this is an important use case (and one that some ICE implementa=
tions don't support very well). =20
> It would also be wise to reference RFC6540 to underscore that IPv6 use
> is a BCP of the IETF.
 		 	   		  =

--_72251298-e3a3-42af-90dd-d4acfa7eeea6_
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<html>
<head>
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{
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<body class=3D'hmmessage'><div dir=3D'ltr'><div>&gt=3B I would specifically=
 state that much of the challenges with NATs are<br>&gt=3B resolved by pref=
erring IPv6.<br>&nbsp=3B</div><div>[BA] That isn't what we've seen in pract=
ice=2C since IPv6 addresses (particularly tunnel addresses) may not be rout=
able.&nbsp=3B</div><div>So in practice=2C things seem to work better if IPv=
4 addresses are preferred and in fact=2C government profiles such as DISA U=
CR 2008 require the ability to configure an IPv4 preference. &nbsp=3B</div>=
<div><br>&gt=3B I would also cover the use case of using TURN to bridge IPv=
6-only<br>&gt=3B clients to communicate with IPv4 clients.</div><div><br></=
div><div>[BA] Yes=2C this is an important use case (and one that some ICE i=
mplementations don't support very well). &nbsp=3B</div><div><br>&gt=3B It w=
ould also be wise to reference RFC6540 to underscore that IPv6 use<br>&gt=
=3B is a BCP of the IETF.<br></div> 		 	   		  </div></body>
</html>=

--_72251298-e3a3-42af-90dd-d4acfa7eeea6_--

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Subject: Re: [rtcweb] draft-ietf-rtcweb-transports-01 TURN/IPV6 RFC 6156.
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On Mon, Sep 9, 2013 at 10:01 AM, Bernard Aboba
<bernard_aboba@hotmail.com> wrote:
>> I would specifically state that much of the challenges with NATs are
>> resolved by preferring IPv6.
>
> [BA] That isn't what we've seen in practice, since IPv6 addresses
> (particularly tunnel addresses) may not be routable.
> So in practice, things seem to work better if IPv4 addresses are preferred
> and in fact, government profiles such as DISA UCR 2008 require the ability
> to configure an IPv4 preference.
>

Is there a citation for this data point about IPv6 not working as well
in production?  I see the DISA thing as not particularly relevant to
the Internet.  That's one way to run  a private network, with knobs to
adjust.

There used to be an issue with 6to4 and maybe Teredo, but i believe
the data shows that this issue has largely been resolved
http://www.google.com/intl/en/ipv6/statistics.html via host treatment
of those tunnels as well as happy-eyeballs in browsers.

I believe happy-eyeballs / RFC6555 would apply to WebRTC interactions
equally well, since the signalling is over the standard HTTP/HTTPS
calls.

Maybe RFC6555 needs to be mentioned in the transport draft as well
since it can be leveraged to mitigate the use of broken IPv6
connections.

CB


>> I would also cover the use case of using TURN to bridge IPv6-only
>> clients to communicate with IPv4 clients.
>
> [BA] Yes, this is an important use case (and one that some ICE
> implementations don't support very well).
>
>> It would also be wise to reference RFC6540 to underscore that IPv6 use
>> is a BCP of the IETF.

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Subject: Re: [rtcweb] draft-ietf-rtcweb-transports-01 TURN/IPV6 RFC 6156.
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--_d80e08cb-6865-4bd2-92ad-a569bcc1bbf1_
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> Is there a citation for this data point about IPv6 not working as well in=
 production?=20
[BA] At IETF 87 (and in several meetings prior to that)=2C we have had a di=
scussion within the MMUSIC WG about "Happy Eyeballs for ICE" (draft-reddy-m=
music-ice-happy-eyeballs):http://www.ietf.org/proceedings/87/slides/slides-=
87-mmusic-0.pdf
> There used to be an issue with 6to4 and maybe Teredo=2C but i believe
> the data shows that this issue has largely been resolved
> http://www.google.com/intl/en/ipv6/statistics.html via host treatment
> of those tunnels as well as happy-eyeballs in browsers.
[BA] Those statistics do not relate to ICEv6 usage=2C and in particular=2C =
they don't cover the range of behavior that we see in enterprise and carrie=
r networks.  As an example=2C in deployments we have encountered IPv6 NATs =
and firewalls with a wide range of behavior=2C IPv6-only carrier networks w=
here connectivity to IPv4 endpoints is needed=2C etc. =20
> I believe happy-eyeballs / RFC6555 would apply to WebRTC interactions
> equally well=2C since the signalling is over the standard HTTP/HTTPS call=
s.
[BA] RFC 6555 doesn't directly apply to ICE=2C which is why we've been havi=
ng a distinct "ICE Happy Eyeballs" discussion.=20
 		 	   		  =

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<html>
<head>
<style><!--
.hmmessage P
{
margin:0px=3B
padding:0px
}
body.hmmessage
{
font-size: 12pt=3B
font-family:Calibri
}
--></style></head>
<body class=3D'hmmessage'><div dir=3D'ltr'><div>&gt=3B Is there a citation =
for this data point about IPv6 not working as well&nbsp=3Bin production?&nb=
sp=3B</div><div><br></div><div>[BA] At IETF 87 (and in several meetings pri=
or to that)=2C we have had a discussion within the MMUSIC WG about "Happy E=
yeballs for ICE" (draft-reddy-mmusic-ice-happy-eyeballs):</div><div><a href=
=3D"http://www.ietf.org/proceedings/87/slides/slides-87-mmusic-0.pdf" targe=
t=3D"_blank">http://www.ietf.org/proceedings/87/slides/slides-87-mmusic-0.p=
df</a></div><div><br></div><div>&gt=3B There used to be an issue with 6to4 =
and maybe Teredo=2C but i believe<br>&gt=3B the data shows that this issue =
has largely been resolved<br>&gt=3B http://www.google.com/intl/en/ipv6/stat=
istics.html via host treatment<br>&gt=3B of those tunnels as well as happy-=
eyeballs in browsers.</div><div><br></div><div>[BA] Those statistics do not=
 relate to ICEv6 usage=2C and in particular=2C they don't cover the range o=
f behavior that we see in enterprise and carrier networks. &nbsp=3BAs an ex=
ample=2C in deployments we have encountered IPv6 NATs and firewalls with a =
wide range of behavior=2C IPv6-only carrier networks where connectivity to =
IPv4 endpoints is needed=2C etc. &nbsp=3B</div><div><br></div><div>&gt=3B I=
 believe happy-eyeballs / RFC6555 would apply to WebRTC interactions<br>&gt=
=3B equally well=2C since the signalling is over the standard HTTP/HTTPS&nb=
sp=3Bcalls.</div><div><br></div><div>[BA] RFC 6555 doesn't directly apply t=
o ICE=2C which is why we've been having a distinct "ICE Happy Eyeballs" dis=
cussion.&nbsp=3B</div><div><br></div> 		 	   		  </div></body>
</html>=

--_d80e08cb-6865-4bd2-92ad-a569bcc1bbf1_--

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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Adopting draft-muthu-behave-consent-freshness?
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Subject: Re: [rtcweb] Adopting draft-muthu-behave-consent-freshness?
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I support adoption of this draft as a WG item for the given milestone.

Cheers,
Charles

On 9/9/13 1:36 AM, "Magnus Westerlund" <magnus.westerlund@ericsson.com>
wrote:

>WG,
>
>This is a call for WG adoption of STUN Usage for Consent Freshness
>(draft-muthu-behave-consent-freshness-04). This document defines a STUN
>usage for consent freshness. As this requires no protocol extensions we
>as intended users can define this usage in our WG. Such work also
>matches our charter. The draft-ietf-rtcweb-security-arch-07 is
>normatively dependent on this STUN usage.
>
>Document:
>https://datatracker.ietf.org/doc/draft-muthu-behave-consent-freshness/
>
>WG, please indicate your support or issues with adopting this document
>as WG item with a proposed milestone:
>
>Mar 2014 Send STUN Usage for Consent Freshness to IESG for publication
>as proposed standard.
>
>Cheers
>
>Magnus Westerlund
>
>----------------------------------------------------------------------
>Multimedia Technologies, Ericsson Research EAB/TVM
>----------------------------------------------------------------------
>Ericsson AB                | Phone  +46 10 7148287
>F=E4r=F6gatan 6                | Mobile +46 73 0949079
>SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>----------------------------------------------------------------------
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb


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Subject: Re: [rtcweb] Consent freshness in the light of no-SDES (Re: Adopting draft-muthu-behave-consent-freshness?)
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On 9 September 2013 05:27, Harald Alvestrand <harald@alvestrand.no> wrote:
> Personal opinion: I think SRTCP is still unsuitable for consent freshness.
> One reason (apart from any aspect that deals with security, on which I don't
> want to comment) is that if one sets up a connection with only a data
> channel, there will be no SSRCs to send SRTCP on.

This is my opinion too.

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Subject: Re: [rtcweb] Consent freshness in the light of no-SDES (Re: Adopting draft-muthu-behave-consent-freshness?)
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On 9 September 2013 05:44, Christer Holmberg
<christer.holmberg@ericsson.com> wrote:
> And, from a security perspective (the backward compatibility is a separate issue) the requirement to integrity protect the STUN binding requests are not affected by the SDES decision?

Integrity protection prevents an on-path attacker who is ignorant of
ICE credentials from artificially extending consent.  That's a fairly
restricted set.

The whole SDES interaction with consent is pretty weak, if you ask me.

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Subject: Re: [rtcweb] Adopting draft-muthu-behave-consent-freshness?
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On 9 September 2013 01:36, Magnus Westerlund
<magnus.westerlund@ericsson.com> wrote:
> Document:
> https://datatracker.ietf.org/doc/draft-muthu-behave-consent-freshness/
>
> WG, please indicate your support or issues with adopting this document
> as WG item with a proposed milestone:

I think that we should adopt this draft, then fix it.  The proposed
algorithm is still quite confusingly specified.  It also unnecessarily
conflates liveness testing with consent.

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+1

On Sep 9, 2013, at 1:54 PM, Charles Eckel (eckelcu) <eckelcu@cisco.com> wro=
te:

> I support adoption of this draft as a WG item for the given milestone.
>=20
> Cheers,
> Charles
>=20
> On 9/9/13 1:36 AM, "Magnus Westerlund" <magnus.westerlund@ericsson.com>
> wrote:
>=20
>> WG,
>>=20
>> This is a call for WG adoption of STUN Usage for Consent Freshness
>> (draft-muthu-behave-consent-freshness-04). This document defines a STUN
>> usage for consent freshness. As this requires no protocol extensions we
>> as intended users can define this usage in our WG. Such work also
>> matches our charter. The draft-ietf-rtcweb-security-arch-07 is
>> normatively dependent on this STUN usage.
>>=20
>> Document:
>> https://datatracker.ietf.org/doc/draft-muthu-behave-consent-freshness/
>>=20
>> WG, please indicate your support or issues with adopting this document
>> as WG item with a proposed milestone:
>>=20
>> Mar 2014 Send STUN Usage for Consent Freshness to IESG for publication
>> as proposed standard.
>>=20
>> Cheers
>>=20
>> Magnus Westerlund
>>=20
>> ----------------------------------------------------------------------
>> Multimedia Technologies, Ericsson Research EAB/TVM
>> ----------------------------------------------------------------------
>> Ericsson AB                | Phone  +46 10 7148287
>> F=E4r=F6gatan 6                | Mobile +46 73 0949079
>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>> ----------------------------------------------------------------------
>>=20
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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+1

On Sep 9, 2013, at 7:54 PM, "Charles Eckel (eckelcu)" <eckelcu@cisco.com> wr=
ote:

> I support adoption of this draft as a WG item for the given milestone.
>=20
> Cheers,
> Charles
>=20
> On 9/9/13 1:36 AM, "Magnus Westerlund" <magnus.westerlund@ericsson.com>
> wrote:
>=20
>> WG,
>>=20
>> This is a call for WG adoption of STUN Usage for Consent Freshness
>> (draft-muthu-behave-consent-freshness-04). This document defines a STUN
>> usage for consent freshness. As this requires no protocol extensions we
>> as intended users can define this usage in our WG. Such work also
>> matches our charter. The draft-ietf-rtcweb-security-arch-07 is
>> normatively dependent on this STUN usage.
>>=20
>> Document:
>> https://datatracker.ietf.org/doc/draft-muthu-behave-consent-freshness/
>>=20
>> WG, please indicate your support or issues with adopting this document
>> as WG item with a proposed milestone:
>>=20
>> Mar 2014 Send STUN Usage for Consent Freshness to IESG for publication
>> as proposed standard.
>>=20
>> Cheers
>>=20
>> Magnus Westerlund
>>=20
>> ----------------------------------------------------------------------
>> Multimedia Technologies, Ericsson Research EAB/TVM
>> ----------------------------------------------------------------------
>> Ericsson AB                | Phone  +46 10 7148287
>> F=C3=A4r=C3=B6gatan 6                | Mobile +46 73 0949079
>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>> ----------------------------------------------------------------------
>>=20
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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Martin Thomson wrote:
> I think that we should adopt this draft, then fix it.  The proposed

+1


From victor.pascual.avila@gmail.com  Tue Sep 10 06:47:39 2013
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From: Victor Pascual Avila <victor.pascual.avila@gmail.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: [rtcweb] Plan for MTI video codec?
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Dear WG Chairs,

What's the plan for the MTI video codec discussion? Are we planning to
discuss this during our next meeting? Is there a reason to delay this
decision?

My apologies if this has already been discussed but could not find it
in the archives.

Thanks,
-Victor

From ekr@rtfm.com  Tue Sep 10 07:16:56 2013
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+1


On Tue, Sep 10, 2013 at 12:45 AM, Timothy B. Terriberry <
tterriberry@mozilla.com> wrote:

> Martin Thomson wrote:
>
>> I think that we should adopt this draft, then fix it.  The proposed
>>
>
> +1
>
>
> ______________________________**_________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/mailman/listinfo/rtcweb>
>

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<div dir=3D"ltr">+1</div><div class=3D"gmail_extra"><br><br><div class=3D"g=
mail_quote">On Tue, Sep 10, 2013 at 12:45 AM, Timothy B. Terriberry <span d=
ir=3D"ltr">&lt;<a href=3D"mailto:tterriberry@mozilla.com" target=3D"_blank"=
>tterriberry@mozilla.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"im">Martin Thomson wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
I think that we should adopt this draft, then fix it. =A0The proposed<br>
</blockquote>
<br></div>
+1<div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>

--047d7b6766f8a3948b04e6082723--

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--001a1133c690aaac8e04e608bd76
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Coming from an implementor's perspective (typically), and having read quite
a few RFCs, I strongly concur with your statement, about using SHOULD in an
RFC requiring spelling out the circumstances where it's reasonable to
ignore.

As Martin said, and I agree, usually implementors take liberties to ignore
the "SHOULDs, RECOMMENDEDs, MAYs, and OPTIONALs" until they get bit by
something (usually in interop).  Then they curse the spec for using non
mandatory language for whatever bit them :-)

Matthew Fredrickson



On Fri, Sep 6, 2013 at 5:27 PM, Harald Alvestrand <harald@alvestrand.no>wrote:

> On 09/06/2013 11:42 PM, Mo Zanaty (mzanaty) wrote:
>
>> True. The same can be said about every SHOULD, but we still bother to
>> specify it.
>> Maybe we should update RFC 2119 to state:
>>
>> Obey: "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT"
>> Ignore: "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", "OPTIONAL"
>>
>
> At the time 2119 was written (I remember discussing this with Scott
> Bradner several times), the intent of the text for "SHOULD" was "you'd
> better do this if you don't have a real good reason why it's not a
> reasonable thing to do in your particular case".
>
> Quoth:
>
> 3. SHOULD   This word, or the adjective "RECOMMENDED", mean that there
>    may exist valid reasons in particular circumstances to ignore a
>    particular item, but the full implications must be understood and
>    carefully weighed before choosing a different course.
>
> Of course, people who don't want to cover all the SHOULDs have pushed
> towards interpreting it as "weak recommendation that we can ignore if we
> feel like it", but that wasn't the original intent of 2119, and some of us
> still want to have that word available for use in the stronger meaning.
>
> At one time there was push towards saying that "if you write SHOULD in an
> RFC, you need to spell out the circumstances where it'll be reasonable to
> ignore it - otherwise, write MUST or MAY". That push has petered out at
> this time, but I felt sympathetic to the idea.
>
> That's why I'm so reluctant to use the word in cases where I think a large
> part of the implementors are going to ignore it, and where (in my opinion)
> no great harm comes to interoperability when they do.
>
> But this instance not something I storm barricades over; if the consensus
> of the WG is to use "RECOMMENDED" rather than "recommended", I'll note that
> I'm the rough part of the consensus, and live with it.
>
>
>
>
>> -----Original Message-----
>> From: Martin Thomson [mailto:martin.thomson@gmail.**com<martin.thomson@gmail.com>
>> ]
>> Sent: Friday, September 06, 2013 4:38 PM
>> To: Mo Zanaty (mzanaty)
>> Cc: Jean-Marc Valin; Bo Burman; rtcweb@ietf.org
>> Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
>>
>> On 6 September 2013 13:28, Mo Zanaty (mzanaty) <mzanaty@cisco.com> wrote:
>>
>>> If the will is hostile to other codecs, this text won't matter.
>>>
>> The natural disposition of the implementer is hostility toward more
>> work, so I don't see the text making much difference either way.
>> ______________________________**_________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/mailman/listinfo/rtcweb>
>>
>
> ______________________________**_________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/mailman/listinfo/rtcweb>
>

--001a1133c690aaac8e04e608bd76
Content-Type: text/html; charset=ISO-8859-1
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<div dir=3D"ltr">Coming from an implementor&#39;s perspective (typically), =
and having read quite a few RFCs, I strongly concur with your statement, ab=
out using SHOULD in an RFC requiring spelling out the circumstances where i=
t&#39;s reasonable to ignore.<div>
<br></div><div>As Martin said, and I agree, usually implementors take liber=
ties to ignore the &quot;SHOULDs, RECOMMENDEDs, MAYs, and OPTIONALs&quot; u=
ntil they get bit by something (usually in interop). =A0Then they curse the=
 spec for using non mandatory language for whatever bit them :-)</div>
<div><br></div><div>Matthew Fredrickson</div><div><br></div></div><div clas=
s=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Fri, Sep 6, 2013 at=
 5:27 PM, Harald Alvestrand <span dir=3D"ltr">&lt;<a href=3D"mailto:harald@=
alvestrand.no" target=3D"_blank">harald@alvestrand.no</a>&gt;</span> wrote:=
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"im">On 09/06/2013 11:42 PM, Mo=
 Zanaty (mzanaty) wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
True. The same can be said about every SHOULD, but we still bother to speci=
fy it.<br>
Maybe we should update RFC 2119 to state:<br>
<br>
Obey: &quot;MUST&quot;, &quot;MUST NOT&quot;, &quot;REQUIRED&quot;, &quot;S=
HALL&quot;, &quot;SHALL NOT&quot;<br>
Ignore: &quot;SHOULD&quot;, &quot;SHOULD NOT&quot;, &quot;RECOMMENDED&quot;=
, &quot;MAY&quot;, &quot;OPTIONAL&quot;<br>
</blockquote>
<br></div>
At the time 2119 was written (I remember discussing this with Scott Bradner=
 several times), the intent of the text for &quot;SHOULD&quot; was &quot;yo=
u&#39;d better do this if you don&#39;t have a real good reason why it&#39;=
s not a reasonable thing to do in your particular case&quot;.<br>

<br>
Quoth:<br>
<br>
3. SHOULD =A0 This word, or the adjective &quot;RECOMMENDED&quot;, mean tha=
t there<br>
=A0 =A0may exist valid reasons in particular circumstances to ignore a<br>
=A0 =A0particular item, but the full implications must be understood and<br=
>
=A0 =A0carefully weighed before choosing a different course.<br>
<br>
Of course, people who don&#39;t want to cover all the SHOULDs have pushed t=
owards interpreting it as &quot;weak recommendation that we can ignore if w=
e feel like it&quot;, but that wasn&#39;t the original intent of 2119, and =
some of us still want to have that word available for use in the stronger m=
eaning.<br>

<br>
At one time there was push towards saying that &quot;if you write SHOULD in=
 an RFC, you need to spell out the circumstances where it&#39;ll be reasona=
ble to ignore it - otherwise, write MUST or MAY&quot;. That push has petere=
d out at this time, but I felt sympathetic to the idea.<br>

<br>
That&#39;s why I&#39;m so reluctant to use the word in cases where I think =
a large part of the implementors are going to ignore it, and where (in my o=
pinion) no great harm comes to interoperability when they do.<br>
<br>
But this instance not something I storm barricades over; if the consensus o=
f the WG is to use &quot;RECOMMENDED&quot; rather than &quot;recommended&qu=
ot;, I&#39;ll note that I&#39;m the rough part of the consensus, and live w=
ith it.<div class=3D"HOEnZb">
<div class=3D"h5"><br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
-----Original Message-----<br>
From: Martin Thomson [mailto:<a href=3D"mailto:martin.thomson@gmail.com" ta=
rget=3D"_blank">martin.thomson@gmail.<u></u>com</a>]<br>
Sent: Friday, September 06, 2013 4:38 PM<br>
To: Mo Zanaty (mzanaty)<br>
Cc: Jean-Marc Valin; Bo Burman; <a href=3D"mailto:rtcweb@ietf.org" target=
=3D"_blank">rtcweb@ietf.org</a><br>
Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt<br>
<br>
On 6 September 2013 13:28, Mo Zanaty (mzanaty) &lt;<a href=3D"mailto:mzanat=
y@cisco.com" target=3D"_blank">mzanaty@cisco.com</a>&gt; wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
If the will is hostile to other codecs, this text won&#39;t matter.<br>
</blockquote>
The natural disposition of the implementer is hostility toward more<br>
work, so I don&#39;t see the text making much difference either way.<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</blockquote>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>

--001a1133c690aaac8e04e608bd76--

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From: Matt Fredrickson <creslin@digium.com>
To: Bo Burman <bo.burman@ericsson.com>
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--001a11c3799ca7a69d04e608c824
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I support it as well.

Matthew Fredrickson


On Thu, Sep 5, 2013 at 8:05 AM, Bo Burman <bo.burman@ericsson.com> wrote:

> I also support it.
> /Bo
>
> > -----Original Message-----
> > From: Rauschenbach, Uwe (NSN - DE/Munich) [mailto:
> uwe.rauschenbach@nsn.com]
> > Sent: den 5 september 2013 12:14
> > To: ext Mo Zanaty (mzanaty); Bo Burman; rtcweb@ietf.org
> > Subject: RE: I-D Action: draft-ietf-rtcweb-audio-02.txt
> >
> > I support this proposal.
> >
> > Kind regards,
> > Uwe
> >
> >
> > > -----Original Message-----
> > > From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> > > Behalf Of ext Mo Zanaty (mzanaty)
> > > Sent: Wednesday, September 04, 2013 10:03 PM
> > > To: Bo Burman; rtcweb@ietf.org
> > > Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
> > >
> > > I support the proposed text addition. But rather than adding at the
> > > end of section 3, replace the beginning of section 3 as below.
> > >
> > > OLD:
> > >    To ensure a baseline level of interoperability between WebRTC
> > >    clients, a minimum set of required codecs are specified below.
> > >    While this section specifies the codecs that will be mandated for
> > > all
> > >    WebRTC client implementations, it leaves the question of supporting
> > >    additional codecs to the will of the implementer.
> > >
> > > NEW:
> > >    To ensure a baseline level of interoperability between WebRTC
> > >    clients, a minimum set of required codecs are specified below.
> > >    If other suitable audio codecs are available to the browser to use,
> > >    it is RECOMMENDED that they are also included in the offer in order
> > >    to maximize the possibility to establish the session without the
> > > need
> > >    for audio transcoding.
> > >
> > > Mo
> > >
> > > -----Original Message-----
> > > From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> > > Behalf Of Bo Burman
> > > Sent: Tuesday, August 27, 2013 10:53 AM
> > > To: rtcweb@ietf.org
> > > Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
> > >
> > > Based on the previous discussion thread http://www.ietf.org/mail-
> > > archive/web/rtcweb/current/msg08501.html, I believe there is support
> > > to add some text into this document.
> > >
> > > I thus propose to add the following text at the end of section 3:
> > >
> > > If other suitable audio codecs are available to the browser to use, it
> > > is RECOMMENDED that they are also included in the offer in order to
> > > maximize the possibility to establish the session without the need for
> > > audio transcoding
> > >
> > > Cheers,
> > > Bo
> > >
> > > > -----Original Message-----
> > > > From: i-d-announce-bounces@ietf.org [mailto:i-d-announce-
> > > bounces@ietf.org] On Behalf Of internet-drafts@ietf.org
> > > > Sent: den 2 augusti 2013 18:30
> > > > To: i-d-announce@ietf.org
> > > > Cc: rtcweb@ietf.org
> > > > Subject: I-D Action: draft-ietf-rtcweb-audio-02.txt
> > > >
> > > >
> > > > A New Internet-Draft is available from the on-line Internet-Drafts
> > > directories.
> > > >  This draft is a work item of the Real-Time Communication in WEB-
> > > browsers Working Group of the IETF.
> > > >
> > > >   Title           : WebRTC Audio Codec and Processing Requirements
> > > >   Author(s)       : Jean-Marc Valin
> > > >                           Cary Bran
> > > >   Filename        : draft-ietf-rtcweb-audio-02.txt
> > > >   Pages           : 6
> > > >   Date            : 2013-08-02
> > > >
> > > > Abstract:
> > > >    This document outlines the audio codec and processing requirements
> > > >    for WebRTC client application and endpoint devices.
> > > >
> > > >
> > > > The IETF datatracker status page for this draft is:
> > > > https://datatracker.ietf.org/doc/draft-ietf-rtcweb-audio
> > > >
> > > > There's also a htmlized version available at:
> > > > http://tools.ietf.org/html/draft-ietf-rtcweb-audio-02
> > > >
> > > > A diff from the previous version is available at:
> > > > http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-audio-02
> > > >
> > > >
> > > > Please note that it may take a couple of minutes from the time of
> > > submission until the htmlized version and diff are
> > > > available at tools.ietf.org.
> > > >
> > > > Internet-Drafts are also available by anonymous FTP at:
> > > > ftp://ftp.ietf.org/internet-drafts/
> > > >
> > > > _______________________________________________
> > > > I-D-Announce mailing list
> > > > I-D-Announce@ietf.org
> > > > https://www.ietf.org/mailman/listinfo/i-d-announce
> > > > Internet-Draft directories: http://www.ietf.org/shadow.html or
> > > ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> > > _______________________________________________
> > > rtcweb mailing list
> > > rtcweb@ietf.org
> > > https://www.ietf.org/mailman/listinfo/rtcweb
> > > _______________________________________________
> > > rtcweb mailing list
> > > rtcweb@ietf.org
> > > https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--001a11c3799ca7a69d04e608c824
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">I support it as well.<div><br>Matthew Fredrickson</div></d=
iv><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Thu, Se=
p 5, 2013 at 8:05 AM, Bo Burman <span dir=3D"ltr">&lt;<a href=3D"mailto:bo.=
burman@ericsson.com" target=3D"_blank">bo.burman@ericsson.com</a>&gt;</span=
> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">I also support it.<br>
<span class=3D"HOEnZb"><font color=3D"#888888">/Bo<br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
&gt; -----Original Message-----<br>
&gt; From: Rauschenbach, Uwe (NSN - DE/Munich) [mailto:<a href=3D"mailto:uw=
e.rauschenbach@nsn.com">uwe.rauschenbach@nsn.com</a>]<br>
&gt; Sent: den 5 september 2013 12:14<br>
&gt; To: ext Mo Zanaty (mzanaty); Bo Burman; <a href=3D"mailto:rtcweb@ietf.=
org">rtcweb@ietf.org</a><br>
&gt; Subject: RE: I-D Action: draft-ietf-rtcweb-audio-02.txt<br>
&gt;<br>
&gt; I support this proposal.<br>
&gt;<br>
&gt; Kind regards,<br>
&gt; Uwe<br>
&gt;<br>
&gt;<br>
&gt; &gt; -----Original Message-----<br>
&gt; &gt; From: <a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@i=
etf.org</a> [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounc=
es@ietf.org</a>] On<br>
&gt; &gt; Behalf Of ext Mo Zanaty (mzanaty)<br>
&gt; &gt; Sent: Wednesday, September 04, 2013 10:03 PM<br>
&gt; &gt; To: Bo Burman; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org=
</a><br>
&gt; &gt; Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt<=
br>
&gt; &gt;<br>
&gt; &gt; I support the proposed text addition. But rather than adding at t=
he<br>
&gt; &gt; end of section 3, replace the beginning of section 3 as below.<br=
>
&gt; &gt;<br>
&gt; &gt; OLD:<br>
&gt; &gt; =A0 =A0To ensure a baseline level of interoperability between Web=
RTC<br>
&gt; &gt; =A0 =A0clients, a minimum set of required codecs are specified be=
low.<br>
&gt; &gt; =A0 =A0While this section specifies the codecs that will be manda=
ted for<br>
&gt; &gt; all<br>
&gt; &gt; =A0 =A0WebRTC client implementations, it leaves the question of s=
upporting<br>
&gt; &gt; =A0 =A0additional codecs to the will of the implementer.<br>
&gt; &gt;<br>
&gt; &gt; NEW:<br>
&gt; &gt; =A0 =A0To ensure a baseline level of interoperability between Web=
RTC<br>
&gt; &gt; =A0 =A0clients, a minimum set of required codecs are specified be=
low.<br>
&gt; &gt; =A0 =A0If other suitable audio codecs are available to the browse=
r to use,<br>
&gt; &gt; =A0 =A0it is RECOMMENDED that they are also included in the offer=
 in order<br>
&gt; &gt; =A0 =A0to maximize the possibility to establish the session witho=
ut the<br>
&gt; &gt; need<br>
&gt; &gt; =A0 =A0for audio transcoding.<br>
&gt; &gt;<br>
&gt; &gt; Mo<br>
&gt; &gt;<br>
&gt; &gt; -----Original Message-----<br>
&gt; &gt; From: <a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@i=
etf.org</a> [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounc=
es@ietf.org</a>] On<br>
&gt; &gt; Behalf Of Bo Burman<br>
&gt; &gt; Sent: Tuesday, August 27, 2013 10:53 AM<br>
&gt; &gt; To: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; &gt; Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt<=
br>
&gt; &gt;<br>
&gt; &gt; Based on the previous discussion thread <a href=3D"http://www.iet=
f.org/mail-" target=3D"_blank">http://www.ietf.org/mail-</a><br>
&gt; &gt; archive/web/rtcweb/current/msg08501.html, I believe there is supp=
ort<br>
&gt; &gt; to add some text into this document.<br>
&gt; &gt;<br>
&gt; &gt; I thus propose to add the following text at the end of section 3:=
<br>
&gt; &gt;<br>
&gt; &gt; If other suitable audio codecs are available to the browser to us=
e, it<br>
&gt; &gt; is RECOMMENDED that they are also included in the offer in order =
to<br>
&gt; &gt; maximize the possibility to establish the session without the nee=
d for<br>
&gt; &gt; audio transcoding<br>
&gt; &gt;<br>
&gt; &gt; Cheers,<br>
&gt; &gt; Bo<br>
&gt; &gt;<br>
&gt; &gt; &gt; -----Original Message-----<br>
&gt; &gt; &gt; From: <a href=3D"mailto:i-d-announce-bounces@ietf.org">i-d-a=
nnounce-bounces@ietf.org</a> [mailto:<a href=3D"mailto:i-d-announce-">i-d-a=
nnounce-</a><br>
&gt; &gt; <a href=3D"mailto:bounces@ietf.org">bounces@ietf.org</a>] On Beha=
lf Of <a href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org<=
/a><br>
&gt; &gt; &gt; Sent: den 2 augusti 2013 18:30<br>
&gt; &gt; &gt; To: <a href=3D"mailto:i-d-announce@ietf.org">i-d-announce@ie=
tf.org</a><br>
&gt; &gt; &gt; Cc: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><b=
r>
&gt; &gt; &gt; Subject: I-D Action: draft-ietf-rtcweb-audio-02.txt<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; A New Internet-Draft is available from the on-line Internet-=
Drafts<br>
&gt; &gt; directories.<br>
&gt; &gt; &gt; =A0This draft is a work item of the Real-Time Communication =
in WEB-<br>
&gt; &gt; browsers Working Group of the IETF.<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; =A0 Title =A0 =A0 =A0 =A0 =A0 : WebRTC Audio Codec and Proce=
ssing Requirements<br>
&gt; &gt; &gt; =A0 Author(s) =A0 =A0 =A0 : Jean-Marc Valin<br>
&gt; &gt; &gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Cary Bra=
n<br>
&gt; &gt; &gt; =A0 Filename =A0 =A0 =A0 =A0: draft-ietf-rtcweb-audio-02.txt=
<br>
&gt; &gt; &gt; =A0 Pages =A0 =A0 =A0 =A0 =A0 : 6<br>
&gt; &gt; &gt; =A0 Date =A0 =A0 =A0 =A0 =A0 =A0: 2013-08-02<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; Abstract:<br>
&gt; &gt; &gt; =A0 =A0This document outlines the audio codec and processing=
 requirements<br>
&gt; &gt; &gt; =A0 =A0for WebRTC client application and endpoint devices.<b=
r>
&gt; &gt; &gt;<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; The IETF datatracker status page for this draft is:<br>
&gt; &gt; &gt; <a href=3D"https://datatracker.ietf.org/doc/draft-ietf-rtcwe=
b-audio" target=3D"_blank">https://datatracker.ietf.org/doc/draft-ietf-rtcw=
eb-audio</a><br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; There&#39;s also a htmlized version available at:<br>
&gt; &gt; &gt; <a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-audi=
o-02" target=3D"_blank">http://tools.ietf.org/html/draft-ietf-rtcweb-audio-=
02</a><br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; A diff from the previous version is available at:<br>
&gt; &gt; &gt; <a href=3D"http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtc=
web-audio-02" target=3D"_blank">http://www.ietf.org/rfcdiff?url2=3Ddraft-ie=
tf-rtcweb-audio-02</a><br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; Please note that it may take a couple of minutes from the ti=
me of<br>
&gt; &gt; submission until the htmlized version and diff are<br>
&gt; &gt; &gt; available at <a href=3D"http://tools.ietf.org" target=3D"_bl=
ank">tools.ietf.org</a>.<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; Internet-Drafts are also available by anonymous FTP at:<br>
&gt; &gt; &gt; <a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=3D"_b=
lank">ftp://ftp.ietf.org/internet-drafts/</a><br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; _______________________________________________<br>
&gt; &gt; &gt; I-D-Announce mailing list<br>
&gt; &gt; &gt; <a href=3D"mailto:I-D-Announce@ietf.org">I-D-Announce@ietf.o=
rg</a><br>
&gt; &gt; &gt; <a href=3D"https://www.ietf.org/mailman/listinfo/i-d-announc=
e" target=3D"_blank">https://www.ietf.org/mailman/listinfo/i-d-announce</a>=
<br>
&gt; &gt; &gt; Internet-Draft directories: <a href=3D"http://www.ietf.org/s=
hadow.html" target=3D"_blank">http://www.ietf.org/shadow.html</a> or<br>
&gt; &gt; <a href=3D"ftp://ftp.ietf.org/ietf/1shadow-sites.txt" target=3D"_=
blank">ftp://ftp.ietf.org/ietf/1shadow-sites.txt</a><br>
&gt; &gt; _______________________________________________<br>
&gt; &gt; rtcweb mailing list<br>
&gt; &gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; &gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=
=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt; &gt; _______________________________________________<br>
&gt; &gt; rtcweb mailing list<br>
&gt; &gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; &gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=
=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>

--001a11c3799ca7a69d04e608c824--

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From: Jean-Marc Valin <jmvalin@mozilla.com>
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
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> To ensure a baseline level of interoperability between WebRTC
> clients, a minimum set of required codecs are specified below.
> If other suitable audio codecs are available to the browser to use,
> it is RECOMMENDED that they are also included in the offer in order
> to maximize the possibility to establish the session without the need
> for audio transcoding.

Let's look at the practical effect here, assuming people were to
actually follow such a recommendation. The only audio codecs that are
implemented in browsers but not available for WebRTC are the ones used
for HTML5. According to http://html5test.com/ these codecs are Vorbis,
MP3, AAC, and uncompressed PCM. These are the codecs that would be
recommended under such text. None of them is really ideal for
interactive audio, none of them is implemented in old phones AFAIK, but
all of them would require time and effort to add to a browser's RTP
stack. I really don't see the point here.

	Jean-Marc

From roman@telurix.com  Tue Sep 10 09:54:49 2013
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--001a11c278147ba2a304e60a5c9a
Content-Type: text/plain; charset=ISO-8859-1

On Tue, Sep 10, 2013 at 12:26 PM, Jean-Marc Valin <jmvalin@mozilla.com>wrote:

> > To ensure a baseline level of interoperability between WebRTC
> > clients, a minimum set of required codecs are specified below.
> > If other suitable audio codecs are available to the browser to use,
> > it is RECOMMENDED that they are also included in the offer in order
> > to maximize the possibility to establish the session without the need
> > for audio transcoding.
>
> Let's look at the practical effect here, assuming people were to
> actually follow such a recommendation. The only audio codecs that are
> implemented in browsers but not available for WebRTC are the ones used
> for HTML5. According to http://html5test.com/ these codecs are Vorbis,
> MP3, AAC, and uncompressed PCM. These are the codecs that would be
> recommended under such text. None of them is really ideal for
> interactive audio, none of them is implemented in old phones AFAIK, but
> all of them would require time and effort to add to a browser's RTP
> stack. I really don't see the point here.
>
>
My biggest problem with the recommendation text is "other suitable audio
codecs are available to the browser to use". Jean-Marc's comment underlines
the problem. His interpretation is that this refers to codecs already
implemented in the browser. Another interpretation is that this refers to
codecs already implemented in the underlying OS or platform, such as AMR-WB
or H.264. Yet another interpretation implies codecs with no associated
patents, such as G.722 or G.726. Yet another interpretation is that implies
codecs with patent licenses which allows free implementation such as iLBC
or iSAC. What is the intent here? With so many interpretations this
language is completely useless.

I think original intent was to encourage browser implementers to add
support for G.722, AMR and AMR-WB if available in underlying platform, and
H.264, if available in underlying platform or browser. In this regards this
language fails completely. Don't you think it would be better to state
exactly what the intent is and provide the language when implementation of
each of those codecs is not required (meaning if additional license fees
will need to be paid or due to incompatible licensing policy).
_____________
Roman Shpount

--001a11c278147ba2a304e60a5c9a
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<div dir=3D"ltr"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">=
On Tue, Sep 10, 2013 at 12:26 PM, Jean-Marc Valin <span dir=3D"ltr">&lt;<a =
href=3D"mailto:jmvalin@mozilla.com" target=3D"_blank">jmvalin@mozilla.com</=
a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left:1px solid rgb(204,204,204);padding-left:1ex"><div class=3D"im">&gt; To=
 ensure a baseline level of interoperability between WebRTC<br>
&gt; clients, a minimum set of required codecs are specified below.<br>
&gt; If other suitable audio codecs are available to the browser to use,<br=
>
&gt; it is RECOMMENDED that they are also included in the offer in order<br=
>
&gt; to maximize the possibility to establish the session without the need<=
br>
&gt; for audio transcoding.<br>
<br>
</div>Let&#39;s look at the practical effect here, assuming people were to<=
br>
actually follow such a recommendation. The only audio codecs that are<br>
implemented in browsers but not available for WebRTC are the ones used<br>
for HTML5. According to <a href=3D"http://html5test.com/" target=3D"_blank"=
>http://html5test.com/</a> these codecs are Vorbis,<br>
MP3, AAC, and uncompressed PCM. These are the codecs that would be<br>
recommended under such text. None of them is really ideal for<br>
interactive audio, none of them is implemented in old phones AFAIK, but<br>
all of them would require time and effort to add to a browser&#39;s RTP<br>
stack. I really don&#39;t see the point here.<br>
<span class=3D""><font color=3D"#888888"></font></span><br></blockquote></d=
iv><br>My biggest problem with the recommendation text is &quot;other suita=
ble audio codecs are available to the browser to use&quot;. Jean-Marc&#39;s=
 comment underlines the problem. His interpretation is that this refers to =
codecs already implemented in the browser. Another interpretation is that t=
his refers to codecs already implemented in the underlying OS or platform, =
such as AMR-WB or H.264. Yet another interpretation implies codecs with no =
associated patents, such as G.722 or G.726. Yet another interpretation is t=
hat implies codecs with patent licenses which allows free implementation su=
ch as iLBC or iSAC. What is the intent here? With so many interpretations t=
his language is completely useless.<br>
<br></div><div class=3D"gmail_extra">I think original intent was to encoura=
ge browser implementers to add support for G.722, AMR and AMR-WB if availab=
le in underlying platform, and H.264, if available in underlying platform o=
r browser. In this regards this language fails completely. Don&#39;t you th=
ink it would be better to state exactly what the intent is and provide the =
language when implementation of each of those codecs is not required (meani=
ng if additional license fees will need to be paid or due to incompatible l=
icensing policy).<br>
</div><div class=3D"gmail_extra"><div>_____________<br>Roman Shpount</div>
<br></div></div>

--001a11c278147ba2a304e60a5c9a--

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Date: Tue, 10 Sep 2013 15:08:26 -0400
Thread-Topic: [rtcweb] Adopting draft-muthu-behave-consent-freshness?
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Muthu et al --=20


In the abstract, second paragraph, the text states "how" a WebRTC browser c=
an verify peer consent and as I read through the rest of the document I als=
o understood "why" this is needed. So, just a knit pick, it may be clear to=
 the reader if the statement also includes why within the body of the Abstr=
act.

One minor typo in section 7. Please change "in-intended" to "un-intended".

Thanks
Sanjay

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Magnus Westerlund
Sent: Monday, September 09, 2013 4:37 AM
To: rtcweb@ietf.org
Subject: [rtcweb] Adopting draft-muthu-behave-consent-freshness?

WG,

This is a call for WG adoption of STUN Usage for Consent Freshness (draft-m=
uthu-behave-consent-freshness-04). This document defines a STUN usage for c=
onsent freshness. As this requires no protocol extensions we as intended us=
ers can define this usage in our WG. Such work also matches our charter. Th=
e draft-ietf-rtcweb-security-arch-07 is normatively dependent on this STUN =
usage.

Document:
https://datatracker.ietf.org/doc/draft-muthu-behave-consent-freshness/

WG, please indicate your support or issues with adopting this document as W=
G item with a proposed milestone:

Mar 2014 Send STUN Usage for Consent Freshness to IESG for publication as p=
roposed standard.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
F=E4r=F6gatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------

_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

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To: Bernard Aboba <bernard_aboba@hotmail.com>
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Subject: Re: [rtcweb] consent freshness vs. circuit breakers
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On Sep 9, 2013, at 8:45 AM, Bernard Aboba <bernard_aboba@hotmail.com> =
wrote:

>=20
> Christer said:=20
>=20
> "Now, as we have decided to not use SDES, I guess that can be =
removed."
>=20
> [BA] Actually, I believe that a sentence or two describing why the =
needs of consent freshness are not met by mechanisms such as SRTCP is =
worth keeping, and possibly should even be expanded upon.  One reason is =
that we also have the "circuit breakers" mechanism which *does* rely (in =
part) on SRTCP.  =20
>=20
> Christer said:=20
>=20
> "But, based on that, I'd just like to verify whether there is still a =
need for the draft :)"
>=20
> [BA] I do think that the draft is still necessary, even without =
SDES/SRTP.   The usage of ICE for consent seems to me to be less =
susceptible to a variety of issues than a dependency on SRTCP might be.=20=

>=20
> However, I am concerned about the interaction between circuit breakers =
and consent freshness.  At IETF 87, we heard that the circuit breakers =
mechanism could fire even when the overall loss rate was quite low, due =
to burst losses.  That shouldn't be true of consent, since loss of =
consent requires up to 30 consecutive losses over a 15 second period.  =
However, it would appear to me that the current circuit breakers =
algorithms will cause media sending to be curtailed prior to loss of =
consent in many cases. =20

Yes, consent needs to avoid the same pitfalls that were explained for =
circuit breakers. =20

(For others that might be interested, the AVT presentation describing =
the research on circuit breakers is at =
http://www.ietf.org/proceedings/87/slides/slides-87-avtcore-2.pdf.)

-d


>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Diso-8859-1"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
"><br><div><div>On Sep 9, 2013, at 8:45 AM, Bernard Aboba &lt;<a =
href=3D"mailto:bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a>&gt=
; wrote:</div><br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite">
<meta http-equiv=3D"Content-Type" content=3D"text/html; =
charset=3Diso-8859-1">
<div class=3D"hmmessage"><div dir=3D"ltr">

<style><!--
.hmmessage P
{
margin:0px;
padding:0px
}
body.hmmessage
{
font-size: 12pt;
font-family:Calibri
}
--></style>
<div dir=3D"ltr"><br><span style=3D"font-size: 12pt; ">Christer =
said:&nbsp;</span><div><div><br></div><div>"Now, as we have decided to =
not use SDES, I guess that can be =
removed."</div><div><br></div><div>[BA] Actually, I believe that a =
sentence or two describing why the needs of consent freshness are not =
met by mechanisms such as SRTCP is worth keeping, and possibly should =
even be expanded upon. &nbsp;One reason is that we also have the =
"circuit breakers" mechanism which *does* rely (in part) on SRTCP. =
&nbsp;&nbsp;</div><div><br>Christer =
said:&nbsp;</div><div><br></div><div>"But, based on that, I'd just like =
to verify whether there is still a need for the draft =
:)"</div><div><br></div><div>[BA] I do think that the draft is still =
necessary, even without SDES/SRTP. &nbsp; The usage of ICE for consent =
seems to me to be less susceptible to a variety of issues than a =
dependency on SRTCP might be.&nbsp;</div><div><br></div><div>However, I =
am concerned about the interaction between circuit breakers and consent =
freshness. &nbsp;At IETF 87, we heard that the circuit breakers =
mechanism could fire even when the overall loss rate was quite low, due =
to burst losses. &nbsp;That shouldn't be true of consent, since loss of =
consent requires up to 30 consecutive losses over a 15 second period. =
&nbsp;However, it would appear to me that the current circuit breakers =
algorithms will cause media sending to be curtailed prior to loss of =
consent in many cases. =
&nbsp;<br></div></div></div></div></div></blockquote><div><br></div><div>Y=
es, consent needs to avoid the same pitfalls that were explained for =
circuit breakers. &nbsp;</div><div><br></div><div>(For others that might =
be interested, the AVT presentation describing the research on circuit =
breakers is at&nbsp;<a =
href=3D"http://www.ietf.org/proceedings/87/slides/slides-87-avtcore-2.pdf"=
>http://www.ietf.org/proceedings/87/slides/slides-87-avtcore-2.pdf</a>.)</=
div><div><br></div><div>-d</div><div><br></div><br><blockquote =
type=3D"cite"><div class=3D"hmmessage"><div dir=3D"ltr"><div =
dir=3D"ltr"><div><div><br></div></div></div>
 		 	   		  </div></div>
_______________________________________________<br>rtcweb mailing =
list<br><a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>https://www.ietf.or=
g/mailman/listinfo/rtcweb<br></blockquote></div><br></body></html>=

--Apple-Mail=_970D3A7B-7DA7-431D-B846-4DDF881D6F30--

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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: "Mishra, Sanjay" <sanjay.mishra@verizon.com>, "Dan Wing (dwing)" <dwing@cisco.com>, "Ram Mohan R (rmohanr)" <rmohanr@cisco.com>, "Tirumaleswar Reddy (tireddy)" <tireddy@cisco.com>
Thread-Topic: [rtcweb] Adopting draft-muthu-behave-consent-freshness?
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Subject: Re: [rtcweb] Adopting draft-muthu-behave-consent-freshness?
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|In the abstract, second paragraph, the text states "how" a WebRTC=20
|browser can verify peer consent and as I read through the rest of=20
|the document I also understood "why" this is needed. So, just a knit
|pick, it may be clear to the reader if the statement also includes=20
|why within the body of the Abstract.

The "why" part is mentioned in the first paragraph of the abstract:
   Verification of peer consent before sending traffic is necessary in
   WebRTC deployments to ensure that a malicious JavaScript cannot use
   the browser as a platform for launching attacks.

Are you suggesting expanding this? The abstract needs to be concise, howeve=
r.

|One minor typo in section 7. Please change "in-intended" to "un-intended".

Will fix it.

Thanks for the review..

Muthu=20

|-----Original Message-----
|From: Mishra, Sanjay [mailto:sanjay.mishra@verizon.com]
|Sent: Wednesday, September 11, 2013 12:38 AM
|To: Muthu Arul Mozhi Perumal (mperumal); Dan Wing (dwing); Ram Mohan R (rm=
ohanr); Tirumaleswar Reddy
|(tireddy)
|Cc: rtcweb@ietf.org
|Subject: RE: [rtcweb] Adopting draft-muthu-behave-consent-freshness?
|
|Muthu et al --
|
|
|In the abstract, second paragraph, the text states "how" a WebRTC browser =
can verify peer consent and
|as I read through the rest of the document I also understood "why" this is=
 needed. So, just a knit
|pick, it may be clear to the reader if the statement also includes why wit=
hin the body of the
|Abstract.
|
|One minor typo in section 7. Please change "in-intended" to "un-intended".
|
|Thanks
|Sanjay
|
|-----Original Message-----
|From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf O=
f Magnus Westerlund
|Sent: Monday, September 09, 2013 4:37 AM
|To: rtcweb@ietf.org
|Subject: [rtcweb] Adopting draft-muthu-behave-consent-freshness?
|
|WG,
|
|This is a call for WG adoption of STUN Usage for Consent Freshness (draft-=
muthu-behave-consent-
|freshness-04). This document defines a STUN usage for consent freshness. A=
s this requires no protocol
|extensions we as intended users can define this usage in our WG. Such work=
 also matches our charter.
|The draft-ietf-rtcweb-security-arch-07 is normatively dependent on this ST=
UN usage.
|
|Document:
|https://datatracker.ietf.org/doc/draft-muthu-behave-consent-freshness/
|
|WG, please indicate your support or issues with adopting this document as =
WG item with a proposed
|milestone:
|
|Mar 2014 Send STUN Usage for Consent Freshness to IESG for publication as =
proposed standard.
|
|Cheers
|
|Magnus Westerlund
|
|----------------------------------------------------------------------
|Multimedia Technologies, Ericsson Research EAB/TVM
|----------------------------------------------------------------------
|Ericsson AB                | Phone  +46 10 7148287
|F=E4r=F6gatan 6                | Mobile +46 73 0949079
|SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
|----------------------------------------------------------------------
|
|_______________________________________________
|rtcweb mailing list
|rtcweb@ietf.org
|https://www.ietf.org/mailman/listinfo/rtcweb

From magnus.westerlund@ericsson.com  Tue Sep 10 22:59:18 2013
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Subject: [rtcweb] JSEP Walk Through on Wednesday the 18th of September at 17.00 CEST (8 AM PDT)
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WG,

There will be a new version of JSEP draft coming out Monday. As a follow
up to get as much progress on this topic as possible. The chairs and
authors are offering a informal walk through session on Wednesday the
18th. People will be strongly recommended to have reviewed the new draft.

The intention of the session is enable the authors to inform you about
the latest changes, any issues to consider and what needs further
discussion. The will also enable you as participant after your review of
the document to ask clarifying questions or for motivations behind
choices, and raise issues you have spotted.

The session is not intended to solve issues or judge consensus on part
of the draft. Discussions of what is the best solution etc, will to be
cut off.

This will be a webex conference per details below.

Cheers

Magnus Westerlund

Topic: RTCWeb - JSEP
Date: Wednesday, September 18, 2013
Time: 8:00 am, Pacific Daylight Time (San Francisco, GMT-07:00)
Meeting Number: 302 061 238
Meeting Password: ietf

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To view in other time zones or languages, please click the link:
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Access Code followed by the # sign.

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----------------------------------------------------------------
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(408) OR (919) AREA CODES
----------------------------------------------------------------
Please dial the local access number for your area from the list below:
- San Jose/Milpitas (408) area: 525-6800
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-- 

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From stephane.proust@orange.com  Wed Sep 11 02:19:50 2013
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From: <stephane.proust@orange.com>
To: 'Jean-Marc Valin' <jmvalin@mozilla.com>, "'Mo Zanaty (mzanaty)'" <mzanaty@cisco.com>
Thread-Topic: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
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Date: Wed, 11 Sep 2013 09:19:41 +0000
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
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>These are the codecs that would be recommended under such text. None of th=
em is really ideal for interactive audio

That is the reason why the proposed text only applies to "suitable" audio c=
odecs
" If other suitable audio codecs..."

That's also why it cannot be interpreted as a strictly normative text and w=
hy I also suggested (as it was previously suggested by the Chairs) to have =
some additional informative (and separate) text about what are those suitab=
le codecs to be considered and what is the rationale behind this (http://ww=
w.ietf.org/mail-archive/web/rtcweb/current/msg08501.html)

St=E9phane


-----Message d'origine-----
De=A0: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] De la part =
de Jean-Marc Valin
Envoy=E9=A0: mardi 10 septembre 2013 18:27
=C0=A0: Mo Zanaty (mzanaty)
Cc=A0: rtcweb@ietf.org
Objet=A0: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt

> To ensure a baseline level of interoperability between WebRTC clients,=20
> a minimum set of required codecs are specified below.
> If other suitable audio codecs are available to the browser to use, it=20
> is RECOMMENDED that they are also included in the offer in order to=20
> maximize the possibility to establish the session without the need for=20
> audio transcoding.

Let's look at the practical effect here, assuming people were to actually f=
ollow such a recommendation. The only audio codecs that are implemented in =
browsers but not available for WebRTC are the ones used for HTML5. Accordin=
g to http://html5test.com/ these codecs are Vorbis, MP3, AAC, and uncompres=
sed PCM. These are the codecs that would be recommended under such text. No=
ne of them is really ideal for interactive audio, none of them is implement=
ed in old phones AFAIK, but all of them would require time and effort to ad=
d to a browser's RTP stack. I really don't see the point here.

	Jean-Marc
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

___________________________________________________________________________=
______________________________________________

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Thank you.


From sanjay.mishra@verizon.com  Wed Sep 11 05:53:20 2013
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To: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>, "Dan Wing (dwing)" <dwing@cisco.com>, "Ram Mohan R (rmohanr)" <rmohanr@cisco.com>, "Tirumaleswar Reddy (tireddy)" <tireddy@cisco.com>
Date: Wed, 11 Sep 2013 08:53:03 -0400
Thread-Topic: [rtcweb] Adopting draft-muthu-behave-consent-freshness?
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Subject: Re: [rtcweb] Adopting draft-muthu-behave-consent-freshness?
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On Tuesday, September 10, 2013 11:51 PM Muthu Arul Mozhi Perumal <mperumal@=
cisco.com> wrote:
>The "why" part is mentioned in the first paragraph of the abstract:
>  Verification of peer consent before sending traffic is necessary in
>  WebRTC deployments to ensure that a malicious JavaScript cannot use
>  the browser as a platform for launching attacks.

> Are you suggesting expanding this? The abstract needs to be concise, howe=
ver.

I agree the first paragraph states the premise and that is good, however, t=
he second paragraph which lays out the purpose of this document, and since =
the purpose is both how and why, the second paragraph could state something=
 like this:

"This document describes both how and why a webRTC browser can and must ver=
ify peer consent to continue sending traffic and detect connection failure.=
"

Thanks
Sanjay

-----Original Message-----
From: Muthu Arul Mozhi Perumal (mperumal) [mailto:mperumal@cisco.com]=20
Sent: Tuesday, September 10, 2013 11:51 PM
To: Mishra, Sanjay; Dan Wing (dwing); Ram Mohan R (rmohanr); Tirumaleswar R=
eddy (tireddy)
Cc: rtcweb@ietf.org
Subject: RE: [rtcweb] Adopting draft-muthu-behave-consent-freshness?

|In the abstract, second paragraph, the text states "how" a WebRTC=20
|browser can verify peer consent and as I read through the rest of the=20
|document I also understood "why" this is needed. So, just a knit pick,=20
|it may be clear to the reader if the statement also includes why within=20
|the body of the Abstract.

The "why" part is mentioned in the first paragraph of the abstract:
   Verification of peer consent before sending traffic is necessary in
   WebRTC deployments to ensure that a malicious JavaScript cannot use
   the browser as a platform for launching attacks.

Are you suggesting expanding this? The abstract needs to be concise, howeve=
r.

|One minor typo in section 7. Please change "in-intended" to "un-intended".

Will fix it.

Thanks for the review..

Muthu=20

|-----Original Message-----
|From: Mishra, Sanjay [mailto:sanjay.mishra@verizon.com]
|Sent: Wednesday, September 11, 2013 12:38 AM
|To: Muthu Arul Mozhi Perumal (mperumal); Dan Wing (dwing); Ram Mohan R=20
|(rmohanr); Tirumaleswar Reddy
|(tireddy)
|Cc: rtcweb@ietf.org
|Subject: RE: [rtcweb] Adopting draft-muthu-behave-consent-freshness?
|
|Muthu et al --
|
|
|In the abstract, second paragraph, the text states "how" a WebRTC=20
|browser can verify peer consent and as I read through the rest of the=20
|document I also understood "why" this is needed. So, just a knit pick,=20
|it may be clear to the reader if the statement also includes why within th=
e body of the Abstract.
|
|One minor typo in section 7. Please change "in-intended" to "un-intended".
|
|Thanks
|Sanjay
|
|-----Original Message-----
|From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On=20
|Behalf Of Magnus Westerlund
|Sent: Monday, September 09, 2013 4:37 AM
|To: rtcweb@ietf.org
|Subject: [rtcweb] Adopting draft-muthu-behave-consent-freshness?
|
|WG,
|
|This is a call for WG adoption of STUN Usage for Consent Freshness=20
|(draft-muthu-behave-consent- freshness-04). This document defines a=20
|STUN usage for consent freshness. As this requires no protocol extensions =
we as intended users can define this usage in our WG. Such work also matche=
s our charter.
|The draft-ietf-rtcweb-security-arch-07 is normatively dependent on this ST=
UN usage.
|
|Document:
|https://datatracker.ietf.org/doc/draft-muthu-behave-consent-freshness/
|
|WG, please indicate your support or issues with adopting this document=20
|as WG item with a proposed
|milestone:
|
|Mar 2014 Send STUN Usage for Consent Freshness to IESG for publication as =
proposed standard.
|
|Cheers
|
|Magnus Westerlund
|
|----------------------------------------------------------------------
|Multimedia Technologies, Ericsson Research EAB/TVM
|----------------------------------------------------------------------
|Ericsson AB                | Phone  +46 10 7148287
|F=E4r=F6gatan 6                | Mobile +46 73 0949079
|SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
|----------------------------------------------------------------------
|
|_______________________________________________
|rtcweb mailing list
|rtcweb@ietf.org
|https://www.ietf.org/mailman/listinfo/rtcweb

From lijing80@huawei.com  Thu Sep 12 00:58:35 2013
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From: "Lijing (Jessie, Huawei)" <lijing80@huawei.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Adopting draft-muthu-behave-consent-freshness?
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Subject: [rtcweb] FW:  Adopting draft-muthu-behave-consent-freshness?
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Hi all,

As a newcomer to RTCWEB, I have some doubts after I have read the draft.=20

1. in "4.  Solution Overview", may be it would be better to clarify when to=
 send a STUN Binding Request(in the middle of a media session or before sen=
ding traffic, during sending traffic or only during silence periods) and wh=
ich side to send the request(controlling agent or both sides)?

2. in " 7.  Security Considerations", there are the following words. As whe=
n one agent receives STUN Binding Request, unlike the processing of indicat=
ion message, it have to do more processes to send the Response message, I a=
m not sure whether it is appreciate not to authenticate the source and resp=
ond directly. Is it more open to malicious attacks?

  "Once that connection to the remote
   peer has been established with ICE, the consent to continue sending
   traffic does not benefit from re-asserting that same username and
   password, so long as the senders and receiver's IP addresses remain
   the same (as they usually do)."

Neglect my words, if my understandings are wrong.

Best regards,

Jessie

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Magnus Westerlund
Sent: Monday, September 09, 2013 4:37 PM
To: rtcweb@ietf.org
Subject: [rtcweb] Adopting draft-muthu-behave-consent-freshness?

WG,

This is a call for WG adoption of STUN Usage for Consent Freshness
(draft-muthu-behave-consent-freshness-04). This document defines a STUN
usage for consent freshness. As this requires no protocol extensions we
as intended users can define this usage in our WG. Such work also
matches our charter. The draft-ietf-rtcweb-security-arch-07 is
normatively dependent on this STUN usage.

Document:
https://datatracker.ietf.org/doc/draft-muthu-behave-consent-freshness/

WG, please indicate your support or issues with adopting this document
as WG item with a proposed milestone:

Mar 2014 Send STUN Usage for Consent Freshness to IESG for publication
as proposed standard.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
F=E4r=F6gatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------

_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

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To: <stephane.proust@orange.com> <stephane.proust@orange.com>
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As I said last time this came up, we need to include language that =
ensures that these non-mandatory codecs don't=20
get used unless the mandatory ones are unavailable, or the web developer =
explicitly selected them.

I don't want to find that I'm getting amr-nb between a pair of phone =
based browsers and getting narrow band because
both the phones happen to have that codec available.


How about:=20

To ensure a baseline level of interoperability between WebRTC clients,=20=

a minimum set of required codecs are specified below.
If other suitable audio codecs are available to the browser to use, it=20=

is advised that they are also included in the offer at a lower priority =
than the required codecs in order to=20
maximize the possibility to establish the session without the need for=20=

audio transcoding when the required codecs are unavailable to one party.


- Of course this whole discussion highlights the need for a set of =
constraints (or an API) that let the web developer=20
express her preferences wrt codec selection.


Tim.

On 11 Sep 2013, at 10:19, <stephane.proust@orange.com> =
<stephane.proust@orange.com> wrote:

>> These are the codecs that would be recommended under such text. None =
of them is really ideal for interactive audio
>=20
> That is the reason why the proposed text only applies to "suitable" =
audio codecs
> " If other suitable audio codecs..."
>=20
> That's also why it cannot be interpreted as a strictly normative text =
and why I also suggested (as it was previously suggested by the Chairs) =
to have some additional informative (and separate) text about what are =
those suitable codecs to be considered and what is the rationale behind =
this (http://www.ietf.org/mail-archive/web/rtcweb/current/msg08501.html)
>=20
> St=E9phane
>=20
>=20
> -----Message d'origine-----
> De : rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] De la =
part de Jean-Marc Valin
> Envoy=E9 : mardi 10 septembre 2013 18:27
> =C0 : Mo Zanaty (mzanaty)
> Cc : rtcweb@ietf.org
> Objet : Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
>=20
>> To ensure a baseline level of interoperability between WebRTC =
clients,=20
>> a minimum set of required codecs are specified below.
>> If other suitable audio codecs are available to the browser to use, =
it=20
>> is RECOMMENDED that they are also included in the offer in order to=20=

>> maximize the possibility to establish the session without the need =
for=20
>> audio transcoding.
>=20
> Let's look at the practical effect here, assuming people were to =
actually follow such a recommendation. The only audio codecs that are =
implemented in browsers but not available for WebRTC are the ones used =
for HTML5. According to http://html5test.com/ these codecs are Vorbis, =
MP3, AAC, and uncompressed PCM. These are the codecs that would be =
recommended under such text. None of them is really ideal for =
interactive audio, none of them is implemented in old phones AFAIK, but =
all of them would require time and effort to add to a browser's RTP =
stack. I really don't see the point here.
>=20
> 	Jean-Marc
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> =
__________________________________________________________________________=
_______________________________________________
>=20
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>=20
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been modified, changed or falsified.
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>=20
> _______________________________________________
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> https://www.ietf.org/mailman/listinfo/rtcweb


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Subject: [rtcweb] Consensus on text around additional audio codecs
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I have reviewed all the emails about this since January 1. Many people =
have spoken in favor of the "recommended to support whatever the =
platform has that makes sense" style text and there has been close to no =
strong objections to this. My belief is that based on the email record, =
there is WG consensus to add text around that. Now there is some =
possibility there are a bunch of people that object given there has been =
no formal consensus call so here's what I would like to have happen.=20

I would like the editors of  to make the following change to =
draft-ietf-rtcweb-audio


---------------------
OLD:
=20
To ensure a baseline level of interoperability between WebRTC
clients, a minimum set of required codecs are specified below.
While this section specifies the codecs that will be mandated for all
WebRTC client implementations, it leaves the question of supporting
additional codecs to the will of the implementer.
=20
WebRTC clients are REQUIRED to implement the following audio codecs.
=20
NEW:
=20
To ensure a baseline level of interoperability between WebRTC
clients, a minimum set of required codecs are specified below.
If other suitable audio codecs are available for the browser to use,
it is RECOMMENDED that they are also be included in the offer in order
to maximize the possibility to establish the session without the need
for audio transcoding.
=20
WebRTC clients are REQUIRED to implement the following audio codecs.

----------------------

This will then get publish as a new draft. The WG can comment on that =
draft and if lots of people have an issue with that text, then we can =
allocate time to discuss it at a future meeting and, if needed, make a =
more formal consensus call so that no one misses it.=20

My hope is that the people that care about this have expressed their =
opinions on the list and that when the draft comes out, the consensus =
will not change. If that turns out to be wrong, folks can still object =
about text in the draft.=20

I will note that several people have pointed out this text is fairly =
weak. However, at this point in time, I don't think there is consensus =
to add stronger language. (Note: I also don't see consensus to not add =
stronger language).=20

Thanks,
Cullen <with my co-chair hat on>


PS - I have received off list suggestions that this is an ideal place to =
use rfc6919 language - as chair I MUST strongly encourage people to =
stick with the the more traditional rfc2119 language. =20



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From: Bernard Aboba <bernard_aboba@hotmail.com>
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--_9f76e178-2df1-4484-9753-405e59534448_
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Tim said:=20

> - Of course this whole discussion highlights the need for a set of constr=
aints (or an API) that let the web developer=20
> express her preferences wrt codec selection.

[BA] This - and the need to specify the default ordering of codecs within t=
he API=2C so that by default the MTI codecs will be offered at a higher pri=
ority.=20

> How about:=20
>=20
> To ensure a baseline level of interoperability between WebRTC clients=2C=
=20
> a minimum set of required codecs are specified below.
> If other suitable audio codecs are available to the browser to use=2C it=
=20
> is advised that they are also included in the offer at a lower priority t=
han the required codecs in order to=20
> maximize the possibility to establish the session without the need for=20
> audio transcoding when the required codecs are unavailable to one party.
=20
[BA] On reading this I'm unsure whether by "offer" you mean the SDP used in=
 the API=2C of if you're talking about an offer sent over the wire.   The t=
ext makes more sense to me if you're talking about the API.=20
=20
If the developer decides they want to offer amr-nb as the preferred codec i=
n an Offer sent over the wire=2C that's their choice=2C as long as they don=
't delete the MTI codecs from the offer=2C which could potentially cause an=
 interoperability failure.   However=2C if the developer is just trusting t=
he API to do something sensible=2C then by default I'd say that one of the =
MTI codecs should be negotiated by default.  In particular=2C in the browse=
r-browser case I'd advocate that Opus by negotiated by default=2C for audio=
. =20

 		 	   		  =

--_9f76e178-2df1-4484-9753-405e59534448_
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<html>
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margin:0px=3B
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body.hmmessage
{
font-size: 12pt=3B
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--></style></head>
<body class=3D'hmmessage'><div dir=3D'ltr'>Tim said: <BR><br>&gt=3B - Of co=
urse this whole discussion highlights the need for a set of constraints (or=
 an API) that let the web developer <br>&gt=3B express her preferences wrt =
codec selection.<br><BR>[BA]&nbsp=3BThis - and the need to specify the defa=
ult ordering of&nbsp=3Bcodecs within the API=2C so that by default the MTI =
codecs will be offered at a higher priority. <BR><br>&gt=3B How about: <br>=
&gt=3B <br>&gt=3B To ensure a baseline level of interoperability between We=
bRTC clients=2C <br>&gt=3B a minimum set of required codecs are specified b=
elow.<br>&gt=3B If other suitable audio codecs are available to the browser=
 to use=2C it <br>&gt=3B is advised that they are also included in the offe=
r at a lower priority than the required codecs in order to <br>&gt=3B maxim=
ize the possibility to establish the session without the need for <br>&gt=
=3B audio transcoding when the required codecs are unavailable to one party=
.<BR>&nbsp=3B<BR>[BA]&nbsp=3BOn reading this I'm unsure whether by&nbsp=3B"=
offer" you mean the SDP used in the API=2C of if you're talking about an of=
fer sent over the wire.&nbsp=3B&nbsp=3B The text makes more sense to me if =
you're talking about the API. <BR>&nbsp=3B<BR>If the developer decides they=
 want to offer&nbsp=3Bamr-nb as the preferred codec in an Offer sent over t=
he wire=2C that's their choice=2C as long as they don't delete the MTI code=
cs from the offer=2C which could potentially cause an interoperability fail=
ure.&nbsp=3B&nbsp=3B However=2C if the developer is just trusting the API t=
o do something sensible=2C then by default I'd say that one of the MTI code=
cs should be negotiated by default.&nbsp=3B In particular=2C in the browser=
-browser case I'd advocate that Opus by negotiated by default=2C for audio.=
&nbsp=3B <br><BR> 		 	   		  </div></body>
</html>=

--_9f76e178-2df1-4484-9753-405e59534448_--

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Date: Thu, 12 Sep 2013 15:49:42 -0400
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From: Roman Shpount <roman@telurix.com>
To: Bernard Aboba <bernard_aboba@hotmail.com>
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
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On Thu, Sep 12, 2013 at 3:43 PM, Bernard Aboba <bernard_aboba@hotmail.com>wrote:

>
> [BA] This - and the need to specify the default ordering of codecs within
> the API, so that by default the MTI codecs will be offered at a higher
> priority.
>

I am not sure that MTI codecs should always be highest priority. If
optional codec would be developed of higher quality then Opus or the yet to
be defined MTI video codec (for instance if someone created Opus-NG or in
case of VP9 or H.265), then there is no reason for this codec not to be
offered at higher priority then MTI.
_____________
Roman Shpount

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On T=
hu, Sep 12, 2013 at 3:43 PM, Bernard Aboba <span dir=3D"ltr">&lt;<a href=3D=
"mailto:bernard_aboba@hotmail.com" target=3D"_blank">bernard_aboba@hotmail.=
com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left:1px solid rgb(204,204,204);padding-left:1ex">


<div><div dir=3D"ltr"><div class=3D"im"><br></div>[BA]=A0This - and the nee=
d to specify the default ordering of=A0codecs within the API, so that by de=
fault the MTI codecs will be offered at a higher priority. <br></div></div>=
</blockquote>
</div><br></div><div class=3D"gmail_extra">I am not sure that MTI codecs sh=
ould always be highest priority. If optional codec would be developed of hi=
gher quality then Opus or the yet to be defined MTI video codec  (for insta=
nce if someone created Opus-NG or in case of VP9 or H.265), then there is n=
o reason for this codec not to be offered at higher priority then MTI.<br>
</div><div class=3D"gmail_extra"><div>_____________<br>Roman Shpount</div>
<br></div></div>

--f46d04447fe504b42204e6350ad1--

From bernard_aboba@hotmail.com  Thu Sep 12 12:57:07 2013
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To: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Thu, 12 Sep 2013 12:57:00 -0700
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Subject: Re: [rtcweb] draft-ietf-rtcweb-transports-01 TURN/IPV6 RFC 6156.
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Andrew Hutton said:=20

> Currently draft-ietf-rtcweb-transports-01 does not say anything about IPV=
6 which I assume it should. Specifically I am thinking that it needs to sta=
te a requirement to support RFC6156 support "Traversal Using Relays around =
NAT (TURN) Extension for IPv6".
>=20
> I am not sure how much we need to say about webrtc client procedures arou=
nd RFC6156 and whether they should be included in the draft-ietf-rtcweb-tra=
nsport or whether it is something we should add to our nat/firewall draft (=
draft-hutton-rtcweb-nat-firewall-considerations).  Any opinions on this?

[BA] Adding it to the NAT/Firewall draft makes the most sense to me.  Among=
 other things=2C it might provide more of an opportunity to get into some o=
f the "Happy Eyeballs" issues that have come up=2C such as relative priorit=
ization of IPv4 versus IPv6. =20

 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>Andrew Hutton said: <BR><br>&gt=
=3B Currently draft-ietf-rtcweb-transports-01 does not say anything about I=
PV6 which I assume it should. Specifically I am thinking that it needs to s=
tate a requirement to support RFC6156 support "Traversal Using Relays aroun=
d NAT (TURN) Extension for IPv6".<br>&gt=3B <br>&gt=3B I am not sure how mu=
ch we need to say about webrtc client procedures around RFC6156 and whether=
 they should be included in the draft-ietf-rtcweb-transport or whether it i=
s something we should add to our nat/firewall draft (draft-hutton-rtcweb-na=
t-firewall-considerations).  Any opinions on this?<BR><br>[BA] Adding it to=
 the NAT/Firewall draft makes the most sense to me.&nbsp=3B Among other thi=
ngs=2C it might provide more of an opportunity to get into some of the "Hap=
py Eyeballs" issues that have come up=2C such as relative&nbsp=3Bprioritiza=
tion of IPv4 versus IPv6.&nbsp=3B&nbsp=3B<br><BR> 		 	   		  </div></body>
</html>=

--_a796e556-5b05-4be3-9481-4be2829ec8f9_--

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Andrew Hutton said:=20

> Currently draft-ietf-rtcweb-transports-01 does not say anything about IPV=
6 which I assume it should. Specifically I am thinking that it needs to sta=
te a requirement to support RFC6156 support "Traversal Using Relays around =
NAT (TURN) Extension for IPv6".
>=20
> I am not sure how much we need to say about webrtc client procedures arou=
nd RFC6156 and whether they should be included in the draft-ietf-rtcweb-tra=
nsport or whether it is something we should add to our nat/firewall draft (=
draft-hutton-rtcweb-nat-firewall-considerations).  Any opinions on this?

[BA] Adding it to the NAT/Firewall draft makes the most sense to me.  Among=
 other things=2C it might provide more of an opportunity to get into some o=
f the "Happy Eyeballs" issues that have come up=2C such as relative priorit=
ization of IPv4 versus IPv6. =20

 		 	   		  =

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--></style></head>
<body class=3D'hmmessage'><div dir=3D'ltr'>Andrew Hutton said: <BR><br>&gt=
=3B Currently draft-ietf-rtcweb-transports-01 does not say anything about I=
PV6 which I assume it should. Specifically I am thinking that it needs to s=
tate a requirement to support RFC6156 support "Traversal Using Relays aroun=
d NAT (TURN) Extension for IPv6".<br>&gt=3B <br>&gt=3B I am not sure how mu=
ch we need to say about webrtc client procedures around RFC6156 and whether=
 they should be included in the draft-ietf-rtcweb-transport or whether it i=
s something we should add to our nat/firewall draft (draft-hutton-rtcweb-na=
t-firewall-considerations).  Any opinions on this?<BR><br>[BA] Adding it to=
 the NAT/Firewall draft makes the most sense to me.&nbsp=3B Among other thi=
ngs=2C it might provide more of an opportunity to get into some of the "Hap=
py Eyeballs" issues that have come up=2C such as relative&nbsp=3Bprioritiza=
tion of IPv4 versus IPv6.&nbsp=3B&nbsp=3B<br><BR> 		 	   		  </div></body>
</html>=

--_d6856e87-7dac-4d49-a6ee-a09945305943_--

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Subject: [rtcweb]  #30: Wiretapping
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#30: Wiretapping

 In several sections of the document, the phrase "It is essential
 that the communication cannot be wiretapped [RFC2804]" is used.
 The phrase is used in Sections 3.2.1.1, 3.2.11.1, 3.2.12.1, 3.2.13.1,
 3.3.1.1 and 3.2.3.1, but not in 3.2.14.1 (which also does not
 reference F20).

 Given the recent revelations, and the discussion of SRTP/SDES at
 IETF 87, I would suggest the following:

 a. Use of more precise terminology than what is in F20.  For example,
 I think what we are asking for in many of the F20 scenarios is
 per-packet encryption and integrity protection of media,
 utilizing keys known only by the endpoints, as well as support
 for perfect forward secrecy.

 b. Inclusion of a reference to F20 in Section 3.2.14.1 (Distributed
 Music Band).  Not sure why protection against snooping wouldn't be
 relevant in this use case (there are countries where musicians
 have been severely punished).

 c. Consideration of the requirement in gateway scenarios. For gateway
 scenarios such as 3.3.1.1, the e2e key management
 requirement probably isn't realistic, so maybe we need to
 just cite F35/F36 for that case.

-- 
-------------------------------------+-------------------------------------
 Reporter:                           |      Owner:  draft-ietf-rtcweb-use-
  bernard_aboba@hotmail.com          |  cases-and-
     Type:  defect                   |  requirements@tools.ietf.org
 Priority:  critical                 |     Status:  new
Component:  rtp-usage                |  Milestone:  milestone1
 Severity:  In WG Last Call          |    Version:  1.0
                                     |   Keywords:
-------------------------------------+-------------------------------------

Ticket URL: <http://tools.ietf.org/wg/rtcweb/trac/ticket/30>
rtcweb <http://tools.ietf.org/rtcweb/>


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#30: Wiretapping

Changes (by bernard_aboba@hotmail.com):

 * component:  rtp-usage => use-cases-and-requirements


-- 
-------------------------------------+-------------------------------------
 Reporter:                           |       Owner:  draft-ietf-rtcweb-use-
  bernard_aboba@hotmail.com          |  cases-and-
     Type:  defect                   |  requirements@tools.ietf.org
 Priority:  critical                 |      Status:  new
Component:  use-cases-and-           |   Milestone:  milestone1
  requirements                       |     Version:  1.0
 Severity:  In WG Last Call          |  Resolution:
 Keywords:                           |
-------------------------------------+-------------------------------------

Ticket URL: <https://tools.ietf.org/wg/rtcweb/trac/ticket/30#comment:1>
rtcweb <http://tools.ietf.org/rtcweb/>


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#31: Miscellaneous

 Section 3.2.1.1

 "The application gives the users the opportunity to stop it from exposing
 the host IP address to the application of the other user."

 [BA] This text suggests a requirement for the browser to ask user
 consent, which seems different from what is in F39:

 "The browser must make it possible to set up a call between two parties
 without one party learning the other party's host IP address."

 IMHO, F39 seems somewhat closer to the requirement we want.

 Section 3.2.10.1

 "NOTE: More profiling of what this means may be needed."

 [BA] Section 3.3.1.1 already describes a telephony gateway scenario,
 but for some reason that section doesn't cite requirement F27 (SIP)
 or A20.  FWIW, I'm not sure what this scenario adds, because RFC 3261
 inter-domain federation isn't widely deployed, as opposed to E.164
 inter-connect.

-- 
-------------------------------------+-------------------------------------
 Reporter:                           |      Owner:  draft-ietf-rtcweb-use-
  bernard_aboba@hotmail.com          |  cases-and-
     Type:  defect                   |  requirements@tools.ietf.org
 Priority:  minor                    |     Status:  new
Component:  use-cases-and-           |  Milestone:  milestone1
  requirements                       |    Version:  1.0
 Severity:  In WG Last Call          |   Keywords:
-------------------------------------+-------------------------------------

Ticket URL: <http://tools.ietf.org/wg/rtcweb/trac/ticket/31>
rtcweb <http://tools.ietf.org/rtcweb/>


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#32: Section 3.2.1.1

 "One user has an unreliable Internet connection.  It sometimes loses
 packets, and sometimes goes down completely."

 [BA] I assume that this is generating requirement F8 (The browser must
 detect when a stream from a peer is not received anymore).  Since we
 are using a consent mechanism to send mechanism,  you might
 consider rephrasing F8 to "The browser must detect peer loss of
 connectivity."

 However, if we just focus at the packet loss issue, this might
 suggest a requirement for statistics (F38) or repair of some kind
 (e.g. either RTX or FEC).

-- 
-------------------------------------+-------------------------------------
 Reporter:                           |      Owner:  draft-ietf-rtcweb-use-
  bernard_aboba@hotmail.com          |  cases-and-
     Type:  defect                   |  requirements@tools.ietf.org
 Priority:  minor                    |     Status:  new
Component:  use-cases-and-           |  Milestone:  milestone1
  requirements                       |    Version:
 Severity:  In WG Last Call          |   Keywords:
-------------------------------------+-------------------------------------

Ticket URL: <http://tools.ietf.org/wg/rtcweb/trac/ticket/32>
rtcweb <http://tools.ietf.org/rtcweb/>


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From: "Lijing (Jessie, Huawei)" <lijing80@huawei.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Adopting draft-muthu-behave-consent-freshness?
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Hi all,

As a newcomer to RTCWEB, I have some doubts after I have read the draft.=20

1. in "4.  Solution Overview", may be it would be better to clarify when to=
 send a STUN Binding Request(in the middle of a media session or before sen=
ding traffic, during sending traffic or only during silence periods) and wh=
ich side to send the request(controlling agent or both sides)? During the s=
ession, there are RTP/RTCP media streams which can keep the NAT mapping and=
 confirm the connectivity state of the media plane. Before the session, we =
can use the ICE connectivity checks.=20
So according to my understanding, the STUN Binding Request should only be s=
ent during the silence periods. A media connectivity check with a request a=
nd a response during a session, is this the main idea of this document?

2. in " 7.  Security Considerations", there are the following words.=20
  "Once that connection to the remote
   peer has been established with ICE, the consent to continue sending
   traffic does not benefit from re-asserting that same username and
   password, so long as the senders and receiver's IP addresses remain
   the same (as they usually do)."

   As when one agent receives STUN Binding Request, unlike the processing o=
f STUN Binding Indication message, it have to do more processes to send the=
 Response message, I am not sure whether it is appreciate not to authentica=
te the source and respond directly. Is it more likely to receive malicious =
attacks?

Neglect my words, if my understandings are wrong.

Best regards,

Jessie


-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Magnus Westerlund
Sent: Monday, September 09, 2013 4:37 PM
To: rtcweb@ietf.org
Subject: [rtcweb] Adopting draft-muthu-behave-consent-freshness?

WG,

This is a call for WG adoption of STUN Usage for Consent Freshness
(draft-muthu-behave-consent-freshness-04). This document defines a STUN
usage for consent freshness. As this requires no protocol extensions we
as intended users can define this usage in our WG. Such work also
matches our charter. The draft-ietf-rtcweb-security-arch-07 is
normatively dependent on this STUN usage.

Document:
https://datatracker.ietf.org/doc/draft-muthu-behave-consent-freshness/

WG, please indicate your support or issues with adopting this document
as WG item with a proposed milestone:

Mar 2014 Send STUN Usage for Consent Freshness to IESG for publication
as proposed standard.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
F=E4r=F6gatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------

_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

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On 09/13/2013 02:08 AM, rtcweb issue tracker wrote:
> #30: Wiretapping
>
>  In several sections of the document, the phrase "It is essential
>  that the communication cannot be wiretapped [RFC2804]" is used.
>  The phrase is used in Sections 3.2.1.1, 3.2.11.1, 3.2.12.1, 3.2.13.1,
>  3.3.1.1 and 3.2.3.1, but not in 3.2.14.1 (which also does not
>  reference F20).
>
>  Given the recent revelations, and the discussion of SRTP/SDES at
>  IETF 87, I would suggest the following:
>
>  a. Use of more precise terminology than what is in F20.  For example,
>  I think what we are asking for in many of the F20 scenarios is
>  per-packet encryption and integrity protection of media,
>  utilizing keys known only by the endpoints, as well as support
>  for perfect forward secrecy.
I think this is a too low level of detail for an use cases document. It
does belong in the security / security-arch document, where we detail
both what the requirement implies and how the requirement is implemented.

If we want to add details here, I suggest adding "against an adversary
that has access to the signalling path and/or access to one of the end
systems after the communication has occured" (which would lead to a
requirement for DTLS key exchange and PFS).

>
>  b. Inclusion of a reference to F20 in Section 3.2.14.1 (Distributed
>  Music Band).  Not sure why protection against snooping wouldn't be
>  relevant in this use case (there are countries where musicians
>  have been severely punished).
>
>  c. Consideration of the requirement in gateway scenarios. For gateway
>  scenarios such as 3.3.1.1, the e2e key management
>  requirement probably isn't realistic, so maybe we need to
>  just cite F35/F36 for that case.
>
If we use the language I suggested above, noting that this spec can do
nothing to secure things beyond the gateway is probably enough ("e2e"
key management in this case would mean "end to gateway" - it's up to the
gateway and what's beyond it to protect the keys after that).


-- 
Surveillance is pervasive. Go Dark.


From simon.perreault@viagenie.ca  Fri Sep 13 02:23:52 2013
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Subject: Re: [rtcweb] [BEHAVE] New Version Notification for	draft-chenxin-behave-turn-websocket-01.txt
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Le 2013-09-13 10:58, Chenxin (Xin) a écrit :
> 	We have been working on a new version of the TURN over Websocket draft, which is now available at:
>
> 	http://www.ietf.org/id/draft-chenxin-behave-turn-websocket-01.txt

There are two major things missing in this draft:

- How are TURN UDP "channels" supported? (See RFC 5766 section 2.5.)

- How are TURN TCP "data connections" supported? (See RFC 6062 section 3.)

I think the answers to these questions will greatly affect the overall 
design of TURN-over-WebSocket.

Simon
-- 
DTN made easy, lean, and smart --> http://postellation.viagenie.ca
NAT64/DNS64 open-source        --> http://ecdysis.viagenie.ca
STUN/TURN server               --> http://numb.viagenie.ca

From ted.ietf@gmail.com  Fri Sep 13 09:52:28 2013
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From: Ted Hardie <ted.ietf@gmail.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>, Cullen Jennings <fluffy@cisco.com>,  Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: [rtcweb] Video Codec Selection Plan
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--bcaec519694bc5e53004e646adb0
Content-Type: text/plain; charset=ISO-8859-1

WG,

The chairs have created a plan for how to perform the Video Codec
selection in our WG. The chairs are asking for review of our plan on
how to undertake the mandatory-to-implement video codec selection.
We'd much prefer to have comments on the mechanics before they begin,
so please review now.  Proponents of a particular proposal should
note both the actions required and the timelines proposed.

The main goal of this plan is to hold a consensus call on which of
the proposed alternatives we as a WG should select at one of the WG
sessions in Vancouver. Such a consensus call will of course be
verified on the mailing list for anyone who can't participate. The
chairs will recuse themselves from judging this particular
consensus.

In the WG session each codec proposal will be allowed an equal amount
of time to highlight the arguments for their proposal. After that a
there will be a slot for discussion and clarifying questions.

To enable the WG participants to get answers to any questions, the
proposals in draft form and any supporting material MUST be made
available by 6th of October. This is to ensure that the WG
participants can verify or object to any claims or statements in
the proposal material prior to the WG session. We chairs would really
not like to see the proponents bring up new arguments at their
presentation. Also the WG participants are expected to raise any
arguments on the list ahead of time to enable the proponents to
respond to such arguments.

The proposed consensus questions will be of the following form:

1. If you support H.264 as the mandatory to implement codec or are
willing to live with it as the MTI, please raise your hand now.

2. If you support VP8 as the mandatory to implement codec or are
willing to live with it as the MTI, please raise your hand now.

You may indicate support on both questions and we encourage you to do
so if you can live with either, even if you have a preference for one
over the other.

Additional proposals than the previous ones are welcome, but must be
submitted as draft and their proponents must notify the chairs no later
than the 6th of October that they also have a candidate proposal.

In case the WG fails to reach consensus we chairs propose that we use
the alternative decision process as discussed in RFC3929. The method
and its usage will be discussed on the list should the WG not
establish consensus on a proposal for mandatory to implement video codec.

regards,

Magnus,  Cullen, and Ted

--bcaec519694bc5e53004e646adb0
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div>WG,<br><br>The chairs have created a plan for how to =
perform the Video Codec<br>selection in our WG. The chairs are asking for r=
eview of our plan on<br>how to undertake the mandatory-to-implement video c=
odec selection.<br>
We&#39;d much prefer to have comments on the mechanics before they begin,<b=
r>so please review now.=A0 Proponents of a particular proposal should<br>no=
te both the actions required and the timelines proposed.<br><br>The main go=
al of this plan is to hold a consensus call on which of<br>
the proposed alternatives we as a WG should select at one of the WG<br>sess=
ions in Vancouver. Such a consensus call will of course be<br>verified on t=
he mailing list for anyone who can&#39;t participate. The<br>chairs will re=
cuse themselves from judging this particular<br>
consensus.<br><br>In the WG session each codec proposal will be allowed an =
equal amount<br>of time to highlight the arguments for their proposal. Afte=
r that a<br>there will be a slot for discussion and clarifying questions.<b=
r>
<br>To enable the WG participants to get answers to any questions, the<br>p=
roposals in draft form and any supporting material MUST be made<br>availabl=
e by 6th of October. This is to ensure that the WG<br>participants can veri=
fy or object to any claims or statements in<br>
the proposal material prior to the WG session. We chairs would really<br>no=
t like to see the proponents bring up new arguments at their<br>presentatio=
n. Also the WG participants are expected to raise any<br>arguments on the l=
ist ahead of time to enable the proponents to<br>
respond to such arguments.<br><br>The proposed consensus questions will be =
of the following form:<br><br>1. If you support H.264 as the mandatory to i=
mplement codec or are<br>willing to live with it as the MTI, please raise y=
our hand now.<br>
<br>2. If you support VP8 as the mandatory to implement codec or are<br>wil=
ling to live with it as the MTI, please raise your hand now.<br><br>You may=
 indicate support on both questions and we encourage you to do<br>so if you=
 can live with either, even if you have a preference for one<br>
over the other.<br><br>Additional proposals than the previous ones are welc=
ome, but must be<br>submitted as draft and their proponents must notify the=
 chairs no later<br>than the 6th of October that they also have a candidate=
 proposal.<br>
<br>In case the WG fails to reach consensus we chairs propose that we use<b=
r>the alternative decision process as discussed in RFC3929. The method<br>a=
nd its usage will be discussed on the list should the WG not<br>establish c=
onsensus on a proposal for mandatory to implement video codec.<br>
<br></div><div>regards,<br><br></div>Magnus,=A0 Cullen, and Ted<br></div>

--bcaec519694bc5e53004e646adb0--

From matthew@matthew.at  Fri Sep 13 09:59:33 2013
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From: Matthew Kaufman <matthew@matthew.at>
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Subject: Re: [rtcweb] Video Codec Selection Plan
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This is a multi-part message in MIME format.
--------------030600080309060108030304
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1. Please explain why this decision is an IETF decision and not a W3C 
decision, given that it is a choice about what to mandate in the 
presentation layer as codec, NOT what to mandate for the on-the-wire 
transport format (which has already been selected as DTLS-SRTP with an 
appropriate IETF payload based on the application)

2. Should I expect "room-packing" for this "show of hands" (which I 
don't believe is the typical "hum" process for the IETF, either) as 
happened during the SDES discussion last time, and therefore need to 
bring as many people who've not participated in RTCWEB previously but 
who want my codec to succeed as I can afford to fly to Vancouver?

Matthew Kaufman

On 9/13/2013 9:52 AM, Ted Hardie wrote:
> WG,
>
> The chairs have created a plan for how to perform the Video Codec
> selection in our WG. The chairs are asking for review of our plan on
> how to undertake the mandatory-to-implement video codec selection.
> We'd much prefer to have comments on the mechanics before they begin,
> so please review now.  Proponents of a particular proposal should
> note both the actions required and the timelines proposed.
>
> The main goal of this plan is to hold a consensus call on which of
> the proposed alternatives we as a WG should select at one of the WG
> sessions in Vancouver. Such a consensus call will of course be
> verified on the mailing list for anyone who can't participate. The
> chairs will recuse themselves from judging this particular
> consensus.
>
> In the WG session each codec proposal will be allowed an equal amount
> of time to highlight the arguments for their proposal. After that a
> there will be a slot for discussion and clarifying questions.
>
> To enable the WG participants to get answers to any questions, the
> proposals in draft form and any supporting material MUST be made
> available by 6th of October. This is to ensure that the WG
> participants can verify or object to any claims or statements in
> the proposal material prior to the WG session. We chairs would really
> not like to see the proponents bring up new arguments at their
> presentation. Also the WG participants are expected to raise any
> arguments on the list ahead of time to enable the proponents to
> respond to such arguments.
>
> The proposed consensus questions will be of the following form:
>
> 1. If you support H.264 as the mandatory to implement codec or are
> willing to live with it as the MTI, please raise your hand now.
>
> 2. If you support VP8 as the mandatory to implement codec or are
> willing to live with it as the MTI, please raise your hand now.
>
> You may indicate support on both questions and we encourage you to do
> so if you can live with either, even if you have a preference for one
> over the other.
>
> Additional proposals than the previous ones are welcome, but must be
> submitted as draft and their proponents must notify the chairs no later
> than the 6th of October that they also have a candidate proposal.
>
> In case the WG fails to reach consensus we chairs propose that we use
> the alternative decision process as discussed in RFC3929. The method
> and its usage will be discussed on the list should the WG not
> establish consensus on a proposal for mandatory to implement video codec.
>
> regards,
>
> Magnus,  Cullen, and Ted
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


--------------030600080309060108030304
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<html>
  <head>
    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
  </head>
  <body text="#000000" bgcolor="#FFFFFF">
    <div class="moz-cite-prefix">1. Please explain why this decision is
      an IETF decision and not a W3C decision, given that it is a choice
      about what to mandate in the presentation layer as codec, NOT what
      to mandate for the on-the-wire transport format (which has already
      been selected as DTLS-SRTP with an appropriate IETF payload based
      on the application)<br>
      <br>
      2. Should I expect "room-packing" for this "show of hands" (which
      I don't believe is the typical "hum" process for the IETF, either)
      as happened during the SDES discussion last time, and therefore
      need to bring as many people who've not participated in RTCWEB
      previously but who want my codec to succeed as I can afford to fly
      to Vancouver?<br>
      <br>
      Matthew Kaufman<br>
      <br>
      On 9/13/2013 9:52 AM, Ted Hardie wrote:<br>
    </div>
    <blockquote
cite="mid:CA+9kkMAvdtq_gufKmDNCNCL+kKcxyi0MGUoVHetd9_DzbEdEnA@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div>WG,<br>
          <br>
          The chairs have created a plan for how to perform the Video
          Codec<br>
          selection in our WG. The chairs are asking for review of our
          plan on<br>
          how to undertake the mandatory-to-implement video codec
          selection.<br>
          We'd much prefer to have comments on the mechanics before they
          begin,<br>
          so please review now.&nbsp; Proponents of a particular proposal
          should<br>
          note both the actions required and the timelines proposed.<br>
          <br>
          The main goal of this plan is to hold a consensus call on
          which of<br>
          the proposed alternatives we as a WG should select at one of
          the WG<br>
          sessions in Vancouver. Such a consensus call will of course be<br>
          verified on the mailing list for anyone who can't participate.
          The<br>
          chairs will recuse themselves from judging this particular<br>
          consensus.<br>
          <br>
          In the WG session each codec proposal will be allowed an equal
          amount<br>
          of time to highlight the arguments for their proposal. After
          that a<br>
          there will be a slot for discussion and clarifying questions.<br>
          <br>
          To enable the WG participants to get answers to any questions,
          the<br>
          proposals in draft form and any supporting material MUST be
          made<br>
          available by 6th of October. This is to ensure that the WG<br>
          participants can verify or object to any claims or statements
          in<br>
          the proposal material prior to the WG session. We chairs would
          really<br>
          not like to see the proponents bring up new arguments at their<br>
          presentation. Also the WG participants are expected to raise
          any<br>
          arguments on the list ahead of time to enable the proponents
          to<br>
          respond to such arguments.<br>
          <br>
          The proposed consensus questions will be of the following
          form:<br>
          <br>
          1. If you support H.264 as the mandatory to implement codec or
          are<br>
          willing to live with it as the MTI, please raise your hand
          now.<br>
          <br>
          2. If you support VP8 as the mandatory to implement codec or
          are<br>
          willing to live with it as the MTI, please raise your hand
          now.<br>
          <br>
          You may indicate support on both questions and we encourage
          you to do<br>
          so if you can live with either, even if you have a preference
          for one<br>
          over the other.<br>
          <br>
          Additional proposals than the previous ones are welcome, but
          must be<br>
          submitted as draft and their proponents must notify the chairs
          no later<br>
          than the 6th of October that they also have a candidate
          proposal.<br>
          <br>
          In case the WG fails to reach consensus we chairs propose that
          we use<br>
          the alternative decision process as discussed in RFC3929. The
          method<br>
          and its usage will be discussed on the list should the WG not<br>
          establish consensus on a proposal for mandatory to implement
          video codec.<br>
          <br>
        </div>
        <div>regards,<br>
          <br>
        </div>
        Magnus,&nbsp; Cullen, and Ted<br>
      </div>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
  </body>
</html>

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Subject: Re: [rtcweb] #31: Miscellaneous
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On 12 September 2013 17:18, rtcweb issue tracker
<trac+rtcweb@trac.tools.ietf.org> wrote:
>  [BA] This text suggests a requirement for the browser to ask user
>  consent, which seems different from what is in F39:

I don't know what question I would ask in this case.  Well, at least
as far as being able to be understood goes, that is.  I'd recommend
avoiding this sort of formulation, and would prefer that it simply be
possible to avoid exposing IP addresses.

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Subject: Re: [rtcweb] Video Codec Selection Plan
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Ted --
The H.264 vs. VP8 discussion has at this point been "overtaken by events"=
=2C so that the proposed plan below would be a waste of everybody's time=2C=
 about as relevant as debating the fashion merits of bell bottoms versus ne=
hru jackets. =20
With Google having announced plans to implement VP9 (potentially with SVC)=
=2C and with the recent ratification of H.265=2C the industry has moved on=
=2C and so should RTCWEB.=20
Date: Fri=2C 13 Sep 2013 09:52:24 -0700
From: ted.ietf@gmail.com
To: rtcweb@ietf.org=3B fluffy@cisco.com=3B magnus.westerlund@ericsson.com
Subject: [rtcweb] Video Codec Selection Plan

WG=2C

The chairs have created a plan for how to perform the Video Codec
selection in our WG. The chairs are asking for review of our plan on
how to undertake the mandatory-to-implement video codec selection.
=0A=
We'd much prefer to have comments on the mechanics before they begin=2C
so please review now.  Proponents of a particular proposal should
note both the actions required and the timelines proposed.

The main goal of this plan is to hold a consensus call on which of
=0A=
the proposed alternatives we as a WG should select at one of the WG
sessions in Vancouver. Such a consensus call will of course be
verified on the mailing list for anyone who can't participate. The
chairs will recuse themselves from judging this particular
=0A=
consensus.

In the WG session each codec proposal will be allowed an equal amount
of time to highlight the arguments for their proposal. After that a
there will be a slot for discussion and clarifying questions.
=0A=

To enable the WG participants to get answers to any questions=2C the
proposals in draft form and any supporting material MUST be made
available by 6th of October. This is to ensure that the WG
participants can verify or object to any claims or statements in
=0A=
the proposal material prior to the WG session. We chairs would really
not like to see the proponents bring up new arguments at their
presentation. Also the WG participants are expected to raise any
arguments on the list ahead of time to enable the proponents to
=0A=
respond to such arguments.

The proposed consensus questions will be of the following form:

1. If you support H.264 as the mandatory to implement codec or are
willing to live with it as the MTI=2C please raise your hand now.
=0A=

2. If you support VP8 as the mandatory to implement codec or are
willing to live with it as the MTI=2C please raise your hand now.

You may indicate support on both questions and we encourage you to do
so if you can live with either=2C even if you have a preference for one
=0A=
over the other.

Additional proposals than the previous ones are welcome=2C but must be
submitted as draft and their proponents must notify the chairs no later
than the 6th of October that they also have a candidate proposal.
=0A=

In case the WG fails to reach consensus we chairs propose that we use
the alternative decision process as discussed in RFC3929. The method
and its usage will be discussed on the list should the WG not
establish consensus on a proposal for mandatory to implement video codec.
=0A=

regards=2C

Magnus=2C  Cullen=2C and Ted
=0A=

_______________________________________________=0A=
rtcweb mailing list=0A=
rtcweb@ietf.org=0A=
https://www.ietf.org/mailman/listinfo/rtcweb 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>Ted --<div><br></div><div>The H.=
264 vs. VP8 discussion has at this point been "overtaken by events"=2C so t=
hat the proposed plan below would be a waste of everybody's time=2C about a=
s relevant as debating the fashion merits of bell bottoms versus nehru jack=
ets. &nbsp=3B</div><div><br></div><div>With Google having announced plans t=
o implement VP9 (potentially with SVC)=2C and with the recent ratification =
of H.265=2C the industry has moved on=2C and so should RTCWEB.&nbsp=3B</div=
><div><br><div><hr id=3D"stopSpelling">Date: Fri=2C 13 Sep 2013 09:52:24 -0=
700<br>From: ted.ietf@gmail.com<br>To: rtcweb@ietf.org=3B fluffy@cisco.com=
=3B magnus.westerlund@ericsson.com<br>Subject: [rtcweb] Video Codec Selecti=
on Plan<br><br><div dir=3D"ltr"><div>WG=2C<br><br>The chairs have created a=
 plan for how to perform the Video Codec<br>selection in our WG. The chairs=
 are asking for review of our plan on<br>how to undertake the mandatory-to-=
implement video codec selection.<br>=0A=
We'd much prefer to have comments on the mechanics before they begin=2C<br>=
so please review now.&nbsp=3B Proponents of a particular proposal should<br=
>note both the actions required and the timelines proposed.<br><br>The main=
 goal of this plan is to hold a consensus call on which of<br>=0A=
the proposed alternatives we as a WG should select at one of the WG<br>sess=
ions in Vancouver. Such a consensus call will of course be<br>verified on t=
he mailing list for anyone who can't participate. The<br>chairs will recuse=
 themselves from judging this particular<br>=0A=
consensus.<br><br>In the WG session each codec proposal will be allowed an =
equal amount<br>of time to highlight the arguments for their proposal. Afte=
r that a<br>there will be a slot for discussion and clarifying questions.<b=
r>=0A=
<br>To enable the WG participants to get answers to any questions=2C the<br=
>proposals in draft form and any supporting material MUST be made<br>availa=
ble by 6th of October. This is to ensure that the WG<br>participants can ve=
rify or object to any claims or statements in<br>=0A=
the proposal material prior to the WG session. We chairs would really<br>no=
t like to see the proponents bring up new arguments at their<br>presentatio=
n. Also the WG participants are expected to raise any<br>arguments on the l=
ist ahead of time to enable the proponents to<br>=0A=
respond to such arguments.<br><br>The proposed consensus questions will be =
of the following form:<br><br>1. If you support H.264 as the mandatory to i=
mplement codec or are<br>willing to live with it as the MTI=2C please raise=
 your hand now.<br>=0A=
<br>2. If you support VP8 as the mandatory to implement codec or are<br>wil=
ling to live with it as the MTI=2C please raise your hand now.<br><br>You m=
ay indicate support on both questions and we encourage you to do<br>so if y=
ou can live with either=2C even if you have a preference for one<br>=0A=
over the other.<br><br>Additional proposals than the previous ones are welc=
ome=2C but must be<br>submitted as draft and their proponents must notify t=
he chairs no later<br>than the 6th of October that they also have a candida=
te proposal.<br>=0A=
<br>In case the WG fails to reach consensus we chairs propose that we use<b=
r>the alternative decision process as discussed in RFC3929. The method<br>a=
nd its usage will be discussed on the list should the WG not<br>establish c=
onsensus on a proposal for mandatory to implement video codec.<br>=0A=
<br></div><div>regards=2C<br><br></div>Magnus=2C&nbsp=3B Cullen=2C and Ted<=
br></div>=0A=
<br>_______________________________________________=0A=
rtcweb mailing list=0A=
rtcweb@ietf.org=0A=
https://www.ietf.org/mailman/listinfo/rtcweb</div></div> 		 	   		  </div><=
/body>
</html>=

--_d6f7ac9e-4613-4859-b50b-033764841a88_--

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Subject: Re: [rtcweb] #31: Miscellaneous
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Martin said:=20
> I don't know what question I would ask in this case.  Well=2C at least
> as far as being able to be understood goes=2C that is.  I'd recommend
> avoiding this sort of formulation=2C and would prefer that it simply be
> possible to avoid exposing IP addresses.

[BA] That is my conclusion as well.  		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'><div>Martin said:&nbsp=3B</div><=
div><br>&gt=3B I don't know what question I would ask in this case.  Well=
=2C at least<br>&gt=3B as far as being able to be understood goes=2C that i=
s.  I'd recommend<br>&gt=3B avoiding this sort of formulation=2C and would =
prefer that it simply be<br>&gt=3B possible to avoid exposing IP addresses.=
<br></div><div><br></div><div>[BA] That is my conclusion as well.&nbsp=3B</=
div> 		 	   		  </div></body>
</html>=

--_bbd9b21e-c2a7-4b50-bb2b-dc0265c0c622_--

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Date: Fri, 13 Sep 2013 19:21:39 +0200
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
To: Ted Hardie <ted.ietf@gmail.com>
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Subject: Re: [rtcweb] Video Codec Selection Plan
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Do not take me wrong, but IMHO this decision should be taken by the W3C
WebRTC group.

--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>
El 13/09/2013 18:52, "Ted Hardie" <ted.ietf@gmail.com> escribi=C3=B3:

> WG,
>
> The chairs have created a plan for how to perform the Video Codec
> selection in our WG. The chairs are asking for review of our plan on
> how to undertake the mandatory-to-implement video codec selection.
> We'd much prefer to have comments on the mechanics before they begin,
> so please review now.  Proponents of a particular proposal should
> note both the actions required and the timelines proposed.
>
> The main goal of this plan is to hold a consensus call on which of
> the proposed alternatives we as a WG should select at one of the WG
> sessions in Vancouver. Such a consensus call will of course be
> verified on the mailing list for anyone who can't participate. The
> chairs will recuse themselves from judging this particular
> consensus.
>
> In the WG session each codec proposal will be allowed an equal amount
> of time to highlight the arguments for their proposal. After that a
> there will be a slot for discussion and clarifying questions.
>
> To enable the WG participants to get answers to any questions, the
> proposals in draft form and any supporting material MUST be made
> available by 6th of October. This is to ensure that the WG
> participants can verify or object to any claims or statements in
> the proposal material prior to the WG session. We chairs would really
> not like to see the proponents bring up new arguments at their
> presentation. Also the WG participants are expected to raise any
> arguments on the list ahead of time to enable the proponents to
> respond to such arguments.
>
> The proposed consensus questions will be of the following form:
>
> 1. If you support H.264 as the mandatory to implement codec or are
> willing to live with it as the MTI, please raise your hand now.
>
> 2. If you support VP8 as the mandatory to implement codec or are
> willing to live with it as the MTI, please raise your hand now.
>
> You may indicate support on both questions and we encourage you to do
> so if you can live with either, even if you have a preference for one
> over the other.
>
> Additional proposals than the previous ones are welcome, but must be
> submitted as draft and their proponents must notify the chairs no later
> than the 6th of October that they also have a candidate proposal.
>
> In case the WG fails to reach consensus we chairs propose that we use
> the alternative decision process as discussed in RFC3929. The method
> and its usage will be discussed on the list should the WG not
> establish consensus on a proposal for mandatory to implement video codec.
>
> regards,
>
> Magnus,  Cullen, and Ted
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<p dir=3D"ltr">Do not take me wrong, but IMHO this decision should be taken=
 by the W3C WebRTC group.<br></p>
<p dir=3D"ltr">--<br>
I=C3=B1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;</p>
<div class=3D"gmail_quote">El 13/09/2013 18:52, &quot;Ted Hardie&quot; &lt;=
<a href=3D"mailto:ted.ietf@gmail.com">ted.ietf@gmail.com</a>&gt; escribi=C3=
=B3:<br type=3D"attribution"><blockquote class=3D"gmail_quote" style=3D"mar=
gin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir=3D"ltr"><div>WG,<br><br>The chairs have created a plan for how to =
perform the Video Codec<br>selection in our WG. The chairs are asking for r=
eview of our plan on<br>how to undertake the mandatory-to-implement video c=
odec selection.<br>

We&#39;d much prefer to have comments on the mechanics before they begin,<b=
r>so please review now.=C2=A0 Proponents of a particular proposal should<br=
>note both the actions required and the timelines proposed.<br><br>The main=
 goal of this plan is to hold a consensus call on which of<br>

the proposed alternatives we as a WG should select at one of the WG<br>sess=
ions in Vancouver. Such a consensus call will of course be<br>verified on t=
he mailing list for anyone who can&#39;t participate. The<br>chairs will re=
cuse themselves from judging this particular<br>

consensus.<br><br>In the WG session each codec proposal will be allowed an =
equal amount<br>of time to highlight the arguments for their proposal. Afte=
r that a<br>there will be a slot for discussion and clarifying questions.<b=
r>

<br>To enable the WG participants to get answers to any questions, the<br>p=
roposals in draft form and any supporting material MUST be made<br>availabl=
e by 6th of October. This is to ensure that the WG<br>participants can veri=
fy or object to any claims or statements in<br>

the proposal material prior to the WG session. We chairs would really<br>no=
t like to see the proponents bring up new arguments at their<br>presentatio=
n. Also the WG participants are expected to raise any<br>arguments on the l=
ist ahead of time to enable the proponents to<br>

respond to such arguments.<br><br>The proposed consensus questions will be =
of the following form:<br><br>1. If you support H.264 as the mandatory to i=
mplement codec or are<br>willing to live with it as the MTI, please raise y=
our hand now.<br>

<br>2. If you support VP8 as the mandatory to implement codec or are<br>wil=
ling to live with it as the MTI, please raise your hand now.<br><br>You may=
 indicate support on both questions and we encourage you to do<br>so if you=
 can live with either, even if you have a preference for one<br>

over the other.<br><br>Additional proposals than the previous ones are welc=
ome, but must be<br>submitted as draft and their proponents must notify the=
 chairs no later<br>than the 6th of October that they also have a candidate=
 proposal.<br>

<br>In case the WG fails to reach consensus we chairs propose that we use<b=
r>the alternative decision process as discussed in RFC3929. The method<br>a=
nd its usage will be discussed on the list should the WG not<br>establish c=
onsensus on a proposal for mandatory to implement video codec.<br>

<br></div><div>regards,<br><br></div>Magnus,=C2=A0 Cullen, and Ted<br></div=
>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div>

--047d7bd76bb462972304e64716fe--

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Date: Fri, 13 Sep 2013 10:41:28 -0700
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Subject: Re: [rtcweb] Video Codec Selection Plan
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On 13-09-13 9:59 AM, Matthew Kaufman wrote:
> 1. Please explain why this decision is an IETF decision and not a W3C
> decision, given that it is a choice about what to mandate in the
> presentation layer as codec, NOT what to mandate for the on-the-wire
> transport format (which has already been selected as DTLS-SRTP with an
> appropriate IETF payload based on the application)

This working group selected OPUS and G711 as MTI audio codecs to avoid
session negotiation failure. Session negotiation being part of the
WebRTC protocol suite being specified within the IETF.

The same arguments apply to a video codec. If we can find rough
consensus to specify one it is useful to do so.

 -r


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On Fri, Sep 13, 2013 at 10:11 AM, Bernard Aboba
<bernard_aboba@hotmail.com>wrote:

> Ted --
>
> The H.264 vs. VP8 discussion has at this point been "overtaken by events",
> so that the proposed plan below would be a waste of everybody's time, about
> as relevant as debating the fashion merits of bell bottoms versus nehru
> jackets.
>
> With Google having announced plans to implement VP9 (potentially with
> SVC), and with the recent ratification of H.265, the industry has moved on,
> and so should RTCWEB.
>
>
Hi Bernard,

The agreement to specify a mandatory-to-implement video codec to avoid
negotiation failure is a very long standing one.  If you believe the MTI
should be VP9 or H.265, you are welcome to propose either; please do so,
however, in the form of a draft by October 6th so we can move forward
knowing that it is your proposal.

regards,

Ted Hardie



> ------------------------------
> Date: Fri, 13 Sep 2013 09:52:24 -0700
> From: ted.ietf@gmail.com
> To: rtcweb@ietf.org; fluffy@cisco.com; magnus.westerlund@ericsson.com
> Subject: [rtcweb] Video Codec Selection Plan
>
>
> WG,
>
> The chairs have created a plan for how to perform the Video Codec
> selection in our WG. The chairs are asking for review of our plan on
> how to undertake the mandatory-to-implement video codec selection.
> We'd much prefer to have comments on the mechanics before they begin,
> so please review now.  Proponents of a particular proposal should
> note both the actions required and the timelines proposed.
>
> The main goal of this plan is to hold a consensus call on which of
> the proposed alternatives we as a WG should select at one of the WG
> sessions in Vancouver. Such a consensus call will of course be
> verified on the mailing list for anyone who can't participate. The
> chairs will recuse themselves from judging this particular
> consensus.
>
> In the WG session each codec proposal will be allowed an equal amount
> of time to highlight the arguments for their proposal. After that a
> there will be a slot for discussion and clarifying questions.
>
> To enable the WG participants to get answers to any questions, the
> proposals in draft form and any supporting material MUST be made
> available by 6th of October. This is to ensure that the WG
> participants can verify or object to any claims or statements in
> the proposal material prior to the WG session. We chairs would really
> not like to see the proponents bring up new arguments at their
> presentation. Also the WG participants are expected to raise any
> arguments on the list ahead of time to enable the proponents to
> respond to such arguments.
>
> The proposed consensus questions will be of the following form:
>
> 1. If you support H.264 as the mandatory to implement codec or are
> willing to live with it as the MTI, please raise your hand now.
>
> 2. If you support VP8 as the mandatory to implement codec or are
> willing to live with it as the MTI, please raise your hand now.
>
> You may indicate support on both questions and we encourage you to do
> so if you can live with either, even if you have a preference for one
> over the other.
>
> Additional proposals than the previous ones are welcome, but must be
> submitted as draft and their proponents must notify the chairs no later
> than the 6th of October that they also have a candidate proposal.
>
> In case the WG fails to reach consensus we chairs propose that we use
> the alternative decision process as discussed in RFC3929. The method
> and its usage will be discussed on the list should the WG not
> establish consensus on a proposal for mandatory to implement video codec.
>
> regards,
>
> Magnus,  Cullen, and Ted
>
> _______________________________________________ rtcweb mailing list
> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
>

--047d7b4142d6178d5e04e6477451
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<div dir=3D"ltr">On Fri, Sep 13, 2013 at 10:11 AM, Bernard Aboba <span dir=
=3D"ltr">&lt;<a href=3D"mailto:bernard_aboba@hotmail.com" target=3D"_blank"=
>bernard_aboba@hotmail.com</a>&gt;</span> wrote:<br><div class=3D"gmail_ext=
ra"><div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">


<div><div dir=3D"ltr">Ted --<div><br></div><div>The H.264 vs. VP8 discussio=
n has at this point been &quot;overtaken by events&quot;, so that the propo=
sed plan below would be a waste of everybody&#39;s time, about as relevant =
as debating the fashion merits of bell bottoms versus nehru jackets. =A0</d=
iv>
<div><br></div><div>With Google having announced plans to implement VP9 (po=
tentially with SVC), and with the recent ratification of H.265, the industr=
y has moved on, and so should RTCWEB.=A0</div><div><br></div></div></div>
</blockquote><div><br></div><div>Hi Bernard,<br><br>The agreement to specif=
y a mandatory-to-implement video codec to avoid negotiation failure is a ve=
ry long standing one.=A0 If you believe the MTI should be VP9 or H.265, you=
 are welcome to propose either; please do so, however, in the form of a dra=
ft by October 6th so we can move forward knowing that it is your proposal.<=
br>
<br>regards,<br><br>Ted Hardie<br></div><div><br>=A0</div><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex"><div><div dir=3D"ltr"><div><div><hr>Date: Fri, 13 Sep 2013 09=
:52:24 -0700<br>
From: <a href=3D"mailto:ted.ietf@gmail.com" target=3D"_blank">ted.ietf@gmai=
l.com</a><br>To: <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcwe=
b@ietf.org</a>; <a href=3D"mailto:fluffy@cisco.com" target=3D"_blank">fluff=
y@cisco.com</a>; <a href=3D"mailto:magnus.westerlund@ericsson.com" target=
=3D"_blank">magnus.westerlund@ericsson.com</a><br>
Subject: [rtcweb] Video Codec Selection Plan<div><div class=3D"h5"><br><br>=
<div dir=3D"ltr"><div>WG,<br><br>The chairs have created a plan for how to =
perform the Video Codec<br>selection in our WG. The chairs are asking for r=
eview of our plan on<br>
how to undertake the mandatory-to-implement video codec selection.<br>
We&#39;d much prefer to have comments on the mechanics before they begin,<b=
r>so please review now.=A0 Proponents of a particular proposal should<br>no=
te both the actions required and the timelines proposed.<br><br>The main go=
al of this plan is to hold a consensus call on which of<br>

the proposed alternatives we as a WG should select at one of the WG<br>sess=
ions in Vancouver. Such a consensus call will of course be<br>verified on t=
he mailing list for anyone who can&#39;t participate. The<br>chairs will re=
cuse themselves from judging this particular<br>

consensus.<br><br>In the WG session each codec proposal will be allowed an =
equal amount<br>of time to highlight the arguments for their proposal. Afte=
r that a<br>there will be a slot for discussion and clarifying questions.<b=
r>

<br>To enable the WG participants to get answers to any questions, the<br>p=
roposals in draft form and any supporting material MUST be made<br>availabl=
e by 6th of October. This is to ensure that the WG<br>participants can veri=
fy or object to any claims or statements in<br>

the proposal material prior to the WG session. We chairs would really<br>no=
t like to see the proponents bring up new arguments at their<br>presentatio=
n. Also the WG participants are expected to raise any<br>arguments on the l=
ist ahead of time to enable the proponents to<br>

respond to such arguments.<br><br>The proposed consensus questions will be =
of the following form:<br><br>1. If you support H.264 as the mandatory to i=
mplement codec or are<br>willing to live with it as the MTI, please raise y=
our hand now.<br>

<br>2. If you support VP8 as the mandatory to implement codec or are<br>wil=
ling to live with it as the MTI, please raise your hand now.<br><br>You may=
 indicate support on both questions and we encourage you to do<br>so if you=
 can live with either, even if you have a preference for one<br>

over the other.<br><br>Additional proposals than the previous ones are welc=
ome, but must be<br>submitted as draft and their proponents must notify the=
 chairs no later<br>than the 6th of October that they also have a candidate=
 proposal.<br>

<br>In case the WG fails to reach consensus we chairs propose that we use<b=
r>the alternative decision process as discussed in RFC3929. The method<br>a=
nd its usage will be discussed on the list should the WG not<br>establish c=
onsensus on a proposal for mandatory to implement video codec.<br>

<br></div><div>regards,<br><br></div>Magnus,=A0 Cullen, and Ted<br></div>
<br></div></div><div class=3D"im">_________________________________________=
______
rtcweb mailing list
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a></div></div></div> 		 	   	=
	  </div></div>
</blockquote></div><br></div></div>

--047d7b4142d6178d5e04e6477451--

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Date: Fri, 13 Sep 2013 20:00:05 +0200
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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El 13/09/2013 19:41, "Ralph Giles" <giles@thaumas.net> escribi=C3=B3:
>
> On 13-09-13 9:59 AM, Matthew Kaufman wrote:
> > 1. Please explain why this decision is an IETF decision and not a W3C
> > decision, given that it is a choice about what to mandate in the
> > presentation layer as codec, NOT what to mandate for the on-the-wire
> > transport format (which has already been selected as DTLS-SRTP with an
> > appropriate IETF payload based on the application)
>
> This working group selected OPUS and G711 as MTI audio codecs to avoid
> session negotiation failure. Session negotiation being part of the
> WebRTC protocol suite being specified within the IETF.
>
> The same arguments apply to a video codec. If we can find rough
> consensus to specify one it is useful to do so.

One error does not justify another one.

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<p dir=3D"ltr"><br>
El 13/09/2013 19:41, &quot;Ralph Giles&quot; &lt;<a href=3D"mailto:giles@th=
aumas.net">giles@thaumas.net</a>&gt; escribi=C3=B3:<br>
&gt;<br>
&gt; On 13-09-13 9:59 AM, Matthew Kaufman wrote:<br>
&gt; &gt; 1. Please explain why this decision is an IETF decision and not a=
 W3C<br>
&gt; &gt; decision, given that it is a choice about what to mandate in the<=
br>
&gt; &gt; presentation layer as codec, NOT what to mandate for the on-the-w=
ire<br>
&gt; &gt; transport format (which has already been selected as DTLS-SRTP wi=
th an<br>
&gt; &gt; appropriate IETF payload based on the application)<br>
&gt;<br>
&gt; This working group selected OPUS and G711 as MTI audio codecs to avoid=
<br>
&gt; session negotiation failure. Session negotiation being part of the<br>
&gt; WebRTC protocol suite being specified within the IETF.<br>
&gt;<br>
&gt; The same arguments apply to a video codec. If we can find rough<br>
&gt; consensus to specify one it is useful to do so.<br></p>
<p dir=3D"ltr">One error does not justify another one.<br>
</p>

--001a1133b8fcd652e404e6479fdd--

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Ted said:
"If you believe the MTI should be VP9 or H.265=2C you are welcome to propos=
e either=3B please do so"
[BA] I am not prepared to argue for (or against) either VP9 or H.265 at thi=
s time=2C and I believe that many industry colleagues would say the same.  =
In a way=2C that is the point -- minds have not yet been made up=2C as in (=
many cases) they have been on the topic of VP8 vs. H.264. =20
"It's not only generals who fight the last war" -- Dan Gardner 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'><span style=3D"font-size: 12pt=
=3B">Ted said:</span><div><span style=3D"font-size: 12pt=3B"><br></span></d=
iv><div><span style=3D"font-size: 12pt=3B">"If you believe the MTI should b=
e VP9 or H.265=2C you are welcome to propose either=3B please do so"</span>=
</div><div><br></div><div>[BA] I am not prepared to argue for (or against) =
either VP9 or H.265 at this time=2C and I believe that many industry collea=
gues would say the same. &nbsp=3BIn a way=2C that is the point -- minds hav=
e not yet been made up=2C as in (many cases) they have been on the topic of=
 VP8 vs. H.264. &nbsp=3B</div><div><br></div><div>"It's not only generals w=
ho fight the last war" -- Dan Gardner</div> 		 	   		  </div></body>
</html>=

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On 9/13/13 12:11, Bernard Aboba wrote:
> With Google having announced plans to implement VP9 (potentially with 
> SVC), and with the recent ratification of H.265, the industry has 
> moved on...

At which point we need to ask whether we're trying to pick "the best 
possible codec" or "a suitable codec." If the former, then your point is 
relevant. But I don't think there's consensus (rough or otherwise) 
around "best possible codec" being the goal.

/a


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    <div class="moz-cite-prefix">On 9/13/13 12:11, Bernard Aboba wrote:<br>
    </div>
    <blockquote cite="mid:BLU169-W111E8BA8C5A1AF65A7031F4933B0@phx.gbl"
      type="cite">
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      <div dir="ltr">With Google having announced plans to implement VP9
        (potentially with SVC), and with the recent ratification of
        H.265, the industry has moved on...</div>
    </blockquote>
    <br>
    At which point we need to ask whether we're trying to pick "the best
    possible codec" or "a suitable codec." If the former, then your
    point is relevant. But I don't think there's consensus (rough or
    otherwise) around "best possible codec" being the goal.<br>
    <br>
    /a<br>
    <br>
  </body>
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--------------030706050809060502040705--

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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: =?iso-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>
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That topic was discussed between W3C and IETF when forming the two WG and v=
ery clear decisions was made that the W3C would not make that decision. Par=
t of the reasons had to do with the history of a very similar argument at W=
3C around the video tag.=20

On Sep 13, 2013, at 11:21 AM, I=F1aki Baz Castillo <ibc@aliax.net> wrote:

> Do not take me wrong, but IMHO this decision should be taken by the W3C W=
ebRTC group.
>=20
>=20


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From: "Matthew Kaufman (SKYPE)" <matthew.kaufman@skype.net>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
Thread-Topic: [rtcweb] Video Codec Selection Plan
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The W3C was the right venue for the video tag codec discussion, just like i=
t is for this video codec discussion.

I'll note that the chairs have repeatedly pointed out that the scope is exp=
licitly "browsers"

Matthew Kaufman

(Sent from my iPhone)

On Sep 13, 2013, at 12:05 PM, "Cullen Jennings (fluffy)" <fluffy@cisco.com>=
 wrote:

>=20
> That topic was discussed between W3C and IETF when forming the two WG and=
 very clear decisions was made that the W3C would not make that decision. P=
art of the reasons had to do with the history of a very similar argument at=
 W3C around the video tag.=20
>=20
> On Sep 13, 2013, at 11:21 AM, I=F1aki Baz Castillo <ibc@aliax.net> wrote:
>=20
>> Do not take me wrong, but IMHO this decision should be taken by the W3C =
WebRTC group.
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20

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Subject: Re: [rtcweb] Video Codec Selection Plan
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Big caveat as I post this: I am speaking strictly as an IETF 
participant. In particular:

- I am not speaking as an AD. I am not the responsible AD for this WG, 
and the chairs can go ahead with their proposed path forward consulting 
with the WG's responsible AD, whatever they think of what I say below. 
No implication of any authority should be taken here.

- Though my comments here are clearly informed by the draft some of you 
know that I'm working on regarding consensus, that draft does not 
represent accepted theory in the IETF, so I'm not appealing to it as 
some sort of "given truth" on this.

I'm just another participant in the room.

All that said, I am concerned about the path being proposed, and I agree 
with Matthew's concern (though I feel somewhat less defeatist than his 
assessment sounds):

On 9/13/13 11:59 AM, Matthew Kaufman wrote:
> 2. Should I expect "room-packing" for this "show of hands" (which I 
> don't believe is the typical "hum" process for the IETF, either) as 
> happened during the SDES discussion last time, and therefore need to 
> bring as many people who've not participated in RTCWEB previously but 
> who want my codec to succeed as I can afford to fly to Vancouver?

I think this is a valid concern. Asking "who wants or can live with 
VP8?" and "who wants or can live with H.264?" might be a good start to 
the discussion, but at that point, I'm going to want to hear *why* 
people *can't* live with one or the other. What I would *not* be 
interested in is yet another presentation from the proponents of either 
of these to tell me why one is better; I want to hear from *opponents* 
of the proposal to hear why the proposal would fail. Only then would I 
want to hear from proponents responding to those objections. And then 
I'd like to see a summary of those objections and those responses by the 
chairs. With that summary, the chairs can make a call of whether there 
is any hope of consensus. I think much of that work can and should take 
place on the mailing list *before* the face-to-face meeting.

But asking for a show of hands in the room for people who "can live 
with" one or the other and deciding whether or not consensus exists 
based on that show of hands is simply taking a vote, it's not judging 
consensus. You cannot know from that result whether the reason that some 
people could not "live with" a proposal was simply because "my Aunt 
Gertrude told me that I should say that I can't live with it." To ask 
for the show of hands and then not find out what it means is not calling 
the consensus. And I think moving to a 3929 alternative on the basis of 
that show of hands would be improper.

Put me down as one voice who objects to the proposed procedure.

pr

> On 9/13/2013 9:52 AM, Ted Hardie wrote:
>> WG,
>>
>> The chairs have created a plan for how to perform the Video Codec
>> selection in our WG. The chairs are asking for review of our plan on
>> how to undertake the mandatory-to-implement video codec selection.
>> We'd much prefer to have comments on the mechanics before they begin,
>> so please review now.  Proponents of a particular proposal should
>> note both the actions required and the timelines proposed.
>>
>> The main goal of this plan is to hold a consensus call on which of
>> the proposed alternatives we as a WG should select at one of the WG
>> sessions in Vancouver. Such a consensus call will of course be
>> verified on the mailing list for anyone who can't participate. The
>> chairs will recuse themselves from judging this particular
>> consensus.
>>
>> In the WG session each codec proposal will be allowed an equal amount
>> of time to highlight the arguments for their proposal. After that a
>> there will be a slot for discussion and clarifying questions.
>>
>> To enable the WG participants to get answers to any questions, the
>> proposals in draft form and any supporting material MUST be made
>> available by 6th of October. This is to ensure that the WG
>> participants can verify or object to any claims or statements in
>> the proposal material prior to the WG session. We chairs would really
>> not like to see the proponents bring up new arguments at their
>> presentation. Also the WG participants are expected to raise any
>> arguments on the list ahead of time to enable the proponents to
>> respond to such arguments.
>>
>> The proposed consensus questions will be of the following form:
>>
>> 1. If you support H.264 as the mandatory to implement codec or are
>> willing to live with it as the MTI, please raise your hand now.
>>
>> 2. If you support VP8 as the mandatory to implement codec or are
>> willing to live with it as the MTI, please raise your hand now.
>>
>> You may indicate support on both questions and we encourage you to do
>> so if you can live with either, even if you have a preference for one
>> over the other.
>>
>> Additional proposals than the previous ones are welcome, but must be
>> submitted as draft and their proponents must notify the chairs no later
>> than the 6th of October that they also have a candidate proposal.
>>
>> In case the WG fails to reach consensus we chairs propose that we use
>> the alternative decision process as discussed in RFC3929. The method
>> and its usage will be discussed on the list should the WG not
>> establish consensus on a proposal for mandatory to implement video codec.
>>
>> regards,
>>
>> Magnus,  Cullen, and Ted

-- 
Pete Resnick<http://www.qualcomm.com/~presnick/>
Qualcomm Technologies, Inc. - +1 (858)651-4478


From sergio.garcia.murillo@gmail.com  Fri Sep 13 13:26:48 2013
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, behave@ietf.org
Subject: Re: [rtcweb] [BEHAVE] New Version Notification for draft-chenxin-behave-turn-websocket-01.txt
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El 13/09/2013 11:23, Simon Perreault escribió:
> Le 2013-09-13 10:58, Chenxin (Xin) a écrit :
>>     We have been working on a new version of the TURN over Websocket 
>> draft, which is now available at:
>>
>>     http://www.ietf.org/id/draft-chenxin-behave-turn-websocket-01.txt
>
> There are two major things missing in this draft:
>
> - How are TURN UDP "channels" supported? (See RFC 5766 section 2.5.)
>
> - How are TURN TCP "data connections" supported? (See RFC 6062 section 
> 3.)
>
> I think the answers to these questions will greatly affect the overall 
> design of TURN-over-WebSocket.
>
>
Hi Simon,

As per RFC 5766:

    For some applications (e.g., Voice over IP), the 36 bytes of overhead
    that a Send indication or Data indication adds to the application
    data can substantially increase the bandwidth required between the
    client and the server.  To remedy this, TURN offers a second way for
    the client and server to associate data with a specific peer.

    This second way uses an alternate packet format known as the
    ChannelData message.  The ChannelData message does not use the STUN
    header used by other TURN messages, but instead has a 4-byte header
    that includes a number known as a channel number.



So, at the end, the ChannelData message is just another TURN message, so 
it will be encapsulated inside a websocket frame. If you feel it is not 
clear, we can add more details about it in next draft.

Regarding RFC 6062, we had some internal discussion regarding if we 
should add websocket support at all to it or not. Mainly, because AFAIK, 
webrtc will use RFC 5766 and not 6062. Anyone please correct me if I am 
wrong.

I agree with you that this has not been well covered in the draft. In 
order to make the minimun changes possible, and align it with the 
general view of changing just the turn client to turn server 
encapsulation protocol, I would propose to extend websockets usage only 
for the turn control connection in RFC 6062 but keep turn data 
connections on TCP. That, or drop RFC 6062 extension altogether and just 
focus on 5766. What do you feel it would be best?

Best regards
Sergio


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Howdy,

I've now had lunch and nice coffee, and I take electrons in hand to answer
your missive, sent earlier today.

On Fri, Sep 13, 2013 at 11:50 AM, Pete Resnick <presnick@qti.qualcomm.com>wrote:

> Big caveat as I post this: I am speaking strictly as an IETF participant.
> In particular:
>
>
<SNIP>


> All that said, I am concerned about the path being proposed, and I agree
> with Matthew's concern (though I feel somewhat less defeatist than his
> assessment sounds):
>
>
> On 9/13/13 11:59 AM, Matthew Kaufman wrote:
>
>> 2. Should I expect "room-packing" for this "show of hands" (which I don't
>> believe is the typical "hum" process for the IETF, either) as happened
>> during the SDES discussion last time, and therefore need to bring as many
>> people who've not participated in RTCWEB previously but who want my codec
>> to succeed as I can afford to fly to Vancouver?
>>
>
> I think this is a valid concern. Asking "who wants or can live with VP8?"
> and "who wants or can live with H.264?" might be a good start to the
> discussion, but at that point, I'm going to want to hear *why* people
> *can't* live with one or the other.


That discussion is supposed to happen before this, not after, so that if
there are issues which might be resolved, they can be.  During the course
of the effort to find a solution here, there have been several such
resolutions (Cisco, for example, graciously agreed to provide an open
source H.264 implementation should it be selected, in order to handle
objections that none were available).  If there are issues which cannot be
resolved, having a second discussion on what they are amounts to re-hashing
and doesn't do anything to move things forward.


> What I would *not* be interested in is yet another presentation from the
> proponents of either of these to tell me why one is better;


One might surmise that this is because you do not intend to actually build,
deploy, or operate a service that relates to this.  If someone puts forward
a codec that is better (in human assessment, lines of code, or operating
requirements) and meets the other criteria for deployment, why wouldn't you
want to hear about it?


> I want to hear from *opponents* of the proposal to hear why the proposal
> would fail.  Only then would I want to hear from proponents responding to
> those objections. And then I'd like to see a summary of those objections
> and those responses by the chairs


Pete, my dear mangel-wurzel and apple of my eye, that's the most process
wonky thing you've said in a long while, which is saying something.  The
working group is trying to identify a common codec that can be used to
avoid negotiation failure.  If there is no common video codec, the network
effect of this system is much less.
If there are people who have stated *they cannot live with this*, they are
not going to build, deploy, or operate services which use the common
codec.  The actual system we are trying to build will suffer.  Judging
consensus is not here an end in itself, it's a way of working out whether
or not the system we're building does or does not have a way to avoid the
negotiation failure.

We would not have spent the amount of time we have unless that were
important, and we would not be considering using RFC 3929 unless we thought
it was worth getting to a resolution.  I implore you to stop treating this
as an academic exercise in consensus theory, and remind yourself that we're
trying to do some engineering here.


>
> With that summary, the chairs can make a call of whether there is any hope
> of consensus. I think much of that work can and should take place on the
> mailing list *before* the face-to-face meeting.
>
>
But asking for a show of hands in the room for people who "can live with"
> one or the other and deciding whether or not consensus exists based on that
> show of hands is simply taking a vote, it's not judging consensus.


That rather presumes you know the count.  In several of our recent calls,
we have had unanimity, which is generally held to be a strong indicator of
consensus.

The last time we had a straw poll which used similar language, we also had
absolutely no question that we had not achieved consensus.

Sometimes signals are strong.


> You cannot know from that result whether the reason that some people could
> not "live with" a proposal was simply because "my Aunt Gertrude told me
> that I should say that I can't live with it." To ask for the show of hands
> and then not find out what it means is not calling the consensus. And I
> think moving to a 3929 alternative on the basis of that show of hands would
> be improper.
>
> If you re-read RFC 3929, you will discover that it requires that you make
an explicit call for its use; the consensus to use it standing in for the
consent on the technical matter.  The working group chairs have no
intention of skipping that step, should RFC 3929 be used.


> Put me down as one voice who objects to the proposed procedure.
>
>
So noted.

fondly,

Ted



> pr
>
>
>  On 9/13/2013 9:52 AM, Ted Hardie wrote:
>>
>>> WG,
>>>
>>> The chairs have created a plan for how to perform the Video Codec
>>> selection in our WG. The chairs are asking for review of our plan on
>>> how to undertake the mandatory-to-implement video codec selection.
>>> We'd much prefer to have comments on the mechanics before they begin,
>>> so please review now.  Proponents of a particular proposal should
>>> note both the actions required and the timelines proposed.
>>>
>>> The main goal of this plan is to hold a consensus call on which of
>>> the proposed alternatives we as a WG should select at one of the WG
>>> sessions in Vancouver. Such a consensus call will of course be
>>> verified on the mailing list for anyone who can't participate. The
>>> chairs will recuse themselves from judging this particular
>>> consensus.
>>>
>>> In the WG session each codec proposal will be allowed an equal amount
>>> of time to highlight the arguments for their proposal. After that a
>>> there will be a slot for discussion and clarifying questions.
>>>
>>> To enable the WG participants to get answers to any questions, the
>>> proposals in draft form and any supporting material MUST be made
>>> available by 6th of October. This is to ensure that the WG
>>> participants can verify or object to any claims or statements in
>>> the proposal material prior to the WG session. We chairs would really
>>> not like to see the proponents bring up new arguments at their
>>> presentation. Also the WG participants are expected to raise any
>>> arguments on the list ahead of time to enable the proponents to
>>> respond to such arguments.
>>>
>>> The proposed consensus questions will be of the following form:
>>>
>>> 1. If you support H.264 as the mandatory to implement codec or are
>>> willing to live with it as the MTI, please raise your hand now.
>>>
>>> 2. If you support VP8 as the mandatory to implement codec or are
>>> willing to live with it as the MTI, please raise your hand now.
>>>
>>> You may indicate support on both questions and we encourage you to do
>>> so if you can live with either, even if you have a preference for one
>>> over the other.
>>>
>>> Additional proposals than the previous ones are welcome, but must be
>>> submitted as draft and their proponents must notify the chairs no later
>>> than the 6th of October that they also have a candidate proposal.
>>>
>>> In case the WG fails to reach consensus we chairs propose that we use
>>> the alternative decision process as discussed in RFC3929. The method
>>> and its usage will be discussed on the list should the WG not
>>> establish consensus on a proposal for mandatory to implement video codec.
>>>
>>> regards,
>>>
>>> Magnus,  Cullen, and Ted
>>>
>>
> --
> Pete Resnick<http://www.qualcomm.**com/~presnick/<http://www.qualcomm.com/~presnick/>
> >
> Qualcomm Technologies, Inc. - +1 (858)651-4478
>
>
> ______________________________**_________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/mailman/listinfo/rtcweb>
>

--047d7b4142d6c8b0c704e649ef1b
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Howdy,<br><br>I&#39;ve now had lunch and nice coffee, and =
I take electrons in hand to answer your missive, sent earlier today.<br><di=
v><br>On Fri, Sep 13, 2013 at 11:50 AM, Pete Resnick <span dir=3D"ltr">&lt;=
<a href=3D"mailto:presnick@qti.qualcomm.com" target=3D"_blank">presnick@qti=
.qualcomm.com</a>&gt;</span> wrote:<br>
<div class=3D"gmail_extra"><div class=3D"gmail_quote"><blockquote class=3D"=
gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(20=
4,204,204);padding-left:1ex">Big caveat as I post this: I am speaking stric=
tly as an IETF participant. In particular:<br>

<br></blockquote><div><br></div><div>&lt;SNIP&gt;<br>=A0<br></div><blockquo=
te class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px =
solid rgb(204,204,204);padding-left:1ex">
All that said, I am concerned about the path being proposed, and I agree wi=
th Matthew&#39;s concern (though I feel somewhat less defeatist than his as=
sessment sounds):<div class=3D"im"><br>
<br>
On 9/13/13 11:59 AM, Matthew Kaufman wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left:1px solid rgb(204,204,204);padding-left:1ex">
2. Should I expect &quot;room-packing&quot; for this &quot;show of hands&qu=
ot; (which I don&#39;t believe is the typical &quot;hum&quot; process for t=
he IETF, either) as happened during the SDES discussion last time, and ther=
efore need to bring as many people who&#39;ve not participated in RTCWEB pr=
eviously but who want my codec to succeed as I can afford to fly to Vancouv=
er?<br>

</blockquote>
<br></div>
I think this is a valid concern. Asking &quot;who wants or can live with VP=
8?&quot; and &quot;who wants or can live with H.264?&quot; might be a good =
start to the discussion, but at that point, I&#39;m going to want to hear *=
why* people *can&#39;t* live with one or the other. </blockquote>
<div><br></div><div>That discussion is supposed to happen before this, not =
after, so that if there are issues which might be resolved, they can be.=A0=
 During the course of the effort to find a solution here, there have been s=
everal such resolutions (Cisco, for example, graciously agreed to provide a=
n open source H.264 implementation should it be selected, in order to handl=
e objections that none were available).=A0 If there are issues which cannot=
 be resolved, having a second discussion on what they are amounts to re-has=
hing and doesn&#39;t do anything to move things forward.<br>
</div><div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0=
px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">What =
I would *not* be interested in is yet another presentation from the propone=
nts of either of these to tell me why one is better;</blockquote>
<div><br></div><div>One might surmise that this is because you do not inten=
d to actually build, deploy, or operate a service that relates to this.=A0 =
If someone puts forward a codec that is better (in human assessment, lines =
of code, or operating requirements) and meets the other criteria for deploy=
ment, why wouldn&#39;t you want to hear about it?<br>
</div><div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0=
px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"> I wa=
nt to hear from *opponents* of the proposal to hear why the proposal would =
fail.=A0 Only then would I want to hear from proponents responding to those=
 objections. And then I&#39;d like to see a summary of those objections and=
 those responses by the chairs </blockquote>
<div>=A0<br>Pete, my dear mangel-wurzel and apple of my eye, that&#39;s the=
 most process wonky thing you&#39;ve said in a long while, which is=20
saying something.=A0 The working group is trying to identify a common=20
codec that can be used to avoid negotiation failure.=A0 If there is no=20
common video codec, the network effect of this system is much less.=A0 <br>=
</div><div>If there are people who have stated *they cannot live with this*=
, they are not going to build, deploy, or operate services which use the co=
mmon codec.=A0 The actual system we are trying to build will suffer.=A0 Jud=
ging consensus is not here an end in itself, it&#39;s a way of working out =
whether or not the system we&#39;re building does or does not have a way to=
 avoid the negotiation failure.<br>
</div><div><br>We=20
would not have spent the amount of time we have unless that were=20
important, and we would not be considering using RFC 3929 unless we thought=
 it was worth getting to a resolution.=A0 I implore you to stop treating th=
is as an academic exercise in consensus theory, and remind yourself that we=
&#39;re trying to do some engineering here.<br>
<br>=A0<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;=
border-left:1px solid rgb(204,204,204);padding-left:1ex">With
 that summary, the chairs can make a call of whether there is any hope=20
of consensus. I think much of that work can and should take place on the
 mailing list *before* the face-to-face meeting.<br>
<br></blockquote><br></div><blockquote class=3D"gmail_quote" style=3D"margi=
n:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex=
">
But asking for a show of hands in the room for people who &quot;can live wi=
th&quot; one or the other and deciding whether or not consensus exists base=
d on that show of hands is simply taking a vote, it&#39;s not judging conse=
nsus. </blockquote>
<div><br></div><div>That rather presumes you know the count.=A0 In several =
of our recent calls, we have had unanimity, which is generally held to be a=
 strong indicator of consensus.=A0 <br><br>The last time we had a straw pol=
l which used similar language, we also had absolutely no question that we h=
ad not achieved consensus.=A0 <br>
<br>Sometimes signals are strong.<br></div><div>=A0</div><blockquote class=
=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rg=
b(204,204,204);padding-left:1ex">You cannot know from that result whether t=
he reason that some people could not &quot;live with&quot; a proposal was s=
imply because &quot;my Aunt Gertrude told me that I should say that I can&#=
39;t live with it.&quot; To ask for the show of hands and then not find out=
 what it means is not calling the consensus. And I think moving to a 3929 a=
lternative on the basis of that show of hands would be improper.<br>

<br></blockquote><div>If you re-read RFC 3929, you will discover that it re=
quires that you make an explicit call for its use; the consensus to use it =
standing in for the consent on the technical matter.=A0 The working group c=
hairs have no intention of skipping that step, should RFC 3929 be used.<br>
=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8e=
x;border-left:1px solid rgb(204,204,204);padding-left:1ex">
Put me down as one voice who objects to the proposed procedure.<br>
<br></blockquote><div><br></div><div>So noted.<br><br></div><div>fondly,<br=
><br>Ted<br></div><div><br>=A0</div><blockquote class=3D"gmail_quote" style=
=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding=
-left:1ex">

pr<div class=3D""><div class=3D"h5"><br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left:1px solid rgb(204,204,204);padding-left:1ex">
On 9/13/2013 9:52 AM, Ted Hardie wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left:1px solid rgb(204,204,204);padding-left:1ex">
WG,<br>
<br>
The chairs have created a plan for how to perform the Video Codec<br>
selection in our WG. The chairs are asking for review of our plan on<br>
how to undertake the mandatory-to-implement video codec selection.<br>
We&#39;d much prefer to have comments on the mechanics before they begin,<b=
r>
so please review now. =A0Proponents of a particular proposal should<br>
note both the actions required and the timelines proposed.<br>
<br>
The main goal of this plan is to hold a consensus call on which of<br>
the proposed alternatives we as a WG should select at one of the WG<br>
sessions in Vancouver. Such a consensus call will of course be<br>
verified on the mailing list for anyone who can&#39;t participate. The<br>
chairs will recuse themselves from judging this particular<br>
consensus.<br>
<br>
In the WG session each codec proposal will be allowed an equal amount<br>
of time to highlight the arguments for their proposal. After that a<br>
there will be a slot for discussion and clarifying questions.<br>
<br>
To enable the WG participants to get answers to any questions, the<br>
proposals in draft form and any supporting material MUST be made<br>
available by 6th of October. This is to ensure that the WG<br>
participants can verify or object to any claims or statements in<br>
the proposal material prior to the WG session. We chairs would really<br>
not like to see the proponents bring up new arguments at their<br>
presentation. Also the WG participants are expected to raise any<br>
arguments on the list ahead of time to enable the proponents to<br>
respond to such arguments.<br>
<br>
The proposed consensus questions will be of the following form:<br>
<br>
1. If you support H.264 as the mandatory to implement codec or are<br>
willing to live with it as the MTI, please raise your hand now.<br>
<br>
2. If you support VP8 as the mandatory to implement codec or are<br>
willing to live with it as the MTI, please raise your hand now.<br>
<br>
You may indicate support on both questions and we encourage you to do<br>
so if you can live with either, even if you have a preference for one<br>
over the other.<br>
<br>
Additional proposals than the previous ones are welcome, but must be<br>
submitted as draft and their proponents must notify the chairs no later<br>
than the 6th of October that they also have a candidate proposal.<br>
<br>
In case the WG fails to reach consensus we chairs propose that we use<br>
the alternative decision process as discussed in RFC3929. The method<br>
and its usage will be discussed on the list should the WG not<br>
establish consensus on a proposal for mandatory to implement video codec.<b=
r>
<br>
regards,<br>
<br>
Magnus, =A0Cullen, and Ted<br>
</blockquote></blockquote>
<br></div></div><span class=3D""><font color=3D"#888888">
-- <br>
Pete Resnick&lt;<a href=3D"http://www.qualcomm.com/~presnick/" target=3D"_b=
lank">http://www.qualcomm.<u></u>com/~presnick/</a>&gt;<br>
Qualcomm Technologies, Inc. - <a href=3D"tel:%2B1%20%28858%29651-4478" valu=
e=3D"+18586514478" target=3D"_blank">+1 (858)651-4478</a></font></span><div=
 class=3D""><div class=3D"h5"><br>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div></div></div>

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On 9/13/13 3:45 PM, Ted Hardie wrote:
> I've now had lunch and nice coffee, and I take electrons in hand to 
> answer your missive, sent earlier today.

Erst kommt das Fressen, dann kommt die Standards.

> On Fri, Sep 13, 2013 at 11:50 AM, Pete Resnick 
> <presnick@qti.qualcomm.com <mailto:presnick@qti.qualcomm.com>> wrote:
>
>     Asking "who wants or can live with VP8?" and "who wants or can
>     live with H.264?" might be a good start to the discussion, but at
>     that point, I'm going to want to hear *why* people *can't* live
>     with one or the other. 
>
>
> That discussion is supposed to happen before this, not after...

That's good to hear, but at least your present "plan" message, and none 
that I can find elsewhere, makes it apparent that you are now soliciting 
the list of outstanding issues and their purported answers, nor that a 
summary of that list and the closed issues would be available before the 
face-to-face. I'm all for that.

> ...so that if there are issues which might be resolved, they can be.

Or, for that matter, if there are issues which are showstoppers that 
can't possibly be resolved, they are noted, because that might leave us 
with only 1 solution anyway.

> If there are issues which cannot be resolved, having a second 
> discussion on what they are amounts to re-hashing and doesn't do 
> anything to move things forward.

Absolutely agree, no second discussion is desirable. Having the list of 
unresolvable issues with the chair's calls on those, would be handy.
>
>     What I would *not* be interested in is yet another presentation
>     from the proponents of either of these to tell me why one is better;
>
>
> One might surmise that this is because you do not intend to actually 
> build, deploy, or operate a service that relates to this.  If someone 
> puts forward a codec that is better (in human assessment, lines of 
> code, or operating requirements) and meets the other criteria for 
> deployment, why wouldn't you want to hear about it?

Ah, you misunderstand my comment. I presume that the cases *for* the 
codecs have all been made quite well already. No doubt, if there is 
something that proponents believe has not been said or understood yet, 
out on the table with it.
>
>     I want to hear from *opponents* of the proposal to hear why the
>     proposal would fail.  Only then would I want to hear from
>     proponents responding to those objections. And then I'd like to
>     see a summary of those objections and those responses by the chairs 
>
>
> Pete, my dear mangel-wurzel and apple of my eye, that's the most 
> process wonky thing you've said in a long while, which is saying 
> something.

Exchange of hugs, kisses, and root vegetables deferred until Vancouver.

> The working group is trying to identify a common codec that can be 
> used to avoid negotiation failure.  If there is no common video codec, 
> the network effect of this system is much less.

Of course.

> If there are people who have stated *they cannot live with this*, they 
> are not going to build, deploy, or operate services which use the 
> common codec.

I call "horse-pucky". And this is exactly why looking at 
skyward-pointing-appendages is a lousy tool in this particular case. You 
don't know that if they state "they cannot live with this" that they are 
not going to deploy. You would only know that if you knew their reasons. 
Because my friend mentioned below who is raising his arm because his 
Aunt Gertrude told him to do so is not affecting deployment one little 
bit. At some point, you have to trust people to be honest, but you need 
to ask the right question. In this case, if I say, "I can't implement 
this in my little open source release (that my employer would never 
sanction) if you use the FOOBAR codec for such-and-so reasons", it's 
reasonable to ask, "Who's in the same boat with Pete?", perhaps even 
following up with "Is the WG really OK with the idea the that folks in 
Pete's circumstances (open-source developers, 1-coder shops, holdover 
Macintosh developers, coders for devices with less than 1K of available 
memory, etc.) are not going to be able to implement this?" Maybe the WG 
will say, "Poor Pete. He's in the rough. We're chartered to make this 
work on a wide-scale, and Pete's sort of implementation is just not 
realistically a part of what we're doing." But then you're asking about 
whether the WG is willing to punt a particular kind of deployment, not 
counting heads (hands, souls, auras, or otherwise).

> The actual system we are trying to build will suffer.  Judging 
> consensus is not here an end in itself, it's a way of working out 
> whether or not the system we're building does or does not have a way 
> to avoid the negotiation failure.

We agree on the goal. We disagree on the methodology.

> We would not have spent the amount of time we have unless that were 
> important, and we would not be considering using RFC 3929 unless we 
> thought it was worth getting to a resolution.  I implore you to stop 
> treating this as an academic exercise in consensus theory, and remind 
> yourself that we're trying to do some engineering here.

Let's avoid questioning each others' motives, shall we? No good down 
that road.

>     But asking for a show of hands in the room for people who "can
>     live with" one or the other and deciding whether or not consensus
>     exists based on that show of hands is simply taking a vote, it's
>     not judging consensus. 
>
>
> That rather presumes you know the count.

Not at all.

> In several of our recent calls, we have had unanimity, which is 
> generally held to be a strong indicator of consensus.

Unanimity is, I concede. Other counts have more or less interesting 
indications.

> The last time we had a straw poll which used similar language, we also 
> had absolutely no question that we had not achieved consensus.

You certainly had the appearance of non-consensus. But I'm not convinced 
the work was done to determine what the showstoppers were for the people 
who "couldn't live with" one choice or the other. (I will put aside the 
fact that some chap -- with a blue dot on his badge, no less -- sitting 
down the row from me during that straw poll, exclaimed upon conclusion 
of it, "Well, it's pretty obvious we have consensus for 'A'". I 
was.....disappointed.)

> Sometimes signals are strong.

Strong signal of something. I'm not always totally sure what.
>
>     You cannot know from that result whether the reason that some
>     people could not "live with" a proposal was simply because "my
>     Aunt Gertrude told me that I should say that I can't live with
>     it." To ask for the show of hands and then not find out what it
>     means is not calling the consensus. And I think moving to a 3929
>     alternative on the basis of that show of hands would be improper.
>
>
> If you re-read RFC 3929, you will discover that it requires that you 
> make an explicit call for its use; the consensus to use it standing in 
> for the consent on the technical matter.  The working group chairs 
> have no intention of skipping that step, should RFC 3929 be used.

As above, I worry about how the determination is made. Once made, all is 
well.

> fondly,

Right back at ya'.

pr

-- 
Pete Resnick<http://www.qualcomm.com/~presnick/>
Qualcomm Technologies, Inc. - +1 (858)651-4478


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On 9/13/13 3:45 PM, Ted Hardie wrote:
<blockquote
 cite="mid:CA+9kkMCuHn3TR7umXtuScWixBfOD9jfS6mu3njxaCOADfG-nvw@mail.gmail.com"
 type="cite">
  <div dir="ltr">I've now had lunch and nice coffee, and I take
electrons in hand to answer your missive, sent earlier today.<br>
  </div>
</blockquote>
<br>
Erst kommt das Fressen, dann kommt die Standards.<br>
<br>
<blockquote
 cite="mid:CA+9kkMCuHn3TR7umXtuScWixBfOD9jfS6mu3njxaCOADfG-nvw@mail.gmail.com"
 type="cite">
  <div dir="ltr">
  <div>On Fri, Sep 13, 2013 at 11:50 AM, Pete Resnick <span dir="ltr">&lt;<a
 moz-do-not-send="true" href="mailto:presnick@qti.qualcomm.com"
 target="_blank">presnick@qti.qualcomm.com</a>&gt;</span> wrote:<br>
  <br>
  <div class="gmail_extra">
  <div class="gmail_quote">
  <blockquote class="gmail_quote"
 style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;">Asking
"who wants or can live with VP8?" and "who wants or can live with
H.264?" might be a good start to the discussion, but at that point, I'm
going to want to hear *why* people *can't* live with one or the other. </blockquote>
  <div><br>
  </div>
  <div>That discussion is supposed to happen before this, not after...</div>
  </div>
  </div>
  </div>
  </div>
</blockquote>
<br>
That's good to hear, but at least your present "plan" message, and none
that I can find elsewhere, makes it apparent that you are now
soliciting the list of outstanding issues and their purported answers,
nor that a summary of that list and the closed issues would be
available before the face-to-face. I'm all for that.<br>
<br>
<blockquote
 cite="mid:CA+9kkMCuHn3TR7umXtuScWixBfOD9jfS6mu3njxaCOADfG-nvw@mail.gmail.com"
 type="cite">
  <div dir="ltr">
  <div>
  <div class="gmail_extra">
  <div class="gmail_quote">
  <div>...so that if there are issues which might be resolved, they can
be.</div>
  </div>
  </div>
  </div>
  </div>
</blockquote>
<br>
Or, for that matter, if there are issues which are showstoppers that
can't possibly be resolved, they are noted, because that might leave us
with only 1 solution anyway.<br>
<br>
<blockquote
 cite="mid:CA+9kkMCuHn3TR7umXtuScWixBfOD9jfS6mu3njxaCOADfG-nvw@mail.gmail.com"
 type="cite">
  <div dir="ltr">
  <div>
  <div class="gmail_extra">
  <div class="gmail_quote">
  <div>If there are issues which cannot be resolved, having a second
discussion on what they are amounts to re-hashing and doesn't do
anything to move things forward.<br>
  </div>
  </div>
  </div>
  </div>
  </div>
</blockquote>
<br>
Absolutely agree, no second discussion is desirable. Having the list of
unresolvable issues with the chair's calls on those, would be handy.<br>
&nbsp;
<blockquote
 cite="mid:CA+9kkMCuHn3TR7umXtuScWixBfOD9jfS6mu3njxaCOADfG-nvw@mail.gmail.com"
 type="cite">
  <div dir="ltr">
  <div>
  <div class="gmail_extra">
  <div class="gmail_quote">
  <div></div>
  <blockquote class="gmail_quote"
 style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;">What
I would *not* be interested in is yet another presentation from the
proponents of either of these to tell me why one is better;</blockquote>
  <div><br>
  </div>
  <div>One might surmise that this is because you do not intend to
actually build, deploy, or operate a service that relates to this.&nbsp; If
someone puts forward a codec that is better (in human assessment, lines
of code, or operating requirements) and meets the other criteria for
deployment, why wouldn't you want to hear about it?<br>
  </div>
  </div>
  </div>
  </div>
  </div>
</blockquote>
<br>
Ah, you misunderstand my comment. I presume that the cases *for* the
codecs have all been made quite well already. No doubt, if there is
something that proponents believe has not been said or understood yet,
out on the table with it.<br>
&nbsp;
<blockquote
 cite="mid:CA+9kkMCuHn3TR7umXtuScWixBfOD9jfS6mu3njxaCOADfG-nvw@mail.gmail.com"
 type="cite">
  <div dir="ltr">
  <div>
  <div class="gmail_extra">
  <div class="gmail_quote">
  <div></div>
  <blockquote class="gmail_quote"
 style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;">
I want to hear from *opponents* of the proposal to hear why the
proposal would fail.&nbsp; Only then would I want to hear from proponents
responding to those objections. And then I'd like to see a summary of
those objections and those responses by the chairs </blockquote>
  <div>&nbsp;<br>
Pete, my dear mangel-wurzel and apple of my eye, that's the most
process wonky thing you've said in a long while, which is saying
something.</div>
  </div>
  </div>
  </div>
  </div>
</blockquote>
<br>
Exchange of hugs, kisses, and root vegetables deferred until Vancouver.<br>
<br>
<blockquote
 cite="mid:CA+9kkMCuHn3TR7umXtuScWixBfOD9jfS6mu3njxaCOADfG-nvw@mail.gmail.com"
 type="cite">
  <div dir="ltr">
  <div>
  <div class="gmail_extra">
  <div class="gmail_quote">
  <div>The working group is trying to identify a common codec that can
be used to avoid negotiation failure.&nbsp; If there is no common video
codec, the network effect of this system is much less.<br>
  </div>
  </div>
  </div>
  </div>
  </div>
</blockquote>
<br>
Of course.<br>
<br>
<blockquote
 cite="mid:CA+9kkMCuHn3TR7umXtuScWixBfOD9jfS6mu3njxaCOADfG-nvw@mail.gmail.com"
 type="cite">
  <div dir="ltr">
  <div>
  <div class="gmail_extra">
  <div class="gmail_quote">
  <div>If there are people who have stated *they cannot live with
this*, they are not going to build, deploy, or operate services which
use the common codec.</div>
  </div>
  </div>
  </div>
  </div>
</blockquote>
<br>
I call "horse-pucky". And this is exactly why looking at
skyward-pointing-appendages is a lousy tool in this particular case.
You don't know that if they state "they cannot live with this" that
they are not going to deploy. You would only know that if you knew
their reasons. Because my friend mentioned below who is raising his arm
because his Aunt Gertrude told him to do so is not affecting deployment
one little bit. At some point, you have to trust people to be honest,
but you need to ask the right question. In this case, if I say, "I
can't implement this in my little open source release (that my employer
would never sanction) if you use the FOOBAR codec for such-and-so
reasons", it's reasonable to ask, "Who's in the same boat with Pete?",
perhaps even following up with "Is the WG really OK with the idea the
that folks in Pete's circumstances (open-source developers, 1-coder
shops, holdover Macintosh developers, coders for devices with less than
1K of available memory, etc.) are not going to be able to implement
this?" Maybe the WG will say, "Poor Pete. He's in the rough. We're
chartered to make this work on a wide-scale, and Pete's sort of
implementation is just not realistically a part of what we're doing."
But then you're asking about whether the WG is willing to punt a
particular kind of deployment, not counting heads (hands, souls, auras,
or otherwise).<br>
<br>
<blockquote
 cite="mid:CA+9kkMCuHn3TR7umXtuScWixBfOD9jfS6mu3njxaCOADfG-nvw@mail.gmail.com"
 type="cite">
  <div dir="ltr">
  <div>
  <div class="gmail_extra">
  <div class="gmail_quote">
  <div>The actual system we are trying to build will suffer.&nbsp; Judging
consensus is not here an end in itself, it's a way of working out
whether or not the system we're building does or does not have a way to
avoid the negotiation failure.<br>
  </div>
  </div>
  </div>
  </div>
  </div>
</blockquote>
<br>
We agree on the goal. We disagree on the methodology.<br>
<br>
<blockquote
 cite="mid:CA+9kkMCuHn3TR7umXtuScWixBfOD9jfS6mu3njxaCOADfG-nvw@mail.gmail.com"
 type="cite">
  <div dir="ltr">
  <div>
  <div class="gmail_extra">
  <div class="gmail_quote">
  <div></div>
  <div>We would not have spent the amount of time we have unless that
were important, and we would not be considering using RFC 3929 unless
we thought it was worth getting to a resolution.&nbsp; I implore you to stop
treating this as an academic exercise in consensus theory, and remind
yourself that we're trying to do some engineering here.<br>
  </div>
  </div>
  </div>
  </div>
  </div>
</blockquote>
<br>
Let's avoid questioning each others' motives, shall we? No good down
that road.<br>
<br>
<blockquote
 cite="mid:CA+9kkMCuHn3TR7umXtuScWixBfOD9jfS6mu3njxaCOADfG-nvw@mail.gmail.com"
 type="cite">
  <div dir="ltr">
  <div>
  <div class="gmail_extra">
  <div class="gmail_quote">
  <blockquote class="gmail_quote"
 style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;">But
asking for a show of hands in the room for people who "can live with"
one or the other and deciding whether or not consensus exists based on
that show of hands is simply taking a vote, it's not judging consensus.
  </blockquote>
  <div><br>
  </div>
  <div>That rather presumes you know the count.</div>
  </div>
  </div>
  </div>
  </div>
</blockquote>
<br>
Not at all.<br>
<br>
<blockquote
 cite="mid:CA+9kkMCuHn3TR7umXtuScWixBfOD9jfS6mu3njxaCOADfG-nvw@mail.gmail.com"
 type="cite">
  <div dir="ltr">
  <div>
  <div class="gmail_extra">
  <div class="gmail_quote">
  <div>In several of our recent calls, we have had unanimity, which is
generally held to be a strong indicator of consensus.<br>
  </div>
  </div>
  </div>
  </div>
  </div>
</blockquote>
<br>
Unanimity is, I concede. Other counts have more or less interesting
indications.<br>
<br>
<blockquote
 cite="mid:CA+9kkMCuHn3TR7umXtuScWixBfOD9jfS6mu3njxaCOADfG-nvw@mail.gmail.com"
 type="cite">
  <div dir="ltr">
  <div>
  <div class="gmail_extra">
  <div class="gmail_quote">
  <div>The last time we had a straw poll which used similar language,
we also had absolutely no question that we had not achieved consensus.<br>
  </div>
  </div>
  </div>
  </div>
  </div>
</blockquote>
<br>
You certainly had the appearance of non-consensus. But I'm not
convinced the work was done to determine what the showstoppers were for
the people who "couldn't live with" one choice or the other. (I will
put aside the fact that some chap -- with a blue dot on his badge, no
less -- sitting down the row from me during that straw poll, exclaimed
upon conclusion of it, "Well, it's pretty obvious we have consensus for
'A'". I was.....disappointed.)<br>
<br>
<blockquote
 cite="mid:CA+9kkMCuHn3TR7umXtuScWixBfOD9jfS6mu3njxaCOADfG-nvw@mail.gmail.com"
 type="cite">
  <div dir="ltr">
  <div>
  <div class="gmail_extra">
  <div class="gmail_quote">
  <div>Sometimes signals are strong.<br>
  </div>
  </div>
  </div>
  </div>
  </div>
</blockquote>
<br>
Strong signal of something. I'm not always totally sure what.<br>
&nbsp;
<blockquote
 cite="mid:CA+9kkMCuHn3TR7umXtuScWixBfOD9jfS6mu3njxaCOADfG-nvw@mail.gmail.com"
 type="cite">
  <div dir="ltr">
  <div>
  <div class="gmail_extra">
  <div class="gmail_quote">
  <blockquote class="gmail_quote"
 style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;">You
cannot know from that result whether the reason that some people could
not "live with" a proposal was simply because "my Aunt Gertrude told me
that I should say that I can't live with it." To ask for the show of
hands and then not find out what it means is not calling the consensus.
And I think moving to a 3929 alternative on the basis of that show of
hands would be improper.<br>
  </blockquote>
  <div><br>
If you re-read RFC 3929, you will discover that it requires that you
make an explicit call for its use; the consensus to use it standing in
for the consent on the technical matter.&nbsp; The working group chairs have
no intention of skipping that step, should RFC 3929 be used.</div>
  </div>
  </div>
  </div>
  </div>
</blockquote>
<br>
As above, I worry about how the determination is made. Once made, all
is well.<br>
<br>
<blockquote
 cite="mid:CA+9kkMCuHn3TR7umXtuScWixBfOD9jfS6mu3njxaCOADfG-nvw@mail.gmail.com"
 type="cite">
  <div dir="ltr">
  <div>
  <div class="gmail_extra">
  <div class="gmail_quote">
  <div>fondly,<br>
  </div>
  </div>
  </div>
  </div>
  </div>
</blockquote>
<br>
Right back at ya'.<br>
<br>
pr<br>
<pre class="moz-signature" cols="72">-- 
Pete Resnick <a class="moz-txt-link-rfc2396E" href="http://www.qualcomm.com/~presnick/">&lt;http://www.qualcomm.com/~presnick/&gt;</a>
Qualcomm Technologies, Inc. - +1 (858)651-4478</pre>
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On Fri, Sep 13, 2013 at 2:34 PM, Pete Resnick <presnick@qti.qualcomm.com>wrote:

> **
> On 9/13/13 3:45 PM, Ted Hardie wrote:
>
> I've now had lunch and nice coffee, and I take electrons in hand to answer
> your missive, sent earlier today.
>
>
> Erst kommt das Fressen, dann kommt die Standards.
>
>
Trust me, the food was too good to guzzle.


 On Fri, Sep 13, 2013 at 11:50 AM, Pete Resnick
<presnick@qti.qualcomm.com>wrote:
>
>  Asking "who wants or can live with VP8?" and "who wants or can live with
>> H.264?" might be a good start to the discussion, but at that point, I'm
>> going to want to hear *why* people *can't* live with one or the other.
>
>
>  That discussion is supposed to happen before this, not after...
>
>
> That's good to hear, but at least your present "plan" message, and none
> that I can find elsewhere, makes it apparent that you are now soliciting
> the list of outstanding issues and their purported answers, nor that a
> summary of that list and the closed issues would be available before the
> face-to-face. I'm all for that.
>
>
Um, did you miss the part where folks were supposed to have their proposals
in by the 6th of October so that objections and discussions could happen in
advance of the meeting?

"To enable the WG participants to get answers to any questions, the
proposals in draft form and any supporting material MUST be made
available by 6th of October. This is to ensure that the WG
participants can verify or object to any claims or statements in
the proposal material prior to the WG session"


> I call "horse-pucky". And this is exactly why looking at
> skyward-pointing-appendages is a lousy tool in this particular case. You
> don't know that if they state "they cannot live with this" that they are
> not going to deploy. You would only know that if you knew their reasons.
> Because my friend mentioned below who is raising his arm because his Aunt
> Gertrude told him to do so is not affecting deployment one little bit.
>

We disagree. A set of working group participants will not use a codec
because they have financial reasons, technical reasons, or a personal
promise to their aunts comes down to the same thing:  they will not
implement or deploy.  The impact to the network effect is the same if the
reasoning is philosophical, financial, or personal.


> At some point, you have to trust people to be honest
>

Exactly.  Or, perhaps more to the point, if you cannot trust their
statements on their future course of action, why would you trust their
statement on their reasons?

but you need to ask the right question. In this case, if I say, "I can't
> implement this in my little open source release (that my employer would
> never sanction) if you use the FOOBAR codec for such-and-so reasons", it's
> reasonable to ask, "Who's in the same boat with Pete?", perhaps even
> following up with "Is the WG really OK with the idea the that folks in
> Pete's circumstances (open-source developers, 1-coder shops, holdover
> Macintosh developers, coders for devices with less than 1K of available
> memory, etc.) are not going to be able to implement this?" Maybe the WG
> will say, "Poor Pete. He's in the rough. We're chartered to make this work
> on a wide-scale, and Pete's sort of implementation is just not
> realistically a part of what we're doing."
>

Actually, what we're doing is set out elsewhere, in the charter and the use
case document.  As you consulted the archives on this topic, perhaps you
saw the WGLC on the latter.


> But then you're asking about whether the WG is willing to punt a
> particular kind of deployment, not counting heads (hands, souls, auras, or
> otherwise).
>
>
>    The actual system we are trying to build will suffer.  Judging
> consensus is not here an end in itself, it's a way of working out whether
> or not the system we're building does or does not have a way to avoid the
> negotiation failure.
>
>
> We agree on the goal. We disagree on the methodology.
>
>
>    We would not have spent the amount of time we have unless that were
> important, and we would not be considering using RFC 3929 unless we thought
> it was worth getting to a resolution.  I implore you to stop treating this
> as an academic exercise in consensus theory, and remind yourself that we're
> trying to do some engineering here.
>
>
> Let's avoid questioning each others' motives, shall we? No good down that
> road.
>
>
Let me implore you, then, to look at the engineering aspects rather than
the process aspects.


>
>    But asking for a show of hands in the room for people who "can live
>> with" one or the other and deciding whether or not consensus exists based
>> on that show of hands is simply taking a vote, it's not judging consensus.
>
>
>  That rather presumes you know the count.
>
>
> Not at all.
>
>
>    In several of our recent calls, we have had unanimity, which is
> generally held to be a strong indicator of consensus.
>
>
> Unanimity is, I concede. Other counts have more or less interesting
> indications.
>
>
>    The last time we had a straw poll which used similar language, we also
> had absolutely no question that we had not achieved consensus.
>
>
> You certainly had the appearance of non-consensus. But I'm not convinced
> the work was done to determine what the showstoppers were for the people
> who "couldn't live with" one choice or the other.
>

Had that been the first time we had visited the topic, I would agree with
you.  But it was not even the first handful.



> (I will put aside the fact that some chap -- with a blue dot on his badge,
> no less -- sitting down the row from me during that straw poll, exclaimed
> upon conclusion of it, "Well, it's pretty obvious we have consensus for
> 'A'". I was.....disappointed.)
>
>
The chairs have recused themselves from calling consensus, but had the most
recent straw poll resulted in an assertion of consensus, I would have been
mightily surprised.


>    Sometimes signals are strong.
>
>
> Strong signal of something. I'm not always totally sure what.
>
>
Well, starting from the assumption that people are answering the question
they were asked seems reasonable to me.


>
>
> If you re-read RFC 3929, you will discover that it requires that you make
> an explicit call for its use; the consensus to use it standing in for the
> consent on the technical matter.  The working group chairs have no
> intention of skipping that step, should RFC 3929 be used.
>
>
> As above, I worry about how the determination is made. Once made, all is
> well.
>
>
Should it become necessary, I'm sure that any suggestion you have on
wording such a question to the working group will be given all due
consideration,

Ted


>    fondly,
>
>
> Right back at ya'.
>
> pr
>
> --
> Pete Resnick <http://www.qualcomm.com/~presnick/> <http://www.qualcomm.com/~presnick/>
>
> Qualcomm Technologies, Inc. - +1 (858)651-4478
>
>

--047d7bd76bb20a055e04e64b0daf
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">On Fri, Sep 13, 2013 at 2:34 PM, Pete Resnick <span dir=3D=
"ltr">&lt;<a href=3D"mailto:presnick@qti.qualcomm.com" target=3D"_blank">pr=
esnick@qti.qualcomm.com</a>&gt;</span> wrote:<br><div class=3D"gmail_extra"=
><div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left:1px solid rgb(204,204,204);padding-left:1ex"><u></u>


 =20

<div text=3D"#000000" bgcolor=3D"#ffffff"><div class=3D"im">
On 9/13/13 3:45 PM, Ted Hardie wrote:
<blockquote type=3D"cite">
  <div dir=3D"ltr">I&#39;ve now had lunch and nice coffee, and I take
electrons in hand to answer your missive, sent earlier today.<br>
  </div>
</blockquote>
<br></div>
Erst kommt das Fressen, dann kommt die Standards.<br>
<br></div></blockquote><div><br></div><div>Trust me, the food was too good =
to guzzle.<br></div><div><br><br></div><blockquote class=3D"gmail_quote" st=
yle=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padd=
ing-left:1ex">
<div text=3D"#000000" bgcolor=3D"#ffffff">
<blockquote type=3D"cite">
  <div dir=3D"ltr">
  <div><div class=3D"im">On Fri, Sep 13, 2013 at 11:50 AM, Pete Resnick <sp=
an dir=3D"ltr">&lt;<a href=3D"mailto:presnick@qti.qualcomm.com" target=3D"_=
blank">presnick@qti.qualcomm.com</a>&gt;</span> wrote:<br>
  <br>
  </div><div class=3D"gmail_extra">
  <div class=3D"gmail_quote"><div class=3D"im">
  <blockquote class=3D"gmail_quote" style=3D"border-left:1px solid rgb(204,=
204,204);margin:0px 0px 0px 0.8ex;padding-left:1ex">Asking
&quot;who wants or can live with VP8?&quot; and &quot;who wants or can live=
 with
H.264?&quot; might be a good start to the discussion, but at that point, I&=
#39;m
going to want to hear *why* people *can&#39;t* live with one or the other. =
</blockquote>
  <div><br>
  </div>
  </div><div>That discussion is supposed to happen before this, not after..=
.</div>
  </div>
  </div>
  </div>
  </div>
</blockquote>
<br>
That&#39;s good to hear, but at least your present &quot;plan&quot; message=
, and none
that I can find elsewhere, makes it apparent that you are now
soliciting the list of outstanding issues and their purported answers,
nor that a summary of that list and the closed issues would be
available before the face-to-face. I&#39;m all for that.<br>
<br></div></blockquote><div><br></div><div>Um, did you miss the part where =
folks were supposed to have their proposals in by the 6th of October so tha=
t objections and discussions could happen in advance of the meeting?<br>
</div><div><br>&quot;To enable the WG participants to get answers to any qu=
estions, the<br>proposals in draft form and any supporting material MUST be=
 made<br>available by 6th of October. This is to ensure that the WG<br>
participants can verify or object to any claims or statements in<br>
the proposal material prior to the WG session&quot;<br>=A0</div><blockquote=
 class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px so=
lid rgb(204,204,204);padding-left:1ex"><div text=3D"#000000" bgcolor=3D"#ff=
ffff">
I call &quot;horse-pucky&quot;. And this is exactly why looking at
skyward-pointing-appendages is a lousy tool in this particular case.
You don&#39;t know that if they state &quot;they cannot live with this&quot=
; that
they are not going to deploy. You would only know that if you knew
their reasons. Because my friend mentioned below who is raising his arm
because his Aunt Gertrude told him to do so is not affecting deployment
one little bit.</div></blockquote><div><br></div><div>We disagree. A set of=
 working group participants will not use a codec because they have financia=
l reasons, technical reasons, or a personal promise to their aunts comes do=
wn to the same thing:=A0 they will not implement or deploy.=A0 The impact t=
o the network effect is the same if the reasoning is philosophical, financi=
al, or personal.<br>
=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8e=
x;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div text=3D"#00=
0000" bgcolor=3D"#ffffff"> At some point, you have to trust people to be ho=
nest</div>
</blockquote><div><br></div><div>Exactly.=A0 Or, perhaps more to the point,=
 if you cannot trust their statements on their future course of action, why=
 would you trust their statement on their reasons?<br></div><div><br></div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left:1px solid rgb(204,204,204);padding-left:1ex"><div text=3D"#000000" bgc=
olor=3D"#ffffff">
but you need to ask the right question. In this case, if I say, &quot;I
can&#39;t implement this in my little open source release (that my employer
would never sanction) if you use the FOOBAR codec for such-and-so
reasons&quot;, it&#39;s reasonable to ask, &quot;Who&#39;s in the same boat=
 with Pete?&quot;,
perhaps even following up with &quot;Is the WG really OK with the idea the
that folks in Pete&#39;s circumstances (open-source developers, 1-coder
shops, holdover Macintosh developers, coders for devices with less than
1K of available memory, etc.) are not going to be able to implement
this?&quot; Maybe the WG will say, &quot;Poor Pete. He&#39;s in the rough. =
We&#39;re
chartered to make this work on a wide-scale, and Pete&#39;s sort of
implementation is just not realistically a part of what we&#39;re doing.&qu=
ot;
</div></blockquote><div><br></div><div>Actually, what we&#39;re doing is se=
t out elsewhere, in the charter and the use case document.=A0 As you consul=
ted the archives on this topic, perhaps you saw the WGLC on the latter.<br>
=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8e=
x;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div text=3D"#00=
0000" bgcolor=3D"#ffffff">But then you&#39;re asking about whether the WG i=
s willing to punt a
particular kind of deployment, not counting heads (hands, souls, auras,
or otherwise).<div class=3D"im"><br>
<br>
<blockquote type=3D"cite">
  <div dir=3D"ltr">
  <div>
  <div class=3D"gmail_extra">
  <div class=3D"gmail_quote">
  <div>The actual system we are trying to build will suffer.=A0 Judging
consensus is not here an end in itself, it&#39;s a way of working out
whether or not the system we&#39;re building does or does not have a way to
avoid the negotiation failure.<br>
  </div>
  </div>
  </div>
  </div>
  </div>
</blockquote>
<br></div>
We agree on the goal. We disagree on the methodology.<div class=3D"im"><br>
<br>
<blockquote type=3D"cite">
  <div dir=3D"ltr">
  <div>
  <div class=3D"gmail_extra">
  <div class=3D"gmail_quote">
  <div></div>
  <div>We would not have spent the amount of time we have unless that
were important, and we would not be considering using RFC 3929 unless
we thought it was worth getting to a resolution.=A0 I implore you to stop
treating this as an academic exercise in consensus theory, and remind
yourself that we&#39;re trying to do some engineering here.<br>
  </div>
  </div>
  </div>
  </div>
  </div>
</blockquote>
<br></div>
Let&#39;s avoid questioning each others&#39; motives, shall we? No good dow=
n
that road.<div class=3D"im"><br></div></div></blockquote><div><br></div><di=
v>Let me implore you, then, to look at the engineering aspects rather than =
the process aspects.<br></div><div>=A0</div><blockquote class=3D"gmail_quot=
e" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204)=
;padding-left:1ex">
<div text=3D"#000000" bgcolor=3D"#ffffff"><div class=3D"im">
<br>
<blockquote type=3D"cite">
  <div dir=3D"ltr">
  <div>
  <div class=3D"gmail_extra">
  <div class=3D"gmail_quote">
  <blockquote class=3D"gmail_quote" style=3D"border-left:1px solid rgb(204,=
204,204);margin:0px 0px 0px 0.8ex;padding-left:1ex">But
asking for a show of hands in the room for people who &quot;can live with&q=
uot;
one or the other and deciding whether or not consensus exists based on
that show of hands is simply taking a vote, it&#39;s not judging consensus.
  </blockquote>
  <div><br>
  </div>
  <div>That rather presumes you know the count.</div>
  </div>
  </div>
  </div>
  </div>
</blockquote>
<br></div>
Not at all.<div class=3D"im"><br>
<br>
<blockquote type=3D"cite">
  <div dir=3D"ltr">
  <div>
  <div class=3D"gmail_extra">
  <div class=3D"gmail_quote">
  <div>In several of our recent calls, we have had unanimity, which is
generally held to be a strong indicator of consensus.<br>
  </div>
  </div>
  </div>
  </div>
  </div>
</blockquote>
<br></div>
Unanimity is, I concede. Other counts have more or less interesting
indications.<div class=3D"im"><br>
<br>
<blockquote type=3D"cite">
  <div dir=3D"ltr">
  <div>
  <div class=3D"gmail_extra">
  <div class=3D"gmail_quote">
  <div>The last time we had a straw poll which used similar language,
we also had absolutely no question that we had not achieved consensus.<br>
  </div>
  </div>
  </div>
  </div>
  </div>
</blockquote>
<br></div>
You certainly had the appearance of non-consensus. But I&#39;m not
convinced the work was done to determine what the showstoppers were for
the people who &quot;couldn&#39;t live with&quot; one choice or the other. =
</div></blockquote><div><br></div><div>Had that been the first time we had =
visited the topic, I would agree with you.=A0 But it was not even the first=
 handful.<br>
</div><div><br>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0=
px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><=
div text=3D"#000000" bgcolor=3D"#ffffff">(I will
put aside the fact that some chap -- with a blue dot on his badge, no
less -- sitting down the row from me during that straw poll, exclaimed
upon conclusion of it, &quot;Well, it&#39;s pretty obvious we have consensu=
s for
&#39;A&#39;&quot;. I was.....disappointed.)<br>
<br></div></blockquote><div><br></div><div>The chairs have recused themselv=
es from calling consensus, but had the most recent straw poll resulted in a=
n assertion of consensus, I would have been mightily surprised.<br></div>
<div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px=
 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div text=
=3D"#000000" bgcolor=3D"#ffffff">
<blockquote type=3D"cite">
  <div dir=3D"ltr">
  <div>
  <div class=3D"gmail_extra">
  <div class=3D"gmail_quote">
  <div>Sometimes signals are strong.<br>
  </div>
  </div>
  </div>
  </div>
  </div>
</blockquote>
<br>
Strong signal of something. I&#39;m not always totally sure what.<div class=
=3D"im"><br></div></div></blockquote><div><br></div><div>Well, starting fro=
m the assumption that people are answering the question they were asked see=
ms reasonable to me.<br>
</div><div><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0px =
0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div=
 text=3D"#000000" bgcolor=3D"#ffffff"><div class=3D"im">=A0<blockquote type=
=3D"cite">
<div dir=3D"ltr"><div><div class=3D"gmail_extra"><div class=3D"gmail_quote"=
><div><br>
If you re-read RFC 3929, you will discover that it requires that you
make an explicit call for its use; the consensus to use it standing in
for the consent on the technical matter.=A0 The working group chairs have
no intention of skipping that step, should RFC 3929 be used.</div>
  </div>
  </div>
  </div>
  </div>
</blockquote>
<br></div>
As above, I worry about how the determination is made. Once made, all
is well.<br>
<br></div></blockquote><div><br></div><div>Should it become necessary, I&#3=
9;m sure that any suggestion you have on wording such a question to the wor=
king group will be given all due consideration,<br><br></div><div>Ted<br>
</div><div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0=
px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div =
text=3D"#000000" bgcolor=3D"#ffffff">
<blockquote type=3D"cite">
  <div dir=3D"ltr">
  <div>
  <div class=3D"gmail_extra">
  <div class=3D"gmail_quote">
  <div>fondly,<br>
  </div>
  </div>
  </div>
  </div>
  </div>
</blockquote>
<br>
Right back at ya&#39;.<span class=3D""><font color=3D"#888888"><br>
<br>
pr<br>
</font></span><pre cols=3D"72"><span class=3D""><font color=3D"#888888">--=
=20
Pete Resnick <a href=3D"http://www.qualcomm.com/~presnick/" target=3D"_blan=
k">&lt;http://www.qualcomm.com/~presnick/&gt;</a></font></span><div class=
=3D"im">
Qualcomm Technologies, Inc. - <a href=3D"tel:%2B1%20%28858%29651-4478" valu=
e=3D"+18586514478" target=3D"_blank">+1 (858)651-4478</a></div></pre>
</div>

</blockquote></div><br></div></div>

--047d7bd76bb20a055e04e64b0daf--

From lorenzo@meetecho.com  Fri Sep 13 15:26:44 2013
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Date: Sat, 14 Sep 2013 00:26:35 +0200
From: Lorenzo Miniero <lorenzo@meetecho.com>
To: Ted Hardie <ted.ietf@gmail.com>
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Subject: Re: [rtcweb] Video Codec Selection Plan
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On Fri, 13 Sep 2013 13:45:36 -0700
Ted Hardie <ted.ietf@gmail.com> wrote:

> Howdy,
> 
> I've now had lunch and nice coffee, and I take electrons in hand to answer
> your missive, sent earlier today.
> 
> On Fri, Sep 13, 2013 at 11:50 AM, Pete Resnick <presnick@qti.qualcomm.com>wrote:
> 
> > Big caveat as I post this: I am speaking strictly as an IETF participant.
> > In particular:
> >
> >
> <SNIP>
> 
> 
> > All that said, I am concerned about the path being proposed, and I agree
> > with Matthew's concern (though I feel somewhat less defeatist than his
> > assessment sounds):
> >
> >
> > On 9/13/13 11:59 AM, Matthew Kaufman wrote:
> >
> >> 2. Should I expect "room-packing" for this "show of hands" (which I don't
> >> believe is the typical "hum" process for the IETF, either) as happened
> >> during the SDES discussion last time, and therefore need to bring as many
> >> people who've not participated in RTCWEB previously but who want my codec
> >> to succeed as I can afford to fly to Vancouver?
> >>
> >
> > I think this is a valid concern. Asking "who wants or can live with VP8?"
> > and "who wants or can live with H.264?" might be a good start to the
> > discussion, but at that point, I'm going to want to hear *why* people
> > *can't* live with one or the other.
> 
> 
> That discussion is supposed to happen before this, not after, so that if
> there are issues which might be resolved, they can be.  During the course
> of the effort to find a solution here, there have been several such
> resolutions (Cisco, for example, graciously agreed to provide an open
> source H.264 implementation should it be selected, in order to handle
> objections that none were available).  If there are issues which cannot be
> resolved, having a second discussion on what they are amounts to re-hashing
> and doesn't do anything to move things forward.
> 


Open source implementations are not the issue (we have x264 for that), licensing is.

Lorenzo


> 
> > What I would *not* be interested in is yet another presentation from the
> > proponents of either of these to tell me why one is better;
> 
> 
> One might surmise that this is because you do not intend to actually build,
> deploy, or operate a service that relates to this.  If someone puts forward
> a codec that is better (in human assessment, lines of code, or operating
> requirements) and meets the other criteria for deployment, why wouldn't you
> want to hear about it?
> 
> 
> > I want to hear from *opponents* of the proposal to hear why the proposal
> > would fail.  Only then would I want to hear from proponents responding to
> > those objections. And then I'd like to see a summary of those objections
> > and those responses by the chairs
> 
> 
> Pete, my dear mangel-wurzel and apple of my eye, that's the most process
> wonky thing you've said in a long while, which is saying something.  The
> working group is trying to identify a common codec that can be used to
> avoid negotiation failure.  If there is no common video codec, the network
> effect of this system is much less.
> If there are people who have stated *they cannot live with this*, they are
> not going to build, deploy, or operate services which use the common
> codec.  The actual system we are trying to build will suffer.  Judging
> consensus is not here an end in itself, it's a way of working out whether
> or not the system we're building does or does not have a way to avoid the
> negotiation failure.
> 
> We would not have spent the amount of time we have unless that were
> important, and we would not be considering using RFC 3929 unless we thought
> it was worth getting to a resolution.  I implore you to stop treating this
> as an academic exercise in consensus theory, and remind yourself that we're
> trying to do some engineering here.
> 
> 
> >
> > With that summary, the chairs can make a call of whether there is any hope
> > of consensus. I think much of that work can and should take place on the
> > mailing list *before* the face-to-face meeting.
> >
> >
> But asking for a show of hands in the room for people who "can live with"
> > one or the other and deciding whether or not consensus exists based on that
> > show of hands is simply taking a vote, it's not judging consensus.
> 
> 
> That rather presumes you know the count.  In several of our recent calls,
> we have had unanimity, which is generally held to be a strong indicator of
> consensus.
> 
> The last time we had a straw poll which used similar language, we also had
> absolutely no question that we had not achieved consensus.
> 
> Sometimes signals are strong.
> 
> 
> > You cannot know from that result whether the reason that some people could
> > not "live with" a proposal was simply because "my Aunt Gertrude told me
> > that I should say that I can't live with it." To ask for the show of hands
> > and then not find out what it means is not calling the consensus. And I
> > think moving to a 3929 alternative on the basis of that show of hands would
> > be improper.
> >
> > If you re-read RFC 3929, you will discover that it requires that you make
> an explicit call for its use; the consensus to use it standing in for the
> consent on the technical matter.  The working group chairs have no
> intention of skipping that step, should RFC 3929 be used.
> 
> 
> > Put me down as one voice who objects to the proposed procedure.
> >
> >
> So noted.
> 
> fondly,
> 
> Ted
> 
> 
> 
> > pr
> >
> >
> >  On 9/13/2013 9:52 AM, Ted Hardie wrote:
> >>
> >>> WG,
> >>>
> >>> The chairs have created a plan for how to perform the Video Codec
> >>> selection in our WG. The chairs are asking for review of our plan on
> >>> how to undertake the mandatory-to-implement video codec selection.
> >>> We'd much prefer to have comments on the mechanics before they begin,
> >>> so please review now.  Proponents of a particular proposal should
> >>> note both the actions required and the timelines proposed.
> >>>
> >>> The main goal of this plan is to hold a consensus call on which of
> >>> the proposed alternatives we as a WG should select at one of the WG
> >>> sessions in Vancouver. Such a consensus call will of course be
> >>> verified on the mailing list for anyone who can't participate. The
> >>> chairs will recuse themselves from judging this particular
> >>> consensus.
> >>>
> >>> In the WG session each codec proposal will be allowed an equal amount
> >>> of time to highlight the arguments for their proposal. After that a
> >>> there will be a slot for discussion and clarifying questions.
> >>>
> >>> To enable the WG participants to get answers to any questions, the
> >>> proposals in draft form and any supporting material MUST be made
> >>> available by 6th of October. This is to ensure that the WG
> >>> participants can verify or object to any claims or statements in
> >>> the proposal material prior to the WG session. We chairs would really
> >>> not like to see the proponents bring up new arguments at their
> >>> presentation. Also the WG participants are expected to raise any
> >>> arguments on the list ahead of time to enable the proponents to
> >>> respond to such arguments.
> >>>
> >>> The proposed consensus questions will be of the following form:
> >>>
> >>> 1. If you support H.264 as the mandatory to implement codec or are
> >>> willing to live with it as the MTI, please raise your hand now.
> >>>
> >>> 2. If you support VP8 as the mandatory to implement codec or are
> >>> willing to live with it as the MTI, please raise your hand now.
> >>>
> >>> You may indicate support on both questions and we encourage you to do
> >>> so if you can live with either, even if you have a preference for one
> >>> over the other.
> >>>
> >>> Additional proposals than the previous ones are welcome, but must be
> >>> submitted as draft and their proponents must notify the chairs no later
> >>> than the 6th of October that they also have a candidate proposal.
> >>>
> >>> In case the WG fails to reach consensus we chairs propose that we use
> >>> the alternative decision process as discussed in RFC3929. The method
> >>> and its usage will be discussed on the list should the WG not
> >>> establish consensus on a proposal for mandatory to implement video codec.
> >>>
> >>> regards,
> >>>
> >>> Magnus,  Cullen, and Ted
> >>>
> >>
> > --
> > Pete Resnick<http://www.qualcomm.**com/~presnick/<http://www.qualcomm.com/~presnick/>
> > >
> > Qualcomm Technologies, Inc. - +1 (858)651-4478
> >
> >
> > ______________________________**_________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/mailman/listinfo/rtcweb>
> >


-- 
Lorenzo Miniero, COB

Meetecho s.r.l.
Web Conferencing and Collaboration Tools
http://www.meetecho.com

From ron.even.tlv@gmail.com  Fri Sep 13 15:38:18 2013
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Ted Hardie'" <ted.ietf@gmail.com>, <rtcweb@ietf.org>, "'Cullen Jennings'" <fluffy@cisco.com>, "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>
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Ted,

What is the difference between the proposed questions and the ones that were
asked in Orlando, we had documents and presentations and could not reach a
decision in Orlando. 

If nothing changed we can start with the proposal to use alternative
decision process.

Roni

 

 

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
Ted Hardie
Sent: 13 September, 2013 7:52 PM
To: rtcweb@ietf.org; Cullen Jennings; Magnus Westerlund
Subject: [rtcweb] Video Codec Selection Plan

 

WG,

The chairs have created a plan for how to perform the Video Codec
selection in our WG. The chairs are asking for review of our plan on
how to undertake the mandatory-to-implement video codec selection.
We'd much prefer to have comments on the mechanics before they begin,
so please review now.  Proponents of a particular proposal should
note both the actions required and the timelines proposed.

The main goal of this plan is to hold a consensus call on which of
the proposed alternatives we as a WG should select at one of the WG
sessions in Vancouver. Such a consensus call will of course be
verified on the mailing list for anyone who can't participate. The
chairs will recuse themselves from judging this particular
consensus.

In the WG session each codec proposal will be allowed an equal amount
of time to highlight the arguments for their proposal. After that a
there will be a slot for discussion and clarifying questions.

To enable the WG participants to get answers to any questions, the
proposals in draft form and any supporting material MUST be made
available by 6th of October. This is to ensure that the WG
participants can verify or object to any claims or statements in
the proposal material prior to the WG session. We chairs would really
not like to see the proponents bring up new arguments at their
presentation. Also the WG participants are expected to raise any
arguments on the list ahead of time to enable the proponents to
respond to such arguments.

The proposed consensus questions will be of the following form:

1. If you support H.264 as the mandatory to implement codec or are
willing to live with it as the MTI, please raise your hand now.

2. If you support VP8 as the mandatory to implement codec or are
willing to live with it as the MTI, please raise your hand now.

You may indicate support on both questions and we encourage you to do
so if you can live with either, even if you have a preference for one
over the other.

Additional proposals than the previous ones are welcome, but must be
submitted as draft and their proponents must notify the chairs no later
than the 6th of October that they also have a candidate proposal.

In case the WG fails to reach consensus we chairs propose that we use
the alternative decision process as discussed in RFC3929. The method
and its usage will be discussed on the list should the WG not
establish consensus on a proposal for mandatory to implement video codec.

regards,

Magnus,  Cullen, and Ted


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D'>Ted,<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>What is the difference between the proposed questions and the ones =
that were asked in Orlando, we had documents and presentations and could =
not reach a decision in Orlando. <o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>If nothing changed we can start with the proposal to use alternative =
decision process.<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
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D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
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1.0pt;padding:3.0pt 0in 0in 0in'><p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> =
rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] <b>On Behalf Of =
</b>Ted Hardie<br><b>Sent:</b> 13 September, 2013 7:52 PM<br><b>To:</b> =
rtcweb@ietf.org; Cullen Jennings; Magnus Westerlund<br><b>Subject:</b> =
[rtcweb] Video Codec Selection Plan<o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><div><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'>WG,<br><br>The chairs have created a plan =
for how to perform the Video Codec<br>selection in our WG. The chairs =
are asking for review of our plan on<br>how to undertake the =
mandatory-to-implement video codec selection.<br>We'd much prefer to =
have comments on the mechanics before they begin,<br>so please review =
now.&nbsp; Proponents of a particular proposal should<br>note both the =
actions required and the timelines proposed.<br><br>The main goal of =
this plan is to hold a consensus call on which of<br>the proposed =
alternatives we as a WG should select at one of the WG<br>sessions in =
Vancouver. Such a consensus call will of course be<br>verified on the =
mailing list for anyone who can't participate. The<br>chairs will recuse =
themselves from judging this particular<br>consensus.<br><br>In the WG =
session each codec proposal will be allowed an equal amount<br>of time =
to highlight the arguments for their proposal. After that a<br>there =
will be a slot for discussion and clarifying questions.<br><br>To enable =
the WG participants to get answers to any questions, the<br>proposals in =
draft form and any supporting material MUST be made<br>available by 6th =
of October. This is to ensure that the WG<br>participants can verify or =
object to any claims or statements in<br>the proposal material prior to =
the WG session. We chairs would really<br>not like to see the proponents =
bring up new arguments at their<br>presentation. Also the WG =
participants are expected to raise any<br>arguments on the list ahead of =
time to enable the proponents to<br>respond to such =
arguments.<br><br>The proposed consensus questions will be of the =
following form:<br><br>1. If you support H.264 as the mandatory to =
implement codec or are<br>willing to live with it as the MTI, please =
raise your hand now.<br><br>2. If you support VP8 as the mandatory to =
implement codec or are<br>willing to live with it as the MTI, please =
raise your hand now.<br><br>You may indicate support on both questions =
and we encourage you to do<br>so if you can live with either, even if =
you have a preference for one<br>over the other.<br><br>Additional =
proposals than the previous ones are welcome, but must be<br>submitted =
as draft and their proponents must notify the chairs no later<br>than =
the 6th of October that they also have a candidate proposal.<br><br>In =
case the WG fails to reach consensus we chairs propose that we =
use<br>the alternative decision process as discussed in RFC3929. The =
method<br>and its usage will be discussed on the list should the WG =
not<br>establish consensus on a proposal for mandatory to implement =
video codec.<o:p></o:p></p></div><div><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'>regards,<o:p></o:p></p></div><p =
class=3DMsoNormal>Magnus,&nbsp; Cullen, and =
Ted<o:p></o:p></p></div></div></div></body></html>
------=_NextPart_000_00B4_01CEB0EA.B28E0210--


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From: Bossiel <bossiel@yahoo.fr>
Date: Sat, 14 Sep 2013 01:19:48 +0200
To: Lorenzo Miniero <lorenzo@meetecho.com>
Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, Pete Resnick <presnick@qti.qualcomm.com>
Subject: Re: [rtcweb] Video Codec Selection Plan
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Sent from my iPhone

On Sep 14, 2013, at 0:26, Lorenzo Miniero <lorenzo@meetecho.com> wrote:

> On Fri, 13 Sep 2013 13:45:36 -0700
> Ted Hardie <ted.ietf@gmail.com> wrote:
>=20
>> Howdy,
>>=20
>> I've now had lunch and nice coffee, and I take electrons in hand to answe=
r
>> your missive, sent earlier today.
>>=20
>> On Fri, Sep 13, 2013 at 11:50 AM, Pete Resnick <presnick@qti.qualcomm.com=
>wrote:
>>=20
>>> Big caveat as I post this: I am speaking strictly as an IETF participant=
.
>>> In particular:
>> <SNIP>
>>=20
>>=20
>>> All that said, I am concerned about the path being proposed, and I agree=

>>> with Matthew's concern (though I feel somewhat less defeatist than his
>>> assessment sounds):
>>>=20
>>>=20
>>> On 9/13/13 11:59 AM, Matthew Kaufman wrote:
>>>=20
>>>> 2. Should I expect "room-packing" for this "show of hands" (which I don=
't
>>>> believe is the typical "hum" process for the IETF, either) as happened
>>>> during the SDES discussion last time, and therefore need to bring as ma=
ny
>>>> people who've not participated in RTCWEB previously but who want my cod=
ec
>>>> to succeed as I can afford to fly to Vancouver?
>>>=20
>>> I think this is a valid concern. Asking "who wants or can live with VP8?=
"
>>> and "who wants or can live with H.264?" might be a good start to the
>>> discussion, but at that point, I'm going to want to hear *why* people
>>> *can't* live with one or the other.
>>=20
>>=20
>> That discussion is supposed to happen before this, not after, so that if
>> there are issues which might be resolved, they can be.  During the course=

>> of the effort to find a solution here, there have been several such
>> resolutions (Cisco, for example, graciously agreed to provide an open
>> source H.264 implementation should it be selected, in order to handle
>> objections that none were available).  If there are issues which cannot b=
e
>> resolved, having a second discussion on what they are amounts to re-hashi=
ng
>> and doesn't do anything to move things forward.
>=20
>=20
> Open source implementations are not the issue (we have x264 for that), lic=
ensing is.
x264 is GPL. I guess we're talking about an open source implementation with p=
ermissive license (e.g BSD)
Mamadou,

>=20
> Lorenzo
>=20
>=20
>>=20
>>> What I would *not* be interested in is yet another presentation from the=

>>> proponents of either of these to tell me why one is better;
>>=20
>>=20
>> One might surmise that this is because you do not intend to actually buil=
d,
>> deploy, or operate a service that relates to this.  If someone puts forwa=
rd
>> a codec that is better (in human assessment, lines of code, or operating
>> requirements) and meets the other criteria for deployment, why wouldn't y=
ou
>> want to hear about it?
>>=20
>>=20
>>> I want to hear from *opponents* of the proposal to hear why the proposal=

>>> would fail.  Only then would I want to hear from proponents responding t=
o
>>> those objections. And then I'd like to see a summary of those objections=

>>> and those responses by the chairs
>>=20
>>=20
>> Pete, my dear mangel-wurzel and apple of my eye, that's the most process
>> wonky thing you've said in a long while, which is saying something.  The
>> working group is trying to identify a common codec that can be used to
>> avoid negotiation failure.  If there is no common video codec, the networ=
k
>> effect of this system is much less.
>> If there are people who have stated *they cannot live with this*, they ar=
e
>> not going to build, deploy, or operate services which use the common
>> codec.  The actual system we are trying to build will suffer.  Judging
>> consensus is not here an end in itself, it's a way of working out whether=

>> or not the system we're building does or does not have a way to avoid the=

>> negotiation failure.
>>=20
>> We would not have spent the amount of time we have unless that were
>> important, and we would not be considering using RFC 3929 unless we thoug=
ht
>> it was worth getting to a resolution.  I implore you to stop treating thi=
s
>> as an academic exercise in consensus theory, and remind yourself that we'=
re
>> trying to do some engineering here.
>>=20
>>=20
>>>=20
>>> With that summary, the chairs can make a call of whether there is any ho=
pe
>>> of consensus. I think much of that work can and should take place on the=

>>> mailing list *before* the face-to-face meeting.
>> But asking for a show of hands in the room for people who "can live with"=

>>> one or the other and deciding whether or not consensus exists based on t=
hat
>>> show of hands is simply taking a vote, it's not judging consensus.
>>=20
>>=20
>> That rather presumes you know the count.  In several of our recent calls,=

>> we have had unanimity, which is generally held to be a strong indicator o=
f
>> consensus.
>>=20
>> The last time we had a straw poll which used similar language, we also ha=
d
>> absolutely no question that we had not achieved consensus.
>>=20
>> Sometimes signals are strong.
>>=20
>>=20
>>> You cannot know from that result whether the reason that some people cou=
ld
>>> not "live with" a proposal was simply because "my Aunt Gertrude told me
>>> that I should say that I can't live with it." To ask for the show of han=
ds
>>> and then not find out what it means is not calling the consensus. And I
>>> think moving to a 3929 alternative on the basis of that show of hands wo=
uld
>>> be improper.
>>>=20
>>> If you re-read RFC 3929, you will discover that it requires that you mak=
e
>> an explicit call for its use; the consensus to use it standing in for the=

>> consent on the technical matter.  The working group chairs have no
>> intention of skipping that step, should RFC 3929 be used.
>>=20
>>=20
>>> Put me down as one voice who objects to the proposed procedure.
>> So noted.
>>=20
>> fondly,
>>=20
>> Ted
>>=20
>>=20
>>=20
>>> pr
>>>=20
>>>=20
>>> On 9/13/2013 9:52 AM, Ted Hardie wrote:
>>>>=20
>>>>> WG,
>>>>>=20
>>>>> The chairs have created a plan for how to perform the Video Codec
>>>>> selection in our WG. The chairs are asking for review of our plan on
>>>>> how to undertake the mandatory-to-implement video codec selection.
>>>>> We'd much prefer to have comments on the mechanics before they begin,
>>>>> so please review now.  Proponents of a particular proposal should
>>>>> note both the actions required and the timelines proposed.
>>>>>=20
>>>>> The main goal of this plan is to hold a consensus call on which of
>>>>> the proposed alternatives we as a WG should select at one of the WG
>>>>> sessions in Vancouver. Such a consensus call will of course be
>>>>> verified on the mailing list for anyone who can't participate. The
>>>>> chairs will recuse themselves from judging this particular
>>>>> consensus.
>>>>>=20
>>>>> In the WG session each codec proposal will be allowed an equal amount
>>>>> of time to highlight the arguments for their proposal. After that a
>>>>> there will be a slot for discussion and clarifying questions.
>>>>>=20
>>>>> To enable the WG participants to get answers to any questions, the
>>>>> proposals in draft form and any supporting material MUST be made
>>>>> available by 6th of October. This is to ensure that the WG
>>>>> participants can verify or object to any claims or statements in
>>>>> the proposal material prior to the WG session. We chairs would really
>>>>> not like to see the proponents bring up new arguments at their
>>>>> presentation. Also the WG participants are expected to raise any
>>>>> arguments on the list ahead of time to enable the proponents to
>>>>> respond to such arguments.
>>>>>=20
>>>>> The proposed consensus questions will be of the following form:
>>>>>=20
>>>>> 1. If you support H.264 as the mandatory to implement codec or are
>>>>> willing to live with it as the MTI, please raise your hand now.
>>>>>=20
>>>>> 2. If you support VP8 as the mandatory to implement codec or are
>>>>> willing to live with it as the MTI, please raise your hand now.
>>>>>=20
>>>>> You may indicate support on both questions and we encourage you to do
>>>>> so if you can live with either, even if you have a preference for one
>>>>> over the other.
>>>>>=20
>>>>> Additional proposals than the previous ones are welcome, but must be
>>>>> submitted as draft and their proponents must notify the chairs no late=
r
>>>>> than the 6th of October that they also have a candidate proposal.
>>>>>=20
>>>>> In case the WG fails to reach consensus we chairs propose that we use
>>>>> the alternative decision process as discussed in RFC3929. The method
>>>>> and its usage will be discussed on the list should the WG not
>>>>> establish consensus on a proposal for mandatory to implement video cod=
ec.
>>>>>=20
>>>>> regards,
>>>>>=20
>>>>> Magnus,  Cullen, and Ted
>>> --
>>> Pete Resnick<http://www.qualcomm.**com/~presnick/<http://www.qualcomm.co=
m/~presnick/>
>>> Qualcomm Technologies, Inc. - +1 (858)651-4478
>>>=20
>>>=20
>>> ______________________________**_________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/mail=
man/listinfo/rtcweb>
>=20
>=20
> --=20
> Lorenzo Miniero, COB
>=20
> Meetecho s.r.l.
> Web Conferencing and Collaboration Tools
> http://www.meetecho.com
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From simon.perreault@viagenie.ca  Fri Sep 13 23:02:48 2013
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Le 2013-09-13 22:26, Sergio Garcia Murillo a écrit :
> So, at the end, the ChannelData message is just another TURN message, so
> it will be encapsulated inside a websocket frame. If you feel it is not
> clear, we can add more details about it in next draft.

You're right, I was mistaken about this. Thanks for the detailed 
explanation. No need for more details.

> Regarding RFC 6062, we had some internal discussion regarding if we
> should add websocket support at all to it or not. Mainly, because AFAIK,
> webrtc will use RFC 5766 and not 6062. Anyone please correct me if I am
> wrong.

I think a better justification would be "because WebRTC data channel 
will use SCTP over DTLS".

> I agree with you that this has not been well covered in the draft. In
> order to make the minimun changes possible, and align it with the
> general view of changing just the turn client to turn server
> encapsulation protocol, I would propose to extend websockets usage only
> for the turn control connection in RFC 6062 but keep turn data
> connections on TCP.

So with an HTTP proxy the control would be over websockets and the data 
would be through the CONNECT method? Yuck...

> That, or drop RFC 6062 extension altogether and just
> focus on 5766. What do you feel it would be best?

I say focus on 5766.

I have a new question: what does the TURN server put in the 
XOR-MAPPED-ADDRESS attribute? The proxy's address or the client's 
address? Can TURN over WebSockets be used to gather server-reflexive 
candidates?

Simon
-- 
DTN made easy, lean, and smart --> http://postellation.viagenie.ca
NAT64/DNS64 open-source        --> http://ecdysis.viagenie.ca
STUN/TURN server               --> http://numb.viagenie.ca

From hangzhou.chenxin@huawei.com  Sat Sep 14 01:53:44 2013
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From: "Chenxin (Xin)" <hangzhou.chenxin@huawei.com>
To: Simon Perreault <simon.perreault@viagenie.ca>, Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
Thread-Topic: [rtcweb] [BEHAVE] New Version Notification for draft-chenxin-behave-turn-websocket-01.txt
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Date: Sat, 14 Sep 2013 08:53:21 +0000
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Subject: Re: [rtcweb] [BEHAVE] New Version Notification for	draft-chenxin-behave-turn-websocket-01.txt
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Hi Simon,

>> I agree with you that this has not been well covered in the draft. In
>> order to make the minimun changes possible, and align it with the
>> general view of changing just the turn client to turn server
>> encapsulation protocol, I would propose to extend websockets usage only
>> for the turn control connection in RFC 6062 but keep turn data
>> connections on TCP.
>
>So with an HTTP proxy the control would be over websockets and the data
>would be through the CONNECT method? Yuck...

Or using websocket multiplexing http://tools.ietf.org/html/draft-ietf-hybi-=
websocket-multiplexing-11 .=20
>
>> That, or drop RFC 6062 extension altogether and just
>> focus on 5766. What do you feel it would be best?
>
>I say focus on 5766.
>
>I have a new question: what does the TURN server put in the
>XOR-MAPPED-ADDRESS attribute? The proxy's address or the client's
>address?=20

The proxy's address. I do not think this attribute will help ICE process.=20

Can TURN over WebSockets be used to gather server-reflexive
>candidates?

Yes, It could do as UDP and TCP. But when there is a http proxy. I think th=
e server reflexive candidates will be the address of proxy, which means not=
hing for the peer. But if there is only NAT, the server reflexive candidate=
s is possible to work in the ICE.


Best Regard,
  Xin
>
>Simon
>--
>DTN made easy, lean, and smart --> http://postellation.viagenie.ca
>NAT64/DNS64 open-source        --> http://ecdysis.viagenie.ca
>STUN/TURN server               --> http://numb.viagenie.ca
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb

From simon.perreault@viagenie.ca  Sat Sep 14 02:14:11 2013
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From: Simon Perreault <simon.perreault@viagenie.ca>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, "behave@ietf.org" <behave@ietf.org>
Subject: Re: [rtcweb] [BEHAVE] New Version Notification for draft-chenxin-behave-turn-websocket-01.txt
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Le 2013-09-14 10:53, Chenxin (Xin) a écrit :
>> I have a new question: what does the TURN server put in the
>> XOR-MAPPED-ADDRESS attribute? The proxy's address or the client's
>> address?
>
> The proxy's address. I do not think this attribute will help ICE process.
>
> Can TURN over WebSockets be used to gather server-reflexive
>> candidates?
>
> Yes, It could do as UDP and TCP. But when there is a http proxy. I think the server reflexive candidates will be the address of proxy, which means nothing for the peer.

Isn't this a big problem? I mean, if the client cannot trust the value 
of the XOR-MAPPED-ADDRESS attribute, doesn't that break STUN/TURN/ICE 
and everything else?

Simon
-- 
DTN made easy, lean, and smart --> http://postellation.viagenie.ca
NAT64/DNS64 open-source        --> http://ecdysis.viagenie.ca
STUN/TURN server               --> http://numb.viagenie.ca

From simon.perreault@viagenie.ca  Sat Sep 14 03:01:41 2013
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Le 2013-09-14 01:19, Bossiel a écrit :
>>> Cisco, for example, graciously agreed to provide an open
>>> source H.264 implementation should it be selected, in order to handle
>>> objections that none were available
>>
>> Open source implementations are not the issue (we have x264 for that), licensing is.
> x264 is GPL. I guess we're talking about an open source implementation with permissive license (e.g BSD)

I understand that the issue is the licensing of the IPR, not of the code.

Simon
-- 
DTN made easy, lean, and smart --> http://postellation.viagenie.ca
NAT64/DNS64 open-source        --> http://ecdysis.viagenie.ca
STUN/TURN server               --> http://numb.viagenie.ca

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From: "Chenxin (Xin)" <hangzhou.chenxin@huawei.com>
To: Simon Perreault <simon.perreault@viagenie.ca>
Thread-Topic: [rtcweb] [BEHAVE] New Version Notification for draft-chenxin-behave-turn-websocket-01.txt
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Hi Simon,
>>> I have a new question: what does the TURN server put in the
>>> XOR-MAPPED-ADDRESS attribute? The proxy's address or the client's
>>> address?
>>
>> The proxy's address. I do not think this attribute will help ICE process=
.
>>
>> Can TURN over WebSockets be used to gather server-reflexive
>>> candidates?
>>
>> Yes, It could do as UDP and TCP. But when there is a http proxy. I think=
 the
>server reflexive candidates will be the address of proxy, which means noth=
ing for
>the peer.
>
>Isn't this a big problem? I mean, if the client cannot trust the value
>of the XOR-MAPPED-ADDRESS attribute, doesn't that break STUN/TURN/ICE
>and everything else?
>
Thanks for your question. You are right about it. We should consider more a=
bout gathering the server reflexive candidates in turn over websocket.
I have checked the RFC 5766 again:
      Reflexive Transport Address:  A transport address learned by a client
      that identifies that client as seen by another host on an IP
      network, typically a STUN server.  When there is an intervening
      NAT between the client and the other host, the reflexive transport
      address represents the mapped address allocated to the client on
      the public side of the NAT.  Reflexive transport addresses are
      learned from the mapped address attribute (MAPPED-ADDRESS or XOR-
      MAPPED-ADDRESS) in STUN responses.
Which restricts that the "Reflexive Transport Address" should be obtained f=
rom NAT. It will be semantic problem if this address is from proxy.

In ICE, I have not found out the harmful scenario yet. It will be possible =
to make ICE Connectivity Checks failed, but it will be possible to be usefu=
l.=20
I am still thinking about how to handle it. Should we just leave it to for =
the usage of gathering candidate from NAT? OR totally forbid it in the turn=
 over websocket?
What is your suggestion?

>Simon
>--
>DTN made easy, lean, and smart --> http://postellation.viagenie.ca
>NAT64/DNS64 open-source        --> http://ecdysis.viagenie.ca
>STUN/TURN server               --> http://numb.viagenie.ca

From bossiel@yahoo.fr  Sat Sep 14 04:00:50 2013
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From: Bossiel <bossiel@yahoo.fr>
Date: Sat, 14 Sep 2013 13:00:43 +0200
To: Simon Perreault <simon.perreault@viagenie.ca>
Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Video Codec Selection Plan
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Sent from my iPhone

On Sep 14, 2013, at 12:01, Simon Perreault <simon.perreault@viagenie.ca> wro=
te:

> Le 2013-09-14 01:19, Bossiel a =C3=A9crit :
>>>> Cisco, for example, graciously agreed to provide an open
>>>> source H.264 implementation should it be selected, in order to handle
>>>> objections that none were available
>>>=20
>>> Open source implementations are not the issue (we have x264 for that), l=
icensing is.
>> x264 is GPL. I guess we're talking about an open source implementation wi=
th permissive license (e.g BSD)
>=20
> I understand that the issue is the licensing of the IPR, not of the code.
Both
>=20
> Simon
> --=20
> DTN made easy, lean, and smart --> http://postellation.viagenie.ca
> NAT64/DNS64 open-source        --> http://ecdysis.viagenie.ca
> STUN/TURN server               --> http://numb.viagenie.ca
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From sergio.garcia.murillo@gmail.com  Sat Sep 14 13:30:36 2013
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Subject: Re: [rtcweb] [BEHAVE] New Version Notification for draft-chenxin-behave-turn-websocket-01.txt
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El 14/09/2013 8:02, Simon Perreault escribió:
>
> I have a new question: what does the TURN server put in the 
> XOR-MAPPED-ADDRESS attribute? The proxy's address or the client's 
> address? Can TURN over WebSockets be used to gather server-reflexive 
> candidates?
>
>
 From my point of view, the value of the XOR-MAPPED-ADDRESS should be 
the same as when using TURN over TLS on the same scenario (if it happens 
to work), that would be either the HTTP proxy address or any NAT 
happening after it.

Best regards
Sergio


From martin.thomson@gmail.com  Sat Sep 14 23:43:00 2013
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On 14 September 2013 02:13, Simon Perreault <simon.perreault@viagenie.ca> wrote:
> Isn't this a big problem? I mean, if the client cannot trust the value of
> the XOR-MAPPED-ADDRESS attribute, doesn't that break STUN/TURN/ICE and
> everything else?

Hardly.  You just lose that option.  It might suck if you needed it,
but given that you are using a relay, it's probably not that bad.

The value of XOR-MAPPED-ADDRESS is degraded by a lesser degree with
TURN over TCP.  Same upside.

From simon.perreault@viagenie.ca  Sun Sep 15 00:24:31 2013
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Le 2013-09-15 08:42, Martin Thomson a Ã©crit :
> On 14 September 2013 02:13, Simon Perreault <simon.perreault@viagenie.ca> wrote:
>> Isn't this a big problem? I mean, if the client cannot trust the value of
>> the XOR-MAPPED-ADDRESS attribute, doesn't that break STUN/TURN/ICE and
>> everything else?
>
> Hardly.  You just lose that option.  It might suck if you needed it,
> but given that you are using a relay, it's probably not that bad.
>
> The value of XOR-MAPPED-ADDRESS is degraded by a lesser degree with
> TURN over TCP.  Same upside.

True.

Then shouldn't server-reflexive candidates always be gathered using 
plain STUN or TURN-over-UDP? That is, the XOR-MAPPED-ADDRESS attribute 
would be ignored (or even not sent) for TURN-over-anything-but-UDP.

Simon
-- 
DTN made easy, lean, and smart --> http://postellation.viagenie.ca
NAT64/DNS64 open-source        --> http://ecdysis.viagenie.ca
STUN/TURN server               --> http://numb.viagenie.ca

From magnus.westerlund@ericsson.com  Sun Sep 15 23:04:33 2013
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On 2013-09-14 00:35, Roni Even wrote:
> Ted,
> 
> What is the difference between the proposed questions and the ones that
> were asked in Orlando, we had documents and presentations and could not
> reach a decision in Orlando.
> 
> If nothing changed we can start with the proposal to use alternative
> decision process.
> 

Roni,

In Orlando we took only a straw poll as it was clear to the chairs that
people would have objected against a consensus call. Reasons for that
include the lack of licensing terms related to the IPR that Google from
other IPR right holders.

We like to avoid be in the same situation and be unable to hold a formal
consensus call. Thus, we have set an early dead line for any proposals
to enable all WG participants to review the proposals and raise any
issues on the mailing list. Thus enabling such issues to be responded to
and possibly resolved.

Cheers

Magnus Westerlund
WG chair

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


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From: Chris Wendt <chris-ietf@chriswendt.net>
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I would argue that because we have selected Opus and G.711, we have =
already satisfied the charter requirements.

  6.  Define a set of media formats that must or should be supported by
        a client to improve interoperability.

Can we move on and focus time on more relevant issues?

-Chris


On Sep 13, 2013, at 1:11 PM, Bernard Aboba <bernard_aboba@hotmail.com> =
wrote:

> Ted --
>=20
> The H.264 vs. VP8 discussion has at this point been "overtaken by =
events", so that the proposed plan below would be a waste of everybody's =
time, about as relevant as debating the fashion merits of bell bottoms =
versus nehru jackets. =20
>=20
> With Google having announced plans to implement VP9 (potentially with =
SVC), and with the recent ratification of H.265, the industry has moved =
on, and so should RTCWEB.=20
>=20
> Date: Fri, 13 Sep 2013 09:52:24 -0700
> From: ted.ietf@gmail.com
> To: rtcweb@ietf.org; fluffy@cisco.com; magnus.westerlund@ericsson.com
> Subject: [rtcweb] Video Codec Selection Plan
>=20
> WG,
>=20
> The chairs have created a plan for how to perform the Video Codec
> selection in our WG. The chairs are asking for review of our plan on
> how to undertake the mandatory-to-implement video codec selection.
> We'd much prefer to have comments on the mechanics before they begin,
> so please review now.  Proponents of a particular proposal should
> note both the actions required and the timelines proposed.
>=20
> The main goal of this plan is to hold a consensus call on which of
> the proposed alternatives we as a WG should select at one of the WG
> sessions in Vancouver. Such a consensus call will of course be
> verified on the mailing list for anyone who can't participate. The
> chairs will recuse themselves from judging this particular
> consensus.
>=20
> In the WG session each codec proposal will be allowed an equal amount
> of time to highlight the arguments for their proposal. After that a
> there will be a slot for discussion and clarifying questions.
>=20
> To enable the WG participants to get answers to any questions, the
> proposals in draft form and any supporting material MUST be made
> available by 6th of October. This is to ensure that the WG
> participants can verify or object to any claims or statements in
> the proposal material prior to the WG session. We chairs would really
> not like to see the proponents bring up new arguments at their
> presentation. Also the WG participants are expected to raise any
> arguments on the list ahead of time to enable the proponents to
> respond to such arguments.
>=20
> The proposed consensus questions will be of the following form:
>=20
> 1. If you support H.264 as the mandatory to implement codec or are
> willing to live with it as the MTI, please raise your hand now.
>=20
> 2. If you support VP8 as the mandatory to implement codec or are
> willing to live with it as the MTI, please raise your hand now.
>=20
> You may indicate support on both questions and we encourage you to do
> so if you can live with either, even if you have a preference for one
> over the other.
>=20
> Additional proposals than the previous ones are welcome, but must be
> submitted as draft and their proponents must notify the chairs no =
later
> than the 6th of October that they also have a candidate proposal.
>=20
> In case the WG fails to reach consensus we chairs propose that we use
> the alternative decision process as discussed in RFC3929. The method
> and its usage will be discussed on the list should the WG not
> establish consensus on a proposal for mandatory to implement video =
codec.
>=20
> regards,
>=20
> Magnus,  Cullen, and Ted
>=20
> _______________________________________________ rtcweb mailing list =
rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


--Apple-Mail=_BF6239A8-1C5E-463D-BFC1-F071D33E5387
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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Diso-8859-1"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space;"><div><br></div><div>I would argue that because we =
have selected Opus and G.711, we have already satisfied the charter =
requirements.</div><div><br><span style=3D"background-color: rgb(255, =
255, 255);">&nbsp; 6.&nbsp; Define a set of media formats that must or =
should be supported by</span><br style=3D"background-color: rgb(255, =
255, 255);"><span style=3D"background-color: rgb(255, 255, =
255);">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a client to improve =
interoperability.</span></div><div><br></div><div>Can we move on and =
focus time on more relevant =
issues?</div><div><br></div><div>-Chris</div><div><br></div><br><div><div>=
On Sep 13, 2013, at 1:11 PM, Bernard Aboba &lt;<a =
href=3D"mailto:bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a>&gt=
; wrote:</div><br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><div class=3D"hmmessage" style=3D"font-size: 12pt; =
font-family: Calibri; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: auto; text-align: start; text-indent: 0px; text-transform: =
none; white-space: normal; widows: auto; word-spacing: 0px; =
-webkit-text-stroke-width: 0px;"><div dir=3D"ltr">Ted =
--<div><br></div><div>The H.264 vs. VP8 discussion has at this point =
been "overtaken by events", so that the proposed plan below would be a =
waste of everybody's time, about as relevant as debating the fashion =
merits of bell bottoms versus nehru jackets. =
&nbsp;</div><div><br></div><div>With Google having announced plans to =
implement VP9 (potentially with SVC), and with the recent ratification =
of H.265, the industry has moved on, and so should =
RTCWEB.&nbsp;</div><div><br><div><hr id=3D"stopSpelling">Date: Fri, 13 =
Sep 2013 09:52:24 -0700<br>From: <a =
href=3D"mailto:ted.ietf@gmail.com">ted.ietf@gmail.com</a><br>To: <a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>; <a =
href=3D"mailto:fluffy@cisco.com">fluffy@cisco.com</a>; <a =
href=3D"mailto:magnus.westerlund@ericsson.com">magnus.westerlund@ericsson.=
com</a><br>Subject: [rtcweb] Video Codec Selection Plan<br><br><div =
dir=3D"ltr"><div>WG,<br><br>The chairs have created a plan for how to =
perform the Video Codec<br>selection in our WG. The chairs are asking =
for review of our plan on<br>how to undertake the mandatory-to-implement =
video codec selection.<br>We'd much prefer to have comments on the =
mechanics before they begin,<br>so please review now.&nbsp; Proponents =
of a particular proposal should<br>note both the actions required and =
the timelines proposed.<br><br>The main goal of this plan is to hold a =
consensus call on which of<br>the proposed alternatives we as a WG =
should select at one of the WG<br>sessions in Vancouver. Such a =
consensus call will of course be<br>verified on the mailing list for =
anyone who can't participate. The<br>chairs will recuse themselves from =
judging this particular<br>consensus.<br><br>In the WG session each =
codec proposal will be allowed an equal amount<br>of time to highlight =
the arguments for their proposal. After that a<br>there will be a slot =
for discussion and clarifying questions.<br><br>To enable the WG =
participants to get answers to any questions, the<br>proposals in draft =
form and any supporting material MUST be made<br>available by 6th of =
October. This is to ensure that the WG<br>participants can verify or =
object to any claims or statements in<br>the proposal material prior to =
the WG session. We chairs would really<br>not like to see the proponents =
bring up new arguments at their<br>presentation. Also the WG =
participants are expected to raise any<br>arguments on the list ahead of =
time to enable the proponents to<br>respond to such =
arguments.<br><br>The proposed consensus questions will be of the =
following form:<br><br>1. If you support H.264 as the mandatory to =
implement codec or are<br>willing to live with it as the MTI, please =
raise your hand now.<br><br>2. If you support VP8 as the mandatory to =
implement codec or are<br>willing to live with it as the MTI, please =
raise your hand now.<br><br>You may indicate support on both questions =
and we encourage you to do<br>so if you can live with either, even if =
you have a preference for one<br>over the other.<br><br>Additional =
proposals than the previous ones are welcome, but must be<br>submitted =
as draft and their proponents must notify the chairs no later<br>than =
the 6th of October that they also have a candidate proposal.<br><br>In =
case the WG fails to reach consensus we chairs propose that we =
use<br>the alternative decision process as discussed in RFC3929. The =
method<br>and its usage will be discussed on the list should the WG =
not<br>establish consensus on a proposal for mandatory to implement =
video codec.<br><br></div><div>regards,<br><br></div>Magnus,&nbsp; =
Cullen, and =
Ted<br></div><br>_______________________________________________ rtcweb =
mailing list <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a> <a =
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org=
/mailman/listinfo/rtcweb</a></div></div></div>____________________________=
___________________<br>rtcweb mailing list<br><a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br><a =
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org=
/mailman/listinfo/rtcweb</a></div></blockquote></div><br></body></html>=

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On Sep 16, 2013 6:22 AM, "Chris Wendt" <chris-ietf@chriswendt.net> wrote:
>
>
> I would argue that because we have selected Opus and G.711, we have
already satisfied the charter requirements.
>
>   6.  Define a set of media formats that must or should be supported by
>         a client to improve interoperability.
>
> Can we move on and focus time on more relevant issues?
>

+1, the mti video is the already agreed audio-only.

If your video does not work, you and the calling/called party can discuss
about the ipr issues and politics that limit your communications

CB
> -Chris
>
>
> On Sep 13, 2013, at 1:11 PM, Bernard Aboba <bernard_aboba@hotmail.com>
wrote:
>
>> Ted --
>>
>> The H.264 vs. VP8 discussion has at this point been "overtaken by
events", so that the proposed plan below would be a waste of everybody's
time, about as relevant as debating the fashion merits of bell bottoms
versus nehru jackets.
>>
>> With Google having announced plans to implement VP9 (potentially with
SVC), and with the recent ratification of H.265, the industry has moved on,
and so should RTCWEB.
>>
>> ________________________________
>> Date: Fri, 13 Sep 2013 09:52:24 -0700
>> From: ted.ietf@gmail.com
>> To: rtcweb@ietf.org; fluffy@cisco.com; magnus.westerlund@ericsson.com
>> Subject: [rtcweb] Video Codec Selection Plan
>>
>> WG,
>>
>> The chairs have created a plan for how to perform the Video Codec
>> selection in our WG. The chairs are asking for review of our plan on
>> how to undertake the mandatory-to-implement video codec selection.
>> We'd much prefer to have comments on the mechanics before they begin,
>> so please review now.  Proponents of a particular proposal should
>> note both the actions required and the timelines proposed.
>>
>> The main goal of this plan is to hold a consensus call on which of
>> the proposed alternatives we as a WG should select at one of the WG
>> sessions in Vancouver. Such a consensus call will of course be
>> verified on the mailing list for anyone who can't participate. The
>> chairs will recuse themselves from judging this particular
>> consensus.
>>
>> In the WG session each codec proposal will be allowed an equal amount
>> of time to highlight the arguments for their proposal. After that a
>> there will be a slot for discussion and clarifying questions.
>>
>> To enable the WG participants to get answers to any questions, the
>> proposals in draft form and any supporting material MUST be made
>> available by 6th of October. This is to ensure that the WG
>> participants can verify or object to any claims or statements in
>> the proposal material prior to the WG session. We chairs would really
>> not like to see the proponents bring up new arguments at their
>> presentation. Also the WG participants are expected to raise any
>> arguments on the list ahead of time to enable the proponents to
>> respond to such arguments.
>>
>> The proposed consensus questions will be of the following form:
>>
>> 1. If you support H.264 as the mandatory to implement codec or are
>> willing to live with it as the MTI, please raise your hand now.
>>
>> 2. If you support VP8 as the mandatory to implement codec or are
>> willing to live with it as the MTI, please raise your hand now.
>>
>> You may indicate support on both questions and we encourage you to do
>> so if you can live with either, even if you have a preference for one
>> over the other.
>>
>> Additional proposals than the previous ones are welcome, but must be
>> submitted as draft and their proponents must notify the chairs no later
>> than the 6th of October that they also have a candidate proposal.
>>
>> In case the WG fails to reach consensus we chairs propose that we use
>> the alternative decision process as discussed in RFC3929. The method
>> and its usage will be discussed on the list should the WG not
>> establish consensus on a proposal for mandatory to implement video codec.
>>
>> regards,
>>
>> Magnus,  Cullen, and Ted
>>
>> _______________________________________________ rtcweb mailing list
rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<p dir=3D"ltr"><br>
On Sep 16, 2013 6:22 AM, &quot;Chris Wendt&quot; &lt;<a href=3D"mailto:chri=
s-ietf@chriswendt.net">chris-ietf@chriswendt.net</a>&gt; wrote:<br>
&gt;<br>
&gt;<br>
&gt; I would argue that because we have selected Opus and G.711, we have al=
ready satisfied the charter requirements.<br>
&gt;<br>
&gt; =A0 6.=A0 Define a set of media formats that must or should be support=
ed by<br>
&gt; =A0=A0=A0=A0=A0=A0=A0 a client to improve interoperability.<br>
&gt;<br>
&gt; Can we move on and focus time on more relevant issues?<br>
&gt;</p>
<p dir=3D"ltr">+1, the mti video is the already agreed audio-only. </p>
<p dir=3D"ltr">If your video does not work, you and the calling/called part=
y can discuss about the ipr issues and politics that limit your communicati=
ons</p>
<p dir=3D"ltr">CB<br>
&gt; -Chris<br>
&gt;<br>
&gt;<br>
&gt; On Sep 13, 2013, at 1:11 PM, Bernard Aboba &lt;<a href=3D"mailto:berna=
rd_aboba@hotmail.com">bernard_aboba@hotmail.com</a>&gt; wrote:<br>
&gt;<br>
&gt;&gt; Ted --<br>
&gt;&gt;<br>
&gt;&gt; The H.264 vs. VP8 discussion has at this point been &quot;overtake=
n by events&quot;, so that the proposed plan below would be a waste of ever=
ybody&#39;s time, about as relevant as debating the fashion merits of bell =
bottoms versus nehru jackets. =A0<br>

&gt;&gt;<br>
&gt;&gt; With Google having announced plans to implement VP9 (potentially w=
ith SVC), and with the recent ratification of H.265, the industry has moved=
 on, and so should RTCWEB.=A0<br>
&gt;&gt;<br>
&gt;&gt; ________________________________<br>
&gt;&gt; Date: Fri, 13 Sep 2013 09:52:24 -0700<br>
&gt;&gt; From: <a href=3D"mailto:ted.ietf@gmail.com">ted.ietf@gmail.com</a>=
<br>
&gt;&gt; To: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>; <a hre=
f=3D"mailto:fluffy@cisco.com">fluffy@cisco.com</a>; <a href=3D"mailto:magnu=
s.westerlund@ericsson.com">magnus.westerlund@ericsson.com</a><br>
&gt;&gt; Subject: [rtcweb] Video Codec Selection Plan<br>
&gt;&gt;<br>
&gt;&gt; WG,<br>
&gt;&gt;<br>
&gt;&gt; The chairs have created a plan for how to perform the Video Codec<=
br>
&gt;&gt; selection in our WG. The chairs are asking for review of our plan =
on<br>
&gt;&gt; how to undertake the mandatory-to-implement video codec selection.=
<br>
&gt;&gt; We&#39;d much prefer to have comments on the mechanics before they=
 begin,<br>
&gt;&gt; so please review now.=A0 Proponents of a particular proposal shoul=
d<br>
&gt;&gt; note both the actions required and the timelines proposed.<br>
&gt;&gt;<br>
&gt;&gt; The main goal of this plan is to hold a consensus call on which of=
<br>
&gt;&gt; the proposed alternatives we as a WG should select at one of the W=
G<br>
&gt;&gt; sessions in Vancouver. Such a consensus call will of course be<br>
&gt;&gt; verified on the mailing list for anyone who can&#39;t participate.=
 The<br>
&gt;&gt; chairs will recuse themselves from judging this particular<br>
&gt;&gt; consensus.<br>
&gt;&gt;<br>
&gt;&gt; In the WG session each codec proposal will be allowed an equal amo=
unt<br>
&gt;&gt; of time to highlight the arguments for their proposal. After that =
a<br>
&gt;&gt; there will be a slot for discussion and clarifying questions.<br>
&gt;&gt;<br>
&gt;&gt; To enable the WG participants to get answers to any questions, the=
<br>
&gt;&gt; proposals in draft form and any supporting material MUST be made<b=
r>
&gt;&gt; available by 6th of October. This is to ensure that the WG<br>
&gt;&gt; participants can verify or object to any claims or statements in<b=
r>
&gt;&gt; the proposal material prior to the WG session. We chairs would rea=
lly<br>
&gt;&gt; not like to see the proponents bring up new arguments at their<br>
&gt;&gt; presentation. Also the WG participants are expected to raise any<b=
r>
&gt;&gt; arguments on the list ahead of time to enable the proponents to<br=
>
&gt;&gt; respond to such arguments.<br>
&gt;&gt;<br>
&gt;&gt; The proposed consensus questions will be of the following form:<br=
>
&gt;&gt;<br>
&gt;&gt; 1. If you support H.264 as the mandatory to implement codec or are=
<br>
&gt;&gt; willing to live with it as the MTI, please raise your hand now.<br=
>
&gt;&gt;<br>
&gt;&gt; 2. If you support VP8 as the mandatory to implement codec or are<b=
r>
&gt;&gt; willing to live with it as the MTI, please raise your hand now.<br=
>
&gt;&gt;<br>
&gt;&gt; You may indicate support on both questions and we encourage you to=
 do<br>
&gt;&gt; so if you can live with either, even if you have a preference for =
one<br>
&gt;&gt; over the other.<br>
&gt;&gt;<br>
&gt;&gt; Additional proposals than the previous ones are welcome, but must =
be<br>
&gt;&gt; submitted as draft and their proponents must notify the chairs no =
later<br>
&gt;&gt; than the 6th of October that they also have a candidate proposal.<=
br>
&gt;&gt;<br>
&gt;&gt; In case the WG fails to reach consensus we chairs propose that we =
use<br>
&gt;&gt; the alternative decision process as discussed in RFC3929. The meth=
od<br>
&gt;&gt; and its usage will be discussed on the list should the WG not<br>
&gt;&gt; establish consensus on a proposal for mandatory to implement video=
 codec.<br>
&gt;&gt;<br>
&gt;&gt; regards,<br>
&gt;&gt;<br>
&gt;&gt; Magnus,=A0 Cullen, and Ted<br>
&gt;&gt;<br>
&gt;&gt; _______________________________________________ rtcweb mailing lis=
t <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a> <a href=3D"https:/=
/www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinf=
o/rtcweb</a><br>

&gt;&gt; _______________________________________________<br>
&gt;&gt; rtcweb mailing list<br>
&gt;&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://w=
ww.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.i=
etf.org/mailman/listinfo/rtcweb</a><br>
&gt;<br>
</p>

--f46d043c7c564b02f804e683aa86--

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References: <20130913005837.14362.66591.idtracker@ietfa.amsl.com> <9E34D50A21D1D1489134B4D770CE03976807F0B0@SZXEMA504-MBX.china.huawei.com> <5232D9A2.8050800@viagenie.ca> <52337505.9000109@gmail.com> <5233FC04.7040509@viagenie.ca> <9E34D50A21D1D1489134B4D770CE03976807F388@SZXEMA504-MBX.china.huawei.com> <523428D2.8050505@viagenie.ca> <CABkgnnVcQ7FZnrBCVzmzkj7PJhwEQomY=e_LACjkThmoCHqhqw@mail.gmail.com> <523560AD.8070109@viagenie.ca>
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From: Matt Fredrickson <creslin@digium.com>
To: Simon Perreault <simon.perreault@viagenie.ca>
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Cc: "behave@ietf.org" <behave@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] [BEHAVE] New Version Notification for draft-chenxin-behave-turn-websocket-01.txt
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--001a11c233ea1f7fa404e6855d84
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On Sun, Sep 15, 2013 at 2:24 AM, Simon Perreault <
simon.perreault@viagenie.ca> wrote:

> Le 2013-09-15 08:42, Martin Thomson a =E9crit :
>
>  On 14 September 2013 02:13, Simon Perreault <simon.perreault@viagenie.ca=
>
>> wrote:
>>
>>> Isn't this a big problem? I mean, if the client cannot trust the value =
of
>>> the XOR-MAPPED-ADDRESS attribute, doesn't that break STUN/TURN/ICE and
>>> everything else?
>>>
>>
>> Hardly.  You just lose that option.  It might suck if you needed it,
>> but given that you are using a relay, it's probably not that bad.
>>
>> The value of XOR-MAPPED-ADDRESS is degraded by a lesser degree with
>> TURN over TCP.  Same upside.
>>
>
> True.
>
> Then shouldn't server-reflexive candidates always be gathered using plain
> STUN or TURN-over-UDP? That is, the XOR-MAPPED-ADDRESS attribute would be
> ignored (or even not sent) for TURN-over-anything-but-UDP.


That's what I was beginning to think as well...  Is there any utility to be
gained by advertising XOR-MAPPED-ADDRESS on transport types that (at least
it would so seem to me) preclude its usage?

Matthew Fredrickson

--001a11c233ea1f7fa404e6855d84
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<div dir=3D"ltr"><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quo=
te">On Sun, Sep 15, 2013 at 2:24 AM, Simon Perreault <span dir=3D"ltr">&lt;=
<a href=3D"mailto:simon.perreault@viagenie.ca" target=3D"_blank">simon.perr=
eault@viagenie.ca</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Le 2013-09-15 08:42, Martin Thomson a =E9cri=
t :<div class=3D"im"><br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On 14 September 2013 02:13, Simon Perreault &lt;<a href=3D"mailto:simon.per=
reault@viagenie.ca" target=3D"_blank">simon.perreault@viagenie.ca</a>&gt; w=
rote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Isn&#39;t this a big problem? I mean, if the client cannot trust the value =
of<br>
the XOR-MAPPED-ADDRESS attribute, doesn&#39;t that break STUN/TURN/ICE and<=
br>
everything else?<br>
</blockquote>
<br>
Hardly. =A0You just lose that option. =A0It might suck if you needed it,<br=
>
but given that you are using a relay, it&#39;s probably not that bad.<br>
<br>
The value of XOR-MAPPED-ADDRESS is degraded by a lesser degree with<br>
TURN over TCP. =A0Same upside.<br>
</blockquote>
<br></div>
True.<br>
<br>
Then shouldn&#39;t server-reflexive candidates always be gathered using pla=
in STUN or TURN-over-UDP? That is, the XOR-MAPPED-ADDRESS attribute would b=
e ignored (or even not sent) for TURN-over-anything-but-UDP.</blockquote>
<div><br></div><div>That&#39;s what I was beginning to think as well... =A0=
Is there any utility to be gained by advertising XOR-MAPPED-ADDRESS on tran=
sport types that (at least it would so seem to me) preclude its usage?</div=
>
<div><br></div><div>Matthew Fredrickson</div></div></div></div>

--001a11c233ea1f7fa404e6855d84--

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From: "Ram Mohan R (rmohanr)" <rmohanr@cisco.com>
To: "Lijing (Jessie, Huawei)" <lijing80@huawei.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] FW:  Adopting draft-muthu-behave-consent-freshness?
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Date: Tue, 17 Sep 2013 05:10:22 +0000
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Subject: Re: [rtcweb] FW:  Adopting draft-muthu-behave-consent-freshness?
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Hi Please see inline

-----Original Message-----
From: Jessie <Lijing>, "Huawei)" <lijing80@huawei.com>
Date: Thursday, 12 September 2013 1:27 PM
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: [rtcweb] FW:  Adopting draft-muthu-behave-consent-freshness?

>Hi all,
>
>As a newcomer to RTCWEB, I have some doubts after I have read the draft.
>
>1. in "4.  Solution Overview", may be it would be better to clarify when
>to send a STUN Binding Request(in the middle of a media session or before
>sending traffic, during sending traffic or only during silence periods)
>and which side to send the request(controlling agent or both sides)?

The introduction section explains when consent is needed. During the
initial call setup ICE connectivity checks are used. This mechanism
described in this document is for periodic consent after initial sessions
is setup.

>
>2. in " 7.  Security Considerations", there are the following words. As
>when one agent receives STUN Binding Request, unlike the processing of
>indication message, it have to do more processes to send the Response
>message, I am not sure whether it is appreciate not to authenticate the
>source and respond directly. Is it more open to malicious attacks?

I am not clear on what you asking here. The below text just tells that
there is no need to re-asserting the username/password that was sent
initially during ICE checks. Are you saying this can lead to attacks ? If
yes can you explain on what you meant in detail ?

Ram

>
>  "Once that connection to the remote
>   peer has been established with ICE, the consent to continue sending
>   traffic does not benefit from re-asserting that same username and
>   password, so long as the senders and receiver's IP addresses remain
>   the same (as they usually do)."
>
>Neglect my words, if my understandings are wrong.
>
>Best regards,
>
>Jessie
>
>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>Of Magnus Westerlund
>Sent: Monday, September 09, 2013 4:37 PM
>To: rtcweb@ietf.org
>Subject: [rtcweb] Adopting draft-muthu-behave-consent-freshness?
>
>WG,
>
>This is a call for WG adoption of STUN Usage for Consent Freshness
>(draft-muthu-behave-consent-freshness-04). This document defines a STUN
>usage for consent freshness. As this requires no protocol extensions we
>as intended users can define this usage in our WG. Such work also
>matches our charter. The draft-ietf-rtcweb-security-arch-07 is
>normatively dependent on this STUN usage.
>
>Document:
>https://datatracker.ietf.org/doc/draft-muthu-behave-consent-freshness/
>
>WG, please indicate your support or issues with adopting this document
>as WG item with a proposed milestone:
>
>Mar 2014 Send STUN Usage for Consent Freshness to IESG for publication
>as proposed standard.
>
>Cheers
>
>Magnus Westerlund
>
>----------------------------------------------------------------------
>Multimedia Technologies, Ericsson Research EAB/TVM
>----------------------------------------------------------------------
>Ericsson AB                | Phone  +46 10 7148287
>F=E4r=F6gatan 6                | Mobile +46 73 0949079
>SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>----------------------------------------------------------------------
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb


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Subject: [rtcweb] Continuing to assert username/password (Re: FW: Adopting draft-muthu-behave-consent-freshness?)
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Changing subject line in order to make life easier for those who try to 
review comments on the original subject....

On 09/17/2013 07:10 AM, Ram Mohan R (rmohanr) wrote:
> Hi Please see inline
>
> -----Original Message-----
> From: Jessie <Lijing>, "Huawei)" <lijing80@huawei.com>
> Date: Thursday, 12 September 2013 1:27 PM
> To: "rtcweb@ietf.org" <rtcweb@ietf.org>
> Subject: [rtcweb] FW:  Adopting draft-muthu-behave-consent-freshness?
>
>> Hi all,
>>
>> As a newcomer to RTCWEB, I have some doubts after I have read the draft.
>>
>> 1. in "4.  Solution Overview", may be it would be better to clarify when
>> to send a STUN Binding Request(in the middle of a media session or before
>> sending traffic, during sending traffic or only during silence periods)
>> and which side to send the request(controlling agent or both sides)?
> The introduction section explains when consent is needed. During the
> initial call setup ICE connectivity checks are used. This mechanism
> described in this document is for periodic consent after initial sessions
> is setup.
>
>> 2. in " 7.  Security Considerations", there are the following words. As
>> when one agent receives STUN Binding Request, unlike the processing of
>> indication message, it have to do more processes to send the Response
>> message, I am not sure whether it is appreciate not to authenticate the
>> source and respond directly. Is it more open to malicious attacks?
> I am not clear on what you asking here. The below text just tells that
> there is no need to re-asserting the username/password that was sent
> initially during ICE checks. Are you saying this can lead to attacks ? If
> yes can you explain on what you meant in detail ?

The text below is wrong.

If, as is commonly the case, an attacker has the ability to inject 
packets on the wire with the same IP address as the victim, the attacker 
can forge ICE packets.

If the victim's correspondent checks MESSAGE-INTEGRITY (which implicitly 
checks username and password), he will detect that these packets don't 
have the correct authorization, and throw them away.

If the victim's correspondent does not check MESSAGE-INTEGRITY, the 
attacker can keep renewing any consent to receive as long as he wants, 
even though the victim wanted to stop consenting.

Since the whole purpose of consent-freshness is to allow the potential 
victim to stop consenting to receiving traffic, checking 
MESSAGE-INTEGRITY (which implictily checks username and password) is 
necessary.

This whole paragraph should be deleted from the "Security 
Considerations" section and replaced with text properly describing the 
situation.

>
> Ram
>
>>   "Once that connection to the remote
>>    peer has been established with ICE, the consent to continue sending
>>    traffic does not benefit from re-asserting that same username and
>>    password, so long as the senders and receiver's IP addresses remain
>>    the same (as they usually do)."
>>
>> Neglect my words, if my understandings are wrong.
>>
>> Best regards,
>>
>> Jessie
>>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>> Of Magnus Westerlund
>> Sent: Monday, September 09, 2013 4:37 PM
>> To: rtcweb@ietf.org
>> Subject: [rtcweb] Adopting draft-muthu-behave-consent-freshness?
>>
>> WG,
>>
>> This is a call for WG adoption of STUN Usage for Consent Freshness
>> (draft-muthu-behave-consent-freshness-04). This document defines a STUN
>> usage for consent freshness. As this requires no protocol extensions we
>> as intended users can define this usage in our WG. Such work also
>> matches our charter. The draft-ietf-rtcweb-security-arch-07 is
>> normatively dependent on this STUN usage.
>>
>> Document:
>> https://datatracker.ietf.org/doc/draft-muthu-behave-consent-freshness/
>>
>> WG, please indicate your support or issues with adopting this document
>> as WG item with a proposed milestone:
>>
>> Mar 2014 Send STUN Usage for Consent Freshness to IESG for publication
>> as proposed standard.
>>
>> Cheers
>>
>> Magnus Westerlund
>>
>> ----------------------------------------------------------------------
>> Multimedia Technologies, Ericsson Research EAB/TVM
>> ----------------------------------------------------------------------
>> Ericsson AB                | Phone  +46 10 7148287
>> Färögatan 6                | Mobile +46 73 0949079
>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>> ----------------------------------------------------------------------
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From magnus.westerlund@ericsson.com  Tue Sep 17 01:37:43 2013
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Subject: Re: [rtcweb] JSEP Walk Through on Wednesday the 18th of September at 17.00 CEST (8 AM PDT)
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WG,

I know that the new JSEP Draft is not yet submitted. I am hoping that it
will show up during the morning hours for the US based people. That way
everyone gets at least some few hours to look at the draft prior to the
call. If it hasn't we will postpone the call, but I really hope we can
avoid this. I will send a notice to the WG if we have to postpone it.

Cheers

Magnus

On 2013-09-11 07:58, Magnus Westerlund wrote:
> WG,
> 
> There will be a new version of JSEP draft coming out Monday. As a follow
> up to get as much progress on this topic as possible. The chairs and
> authors are offering a informal walk through session on Wednesday the
> 18th. People will be strongly recommended to have reviewed the new draft.
> 
> The intention of the session is enable the authors to inform you about
> the latest changes, any issues to consider and what needs further
> discussion. The will also enable you as participant after your review of
> the document to ask clarifying questions or for motivations behind
> choices, and raise issues you have spotted.
> 
> The session is not intended to solve issues or judge consensus on part
> of the draft. Discussions of what is the best solution etc, will to be
> cut off.
> 
> This will be a webex conference per details below.
> 
> Cheers
> 
> Magnus Westerlund
> 
> Topic: RTCWeb - JSEP
> Date: Wednesday, September 18, 2013
> Time: 8:00 am, Pacific Daylight Time (San Francisco, GMT-07:00)
> Meeting Number: 302 061 238
> Meeting Password: ietf
> 
> -------------------------------------------------------
> To join the online meeting (Now from mobile devices!)
> -------------------------------------------------------
> 1. Go to
> https://cisco.webex.com/ciscosales/j.php?ED=206400898&UID=0&PW=NYjk4NmE0ODBj&RT=MiM0
> 
> 2. Enter your name and email address.
> 3. Enter the meeting password: ietf
> 4. Click "Join Now".
> 
> To view in other time zones or languages, please click the link:
> https://cisco.webex.com/ciscosales/j.php?ED=206400898&UID=0&PW=NYjk4NmE0ODBj&ORT=MiM0
> 
> 
> -------------------------------------------------------
> To join the teleconference only
> -------------------------------------------------------
> 1. Dial into Cisco WebEx (view all Global Access Numbers at
> http://cisco.com/en/US/about/doing_business/conferencing/index.html
> 2. Follow the prompts to enter the Meeting Number (listed above) or
> Access Code followed by the # sign.
> 
> San Jose, CA: +1.408.525.6800 RTP: +1.919.392.3330
> 
> US/Canada: +1.866.432.9903 United Kingdom: +44.20.8824.0117
> 
> India: +91.80.4350.1111 Germany: +49.619.6773.9002
> 
> Japan: +81.3.5763.9394 China: +86.10.8515.5666
> 
> 
> ----------------------------------------------------------------
> ALERT – PLEASE READ: DO NOT DIAL THE TOLL FREE NUMBERS FROM WITHIN THE
> (408) OR (919) AREA CODES
> ----------------------------------------------------------------
> Please dial the local access number for your area from the list below:
> - San Jose/Milpitas (408) area: 525-6800
> - RTP (919) area: 392-3330
> 
> Dialing the WebEx toll free numbers from within 408 or 919 area codes is
> not enabled (non-Cisco phones). “ If you dial the toll-free numbers
> within the 408 or 919 area codes you will be instructed to hang up and
> dial the local access number.” Please use the call-back option whenever
> possible and otherwise dial local numbers only. The affected toll free
> numbers are: (866) 432-9903 for the San Jose/Milpitas area and (866)
> 349-3520 for the RTP area.
> 


-- 

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From magnus.westerlund@ericsson.com  Tue Sep 17 01:45:16 2013
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Subject: Re: [rtcweb] Adopting draft-muthu-behave-consent-freshness?
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WG,

More than a week since this call was issued has passed. There has been
strong support for adopting this WG item and the draft as a starting
point. I will therefore request approval for the milestone from the AD.

As pointed out there is clearly some issues in the draft that needs to
be addressed. These will now be the WG concern. So please continue
review and provide proposals for improving the document.

Cheers

Magnus Westerlund
WG Chair

On 2013-09-09 10:36, Magnus Westerlund wrote:
> WG,
> 
> This is a call for WG adoption of STUN Usage for Consent Freshness
> (draft-muthu-behave-consent-freshness-04). This document defines a STUN
> usage for consent freshness. As this requires no protocol extensions we
> as intended users can define this usage in our WG. Such work also
> matches our charter. The draft-ietf-rtcweb-security-arch-07 is
> normatively dependent on this STUN usage.
> 
> Document:
> https://datatracker.ietf.org/doc/draft-muthu-behave-consent-freshness/
> 
> WG, please indicate your support or issues with adopting this document
> as WG item with a proposed milestone:
> 
> Mar 2014 Send STUN Usage for Consent Freshness to IESG for publication
> as proposed standard.
> 
> Cheers
> 
> Magnus Westerlund
> 
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> Färögatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> 
> 


-- 

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


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Subject: Re: [rtcweb] JSEP Walk Through on Wednesday the 18th of September at 17.00 CEST (8 AM PDT)
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Magnus,

what is the expected duration of the call?

On 09/17/2013 10:38 AM, Magnus Westerlund wrote:
> WG,
>
> I know that the new JSEP Draft is not yet submitted. I am hoping that it
> will show up during the morning hours for the US based people. That way
> everyone gets at least some few hours to look at the draft prior to the
> call. If it hasn't we will postpone the call, but I really hope we can
> avoid this. I will send a notice to the WG if we have to postpone it.
>
> Cheers
>
> Magnus
>
> On 2013-09-11 07:58, Magnus Westerlund wrote:
>> WG,
>>
>> There will be a new version of JSEP draft coming out Monday. As a follow
>> up to get as much progress on this topic as possible. The chairs and
>> authors are offering a informal walk through session on Wednesday the
>> 18th. People will be strongly recommended to have reviewed the new draft.
>>
>> The intention of the session is enable the authors to inform you about
>> the latest changes, any issues to consider and what needs further
>> discussion. The will also enable you as participant after your review of
>> the document to ask clarifying questions or for motivations behind
>> choices, and raise issues you have spotted.
>>
>> The session is not intended to solve issues or judge consensus on part
>> of the draft. Discussions of what is the best solution etc, will to be
>> cut off.
>>
>> This will be a webex conference per details below.
>>
>> Cheers
>>
>> Magnus Westerlund
>>
>> Topic: RTCWeb - JSEP
>> Date: Wednesday, September 18, 2013
>> Time: 8:00 am, Pacific Daylight Time (San Francisco, GMT-07:00)
>> Meeting Number: 302 061 238
>> Meeting Password: ietf
>>
>> -------------------------------------------------------
>> To join the online meeting (Now from mobile devices!)
>> -------------------------------------------------------
>> 1. Go to
>> https://cisco.webex.com/ciscosales/j.php?ED=206400898&UID=0&PW=NYjk4NmE0ODBj&RT=MiM0
>>
>> 2. Enter your name and email address.
>> 3. Enter the meeting password: ietf
>> 4. Click "Join Now".
>>
>> To view in other time zones or languages, please click the link:
>> https://cisco.webex.com/ciscosales/j.php?ED=206400898&UID=0&PW=NYjk4NmE0ODBj&ORT=MiM0
>>
>>
>> -------------------------------------------------------
>> To join the teleconference only
>> -------------------------------------------------------
>> 1. Dial into Cisco WebEx (view all Global Access Numbers at
>> http://cisco.com/en/US/about/doing_business/conferencing/index.html
>> 2. Follow the prompts to enter the Meeting Number (listed above) or
>> Access Code followed by the # sign.
>>
>> San Jose, CA: +1.408.525.6800 RTP: +1.919.392.3330
>>
>> US/Canada: +1.866.432.9903 United Kingdom: +44.20.8824.0117
>>
>> India: +91.80.4350.1111 Germany: +49.619.6773.9002
>>
>> Japan: +81.3.5763.9394 China: +86.10.8515.5666
>>
>>
>> ----------------------------------------------------------------
>> ALERT – PLEASE READ: DO NOT DIAL THE TOLL FREE NUMBERS FROM WITHIN THE
>> (408) OR (919) AREA CODES
>> ----------------------------------------------------------------
>> Please dial the local access number for your area from the list below:
>> - San Jose/Milpitas (408) area: 525-6800
>> - RTP (919) area: 392-3330
>>
>> Dialing the WebEx toll free numbers from within 408 or 919 area codes is
>> not enabled (non-Cisco phones). “ If you dial the toll-free numbers
>> within the 408 or 919 area codes you will be instructed to hang up and
>> dial the local access number.” Please use the call-back option whenever
>> possible and otherwise dial local numbers only. The affected toll free
>> numbers are: (866) 432-9903 for the San Jose/Milpitas area and (866)
>> 349-3520 for the RTP area.
>>
>


From magnus.westerlund@ericsson.com  Tue Sep 17 01:50:05 2013
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Date: Tue, 17 Sep 2013 10:51:08 +0200
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] JSEP Walk Through on Wednesday the 18th of September at 17.00 CEST (8 AM PDT)
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On 2013-09-17 10:47, Harald Alvestrand wrote:
> Magnus,
> 
> what is the expected duration of the call?

2 hour maximum

Magnus

> 
> On 09/17/2013 10:38 AM, Magnus Westerlund wrote:
>> WG,
>>
>> I know that the new JSEP Draft is not yet submitted. I am hoping that it
>> will show up during the morning hours for the US based people. That way
>> everyone gets at least some few hours to look at the draft prior to the
>> call. If it hasn't we will postpone the call, but I really hope we can
>> avoid this. I will send a notice to the WG if we have to postpone it.
>>
>> Cheers
>>
>> Magnus
>>
>> On 2013-09-11 07:58, Magnus Westerlund wrote:
>>> WG,
>>>
>>> There will be a new version of JSEP draft coming out Monday. As a follow
>>> up to get as much progress on this topic as possible. The chairs and
>>> authors are offering a informal walk through session on Wednesday the
>>> 18th. People will be strongly recommended to have reviewed the new
>>> draft.
>>>
>>> The intention of the session is enable the authors to inform you about
>>> the latest changes, any issues to consider and what needs further
>>> discussion. The will also enable you as participant after your review of
>>> the document to ask clarifying questions or for motivations behind
>>> choices, and raise issues you have spotted.
>>>
>>> The session is not intended to solve issues or judge consensus on part
>>> of the draft. Discussions of what is the best solution etc, will to be
>>> cut off.
>>>
>>> This will be a webex conference per details below.
>>>
>>> Cheers
>>>
>>> Magnus Westerlund
>>>
>>> Topic: RTCWeb - JSEP
>>> Date: Wednesday, September 18, 2013
>>> Time: 8:00 am, Pacific Daylight Time (San Francisco, GMT-07:00)
>>> Meeting Number: 302 061 238
>>> Meeting Password: ietf
>>>
>>> -------------------------------------------------------
>>> To join the online meeting (Now from mobile devices!)
>>> -------------------------------------------------------
>>> 1. Go to
>>> https://cisco.webex.com/ciscosales/j.php?ED=206400898&UID=0&PW=NYjk4NmE0ODBj&RT=MiM0
>>>
>>>
>>> 2. Enter your name and email address.
>>> 3. Enter the meeting password: ietf
>>> 4. Click "Join Now".
>>>
>>> To view in other time zones or languages, please click the link:
>>> https://cisco.webex.com/ciscosales/j.php?ED=206400898&UID=0&PW=NYjk4NmE0ODBj&ORT=MiM0
>>>
>>>
>>>
>>> -------------------------------------------------------
>>> To join the teleconference only
>>> -------------------------------------------------------
>>> 1. Dial into Cisco WebEx (view all Global Access Numbers at
>>> http://cisco.com/en/US/about/doing_business/conferencing/index.html
>>> 2. Follow the prompts to enter the Meeting Number (listed above) or
>>> Access Code followed by the # sign.
>>>
>>> San Jose, CA: +1.408.525.6800 RTP: +1.919.392.3330
>>>
>>> US/Canada: +1.866.432.9903 United Kingdom: +44.20.8824.0117
>>>
>>> India: +91.80.4350.1111 Germany: +49.619.6773.9002
>>>
>>> Japan: +81.3.5763.9394 China: +86.10.8515.5666
>>>
>>>
>>> ----------------------------------------------------------------
>>> ALERT – PLEASE READ: DO NOT DIAL THE TOLL FREE NUMBERS FROM WITHIN THE
>>> (408) OR (919) AREA CODES
>>> ----------------------------------------------------------------
>>> Please dial the local access number for your area from the list below:
>>> - San Jose/Milpitas (408) area: 525-6800
>>> - RTP (919) area: 392-3330
>>>
>>> Dialing the WebEx toll free numbers from within 408 or 919 area codes is
>>> not enabled (non-Cisco phones). “ If you dial the toll-free numbers
>>> within the 408 or 919 area codes you will be instructed to hang up and
>>> dial the local access number.” Please use the call-back option whenever
>>> possible and otherwise dial local numbers only. The affected toll free
>>> numbers are: (866) 432-9903 for the San Jose/Milpitas area and (866)
>>> 349-3520 for the RTP area.
>>>
>>
> 
> 
> 


-- 

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From ietf-secretariat-reply@ietf.org  Tue Sep 17 01:49:07 2013
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Subject: [rtcweb] Milestones changed for rtcweb WG
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Changed milestone "Send Use Cases document
(draft-ietf-rtcweb-use-cases-and- requirements) to IESG for
publication as Informational", added
draft-ietf-rtcweb-use-cases-and-requirements to milestone.

Changed milestone "Complete Overview (and hold for dependency
resolution) (draft-ietf-rtcweb-overview)", added
draft-ietf-rtcweb-overview to milestone.

Changed milestone "Send Security and Privacy Problem Statement
(draft-ietf- rtcweb-security) to IESG for publication as
Informational", added draft-ietf-rtcweb-security to milestone.

Changed milestone "Send Signalling Negotiation and NAT Traversal
(draft-ietf- rtcweb-jsep) to IESG for publication as Proposed
Standard", added draft-ietf-rtcweb-jsep to milestone.

Changed milestone "Send Security Solution
(draft-ietf-rtcweb-security-arch) to IESG for publication as Proposed
Standard", added draft-ietf-rtcweb-security-arch to milestone.

Changed milestone "Send Media Transport (draft-ietf-rtcweb-rtp-usage)
to IESG for publication as Proposed Standard", added
draft-ietf-rtcweb-rtp-usage to milestone.

Changed milestone "Audio Processing and Audio Codecs
(draft-ietf-rtcweb-audio) to IESG for publication as Proposed
Standard", added draft-ietf-rtcweb-audio to milestone.

Changed milestone "Send Data Stream Transport for non-media data
(draft-ietf- rtcweb-data-channel) to IESG for publication as Proposed
Standard", added draft-ietf-rtcweb-data-channel,
draft-ietf-rtcweb-data-protocol to milestone.

URL: http://datatracker.ietf.org/wg/rtcweb/charter/

From magnus.westerlund@ericsson.com  Tue Sep 17 02:45:51 2013
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Subject: Re: [rtcweb] Milestones changed for rtcweb WG
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WG,

These changes are the just adding the drafts to their corresponding
milestones using the data tracker tool. That way you get clickable links
to the drafts.

An additional message will show up when our AD approves the milestone
for consent freshness that I have requested.

Cheers

Magnus

On 2013-09-17 10:49, IETF Secretariat wrote:
> Changed milestone "Send Use Cases document
> (draft-ietf-rtcweb-use-cases-and- requirements) to IESG for
> publication as Informational", added
> draft-ietf-rtcweb-use-cases-and-requirements to milestone.
> 
> Changed milestone "Complete Overview (and hold for dependency
> resolution) (draft-ietf-rtcweb-overview)", added
> draft-ietf-rtcweb-overview to milestone.
> 
> Changed milestone "Send Security and Privacy Problem Statement
> (draft-ietf- rtcweb-security) to IESG for publication as
> Informational", added draft-ietf-rtcweb-security to milestone.
> 
> Changed milestone "Send Signalling Negotiation and NAT Traversal
> (draft-ietf- rtcweb-jsep) to IESG for publication as Proposed
> Standard", added draft-ietf-rtcweb-jsep to milestone.
> 
> Changed milestone "Send Security Solution
> (draft-ietf-rtcweb-security-arch) to IESG for publication as Proposed
> Standard", added draft-ietf-rtcweb-security-arch to milestone.
> 
> Changed milestone "Send Media Transport (draft-ietf-rtcweb-rtp-usage)
> to IESG for publication as Proposed Standard", added
> draft-ietf-rtcweb-rtp-usage to milestone.
> 
> Changed milestone "Audio Processing and Audio Codecs
> (draft-ietf-rtcweb-audio) to IESG for publication as Proposed
> Standard", added draft-ietf-rtcweb-audio to milestone.
> 
> Changed milestone "Send Data Stream Transport for non-media data
> (draft-ietf- rtcweb-data-channel) to IESG for publication as Proposed
> Standard", added draft-ietf-rtcweb-data-channel,
> draft-ietf-rtcweb-data-protocol to milestone.
> 
> URL: http://datatracker.ietf.org/wg/rtcweb/charter/
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> 
> 


-- 

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
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From: "Tirumaleswar Reddy (tireddy)" <tireddy@cisco.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Continuing to assert username/password (Re: FW: Adopting draft-muthu-behave-consent-freshness?)
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Subject: Re: [rtcweb] Continuing to assert username/password (Re: FW: Adopting draft-muthu-behave-consent-freshness?)
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Hi Harald,

Please see inline

> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf =
Of
> Harald Alvestrand
> Sent: Tuesday, September 17, 2013 2:04 PM
> To: rtcweb@ietf.org
> Subject: [rtcweb] Continuing to assert username/password (Re: FW: Adoptin=
g
> draft-muthu-behave-consent-freshness?)
>=20
> Changing subject line in order to make life easier for those who try to
> review comments on the original subject....
>=20
> On 09/17/2013 07:10 AM, Ram Mohan R (rmohanr) wrote:
> > Hi Please see inline
> >
> > -----Original Message-----
> > From: Jessie <Lijing>, "Huawei)" <lijing80@huawei.com>
> > Date: Thursday, 12 September 2013 1:27 PM
> > To: "rtcweb@ietf.org" <rtcweb@ietf.org>
> > Subject: [rtcweb] FW:  Adopting draft-muthu-behave-consent-freshness?
> >
> >> Hi all,
> >>
> >> As a newcomer to RTCWEB, I have some doubts after I have read the draf=
t.
> >>
> >> 1. in "4.  Solution Overview", may be it would be better to clarify wh=
en
> >> to send a STUN Binding Request(in the middle of a media session or bef=
ore
> >> sending traffic, during sending traffic or only during silence periods=
)
> >> and which side to send the request(controlling agent or both sides)?
> > The introduction section explains when consent is needed. During the
> > initial call setup ICE connectivity checks are used. This mechanism
> > described in this document is for periodic consent after initial sessio=
ns
> > is setup.
> >
> >> 2. in " 7.  Security Considerations", there are the following words. A=
s
> >> when one agent receives STUN Binding Request, unlike the processing of
> >> indication message, it have to do more processes to send the Response
> >> message, I am not sure whether it is appreciate not to authenticate th=
e
> >> source and respond directly. Is it more open to malicious attacks?
> > I am not clear on what you asking here. The below text just tells that
> > there is no need to re-asserting the username/password that was sent
> > initially during ICE checks. Are you saying this can lead to attacks ? =
If
> > yes can you explain on what you meant in detail ?
>=20
> The text below is wrong.
>=20
> If, as is commonly the case, an attacker has the ability to inject
> packets on the wire with the same IP address as the victim, the attacker
> can forge ICE packets.
>=20
> If the victim's correspondent checks MESSAGE-INTEGRITY (which implicitly
> checks username and password), he will detect that these packets don't
> have the correct authorization, and throw them away.
>=20
> If the victim's correspondent does not check MESSAGE-INTEGRITY, the
> attacker can keep renewing any consent to receive as long as he wants,
> even though the victim wanted to stop consenting.

Since each endpoint contributes at least 24 bits of randomness to the ice-u=
frag which provides 48 bits of randomness. It would not be possible for off=
-path attacker to guess those 48 bits to cause the endpoint to perform HMAC=
-SHA1 validation of the MESSAGE-INTEGRITY attribute. Hence my understanding=
 is for consent freshness checking MESSAGE-INTEGRITY is not be required, on=
ly checking IP address and USERNAME attribute looks sufficient to handle of=
f-path attacks.

-Tiru.

>=20
> Since the whole purpose of consent-freshness is to allow the potential
> victim to stop consenting to receiving traffic, checking
> MESSAGE-INTEGRITY (which implictily checks username and password) is
> necessary.
>=20
> This whole paragraph should be deleted from the "Security
> Considerations" section and replaced with text properly describing the
> situation.
>=20
> >
> > Ram
> >
> >>   "Once that connection to the remote
> >>    peer has been established with ICE, the consent to continue sending
> >>    traffic does not benefit from re-asserting that same username and
> >>    password, so long as the senders and receiver's IP addresses remain
> >>    the same (as they usually do)."
> >>
> >> Neglect my words, if my understandings are wrong.
> >>
> >> Best regards,
> >>
> >> Jessie
> >>
> >> -----Original Message-----
> >> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Beha=
lf
> >> Of Magnus Westerlund
> >> Sent: Monday, September 09, 2013 4:37 PM
> >> To: rtcweb@ietf.org
> >> Subject: [rtcweb] Adopting draft-muthu-behave-consent-freshness?
> >>
> >> WG,
> >>
> >> This is a call for WG adoption of STUN Usage for Consent Freshness
> >> (draft-muthu-behave-consent-freshness-04). This document defines a STU=
N
> >> usage for consent freshness. As this requires no protocol extensions w=
e
> >> as intended users can define this usage in our WG. Such work also
> >> matches our charter. The draft-ietf-rtcweb-security-arch-07 is
> >> normatively dependent on this STUN usage.
> >>
> >> Document:
> >> https://datatracker.ietf.org/doc/draft-muthu-behave-consent-freshness/
> >>
> >> WG, please indicate your support or issues with adopting this document
> >> as WG item with a proposed milestone:
> >>
> >> Mar 2014 Send STUN Usage for Consent Freshness to IESG for publication
> >> as proposed standard.
> >>
> >> Cheers
> >>
> >> Magnus Westerlund
> >>
> >> ----------------------------------------------------------------------
> >> Multimedia Technologies, Ericsson Research EAB/TVM
> >> ----------------------------------------------------------------------
> >> Ericsson AB                | Phone  +46 10 7148287
> >> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> >> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> >> ----------------------------------------------------------------------
> >>
> >> _______________________________________________
> >> rtcweb mailing list
> >> rtcweb@ietf.org
> >> https://www.ietf.org/mailman/listinfo/rtcweb
> >> _______________________________________________
> >> rtcweb mailing list
> >> rtcweb@ietf.org
> >> https://www.ietf.org/mailman/listinfo/rtcweb
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From michelg@upperside.fr  Tue Sep 17 04:31:00 2013
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Subject: [rtcweb] WebRTC Paris 2013: The first real-world deployments
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The second edition of the WebRTC conference will be held in Paris Roissy
CDG, from 10 to 12 December, 2013:

Sessions include telco and enterprise testimonies, standardization updates,
performance, QoS & codecs issues. A large part of the second day is
dedicated to implementations.

A panel will focus on Browser/HTML5 vs. Native applications and support for
WebRTC on mobile devices and chipsets.

A startup speed dating session will include descriptions from start-ups and
from new initiatives at larger companies. Each will have a few minutes to
present the company and key benefits.

Application for this session is still open. Organizers will select finalists
to present on stage.

More info:
<http://www.uppersideconferences.com/webrtc2013/webrtc2013intro.html>
http://www.uppersideconferences.com/webrtc2013/webrtc2013intro.html


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CDG, from 10 to 12 December, 2013:</span></span><span lang=3DEN-US =
style=3D'font-size:9.0pt;font-family:"Arial","sans-serif"'><o:p></o:p></s=
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lang=3DEN-US =
style=3D'font-size:9.0pt;font-family:"Arial","sans-serif";color:#262626'>=
Sessions include telco and enterprise testimonies, standardization =
updates, performance, QoS &amp; codecs issues. A large part of the =
second day is dedicated to implementations.</span></span><span =
lang=3DEN-US =
style=3D'font-size:9.0pt;font-family:"Arial","sans-serif"'><o:p></o:p></s=
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class=3Dapple-style-span><span lang=3DEN-US =
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A</span></span><span class=3Dapple-style-span><span lang=3DEN-US =
style=3D'font-size:9.0pt;font-family:"Arial","sans-serif"'> panel<b> =
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style=3D'font-size:9.0pt;font-family:"Arial","sans-serif"'><o:p></o:p></s=
pan></p><p class=3DMsoNormal =
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class=3Dapple-style-span><span lang=3DEN-US =
style=3D'font-size:9.0pt;font-family:"Arial","sans-serif"'>A startup =
speed dating session will include descriptions from start-ups and from =
new initiatives at larger companies. Each will have a few minutes to =
present the company and key benefits.</span></span><span lang=3DEN-US =
style=3D'font-size:9.0pt;font-family:"Arial","sans-serif"'><o:p></o:p></s=
pan></p><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;text-autospac=
e:none'><span class=3Dapple-style-span><span lang=3DEN-US =
style=3D'font-size:9.0pt;font-family:"Arial","sans-serif"'>Application =
for this session is still open. Organizers will select finalists to =
present on stage.</span></span><span lang=3DEN-US =
style=3D'font-size:9.0pt;font-family:"Arial","sans-serif"'><o:p></o:p></s=
pan></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:9.0pt;font-family:"Arial","sans-serif"'>More info: =
</span><span =
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From harald@alvestrand.no  Tue Sep 17 04:33:56 2013
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Subject: Re: [rtcweb] Continuing to assert username/password (Re: FW: Adopting draft-muthu-behave-consent-freshness?)
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On 09/17/2013 12:05 PM, Tirumaleswar Reddy (tireddy) wrote:
> Hi Harald,
>
> Please see inline
>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
>> Harald Alvestrand
>> Sent: Tuesday, September 17, 2013 2:04 PM
>> To: rtcweb@ietf.org
>> Subject: [rtcweb] Continuing to assert username/password (Re: FW: Adopting
>> draft-muthu-behave-consent-freshness?)
>>
>> Changing subject line in order to make life easier for those who try to
>> review comments on the original subject....
>>
>> On 09/17/2013 07:10 AM, Ram Mohan R (rmohanr) wrote:
>>> Hi Please see inline
>>>
>>> -----Original Message-----
>>> From: Jessie <Lijing>, "Huawei)" <lijing80@huawei.com>
>>> Date: Thursday, 12 September 2013 1:27 PM
>>> To: "rtcweb@ietf.org" <rtcweb@ietf.org>
>>> Subject: [rtcweb] FW:  Adopting draft-muthu-behave-consent-freshness?
>>>
>>>> Hi all,
>>>>
>>>> As a newcomer to RTCWEB, I have some doubts after I have read the draft.
>>>>
>>>> 1. in "4.  Solution Overview", may be it would be better to clarify when
>>>> to send a STUN Binding Request(in the middle of a media session or before
>>>> sending traffic, during sending traffic or only during silence periods)
>>>> and which side to send the request(controlling agent or both sides)?
>>> The introduction section explains when consent is needed. During the
>>> initial call setup ICE connectivity checks are used. This mechanism
>>> described in this document is for periodic consent after initial sessions
>>> is setup.
>>>
>>>> 2. in " 7.  Security Considerations", there are the following words. As
>>>> when one agent receives STUN Binding Request, unlike the processing of
>>>> indication message, it have to do more processes to send the Response
>>>> message, I am not sure whether it is appreciate not to authenticate the
>>>> source and respond directly. Is it more open to malicious attacks?
>>> I am not clear on what you asking here. The below text just tells that
>>> there is no need to re-asserting the username/password that was sent
>>> initially during ICE checks. Are you saying this can lead to attacks ? If
>>> yes can you explain on what you meant in detail ?
>> The text below is wrong.
>>
>> If, as is commonly the case, an attacker has the ability to inject
>> packets on the wire with the same IP address as the victim, the attacker
>> can forge ICE packets.
>>
>> If the victim's correspondent checks MESSAGE-INTEGRITY (which implicitly
>> checks username and password), he will detect that these packets don't
>> have the correct authorization, and throw them away.
>>
>> If the victim's correspondent does not check MESSAGE-INTEGRITY, the
>> attacker can keep renewing any consent to receive as long as he wants,
>> even though the victim wanted to stop consenting.
> Since each endpoint contributes at least 24 bits of randomness to the ice-ufrag which provides 48 bits of randomness. It would not be possible for off-path attacker to guess those 48 bits to cause the endpoint to perform HMAC-SHA1 validation of the MESSAGE-INTEGRITY attribute. Hence my understanding is for consent freshness checking MESSAGE-INTEGRITY is not be required, only checking IP address and USERNAME attribute looks sufficient to handle off-path attacks.
>

Are we talking past each other?

In order to perform HMAC-SHA1 validation of the MESSAGE-INTEGRITY 
attribute, the HMAC-SHA1 function takes two inputs: The key and the message.

 From RFC 5389 section 15.4:

For long-term credentials, the key is 16 bytes:

             key = MD5(username ":" realm ":" SASLprep(password))

For short-term credentials:

                           key = SASLprep(password)

In other words: A successful HMAC that produces a match with the 
MESSAGE-INTEGRITY attribute value means that the producer of the 
MESSAGE-INTEGRITY attribute knows the password, and, if long term 
credentials are used, he knows the username.

This implicitly reasserts the password (and possibly the username), and 
the words we're discussing:

   "Once that connection to the remote
    peer has been established with ICE, the consent to continue sending
    traffic does not benefit from re-asserting that same username and
    password, so long as the senders and receiver's IP addresses remain
    the same (as they usually do)."

don't make any sense to me.
If they don't make any sense to other people, they do not belong in the document.






From magnus.westerlund@ericsson.com  Tue Sep 17 04:59:46 2013
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Subject: Re: [rtcweb] WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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Hi,

I have reviewed the Use Case and Requirements draft (v11) in the WG last
call. I have the following comments on this document. I have found a
number of issues that I definitely needs to be fixed prior to any
publication request.


1. First of all I like to bring up a document structure problem. When
reviewing this it is difficult to follow what requirements that are
added for the different use cases. I would propose that each Use Case or
general section that describe something that causes new requirements
that hasn't been listed yet in the document to define the Requirement
there. Something like this:

3.2.2.1.  Description

   This use-case is almost identical to the Simple Video Communication
   Service use-case (Section 3.2.1).  The difference is that one of the
   users is behind a NAT that blocks UDP traffic.

3.2.2.2.  Derived Requirements

   F1, F2, F3, F4, F5, F8, F9, F10, F20, F25, F28

   A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12

   New Requirement:

   F29     The browser must be able to send streams and
           data to a peer in the presence of NATs that
           block UDP traffic.


This makes the incremental use cases much more understandable in how
they add requirements.

This I think will also affect Section 3.1 which will need to include the
requirement definitions created by these general considerations.

However, I do suggest that you leave the collection of all requirements
in place for easy reference.

2. Section 1:

   The document focuses on requirements related to real-time media
   streams and data exchange.  Requirements related to privacy,
   signalling between the browser and web server etc. are currently not
   considered.

I think this needs to be rewritten. The draft actually do consider a
number of high level security and privacy concerns.

3. Section 2:

If you have no definitions, do remove the section, rather then leaving
it TBD.

4. Section 3.1

   o  Clients can be on wideband (10s of Mbits/sec)

   o  Clients can be on narrowband (10s to 100s of Kbits/sec)

Due to the usage of the terms wideband and narrowband I think you need
to be more explicit about what is actually referred to here. My
assumption is network throughput capability. And in these cases maybe
expressing this explicitly is a better choice.

5. Section 3.1:

   o  Clients can be on networks with any type (as described in RFC4787)
      of NAT.

I think "any type" is unclear here, as there are no types defined in
that RFC unless you are trying to express the difference between basic
NAT and NAPT, which I don't believe you are. I think you should use
something like this:

   o  Clients can be on networks with a NAT using any type of Mapping
      and Filtering behaviors (as described in RFC4787).

6. Section 3.2.1.1:

   It is essential that the communication cannot be wiretapped
   [RFC2804].

This sentence returns a number of times in this document. I think it is
a general security requirement that applies to probably all of the use
cases. If there is one use case that don't need to enable protection
against this, then please be explicit about that.

My Suggestion is to move this into the general considerations section.

7. Section 3.2.1.1:

   In addition, it is required that browsers enable the media and data
   security keys to be cryptographically bound to the user identity.

I think this sentence is actually subtly wrong in a couple of ways.
First of all it talks about "the identity", where I think it should be
talking about "a identity". This can both the human users identity, but
also a role identity, like Police Officer.

Secondly, binding it to  the security keys without putting requirements
on the keying and key-exchange mechanism to prevent man in the middles
empties this requirement. I would recommend that you reformulate this to
a high level requirement without going into mechanism. For example:

In addition, it is required that browsers enables binding an identity of
the user's choice to his end of the secured communication and that such
mechanism prevent man in the middle attacks.

8. Section 3.2.1.1

   One user has an unreliable Internet connection.  It sometimes loses
   packets, and sometimes goes down completely.

I guess the requirement here is to handle this in a graceful way from
media transport and rendering perspective. The packet loss is also
unclear as I don't know if the intention is to capture spurious packet
losses due to the link media, or congestion losses. I think some
expansion of this is needed.

9. Section 3.2.1.1:

   One user is located behind a Network Address Translator (NAT).

What is the point being stressed here that isn't covered by the general
consideration in 3.1 that says:

   o  Clients can be on networks with any type (as described in RFC4787)
      of NAT.

Section 3.2.2.2:

3.2.2.2.  Derived Requirements

   F1, F2, F3, F4, F5, F8, F9, F10, F20, F25, F28, F29

   A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12


How come this list of Derived Requirements are shorter than for the
parent use case which has the following list:

3.2.1.2.  Derived Requirements

   F1, F2, F3, F4, F5, F8, F9, F10, F20, F25, F28, F35, F36, F38, F39

   A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A25, A26


If each use case is listing all the requirement then it needs to be
consistently done and actually list all that applies. If there are
requirements from the parent that doesn't apply that may even be worth
explicitly mentioning.

This comments applies to all use cases as there appears to be general
inconsistencies.

10. Section 3.2.3.1:


3.2.3.1.  Description

   This use-case is almost identical to the Simple Video Communication
   Service use-case (Section 3.2.1).  The difference is that one of the
   users is behind a FW that only allows http traffic.

If a firewall only allows HTTP traffic, then can we really assume that
the firewall administrator per default will accept WebRTC Media and Data
traffic?

I am far from certain of this, and think on a requirement level needs to
express a situation where the firewall administrator allows WebRTC
across its FW, or at least can easily configure a rule to block it. Thus
resulting in that any solution for this needs to be easily identifiable
and possible to block.

See also PNTAW@ietf.org mailing list.

11. Section 3.2.8.1:

   But in addition to this, one of the users can share what is being
   displayed on her/his screen with a peer.  The user can choose to
   share the entire screen, part of the screen (part selected by the
   user) or what a selected applicaton displays with the peer.

Does this needs mentioning of additional high level security requirements?

12. Section 3.2.10.1:

  The service providers are interconnected by some means, but exchange
   no more information about the users than what can be carried using
   SIP.

   NOTE: More profiling of what this means may be needed.

The Note, can it be removed, or do we need additional text in this document?

13. Section 3.2.10.1:

   The same issues with connectivity apply.

I don't understand this comment. What is it intended to refer to?

14. Section 3.2.11.1:

   Before the game starts, and during game breaks, the talent scout and
   the manager have a 1-1 audiovisual communication session.  Only the
   rear facing camera of the mobile phone is used.  On the display of
   the mobile phone, the video of the club manager is shown with a
   picture-in-picture thumbnail of the rear facing camera (self-view).
   On the display of the desktop, the video of the talent scout is shown
   with a picture-in-picture thumbnail of the desktop camera (self-
   view).

I get a bit confused with what is the front and rear of my smart phone.
I would consider the display side the front side, and the rear the other
big side of my block shaped mobile phone. I suggest using other terms.

15. Section 3.2.14.1:

   Discussion: This use-case was briefly discussed at the Quebec webrtc
   meeting and it got support.  So far the only concrete requirement
   (A17) derived is that the application must be able to ask the browser
   to treat the audio signal as audio (in contrast to speech).  However,
   the use case should be further analysed to determine other
   requirements (could be e.g. on delay mic->speaker, level control of
   audio signals, etc.).

I think this Discussion points needs to be dealt with before we can
publish this document. I would consider the question if this needs more
analysis a open issue.

16. Section 3.3.2.1:

   Alice uses her web browser with a service that allows her to call
   PSTN numbers.  Alice calls 1-800-gofedex.  Alice should be able to
   hear the initial prompts from the fedex IVR and when the IVR says
   press 1, there should be a way for Alice to navigate the IVR.

Can you please expand IVR.

17. Section 3.3.3.1:

   The organization has an internal network set up with an aggressive
   firewall handling access to the Internet.  If users cannot physically
   access the internal network, they can establish a Virtual Private
   Network (VPN).

I think the requirements or limitations the above paragraph try to
express is not clear. There is some missing assumptions of location of
server related to clients that is not covered. Thus it is not clear what
requirements this results in.

Can you please expand this to something that is understandable.

18. Section 3.3.3.1:

   This use-case adds requirements on support for fast stream
   switches F7, on encryption of media and on ability to traverse very
   restrictive FWs.

Regarding the restricted FWs, my comment 10) applies to also this comment.

19. Section 4.2:

   F14     The browser must be able to measure the level

   F15     The browser must be able to change the level
           in audio streams.


Can you please make explicit what type level you think should be
measured or changed, energy, "voice activity"?

20. Section 4.2

  F19     Streams and data must be able to pass through
           limited middleboxes.

What type of limitations are you considering in this requirement?

21. Section 4.2

I note that F30, and F36 may be affected by earlier comments on the
underlying reason for these requirements and need additional text or
clarifications.

22. Section 4.2:
   F37     The browser must be able to send streams and
           data to a peer in the presence of FWs that only
           allows http(s) traffic.

I think this requirement should have a caveat along the lines: when FW
policy allos WebRTC traffic.

23. Section 5:

Change the TBD to a statement that there are no IANA actions in this
document.

24. Section 6.2:

These security considerations are really security requirements on the
browser or the API. I don't know if would be better to move them to
relevant use case sections, or leave them in place.

25. Section 7:

   13.  Enable company coop without being able to decipher http://
        www.ietf.org/mail-archive/web/rtcweb/current/msg04461.html

What is "coop" in the above?


26. Please run a spell checker on the document prior to submitting the
update.


27. Ensure that the ID nits issues are not real issues:

  == Line 667 has weird spacing: '...resence  of NA...'

  == Line 785 has weird spacing: '...of that  strea...'


Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
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From: "Tirumaleswar Reddy (tireddy)" <tireddy@cisco.com>
To: Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: [rtcweb] Continuing to assert username/password (Re: FW: Adopting draft-muthu-behave-consent-freshness?)
Thread-Index: AQHOs4C/clq2b3F4c0Cq3gToydiwV5nJqo4AgAB1y4CAABEQ4A==
Date: Tue, 17 Sep 2013 17:40:16 +0000
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Subject: Re: [rtcweb] Continuing to assert username/password (Re: FW: Adopting draft-muthu-behave-consent-freshness?)
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> -----Original Message-----
> From: Harald Alvestrand [mailto:harald@alvestrand.no]
> Sent: Tuesday, September 17, 2013 5:04 PM
> To: Tirumaleswar Reddy (tireddy)
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Continuing to assert username/password (Re: FW: Ado=
pting
> draft-muthu-behave-consent-freshness?)
>=20
> On 09/17/2013 12:05 PM, Tirumaleswar Reddy (tireddy) wrote:
> > Hi Harald,
> >
> > Please see inline
> >
> >> -----Original Message-----
> >> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Beha=
lf Of
> >> Harald Alvestrand
> >> Sent: Tuesday, September 17, 2013 2:04 PM
> >> To: rtcweb@ietf.org
> >> Subject: [rtcweb] Continuing to assert username/password (Re: FW: Adop=
ting
> >> draft-muthu-behave-consent-freshness?)
> >>
> >> Changing subject line in order to make life easier for those who try t=
o
> >> review comments on the original subject....
> >>
> >> On 09/17/2013 07:10 AM, Ram Mohan R (rmohanr) wrote:
> >>> Hi Please see inline
> >>>
> >>> -----Original Message-----
> >>> From: Jessie <Lijing>, "Huawei)" <lijing80@huawei.com>
> >>> Date: Thursday, 12 September 2013 1:27 PM
> >>> To: "rtcweb@ietf.org" <rtcweb@ietf.org>
> >>> Subject: [rtcweb] FW:  Adopting draft-muthu-behave-consent-freshness?
> >>>
> >>>> Hi all,
> >>>>
> >>>> As a newcomer to RTCWEB, I have some doubts after I have read the dr=
aft.
> >>>>
> >>>> 1. in "4.  Solution Overview", may be it would be better to clarify =
when
> >>>> to send a STUN Binding Request(in the middle of a media session or b=
efore
> >>>> sending traffic, during sending traffic or only during silence perio=
ds)
> >>>> and which side to send the request(controlling agent or both sides)?
> >>> The introduction section explains when consent is needed. During the
> >>> initial call setup ICE connectivity checks are used. This mechanism
> >>> described in this document is for periodic consent after initial sess=
ions
> >>> is setup.
> >>>
> >>>> 2. in " 7.  Security Considerations", there are the following words.=
 As
> >>>> when one agent receives STUN Binding Request, unlike the processing =
of
> >>>> indication message, it have to do more processes to send the Respons=
e
> >>>> message, I am not sure whether it is appreciate not to authenticate =
the
> >>>> source and respond directly. Is it more open to malicious attacks?
> >>> I am not clear on what you asking here. The below text just tells tha=
t
> >>> there is no need to re-asserting the username/password that was sent
> >>> initially during ICE checks. Are you saying this can lead to attacks =
? If
> >>> yes can you explain on what you meant in detail ?
> >> The text below is wrong.
> >>
> >> If, as is commonly the case, an attacker has the ability to inject
> >> packets on the wire with the same IP address as the victim, the attack=
er
> >> can forge ICE packets.
> >>
> >> If the victim's correspondent checks MESSAGE-INTEGRITY (which implicit=
ly
> >> checks username and password), he will detect that these packets don't
> >> have the correct authorization, and throw them away.
> >>
> >> If the victim's correspondent does not check MESSAGE-INTEGRITY, the
> >> attacker can keep renewing any consent to receive as long as he wants,
> >> even though the victim wanted to stop consenting.
> > Since each endpoint contributes at least 24 bits of randomness to the i=
ce-
> ufrag which provides 48 bits of randomness. It would not be possible for =
off-
> path attacker to guess those 48 bits to cause the endpoint to perform HMA=
C-
> SHA1 validation of the MESSAGE-INTEGRITY attribute. Hence my understandin=
g is
> for consent freshness checking MESSAGE-INTEGRITY is not be required, only
> checking IP address and USERNAME attribute looks sufficient to handle off=
-path
> attacks.
> >
>=20
> Are we talking past each other?
>=20
> In order to perform HMAC-SHA1 validation of the MESSAGE-INTEGRITY
> attribute, the HMAC-SHA1 function takes two inputs: The key and the messa=
ge.
>=20
>  From RFC 5389 section 15.4:
>=20
> For long-term credentials, the key is 16 bytes:
>=20
>              key =3D MD5(username ":" realm ":" SASLprep(password))
>=20
> For short-term credentials:
>=20
>                            key =3D SASLprep(password)
>=20
> In other words: A successful HMAC that produces a match with the
> MESSAGE-INTEGRITY attribute value means that the producer of the
> MESSAGE-INTEGRITY attribute knows the password, and, if long term
> credentials are used, he knows the username.
>=20
> This implicitly reasserts the password (and possibly the username), and
> the words we're discussing:
>=20
>    "Once that connection to the remote
>     peer has been established with ICE, the consent to continue sending
>     traffic does not benefit from re-asserting that same username and
>     password, so long as the senders and receiver's IP addresses remain
>     the same (as they usually do)."

Agreed, the above text in the draft needs to be corrected.=20

-Tiru.

>=20
> don't make any sense to me.
> If they don't make any sense to other people, they do not belong in the
> document.
>=20
>=20
>=20
>=20


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Subject: Re: [rtcweb] Video Codec Selection Plan
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Hi Ted,

     Seeing as this discussion stems from licensing concerns, I like to 
propose the following alternative:

 1. Mandate a video codec whose IPR has expired. I agree that video
    quality will degrade, which brings me to the next point.
 2. Provide a negotiation mechanism which would allow peers to "upgrade"
    to a superior (optionally-implemented) video codec.

     This will allow us to support VP8, VP9, H264, H265 or whatever 
other codec people like without the fear of transcoding or IPR. I 
believe that in most cases negotiation will succeed in upgrading to a 
superior codec. It will also encourage (as opposed to force) vendors to 
support each other's codecs, which is the right way to go in light of 
the political nature of this decision.

Gili


1. If you support H.264 as the mandatory to implement codec or are
willing to live with it as the MTI, please raise your hand now.

2. If you support VP8 as the mandatory to implement codec or are
willing to live with it as the MTI, please raise your hand now.


Gili

On 13/09/2013 12:52 PM, Ted Hardie wrote:
> WG,
>
> The chairs have created a plan for how to perform the Video Codec
> selection in our WG. The chairs are asking for review of our plan on
> how to undertake the mandatory-to-implement video codec selection.
> We'd much prefer to have comments on the mechanics before they begin,
> so please review now.  Proponents of a particular proposal should
> note both the actions required and the timelines proposed.
>
> The main goal of this plan is to hold a consensus call on which of
> the proposed alternatives we as a WG should select at one of the WG
> sessions in Vancouver. Such a consensus call will of course be
> verified on the mailing list for anyone who can't participate. The
> chairs will recuse themselves from judging this particular
> consensus.
>
> In the WG session each codec proposal will be allowed an equal amount
> of time to highlight the arguments for their proposal. After that a
> there will be a slot for discussion and clarifying questions.
>
> To enable the WG participants to get answers to any questions, the
> proposals in draft form and any supporting material MUST be made
> available by 6th of October. This is to ensure that the WG
> participants can verify or object to any claims or statements in
> the proposal material prior to the WG session. We chairs would really
> not like to see the proponents bring up new arguments at their
> presentation. Also the WG participants are expected to raise any
> arguments on the list ahead of time to enable the proponents to
> respond to such arguments.
>
> The proposed consensus questions will be of the following form:
>
> 1. If you support H.264 as the mandatory to implement codec or are
> willing to live with it as the MTI, please raise your hand now.
>
> 2. If you support VP8 as the mandatory to implement codec or are
> willing to live with it as the MTI, please raise your hand now.
>
> You may indicate support on both questions and we encourage you to do
> so if you can live with either, even if you have a preference for one
> over the other.
>
> Additional proposals than the previous ones are welcome, but must be
> submitted as draft and their proponents must notify the chairs no later
> than the 6th of October that they also have a candidate proposal.
>
> In case the WG fails to reach consensus we chairs propose that we use
> the alternative decision process as discussed in RFC3929. The method
> and its usage will be discussed on the list should the WG not
> establish consensus on a proposal for mandatory to implement video codec.
>
> regards,
>
> Magnus,  Cullen, and Ted
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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    <div class="moz-cite-prefix">Hi Ted,<br>
      <br>
      &nbsp;&nbsp;&nbsp; Seeing as this discussion stems from licensing concerns, I
      like to propose the following alternative:<br>
      <ol>
        <li>Mandate a video codec whose IPR has expired. I agree that
          video quality will degrade, which brings me to the next point.</li>
        <li>Provide a negotiation mechanism which would allow peers to
          "upgrade" to a superior (optionally-implemented) video codec.</li>
      </ol>
      <p>&nbsp;&nbsp;&nbsp; This will allow us to support VP8, VP9, H264, H265 or
        whatever other codec people like without the fear of transcoding
        or IPR. I believe that in most cases negotiation will succeed in
        upgrading to a superior codec. It will also encourage (as
        opposed to force) vendors to support each other's codecs, which
        is the right way to go in light of the political nature of this
        decision.<br>
      </p>
      <p>Gili<br>
      </p>
      <br>
      1. If you support H.264 as the mandatory to implement codec or are<br>
      willing to live with it as the MTI, please raise your hand now.<br>
      <br>
      2. If you support VP8 as the mandatory to implement codec or are<br>
      willing to live with it as the MTI, please raise your hand now.<br>
      <br>
      <br>
      Gili<br>
      <br>
      On 13/09/2013 12:52 PM, Ted Hardie wrote:<br>
    </div>
    <blockquote
cite="mid:CA+9kkMAvdtq_gufKmDNCNCL+kKcxyi0MGUoVHetd9_DzbEdEnA@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div>WG,<br>
          <br>
          The chairs have created a plan for how to perform the Video
          Codec<br>
          selection in our WG. The chairs are asking for review of our
          plan on<br>
          how to undertake the mandatory-to-implement video codec
          selection.<br>
          We'd much prefer to have comments on the mechanics before they
          begin,<br>
          so please review now.&nbsp; Proponents of a particular proposal
          should<br>
          note both the actions required and the timelines proposed.<br>
          <br>
          The main goal of this plan is to hold a consensus call on
          which of<br>
          the proposed alternatives we as a WG should select at one of
          the WG<br>
          sessions in Vancouver. Such a consensus call will of course be<br>
          verified on the mailing list for anyone who can't participate.
          The<br>
          chairs will recuse themselves from judging this particular<br>
          consensus.<br>
          <br>
          In the WG session each codec proposal will be allowed an equal
          amount<br>
          of time to highlight the arguments for their proposal. After
          that a<br>
          there will be a slot for discussion and clarifying questions.<br>
          <br>
          To enable the WG participants to get answers to any questions,
          the<br>
          proposals in draft form and any supporting material MUST be
          made<br>
          available by 6th of October. This is to ensure that the WG<br>
          participants can verify or object to any claims or statements
          in<br>
          the proposal material prior to the WG session. We chairs would
          really<br>
          not like to see the proponents bring up new arguments at their<br>
          presentation. Also the WG participants are expected to raise
          any<br>
          arguments on the list ahead of time to enable the proponents
          to<br>
          respond to such arguments.<br>
          <br>
          The proposed consensus questions will be of the following
          form:<br>
          <br>
          1. If you support H.264 as the mandatory to implement codec or
          are<br>
          willing to live with it as the MTI, please raise your hand
          now.<br>
          <br>
          2. If you support VP8 as the mandatory to implement codec or
          are<br>
          willing to live with it as the MTI, please raise your hand
          now.<br>
          <br>
          You may indicate support on both questions and we encourage
          you to do<br>
          so if you can live with either, even if you have a preference
          for one<br>
          over the other.<br>
          <br>
          Additional proposals than the previous ones are welcome, but
          must be<br>
          submitted as draft and their proponents must notify the chairs
          no later<br>
          than the 6th of October that they also have a candidate
          proposal.<br>
          <br>
          In case the WG fails to reach consensus we chairs propose that
          we use<br>
          the alternative decision process as discussed in RFC3929. The
          method<br>
          and its usage will be discussed on the list should the WG not<br>
          establish consensus on a proposal for mandatory to implement
          video codec.<br>
          <br>
        </div>
        <div>regards,<br>
          <br>
        </div>
        Magnus,&nbsp; Cullen, and Ted<br>
      </div>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
  </body>
</html>

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From christer.holmberg@ericsson.com  Tue Sep 17 12:36:37 2013
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Subject: Re: [rtcweb] Video Codec Selection Plan
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Hi,

SDP Offer/Answer is used to negotiate codecs etc, so the API (JSEP) already=
 provides such negotiation mechanism.

(Obviously the peers also need to negotiate with each other, but whatever p=
rotocol is used for that is outside the scope.)

Regards,

Christer

L=E4hett=E4j=E4: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] P=
uolesta cowwoc
L=E4hetetty: 17. syyskuuta 2013 21:54
Vastaanottaja: rtcweb@ietf.org
Aihe: Re: [rtcweb] Video Codec Selection Plan

Hi Ted,

    Seeing as this discussion stems from licensing concerns, I like to prop=
ose the following alternative:

  1.  Mandate a video codec whose IPR has expired. I agree that video quali=
ty will degrade, which brings me to the next point.
  2.  Provide a negotiation mechanism which would allow peers to "upgrade" =
to a superior (optionally-implemented) video codec.

    This will allow us to support VP8, VP9, H264, H265 or whatever other co=
dec people like without the fear of transcoding or IPR. I believe that in m=
ost cases negotiation will succeed in upgrading to a superior codec. It wil=
l also encourage (as opposed to force) vendors to support each other's code=
cs, which is the right way to go in light of the political nature of this d=
ecision.

Gili

1. If you support H.264 as the mandatory to implement codec or are
willing to live with it as the MTI, please raise your hand now.

2. If you support VP8 as the mandatory to implement codec or are
willing to live with it as the MTI, please raise your hand now.


Gili

On 13/09/2013 12:52 PM, Ted Hardie wrote:
WG,

The chairs have created a plan for how to perform the Video Codec
selection in our WG. The chairs are asking for review of our plan on
how to undertake the mandatory-to-implement video codec selection.
We'd much prefer to have comments on the mechanics before they begin,
so please review now.  Proponents of a particular proposal should
note both the actions required and the timelines proposed.

The main goal of this plan is to hold a consensus call on which of
the proposed alternatives we as a WG should select at one of the WG
sessions in Vancouver. Such a consensus call will of course be
verified on the mailing list for anyone who can't participate. The
chairs will recuse themselves from judging this particular
consensus.

In the WG session each codec proposal will be allowed an equal amount
of time to highlight the arguments for their proposal. After that a
there will be a slot for discussion and clarifying questions.

To enable the WG participants to get answers to any questions, the
proposals in draft form and any supporting material MUST be made
available by 6th of October. This is to ensure that the WG
participants can verify or object to any claims or statements in
the proposal material prior to the WG session. We chairs would really
not like to see the proponents bring up new arguments at their
presentation. Also the WG participants are expected to raise any
arguments on the list ahead of time to enable the proponents to
respond to such arguments.

The proposed consensus questions will be of the following form:

1. If you support H.264 as the mandatory to implement codec or are
willing to live with it as the MTI, please raise your hand now.

2. If you support VP8 as the mandatory to implement codec or are
willing to live with it as the MTI, please raise your hand now.

You may indicate support on both questions and we encourage you to do
so if you can live with either, even if you have a preference for one
over the other.

Additional proposals than the previous ones are welcome, but must be
submitted as draft and their proponents must notify the chairs no later
than the 6th of October that they also have a candidate proposal.

In case the WG fails to reach consensus we chairs propose that we use
the alternative decision process as discussed in RFC3929. The method
and its usage will be discussed on the list should the WG not
establish consensus on a proposal for mandatory to implement video codec.
regards,
Magnus,  Cullen, and Ted




_______________________________________________

rtcweb mailing list

rtcweb@ietf.org<mailto:rtcweb@ietf.org>

https://www.ietf.org/mailman/listinfo/rtcweb


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<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">SDP Offer/=
Answer is used to negotiate codecs etc, so the API (JSEP) already provides =
such negotiation mechanism.
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">(Obviously=
 the peers also need to negotiate with each other, but whatever protocol is=
 used for that is outside the scope.)<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Regards,<o=
:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Christer<o=
:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">L=E4hett=E4j=E4:</sp=
an></b><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot=
;sans-serif&quot;;color:windowtext"> rtcweb-bounces@ietf.org [mailto:rtcweb=
-bounces@ietf.org]
<b>Puolesta </b>cowwoc<br>
<b>L=E4hetetty:</b> 17. syyskuuta 2013 21:54<br>
<b>Vastaanottaja:</b> rtcweb@ietf.org<br>
<b>Aihe:</b> Re: [rtcweb] Video Codec Selection Plan<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">Hi Ted,<br>
<br>
&nbsp;&nbsp;&nbsp; Seeing as this discussion stems from licensing concerns,=
 I like to propose the following alternative:<o:p></o:p></p>
<ol start=3D"1" type=3D"1">
<li class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-=
alt:auto;mso-list:l0 level1 lfo1">
Mandate a video codec whose IPR has expired. I agree that video quality wil=
l degrade, which brings me to the next point.<o:p></o:p></li><li class=3D"M=
soNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;mso-l=
ist:l0 level1 lfo1">
Provide a negotiation mechanism which would allow peers to &quot;upgrade&qu=
ot; to a superior (optionally-implemented) video codec.<o:p></o:p></li></ol=
>
<p>&nbsp;&nbsp;&nbsp; This will allow us to support VP8, VP9, H264, H265 or=
 whatever other codec people like without the fear of transcoding or IPR. I=
 believe that in most cases negotiation will succeed in upgrading to a supe=
rior codec. It will also encourage (as opposed
 to force) vendors to support each other's codecs, which is the right way t=
o go in light of the political nature of this decision.<o:p></o:p></p>
<p>Gili<o:p></o:p></p>
<p class=3D"MsoNormal"><br>
1. If you support H.264 as the mandatory to implement codec or are<br>
willing to live with it as the MTI, please raise your hand now.<br>
<br>
2. If you support VP8 as the mandatory to implement codec or are<br>
willing to live with it as the MTI, please raise your hand now.<br>
<br>
<br>
Gili<br>
<br>
On 13/09/2013 12:52 PM, Ted Hardie wrote:<o:p></o:p></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">WG,<br>
<br>
The chairs have created a plan for how to perform the Video Codec<br>
selection in our WG. The chairs are asking for review of our plan on<br>
how to undertake the mandatory-to-implement video codec selection.<br>
We'd much prefer to have comments on the mechanics before they begin,<br>
so please review now.&nbsp; Proponents of a particular proposal should<br>
note both the actions required and the timelines proposed.<br>
<br>
The main goal of this plan is to hold a consensus call on which of<br>
the proposed alternatives we as a WG should select at one of the WG<br>
sessions in Vancouver. Such a consensus call will of course be<br>
verified on the mailing list for anyone who can't participate. The<br>
chairs will recuse themselves from judging this particular<br>
consensus.<br>
<br>
In the WG session each codec proposal will be allowed an equal amount<br>
of time to highlight the arguments for their proposal. After that a<br>
there will be a slot for discussion and clarifying questions.<br>
<br>
To enable the WG participants to get answers to any questions, the<br>
proposals in draft form and any supporting material MUST be made<br>
available by 6th of October. This is to ensure that the WG<br>
participants can verify or object to any claims or statements in<br>
the proposal material prior to the WG session. We chairs would really<br>
not like to see the proponents bring up new arguments at their<br>
presentation. Also the WG participants are expected to raise any<br>
arguments on the list ahead of time to enable the proponents to<br>
respond to such arguments.<br>
<br>
The proposed consensus questions will be of the following form:<br>
<br>
1. If you support H.264 as the mandatory to implement codec or are<br>
willing to live with it as the MTI, please raise your hand now.<br>
<br>
2. If you support VP8 as the mandatory to implement codec or are<br>
willing to live with it as the MTI, please raise your hand now.<br>
<br>
You may indicate support on both questions and we encourage you to do<br>
so if you can live with either, even if you have a preference for one<br>
over the other.<br>
<br>
Additional proposals than the previous ones are welcome, but must be<br>
submitted as draft and their proponents must notify the chairs no later<br>
than the 6th of October that they also have a candidate proposal.<br>
<br>
In case the WG fails to reach consensus we chairs propose that we use<br>
the alternative decision process as discussed in RFC3929. The method<br>
and its usage will be discussed on the list should the WG not<br>
establish consensus on a proposal for mandatory to implement video codec.<o=
:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">regards,<o:p></o:p></=
p>
</div>
<p class=3D"MsoNormal">Magnus,&nbsp; Cullen, and Ted<o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><br>
<br>
<br>
<o:p></o:p></p>
<pre>_______________________________________________<o:p></o:p></pre>
<pre>rtcweb mailing list<o:p></o:p></pre>
<pre><a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><o:p></o:p></pre=
>
<pre><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.i=
etf.org/mailman/listinfo/rtcweb</a><o:p></o:p></pre>
</blockquote>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
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     So then the only question is: is there a reasonable video codec 
whose IPR has expired that we can agree on?

     My goal is to displease everyone equally. By picking a neutral 
codec, acknowledging that it displeases all parties in some way, we 
should be able to defuse most of the politics surrounding the decision.

Gili

On 17/09/2013 3:36 PM, Christer Holmberg wrote:
>
> Hi,
>
> SDP Offer/Answer is used to negotiate codecs etc, so the API (JSEP) 
> already provides such negotiation mechanism.
>
> (Obviously the peers also need to negotiate with each other, but 
> whatever protocol is used for that is outside the scope.)
>
> Regards,
>
> Christer
>
> *Lähettäjä:*rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] 
> *Puolesta *cowwoc
> *Lähetetty:* 17. syyskuuta 2013 21:54
> *Vastaanottaja:* rtcweb@ietf.org
> *Aihe:* Re: [rtcweb] Video Codec Selection Plan
>
> Hi Ted,
>
>     Seeing as this discussion stems from licensing concerns, I like to 
> propose the following alternative:
>
>  1. Mandate a video codec whose IPR has expired. I agree that video
>     quality will degrade, which brings me to the next point.
>  2. Provide a negotiation mechanism which would allow peers to
>     "upgrade" to a superior (optionally-implemented) video codec.
>
>     This will allow us to support VP8, VP9, H264, H265 or whatever 
> other codec people like without the fear of transcoding or IPR. I 
> believe that in most cases negotiation will succeed in upgrading to a 
> superior codec. It will also encourage (as opposed to force) vendors 
> to support each other's codecs, which is the right way to go in light 
> of the political nature of this decision.
>
> Gili
>
>
> 1. If you support H.264 as the mandatory to implement codec or are
> willing to live with it as the MTI, please raise your hand now.
>
> 2. If you support VP8 as the mandatory to implement codec or are
> willing to live with it as the MTI, please raise your hand now.
>
>
> Gili
>
> On 13/09/2013 12:52 PM, Ted Hardie wrote:
>
>     WG,
>
>     The chairs have created a plan for how to perform the Video Codec
>     selection in our WG. The chairs are asking for review of our plan on
>     how to undertake the mandatory-to-implement video codec selection.
>     We'd much prefer to have comments on the mechanics before they begin,
>     so please review now.  Proponents of a particular proposal should
>     note both the actions required and the timelines proposed.
>
>     The main goal of this plan is to hold a consensus call on which of
>     the proposed alternatives we as a WG should select at one of the WG
>     sessions in Vancouver. Such a consensus call will of course be
>     verified on the mailing list for anyone who can't participate. The
>     chairs will recuse themselves from judging this particular
>     consensus.
>
>     In the WG session each codec proposal will be allowed an equal amount
>     of time to highlight the arguments for their proposal. After that a
>     there will be a slot for discussion and clarifying questions.
>
>     To enable the WG participants to get answers to any questions, the
>     proposals in draft form and any supporting material MUST be made
>     available by 6th of October. This is to ensure that the WG
>     participants can verify or object to any claims or statements in
>     the proposal material prior to the WG session. We chairs would really
>     not like to see the proponents bring up new arguments at their
>     presentation. Also the WG participants are expected to raise any
>     arguments on the list ahead of time to enable the proponents to
>     respond to such arguments.
>
>     The proposed consensus questions will be of the following form:
>
>     1. If you support H.264 as the mandatory to implement codec or are
>     willing to live with it as the MTI, please raise your hand now.
>
>     2. If you support VP8 as the mandatory to implement codec or are
>     willing to live with it as the MTI, please raise your hand now.
>
>     You may indicate support on both questions and we encourage you to do
>     so if you can live with either, even if you have a preference for one
>     over the other.
>
>     Additional proposals than the previous ones are welcome, but must be
>     submitted as draft and their proponents must notify the chairs no
>     later
>     than the 6th of October that they also have a candidate proposal.
>
>     In case the WG fails to reach consensus we chairs propose that we use
>     the alternative decision process as discussed in RFC3929. The method
>     and its usage will be discussed on the list should the WG not
>     establish consensus on a proposal for mandatory to implement video
>     codec.
>
>     regards,
>
>     Magnus,  Cullen, and Ted
>
>
>
>
>     _______________________________________________
>
>     rtcweb mailing list
>
>     rtcweb@ietf.org  <mailto:rtcweb@ietf.org>
>
>     https://www.ietf.org/mailman/listinfo/rtcweb
>


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    <div class="moz-cite-prefix"><br>
      &nbsp;&nbsp;&nbsp; So then the only question is: is there a reasonable video
      codec whose IPR has expired that we can agree on?<br>
      <br>
      &nbsp;&nbsp;&nbsp; My goal is to displease everyone equally. By picking a neutral
      codec, acknowledging that it displeases all parties in some way,
      we should be able to defuse most of the politics surrounding the
      decision.<br>
      <br>
      Gili<br>
      <br>
      On 17/09/2013 3:36 PM, Christer Holmberg wrote:<br>
    </div>
    <blockquote
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        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi,<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US">SDP Offer/Answer is used to negotiate codecs
            etc, so the API (JSEP) already provides such negotiation
            mechanism.
            <o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US">(Obviously the peers also need to negotiate
            with each other, but whatever protocol is used for that is
            outside the scope.)<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US">Regards,<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US">Christer<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
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        <div>
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            <p class="MsoNormal"><b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">L&auml;hett&auml;j&auml;:</span></b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">
                <a class="moz-txt-link-abbreviated" href="mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a> [<a class="moz-txt-link-freetext" href="mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org</a>]
                <b>Puolesta </b>cowwoc<br>
                <b>L&auml;hetetty:</b> 17. syyskuuta 2013 21:54<br>
                <b>Vastaanottaja:</b> <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                <b>Aihe:</b> Re: [rtcweb] Video Codec Selection Plan<o:p></o:p></span></p>
          </div>
        </div>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        <div>
          <p class="MsoNormal">Hi Ted,<br>
            <br>
            &nbsp;&nbsp;&nbsp; Seeing as this discussion stems from licensing concerns,
            I like to propose the following alternative:<o:p></o:p></p>
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            <li class="MsoNormal"
              style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;mso-list:l0
              level1 lfo1">
              Mandate a video codec whose IPR has expired. I agree that
              video quality will degrade, which brings me to the next
              point.<o:p></o:p></li>
            <li class="MsoNormal"
              style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;mso-list:l0
              level1 lfo1">
              Provide a negotiation mechanism which would allow peers to
              "upgrade" to a superior (optionally-implemented) video
              codec.<o:p></o:p></li>
          </ol>
          <p>&nbsp;&nbsp;&nbsp; This will allow us to support VP8, VP9, H264, H265 or
            whatever other codec people like without the fear of
            transcoding or IPR. I believe that in most cases negotiation
            will succeed in upgrading to a superior codec. It will also
            encourage (as opposed to force) vendors to support each
            other's codecs, which is the right way to go in light of the
            political nature of this decision.<o:p></o:p></p>
          <p>Gili<o:p></o:p></p>
          <p class="MsoNormal"><br>
            1. If you support H.264 as the mandatory to implement codec
            or are<br>
            willing to live with it as the MTI, please raise your hand
            now.<br>
            <br>
            2. If you support VP8 as the mandatory to implement codec or
            are<br>
            willing to live with it as the MTI, please raise your hand
            now.<br>
            <br>
            <br>
            Gili<br>
            <br>
            On 13/09/2013 12:52 PM, Ted Hardie wrote:<o:p></o:p></p>
        </div>
        <blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
          <div>
            <div>
              <p class="MsoNormal" style="margin-bottom:12.0pt">WG,<br>
                <br>
                The chairs have created a plan for how to perform the
                Video Codec<br>
                selection in our WG. The chairs are asking for review of
                our plan on<br>
                how to undertake the mandatory-to-implement video codec
                selection.<br>
                We'd much prefer to have comments on the mechanics
                before they begin,<br>
                so please review now.&nbsp; Proponents of a particular
                proposal should<br>
                note both the actions required and the timelines
                proposed.<br>
                <br>
                The main goal of this plan is to hold a consensus call
                on which of<br>
                the proposed alternatives we as a WG should select at
                one of the WG<br>
                sessions in Vancouver. Such a consensus call will of
                course be<br>
                verified on the mailing list for anyone who can't
                participate. The<br>
                chairs will recuse themselves from judging this
                particular<br>
                consensus.<br>
                <br>
                In the WG session each codec proposal will be allowed an
                equal amount<br>
                of time to highlight the arguments for their proposal.
                After that a<br>
                there will be a slot for discussion and clarifying
                questions.<br>
                <br>
                To enable the WG participants to get answers to any
                questions, the<br>
                proposals in draft form and any supporting material MUST
                be made<br>
                available by 6th of October. This is to ensure that the
                WG<br>
                participants can verify or object to any claims or
                statements in<br>
                the proposal material prior to the WG session. We chairs
                would really<br>
                not like to see the proponents bring up new arguments at
                their<br>
                presentation. Also the WG participants are expected to
                raise any<br>
                arguments on the list ahead of time to enable the
                proponents to<br>
                respond to such arguments.<br>
                <br>
                The proposed consensus questions will be of the
                following form:<br>
                <br>
                1. If you support H.264 as the mandatory to implement
                codec or are<br>
                willing to live with it as the MTI, please raise your
                hand now.<br>
                <br>
                2. If you support VP8 as the mandatory to implement
                codec or are<br>
                willing to live with it as the MTI, please raise your
                hand now.<br>
                <br>
                You may indicate support on both questions and we
                encourage you to do<br>
                so if you can live with either, even if you have a
                preference for one<br>
                over the other.<br>
                <br>
                Additional proposals than the previous ones are welcome,
                but must be<br>
                submitted as draft and their proponents must notify the
                chairs no later<br>
                than the 6th of October that they also have a candidate
                proposal.<br>
                <br>
                In case the WG fails to reach consensus we chairs
                propose that we use<br>
                the alternative decision process as discussed in
                RFC3929. The method<br>
                and its usage will be discussed on the list should the
                WG not<br>
                establish consensus on a proposal for mandatory to
                implement video codec.<o:p></o:p></p>
            </div>
            <div>
              <p class="MsoNormal" style="margin-bottom:12.0pt">regards,<o:p></o:p></p>
            </div>
            <p class="MsoNormal">Magnus,&nbsp; Cullen, and Ted<o:p></o:p></p>
          </div>
          <p class="MsoNormal"><br>
            <br>
            <br>
            <o:p></o:p></p>
          <pre>_______________________________________________<o:p></o:p></pre>
          <pre>rtcweb mailing list<o:p></o:p></pre>
          <pre><a moz-do-not-send="true" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><o:p></o:p></pre>
          <pre><a moz-do-not-send="true" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></o:p></pre>
        </blockquote>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
      </div>
    </blockquote>
    <br>
  </body>
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From john@jlc.net  Tue Sep 17 12:53:34 2013
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From: John Leslie <john@jlc.net>
To: cowwoc <cowwoc@bbs.darktech.org>
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cowwoc <cowwoc@bbs.darktech.org> wrote:
> 
> Seeing as this discussion stems from licensing concerns, I like to 
> propose the following alternative:
> 
> 1. Mandate a video codec whose IPR has expired. I agree that video
>    quality will degrade, which brings me to the next point.

   This is what I support. It won't be that long before whatever we
mandate will be obsolete anyway...

   But the chairs have determined we shall do one more round of duking
it out between H.264 and VP8. To get any other codec considered this
round, you must write a draft proposing it and stating its benefits.

   I do not choose to do so.

   Perhaps we _will_ choose one of these front-runners...

   Regardless, folks seem determined to argue the technical benefits
of these front-runners; and a number of us hold pretty strong opinions
of which of these is "better". I don't wish to complicate that
argument.

   YMMV, of course...

--
John Leslie <john@jlc.net>

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Date: Tue, 17 Sep 2013 13:10:26 -0700
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From: Ted Hardie <ted.ietf@gmail.com>
To: cowwoc <cowwoc@bbs.darktech.org>
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Subject: Re: [rtcweb] Video Codec Selection Plan
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On Tue, Sep 17, 2013 at 11:54 AM, cowwoc <cowwoc@bbs.darktech.org> wrote:

>  Hi Ted,
>
>     Seeing as this discussion stems from licensing concerns, I like to
> propose the following alternative:
>
>    1. Mandate a video codec whose IPR has expired. I agree that video
>    quality will degrade, which brings me to the next point.
>
>
To propose a video codec, please write a draft before the deadline and put
forward the technical details you're proposing (if there are profiles, for
example, which profile).  Saying you want this on the list isn't enough
information for folks to judge the proposal.  This is why we are insisting
on drafts at a deadline that gives folks the opportunity to review the
proposal.


>    1. Provide a negotiation mechanism which would allow peers to
>    "upgrade" to a superior (optionally-implemented) video codec.
>
>
This is a base part of the system.

Ted



>     This will allow us to support VP8, VP9, H264, H265 or whatever other
> codec people like without the fear of transcoding or IPR. I believe that in
> most cases negotiation will succeed in upgrading to a superior codec. It
> will also encourage (as opposed to force) vendors to support each other's
> codecs, which is the right way to go in light of the political nature of
> this decision.
>
> Gili
>
> 1. If you support H.264 as the mandatory to implement codec or are
> willing to live with it as the MTI, please raise your hand now.
>
> 2. If you support VP8 as the mandatory to implement codec or are
> willing to live with it as the MTI, please raise your hand now.
>
>
> Gili
>
>
> On 13/09/2013 12:52 PM, Ted Hardie wrote:
>
>  WG,
>
> The chairs have created a plan for how to perform the Video Codec
> selection in our WG. The chairs are asking for review of our plan on
> how to undertake the mandatory-to-implement video codec selection.
> We'd much prefer to have comments on the mechanics before they begin,
> so please review now.  Proponents of a particular proposal should
> note both the actions required and the timelines proposed.
>
> The main goal of this plan is to hold a consensus call on which of
> the proposed alternatives we as a WG should select at one of the WG
> sessions in Vancouver. Such a consensus call will of course be
> verified on the mailing list for anyone who can't participate. The
> chairs will recuse themselves from judging this particular
> consensus.
>
> In the WG session each codec proposal will be allowed an equal amount
> of time to highlight the arguments for their proposal. After that a
> there will be a slot for discussion and clarifying questions.
>
> To enable the WG participants to get answers to any questions, the
> proposals in draft form and any supporting material MUST be made
> available by 6th of October. This is to ensure that the WG
> participants can verify or object to any claims or statements in
> the proposal material prior to the WG session. We chairs would really
> not like to see the proponents bring up new arguments at their
> presentation. Also the WG participants are expected to raise any
> arguments on the list ahead of time to enable the proponents to
> respond to such arguments.
>
> The proposed consensus questions will be of the following form:
>
> 1. If you support H.264 as the mandatory to implement codec or are
> willing to live with it as the MTI, please raise your hand now.
>
> 2. If you support VP8 as the mandatory to implement codec or are
> willing to live with it as the MTI, please raise your hand now.
>
> You may indicate support on both questions and we encourage you to do
> so if you can live with either, even if you have a preference for one
> over the other.
>
> Additional proposals than the previous ones are welcome, but must be
> submitted as draft and their proponents must notify the chairs no later
> than the 6th of October that they also have a candidate proposal.
>
> In case the WG fails to reach consensus we chairs propose that we use
> the alternative decision process as discussed in RFC3929. The method
> and its usage will be discussed on the list should the WG not
> establish consensus on a proposal for mandatory to implement video codec.
>
>  regards,
>
>  Magnus,  Cullen, and Ted
>
>
> _______________________________________________
> rtcweb mailing listrtcweb@ietf.orghttps://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr">On Tue, Sep 17, 2013 at 11:54 AM, cowwoc <span dir=3D"ltr"=
>&lt;<a href=3D"mailto:cowwoc@bbs.darktech.org" target=3D"_blank">cowwoc@bb=
s.darktech.org</a>&gt;</span> wrote:<br><div class=3D"gmail_extra"><div cla=
ss=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
 =20
   =20
 =20
  <div text=3D"#000000" bgcolor=3D"#FFFFFF">
    <div>Hi Ted,<br>
      <br>
      =A0=A0=A0 Seeing as this discussion stems from licensing concerns, I
      like to propose the following alternative:<br>
      <ol>
        <li>Mandate a video codec whose IPR has expired. I agree that
          video quality will degrade, which brings me to the next point.</l=
i></ol></div></div></blockquote><div><br></div><div>To propose a video code=
c, please write a draft before the deadline and put forward the technical d=
etails you&#39;re proposing (if there are profiles, for example, which prof=
ile).=A0 Saying you want this on the list isn&#39;t enough information for =
folks to judge the proposal.=A0 This is why we are insisting on drafts at a=
 deadline that gives folks the opportunity to review the proposal.<br>
</div><div><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 =
0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div text=3D"#000000" b=
gcolor=3D"#FFFFFF"><div><ol>
        <li>Provide a negotiation mechanism which would allow peers to
          &quot;upgrade&quot; to a superior (optionally-implemented) video =
codec.</li>
      </ol>
      <p></p></div></div></blockquote><div><br></div><div>This is a base pa=
rt of the system.<br><br>Ted<br></div><div><br>=A0</div><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex">
<div text=3D"#000000" bgcolor=3D"#FFFFFF"><div><p>=A0=A0=A0 This will allow=
 us to support VP8, VP9, H264, H265 or
        whatever other codec people like without the fear of transcoding
        or IPR. I believe that in most cases negotiation will succeed in
        upgrading to a superior codec. It will also encourage (as
        opposed to force) vendors to support each other&#39;s codecs, which
        is the right way to go in light of the political nature of this
        decision.<br>
      </p>
      <p>Gili<br>
      </p><div class=3D"im">
      <br>
      1. If you support H.264 as the mandatory to implement codec or are<br=
>
      willing to live with it as the MTI, please raise your hand now.<br>
      <br>
      2. If you support VP8 as the mandatory to implement codec or are<br>
      willing to live with it as the MTI, please raise your hand now.<br>
      <br>
      <br></div>
      Gili<div><div class=3D"h5"><br>
      <br>
      On 13/09/2013 12:52 PM, Ted Hardie wrote:<br>
    </div></div></div>
    <blockquote type=3D"cite"><div><div class=3D"h5">
      <div dir=3D"ltr">
        <div>WG,<br>
          <br>
          The chairs have created a plan for how to perform the Video
          Codec<br>
          selection in our WG. The chairs are asking for review of our
          plan on<br>
          how to undertake the mandatory-to-implement video codec
          selection.<br>
          We&#39;d much prefer to have comments on the mechanics before the=
y
          begin,<br>
          so please review now.=A0 Proponents of a particular proposal
          should<br>
          note both the actions required and the timelines proposed.<br>
          <br>
          The main goal of this plan is to hold a consensus call on
          which of<br>
          the proposed alternatives we as a WG should select at one of
          the WG<br>
          sessions in Vancouver. Such a consensus call will of course be<br=
>
          verified on the mailing list for anyone who can&#39;t participate=
.
          The<br>
          chairs will recuse themselves from judging this particular<br>
          consensus.<br>
          <br>
          In the WG session each codec proposal will be allowed an equal
          amount<br>
          of time to highlight the arguments for their proposal. After
          that a<br>
          there will be a slot for discussion and clarifying questions.<br>
          <br>
          To enable the WG participants to get answers to any questions,
          the<br>
          proposals in draft form and any supporting material MUST be
          made<br>
          available by 6th of October. This is to ensure that the WG<br>
          participants can verify or object to any claims or statements
          in<br>
          the proposal material prior to the WG session. We chairs would
          really<br>
          not like to see the proponents bring up new arguments at their<br=
>
          presentation. Also the WG participants are expected to raise
          any<br>
          arguments on the list ahead of time to enable the proponents
          to<br>
          respond to such arguments.<br>
          <br>
          The proposed consensus questions will be of the following
          form:<br>
          <br>
          1. If you support H.264 as the mandatory to implement codec or
          are<br>
          willing to live with it as the MTI, please raise your hand
          now.<br>
          <br>
          2. If you support VP8 as the mandatory to implement codec or
          are<br>
          willing to live with it as the MTI, please raise your hand
          now.<br>
          <br>
          You may indicate support on both questions and we encourage
          you to do<br>
          so if you can live with either, even if you have a preference
          for one<br>
          over the other.<br>
          <br>
          Additional proposals than the previous ones are welcome, but
          must be<br>
          submitted as draft and their proponents must notify the chairs
          no later<br>
          than the 6th of October that they also have a candidate
          proposal.<br>
          <br>
          In case the WG fails to reach consensus we chairs propose that
          we use<br>
          the alternative decision process as discussed in RFC3929. The
          method<br>
          and its usage will be discussed on the list should the WG not<br>
          establish consensus on a proposal for mandatory to implement
          video codec.<br>
          <br>
        </div>
        <div>regards,<br>
          <br>
        </div>
        Magnus,=A0 Cullen, and Ted<br>
      </div>
      <br>
      <fieldset></fieldset>
      <br>
      </div></div><div class=3D"im"><pre>__________________________________=
_____________
rtcweb mailing list
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </div></blockquote>
    <br>
  </div>

<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>

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On Sep 17, 2013, at 12:53, "John Leslie" <john@jlc.net> wrote:

> cowwoc <cowwoc@bbs.darktech.org> wrote:
>>=20
>>=20
>> 1. Mandate a video codec whose IPR has expired. I agree that video
>>   quality will degrade, which brings me to the next point.
>=20
>   This is what I support. It won't be that long before whatever we
> mandate will be obsolete anyway...

[BA] I agree that the VP8 vs. H.264 discussion is past it's due date.  Howev=
er, starting a discussion of even older, completely obsolete codecs that no b=
rowser vendor has shown an interest in doesn't seem like a better choice.

>   I do not choose to do so.

[BA] Thank you for your leadership on this. Now if everyone else would follo=
w your lead we can recover the time for more productive pursuits :)=

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Subject: [rtcweb] Milestones changed for rtcweb WG
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Changed milestone "Send STUN Usage for Consent Freshness to IESG for
publication as proposed standard", set state to active from review,
accepting new milestone.

URL: http://datatracker.ietf.org/wg/rtcweb/charter/

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I agree with Magnus's comments in large part.  A few notes:=20

> 1. First of all I like to bring up a document structure problem. When
> reviewing this it is difficult to follow what requirements that are
> added for the different use cases. I would propose that each Use Case or
> general section that describe something that causes new requirements
> that hasn't been listed yet in the document to define the Requirement
> there....>> This makes the incremental use cases much more understandable=
 in how> they add requirements.

[BA] +1 on this.  I found myself flipping back and forth between the requir=
ements definition and the use cases to try to understand what the increment=
al functionality was and sometimes it was less than obvious why a requireme=
nt (such as "no wiretapping") was dropped from the basic use case.  To me=
=2C it makes more sense to lay out basic use cases=2C and then to focus on =
incremental requirement additions based on that. =20
>    The document focuses on requirements related to real-time media
>    streams and data exchange.  Requirements related to privacy=2C
>    signalling between the browser and web server etc. are currently not
>    considered.
>=20
> I think this needs to be rewritten. The draft actually do consider a
> number of high level security and privacy concerns.
[BA] I do think the document covers the security and privacy concerns (modu=
lo some issues I had with the "wiretapping" definition).  So I think we can=
 eliminate that last sentence entirely.=20
>    It is essential that the communication cannot be wiretapped
>    [RFC2804].
>=20
> This sentence returns a number of times in this document. I think it is
> a general security requirement that applies to probably all of the use
> cases. If there is one use case that don't need to enable protection
> against this=2C then please be explicit about that.>> My Suggestion is to=
 move this into the general considerations section.

[BA] There is a use case ("distributed music band") that omits the requirem=
ent=2C but that omission didn't make sense to me.  Personally=2C I would el=
evate this to a universal requirement in the base use case=2C and all other=
 cases.=20
>    One user has an unreliable Internet connection.  It sometimes loses
>    packets=2C and sometimes goes down completely.
>=20
> I guess the requirement here is to handle this in a graceful way from
> media transport and rendering perspective. The packet loss is also
> unclear as I don't know if the intention is to capture spurious packet
> losses due to the link media=2C or congestion losses. I think some
> expansion of this is needed.

[BA] To me=2C connectivity loss and packet loss are somewhat different issu=
es.  Connectivity loss leads to addition of requirements relating to keep-a=
lives and/or consent (e.g. if the peer drops off the Internet=2C the consen=
t mechanism will detect it eventually and stop sending).   Packet loss lead=
s to requirements relating to concealment=2C FEC and/or RTX.  So lumping th=
em together doesn't make sense to me.  > 3.2.2.2.  Derived Requirements
>=20
>    F1=2C F2=2C F3=2C F4=2C F5=2C F8=2C F9=2C F10=2C F20=2C F25=2C F28=2C =
F29
>=20
>    A1=2C A2=2C A3=2C A4=2C A5=2C A6=2C A7=2C A8=2C A9=2C A10=2C A11=2C A1=
2
>=20
> How come this list of Derived Requirements are shorter than for the
> parent use case which has the following list:

[BA] Yes=2C that didn't make sense to me either.  Better to stick with the =
requirements of the base case and add to them as necessary for more advance=
d cases.  If something is deleted=2C I think the document needs to explain =
why=2C but in this case the deletions didn't make sense to me.=20
> If a firewall only allows HTTP traffic=2C then can we really assume that
> the firewall administrator per default will accept WebRTC Media and Data
> traffic?

[BA]  Data traffic sent as SCTP/DTLS/UDP will *not* work in this case.  Web=
sockets might=2C but then I think you need to clarify whether "HTTP only" i=
ncludes websockets (in my experience=2C in most cases it won't).=20
>    But in addition to this=2C one of the users can share what is being
>    displayed on her/his screen with a peer.  The user can choose to
>    share the entire screen=2C part of the screen (part selected by the
>    user) or what a selected applicaton displays with the peer.
>=20
> Does this needs mentioning of additional high level security requirements=
?

[BA]  The requirements may need to be *very* high level (e.g. "without intr=
oducing other vulnerabilities")=2C because this is a tricky area that may e=
nd up requiring significant work.=20
> 12. Section 3.2.10.1:
>=20
>   The service providers are interconnected by some means=2C but exchange
>    no more information about the users than what can be carried using
>    SIP.
>=20
>    NOTE: More profiling of what this means may be needed.
>=20
> The Note=2C can it be removed=2C or do we need additional text in this do=
cument?
[BA] The NOTE tells me that we're not even really sure why this section is =
there.  So perhaps we should figure out what it adds above and beyond the g=
ateway use case.=20
 		 	   		  =

--_8f222632-cb83-4518-886b-d36de25f0e66_
Content-Type: text/html; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<html>
<head>
<style><!--
.hmmessage P
{
margin:0px=3B
padding:0px
}
body.hmmessage
{
font-size: 12pt=3B
font-family:Calibri
}
--></style></head>
<body class=3D'hmmessage'><div dir=3D'ltr'>I agree with Magnus's comments i=
n large part. &nbsp=3BA few notes:&nbsp=3B<br><br><div>&gt=3B 1. First of a=
ll I like to bring up a document structure problem. When<br>&gt=3B reviewin=
g this it is difficult to follow what requirements that are<br>&gt=3B added=
 for the different use cases. I would propose that each Use Case or<br>&gt=
=3B general section that describe something that causes new requirements<br=
>&gt=3B that hasn't been listed yet in the document to define the Requireme=
nt<br>&gt=3B there....</div><div><span style=3D"font-size: 12pt=3B">&gt=3B<=
/span></div><div><span style=3D"font-size: 12pt=3B">&gt=3B This makes the i=
ncremental use cases much more understandable in how</span></div>&gt=3B the=
y add requirements.<div><br></div><div><br></div><div>[BA] +1 on this. &nbs=
p=3BI found myself flipping back and forth between the requirements definit=
ion and the use cases to&nbsp=3B</div><div>try to understand what the incre=
mental functionality was and sometimes it was less than obvious why a requi=
rement (such as "no wiretapping") was dropped from the basic use case. &nbs=
p=3BTo me=2C it makes more sense to lay out basic use cases=2C and then to =
focus on incremental requirement additions based on that. &nbsp=3B</div><di=
v><br></div><div>&gt=3B    The document focuses on requirements related to =
real-time media<br>&gt=3B    streams and data exchange.  Requirements relat=
ed to privacy=2C<br>&gt=3B    signalling between the browser and web server=
 etc. are currently not<br>&gt=3B    considered.<br>&gt=3B <br>&gt=3B I thi=
nk this needs to be rewritten. The draft actually do consider a<br>&gt=3B n=
umber of high level security and privacy concerns.</div><div><br></div><div=
>[BA] I do think the document covers the security and privacy concerns (mod=
ulo some issues I had with the "wiretapping" definition). &nbsp=3BSo I thin=
k we can eliminate that last sentence entirely.&nbsp=3B</div><div><br></div=
><div><span style=3D"font-size: 12pt=3B">&gt=3B    It is essential that the=
 communication cannot be wiretapped</span><br>&gt=3B    [RFC2804].<br>&gt=
=3B <br>&gt=3B This sentence returns a number of times in this document. I =
think it is<br>&gt=3B a general security requirement that applies to probab=
ly all of the use<br>&gt=3B cases. If there is one use case that don't need=
 to enable protection<br>&gt=3B against this=2C then please be explicit abo=
ut that.</div><div>&gt=3B</div><div>&gt=3B My Suggestion is to move this in=
to the general considerations section.<br><br></div><div>[BA] There is a us=
e case ("distributed music band") that omits the requirement=2C but that om=
ission didn't make sense to me. &nbsp=3BPersonally=2C I would elevate this =
to a universal requirement in the base use case=2C and all other cases.&nbs=
p=3B</div><div><br></div><div>&gt=3B    One user has an unreliable Internet=
 connection.  It sometimes loses<br>&gt=3B    packets=2C and sometimes goes=
 down completely.<br>&gt=3B <br>&gt=3B I guess the requirement here is to h=
andle this in a graceful way from<br>&gt=3B media transport and rendering p=
erspective. The packet loss is also<br>&gt=3B unclear as I don't know if th=
e intention is to capture spurious packet<br>&gt=3B losses due to the link =
media=2C or congestion losses. I think some<br>&gt=3B expansion of this is =
needed.<br><br></div><div>[BA] To me=2C connectivity loss and packet loss a=
re somewhat different issues. &nbsp=3BConnectivity loss leads to addition o=
f requirements relating to keep-alives and/or consent (e.g. if the peer dro=
ps off the Internet=2C the consent mechanism will detect it eventually and =
stop sending). &nbsp=3B Packet loss leads to requirements relating to conce=
alment=2C FEC and/or RTX. &nbsp=3BSo lumping them together doesn't make sen=
se to me.&nbsp=3B</div><div><span style=3D"font-size: 12pt=3B">&nbsp=3B</sp=
an></div><div>&gt=3B 3.2.2.2.  Derived Requirements<br>&gt=3B <br>&gt=3B   =
 F1=2C F2=2C F3=2C F4=2C F5=2C F8=2C F9=2C F10=2C F20=2C F25=2C F28=2C F29<=
br>&gt=3B <br>&gt=3B    A1=2C A2=2C A3=2C A4=2C A5=2C A6=2C A7=2C A8=2C A9=
=2C A10=2C A11=2C A12<br>&gt=3B <br>&gt=3B How come this list of Derived Re=
quirements are shorter than for the<br>&gt=3B parent use case which has the=
 following list:<br><br></div><div>[BA] Yes=2C that didn't make sense to me=
 either. &nbsp=3BBetter to stick with the requirements of the base case and=
 add to them as necessary for more advanced cases. &nbsp=3BIf something is =
deleted=2C I think the document needs to explain why=2C but in this case th=
e deletions didn't make sense to me.&nbsp=3B</div><div><br></div><div>&gt=
=3B If a firewall only allows HTTP traffic=2C then can we really assume tha=
t<br>&gt=3B the firewall administrator per default will accept WebRTC Media=
 and Data<br>&gt=3B traffic?<br><br></div><div>[BA] &nbsp=3BData traffic se=
nt as SCTP/DTLS/UDP will *not* work in this case. &nbsp=3BWebsockets might=
=2C but then I think you need to clarify whether "HTTP only" includes webso=
ckets (in my experience=2C in most cases it won't).&nbsp=3B</div><div><br><=
/div><div>&gt=3B    But in addition to this=2C one of the users can share w=
hat is being<br>&gt=3B    displayed on her/his screen with a peer.  The use=
r can choose to<br>&gt=3B    share the entire screen=2C part of the screen =
(part selected by the<br>&gt=3B    user) or what a selected applicaton disp=
lays with the peer.<br>&gt=3B <br>&gt=3B Does this needs mentioning of addi=
tional high level security requirements?<br><br></div><div>[BA] &nbsp=3BThe=
 requirements may need to be *very* high level (e.g. "without introducing o=
ther vulnerabilities")=2C because this is a tricky area that may end up req=
uiring significant work.&nbsp=3B</div><div><br></div><div>&gt=3B 12. Sectio=
n 3.2.10.1:<br>&gt=3B <br>&gt=3B   The service providers are interconnected=
 by some means=2C but exchange<br>&gt=3B    no more information about the u=
sers than what can be carried using<br>&gt=3B    SIP.<br>&gt=3B <br>&gt=3B =
   NOTE: More profiling of what this means may be needed.<br>&gt=3B <br>&gt=
=3B The Note=2C can it be removed=2C or do we need additional text in this =
document?</div><div><br></div><div>[BA] The NOTE tells me that we're not ev=
en really sure why this section is there. &nbsp=3BSo perhaps we should figu=
re out what it adds above and beyond the gateway use case.&nbsp=3B</div><di=
v><br></div> 		 	   		  </div></body>
</html>=

--_8f222632-cb83-4518-886b-d36de25f0e66_--

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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Real-Time Communication in WEB-browsers W=
orking Group of the IETF.

	Title           : Javascript Session Establishment Protocol
	Author(s)       : Justin Uberti
                          Cullen Jennings
	Filename        : draft-ietf-rtcweb-jsep-04.txt
	Pages           : 46
	Date            : 2013-09-17

Abstract:
   This document describes the mechanisms for allowing a Javascript
   application to control the signaling plane of a multimedia session
   via the interface specified in the W3C RTCPeerConnection API, and
   discusses how this relates to existing signaling protocols.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-jsep

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-jsep-04

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-jsep-04


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


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--047d7b2e0abb6f6d4504e69bdae0
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The draft has now been posted at
http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-jsep-04.txt, with 13
new pages of text.

Readers should focus on Section 5, which specifies the behavior of
createOffer/createAnswer, as well as Appendix A.2, which provides examples
of the generated SDP.


On Tue, Sep 17, 2013 at 1:51 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> On 2013-09-17 10:47, Harald Alvestrand wrote:
> > Magnus,
> >
> > what is the expected duration of the call?
>
> 2 hour maximum
>
> Magnus
>
> >
> > On 09/17/2013 10:38 AM, Magnus Westerlund wrote:
> >> WG,
> >>
> >> I know that the new JSEP Draft is not yet submitted. I am hoping that =
it
> >> will show up during the morning hours for the US based people. That wa=
y
> >> everyone gets at least some few hours to look at the draft prior to th=
e
> >> call. If it hasn't we will postpone the call, but I really hope we can
> >> avoid this. I will send a notice to the WG if we have to postpone it.
> >>
> >> Cheers
> >>
> >> Magnus
> >>
> >> On 2013-09-11 07:58, Magnus Westerlund wrote:
> >>> WG,
> >>>
> >>> There will be a new version of JSEP draft coming out Monday. As a
> follow
> >>> up to get as much progress on this topic as possible. The chairs and
> >>> authors are offering a informal walk through session on Wednesday the
> >>> 18th. People will be strongly recommended to have reviewed the new
> >>> draft.
> >>>
> >>> The intention of the session is enable the authors to inform you abou=
t
> >>> the latest changes, any issues to consider and what needs further
> >>> discussion. The will also enable you as participant after your review
> of
> >>> the document to ask clarifying questions or for motivations behind
> >>> choices, and raise issues you have spotted.
> >>>
> >>> The session is not intended to solve issues or judge consensus on par=
t
> >>> of the draft. Discussions of what is the best solution etc, will to b=
e
> >>> cut off.
> >>>
> >>> This will be a webex conference per details below.
> >>>
> >>> Cheers
> >>>
> >>> Magnus Westerlund
> >>>
> >>> Topic: RTCWeb - JSEP
> >>> Date: Wednesday, September 18, 2013
> >>> Time: 8:00 am, Pacific Daylight Time (San Francisco, GMT-07:00)
> >>> Meeting Number: 302 061 238
> >>> Meeting Password: ietf
> >>>
> >>> -------------------------------------------------------
> >>> To join the online meeting (Now from mobile devices!)
> >>> -------------------------------------------------------
> >>> 1. Go to
> >>>
> https://cisco.webex.com/ciscosales/j.php?ED=3D206400898&UID=3D0&PW=3DNYjk=
4NmE0ODBj&RT=3DMiM0
> >>>
> >>>
> >>> 2. Enter your name and email address.
> >>> 3. Enter the meeting password: ietf
> >>> 4. Click "Join Now".
> >>>
> >>> To view in other time zones or languages, please click the link:
> >>>
> https://cisco.webex.com/ciscosales/j.php?ED=3D206400898&UID=3D0&PW=3DNYjk=
4NmE0ODBj&ORT=3DMiM0
> >>>
> >>>
> >>>
> >>> -------------------------------------------------------
> >>> To join the teleconference only
> >>> -------------------------------------------------------
> >>> 1. Dial into Cisco WebEx (view all Global Access Numbers at
> >>> http://cisco.com/en/US/about/doing_business/conferencing/index.html
> >>> 2. Follow the prompts to enter the Meeting Number (listed above) or
> >>> Access Code followed by the # sign.
> >>>
> >>> San Jose, CA: +1.408.525.6800 RTP: +1.919.392.3330
> >>>
> >>> US/Canada: +1.866.432.9903 United Kingdom: +44.20.8824.0117
> >>>
> >>> India: +91.80.4350.1111 Germany: +49.619.6773.9002
> >>>
> >>> Japan: +81.3.5763.9394 China: +86.10.8515.5666
> >>>
> >>>
> >>> ----------------------------------------------------------------
> >>> ALERT =E2=80=93 PLEASE READ: DO NOT DIAL THE TOLL FREE NUMBERS FROM W=
ITHIN THE
> >>> (408) OR (919) AREA CODES
> >>> ----------------------------------------------------------------
> >>> Please dial the local access number for your area from the list below=
:
> >>> - San Jose/Milpitas (408) area: 525-6800
> >>> - RTP (919) area: 392-3330
> >>>
> >>> Dialing the WebEx toll free numbers from within 408 or 919 area codes
> is
> >>> not enabled (non-Cisco phones). =E2=80=9C If you dial the toll-free n=
umbers
> >>> within the 408 or 919 area codes you will be instructed to hang up an=
d
> >>> dial the local access number.=E2=80=9D Please use the call-back optio=
n whenever
> >>> possible and otherwise dial local numbers only. The affected toll fre=
e
> >>> numbers are: (866) 432-9903 for the San Jose/Milpitas area and (866)
> >>> 349-3520 for the RTP area.
> >>>
> >>
> >
> >
> >
>
>
> --
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=C3=A4r=C3=B6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--047d7b2e0abb6f6d4504e69bdae0
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">The draft has now been posted at=C2=A0<a href=3D"http://ww=
w.ietf.org/internet-drafts/draft-ietf-rtcweb-jsep-04.txt" target=3D"_blank"=
>http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-jsep-04.txt</a>, wit=
h 13 new pages of text.<div>

<br></div><div>Readers should focus on Section 5, which specifies the behav=
ior of createOffer/createAnswer, as well as Appendix A.2, which provides ex=
amples of the generated SDP.</div></div><div class=3D"gmail_extra"><br><br>

<div class=3D"gmail_quote">On Tue, Sep 17, 2013 at 1:51 AM, Magnus Westerlu=
nd <span dir=3D"ltr">&lt;<a href=3D"mailto:magnus.westerlund@ericsson.com" =
target=3D"_blank">magnus.westerlund@ericsson.com</a>&gt;</span> wrote:<br><=
blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px=
 #ccc solid;padding-left:1ex">

<div class=3D"im">On 2013-09-17 10:47, Harald Alvestrand wrote:<br>
&gt; Magnus,<br>
&gt;<br>
&gt; what is the expected duration of the call?<br>
<br>
</div>2 hour maximum<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
Magnus<br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
&gt;<br>
&gt; On 09/17/2013 10:38 AM, Magnus Westerlund wrote:<br>
&gt;&gt; WG,<br>
&gt;&gt;<br>
&gt;&gt; I know that the new JSEP Draft is not yet submitted. I am hoping t=
hat it<br>
&gt;&gt; will show up during the morning hours for the US based people. Tha=
t way<br>
&gt;&gt; everyone gets at least some few hours to look at the draft prior t=
o the<br>
&gt;&gt; call. If it hasn&#39;t we will postpone the call, but I really hop=
e we can<br>
&gt;&gt; avoid this. I will send a notice to the WG if we have to postpone =
it.<br>
&gt;&gt;<br>
&gt;&gt; Cheers<br>
&gt;&gt;<br>
&gt;&gt; Magnus<br>
&gt;&gt;<br>
&gt;&gt; On 2013-09-11 07:58, Magnus Westerlund wrote:<br>
&gt;&gt;&gt; WG,<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; There will be a new version of JSEP draft coming out Monday. A=
s a follow<br>
&gt;&gt;&gt; up to get as much progress on this topic as possible. The chai=
rs and<br>
&gt;&gt;&gt; authors are offering a informal walk through session on Wednes=
day the<br>
&gt;&gt;&gt; 18th. People will be strongly recommended to have reviewed the=
 new<br>
&gt;&gt;&gt; draft.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; The intention of the session is enable the authors to inform y=
ou about<br>
&gt;&gt;&gt; the latest changes, any issues to consider and what needs furt=
her<br>
&gt;&gt;&gt; discussion. The will also enable you as participant after your=
 review of<br>
&gt;&gt;&gt; the document to ask clarifying questions or for motivations be=
hind<br>
&gt;&gt;&gt; choices, and raise issues you have spotted.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; The session is not intended to solve issues or judge consensus=
 on part<br>
&gt;&gt;&gt; of the draft. Discussions of what is the best solution etc, wi=
ll to be<br>
&gt;&gt;&gt; cut off.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; This will be a webex conference per details below.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Cheers<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Magnus Westerlund<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Topic: RTCWeb - JSEP<br>
&gt;&gt;&gt; Date: Wednesday, September 18, 2013<br>
&gt;&gt;&gt; Time: 8:00 am, Pacific Daylight Time (San Francisco, GMT-07:00=
)<br>
&gt;&gt;&gt; Meeting Number: 302 061 238<br>
&gt;&gt;&gt; Meeting Password: ietf<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; -------------------------------------------------------<br>
&gt;&gt;&gt; To join the online meeting (Now from mobile devices!)<br>
&gt;&gt;&gt; -------------------------------------------------------<br>
&gt;&gt;&gt; 1. Go to<br>
&gt;&gt;&gt; <a href=3D"https://cisco.webex.com/ciscosales/j.php?ED=3D20640=
0898&amp;UID=3D0&amp;PW=3DNYjk4NmE0ODBj&amp;RT=3DMiM0" target=3D"_blank">ht=
tps://cisco.webex.com/ciscosales/j.php?ED=3D206400898&amp;UID=3D0&amp;PW=3D=
NYjk4NmE0ODBj&amp;RT=3DMiM0</a><br>


&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; 2. Enter your name and email address.<br>
&gt;&gt;&gt; 3. Enter the meeting password: ietf<br>
&gt;&gt;&gt; 4. Click &quot;Join Now&quot;.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; To view in other time zones or languages, please click the lin=
k:<br>
&gt;&gt;&gt; <a href=3D"https://cisco.webex.com/ciscosales/j.php?ED=3D20640=
0898&amp;UID=3D0&amp;PW=3DNYjk4NmE0ODBj&amp;ORT=3DMiM0" target=3D"_blank">h=
ttps://cisco.webex.com/ciscosales/j.php?ED=3D206400898&amp;UID=3D0&amp;PW=
=3DNYjk4NmE0ODBj&amp;ORT=3DMiM0</a><br>


&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; -------------------------------------------------------<br>
&gt;&gt;&gt; To join the teleconference only<br>
&gt;&gt;&gt; -------------------------------------------------------<br>
&gt;&gt;&gt; 1. Dial into Cisco WebEx (view all Global Access Numbers at<br=
>
&gt;&gt;&gt; <a href=3D"http://cisco.com/en/US/about/doing_business/confere=
ncing/index.html" target=3D"_blank">http://cisco.com/en/US/about/doing_busi=
ness/conferencing/index.html</a><br>
&gt;&gt;&gt; 2. Follow the prompts to enter the Meeting Number (listed abov=
e) or<br>
&gt;&gt;&gt; Access Code followed by the # sign.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; San Jose, CA: <a href=3D"tel:%2B1.408.525.6800" value=3D"+1408=
5256800">+1.408.525.6800</a> RTP: <a href=3D"tel:%2B1.919.392.3330" value=
=3D"+19193923330">+1.919.392.3330</a><br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; US/Canada: <a href=3D"tel:%2B1.866.432.9903" value=3D"+1866432=
9903">+1.866.432.9903</a> United Kingdom: <a href=3D"tel:%2B44.20.8824.0117=
" value=3D"+442088240117">+44.20.8824.0117</a><br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; India: <a href=3D"tel:%2B91.80.4350.1111" value=3D"+9180435011=
11">+91.80.4350.1111</a> Germany: <a href=3D"tel:%2B49.619.6773.9002" value=
=3D"+4961967739002">+49.619.6773.9002</a><br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Japan: <a href=3D"tel:%2B81.3.5763.9394" value=3D"+81357639394=
">+81.3.5763.9394</a> China: <a href=3D"tel:%2B86.10.8515.5666" value=3D"+8=
61085155666">+86.10.8515.5666</a><br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; --------------------------------------------------------------=
--<br>
&gt;&gt;&gt; ALERT =E2=80=93 PLEASE READ: DO NOT DIAL THE TOLL FREE NUMBERS=
 FROM WITHIN THE<br>
&gt;&gt;&gt; (408) OR (919) AREA CODES<br>
&gt;&gt;&gt; --------------------------------------------------------------=
--<br>
&gt;&gt;&gt; Please dial the local access number for your area from the lis=
t below:<br>
&gt;&gt;&gt; - San Jose/Milpitas (408) area: 525-6800<br>
&gt;&gt;&gt; - RTP (919) area: 392-3330<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Dialing the WebEx toll free numbers from within 408 or 919 are=
a codes is<br>
&gt;&gt;&gt; not enabled (non-Cisco phones). =E2=80=9C If you dial the toll=
-free numbers<br>
&gt;&gt;&gt; within the 408 or 919 area codes you will be instructed to han=
g up and<br>
&gt;&gt;&gt; dial the local access number.=E2=80=9D Please use the call-bac=
k option whenever<br>
&gt;&gt;&gt; possible and otherwise dial local numbers only. The affected t=
oll free<br>
&gt;&gt;&gt; numbers are: (866) 432-9903 for the San Jose/Milpitas area and=
 (866)<br>
&gt;&gt;&gt; 349-3520 for the RTP area.<br>
&gt;&gt;&gt;<br>
&gt;&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
<br>
<br>
</div></div><div class=3D"im HOEnZb">--<br>
<br>
Magnus Westerlund<br>
<br>
----------------------------------------------------------------------<br>
Multimedia Technologies, Ericsson Research EAB/TVM<br>
----------------------------------------------------------------------<br>
Ericsson AB =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0| Phone =
=C2=A0<a href=3D"tel:%2B46%2010%207148287" value=3D"+46107148287">+46 10 71=
48287</a><br>
F=C3=A4r=C3=B6gatan 6 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0| Mobile <a href=3D"tel:%2B46%2073%200949079" value=3D"+46730949079">+46=
 73 0949079</a><br>
SE-164 80 Stockholm, Sweden| mailto: <a href=3D"mailto:magnus.westerlund@er=
icsson.com">magnus.westerlund@ericsson.com</a><br>
----------------------------------------------------------------------<br>
<br>
</div><div class=3D"HOEnZb"><div class=3D"h5">_____________________________=
__________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>

--047d7b2e0abb6f6d4504e69bdae0--

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Date: Tue, 17 Sep 2013 17:13:05 -0700
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--047d7b6d8afc91b17104e69d4fe7
Content-Type: text/plain; charset=ISO-8859-1

On Tue, Sep 17, 2013 at 11:54 AM, cowwoc <cowwoc@bbs.darktech.org> wrote:

>  Hi Ted,
>
>     Seeing as this discussion stems from licensing concerns, I like to
> propose the following alternative:
>
>    1. Mandate a video codec whose IPR has expired. I agree that video
>    quality will degrade, which brings me to the next point.
>    2. Provide a negotiation mechanism which would allow peers to
>    "upgrade" to a superior (optionally-implemented) video codec.
>
> Negotiation has always been part of the design. The sole question is which
codec is mandatory to implement, not which is the sole codec to be mandated.

-Ekr


    This will allow us to support VP8, VP9, H264, H265 or whatever other
> codec people like without the fear of transcoding or IPR. I believe that in
> most cases negotiation will succeed in upgrading to a superior codec. It
> will also encourage (as opposed to force) vendors to support each other's
> codecs, which is the right way to go in light of the political nature of
> this decision.
>
> Gili
>
> 1. If you support H.264 as the mandatory to implement codec or are
> willing to live with it as the MTI, please raise your hand now.
>
> 2. If you support VP8 as the mandatory to implement codec or are
> willing to live with it as the MTI, please raise your hand now.
>
>
> Gili
>
>
> On 13/09/2013 12:52 PM, Ted Hardie wrote:
>
>  WG,
>
> The chairs have created a plan for how to perform the Video Codec
> selection in our WG. The chairs are asking for review of our plan on
> how to undertake the mandatory-to-implement video codec selection.
> We'd much prefer to have comments on the mechanics before they begin,
> so please review now.  Proponents of a particular proposal should
> note both the actions required and the timelines proposed.
>
> The main goal of this plan is to hold a consensus call on which of
> the proposed alternatives we as a WG should select at one of the WG
> sessions in Vancouver. Such a consensus call will of course be
> verified on the mailing list for anyone who can't participate. The
> chairs will recuse themselves from judging this particular
> consensus.
>
> In the WG session each codec proposal will be allowed an equal amount
> of time to highlight the arguments for their proposal. After that a
> there will be a slot for discussion and clarifying questions.
>
> To enable the WG participants to get answers to any questions, the
> proposals in draft form and any supporting material MUST be made
> available by 6th of October. This is to ensure that the WG
> participants can verify or object to any claims or statements in
> the proposal material prior to the WG session. We chairs would really
> not like to see the proponents bring up new arguments at their
> presentation. Also the WG participants are expected to raise any
> arguments on the list ahead of time to enable the proponents to
> respond to such arguments.
>
> The proposed consensus questions will be of the following form:
>
> 1. If you support H.264 as the mandatory to implement codec or are
> willing to live with it as the MTI, please raise your hand now.
>
> 2. If you support VP8 as the mandatory to implement codec or are
> willing to live with it as the MTI, please raise your hand now.
>
> You may indicate support on both questions and we encourage you to do
> so if you can live with either, even if you have a preference for one
> over the other.
>
> Additional proposals than the previous ones are welcome, but must be
> submitted as draft and their proponents must notify the chairs no later
> than the 6th of October that they also have a candidate proposal.
>
> In case the WG fails to reach consensus we chairs propose that we use
> the alternative decision process as discussed in RFC3929. The method
> and its usage will be discussed on the list should the WG not
> establish consensus on a proposal for mandatory to implement video codec.
>
>  regards,
>
>  Magnus,  Cullen, and Ted
>
>
> _______________________________________________
> rtcweb mailing listrtcweb@ietf.orghttps://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--047d7b6d8afc91b17104e69d4fe7
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Tue, Sep 17, 2013 at 11:54 AM, cowwoc <span dir=3D"ltr">&lt;<a h=
ref=3D"mailto:cowwoc@bbs.darktech.org" target=3D"_blank">cowwoc@bbs.darktec=
h.org</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
 =20
   =20
 =20
  <div text=3D"#000000" bgcolor=3D"#FFFFFF">
    <div>Hi Ted,<br>
      <br>
      =A0=A0=A0 Seeing as this discussion stems from licensing concerns, I
      like to propose the following alternative:<br>
      <ol>
        <li>Mandate a video codec whose IPR has expired. I agree that
          video quality will degrade, which brings me to the next point.</l=
i>
        <li>Provide a negotiation mechanism which would allow peers to
          &quot;upgrade&quot; to a superior (optionally-implemented) video =
codec.</li></ol></div></div></blockquote><div>Negotiation has always been p=
art of the design. The sole question is which</div><div>codec is mandatory =
to implement, not which is the sole codec to be mandated.</div>

<div><br></div><div>-Ekr</div><div><br></div><div><br></div><blockquote cla=
ss=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pa=
dding-left:1ex"><div text=3D"#000000" bgcolor=3D"#FFFFFF"><div><p>=A0 =A0 T=
his will allow us to support VP8, VP9, H264, H265 or
        whatever other codec people like without the fear of transcoding
        or IPR. I believe that in most cases negotiation will succeed in
        upgrading to a superior codec. It will also encourage (as
        opposed to force) vendors to support each other&#39;s codecs, which
        is the right way to go in light of the political nature of this
        decision.<br>
      </p>
      <p>Gili<br>
      </p><div class=3D"im">
      <br>
      1. If you support H.264 as the mandatory to implement codec or are<br=
>
      willing to live with it as the MTI, please raise your hand now.<br>
      <br>
      2. If you support VP8 as the mandatory to implement codec or are<br>
      willing to live with it as the MTI, please raise your hand now.<br>
      <br>
      <br></div>
      Gili<div><div class=3D"h5"><br>
      <br>
      On 13/09/2013 12:52 PM, Ted Hardie wrote:<br>
    </div></div></div>
    <blockquote type=3D"cite"><div><div class=3D"h5">
      <div dir=3D"ltr">
        <div>WG,<br>
          <br>
          The chairs have created a plan for how to perform the Video
          Codec<br>
          selection in our WG. The chairs are asking for review of our
          plan on<br>
          how to undertake the mandatory-to-implement video codec
          selection.<br>
          We&#39;d much prefer to have comments on the mechanics before the=
y
          begin,<br>
          so please review now.=A0 Proponents of a particular proposal
          should<br>
          note both the actions required and the timelines proposed.<br>
          <br>
          The main goal of this plan is to hold a consensus call on
          which of<br>
          the proposed alternatives we as a WG should select at one of
          the WG<br>
          sessions in Vancouver. Such a consensus call will of course be<br=
>
          verified on the mailing list for anyone who can&#39;t participate=
.
          The<br>
          chairs will recuse themselves from judging this particular<br>
          consensus.<br>
          <br>
          In the WG session each codec proposal will be allowed an equal
          amount<br>
          of time to highlight the arguments for their proposal. After
          that a<br>
          there will be a slot for discussion and clarifying questions.<br>
          <br>
          To enable the WG participants to get answers to any questions,
          the<br>
          proposals in draft form and any supporting material MUST be
          made<br>
          available by 6th of October. This is to ensure that the WG<br>
          participants can verify or object to any claims or statements
          in<br>
          the proposal material prior to the WG session. We chairs would
          really<br>
          not like to see the proponents bring up new arguments at their<br=
>
          presentation. Also the WG participants are expected to raise
          any<br>
          arguments on the list ahead of time to enable the proponents
          to<br>
          respond to such arguments.<br>
          <br>
          The proposed consensus questions will be of the following
          form:<br>
          <br>
          1. If you support H.264 as the mandatory to implement codec or
          are<br>
          willing to live with it as the MTI, please raise your hand
          now.<br>
          <br>
          2. If you support VP8 as the mandatory to implement codec or
          are<br>
          willing to live with it as the MTI, please raise your hand
          now.<br>
          <br>
          You may indicate support on both questions and we encourage
          you to do<br>
          so if you can live with either, even if you have a preference
          for one<br>
          over the other.<br>
          <br>
          Additional proposals than the previous ones are welcome, but
          must be<br>
          submitted as draft and their proponents must notify the chairs
          no later<br>
          than the 6th of October that they also have a candidate
          proposal.<br>
          <br>
          In case the WG fails to reach consensus we chairs propose that
          we use<br>
          the alternative decision process as discussed in RFC3929. The
          method<br>
          and its usage will be discussed on the list should the WG not<br>
          establish consensus on a proposal for mandatory to implement
          video codec.<br>
          <br>
        </div>
        <div>regards,<br>
          <br>
        </div>
        Magnus,=A0 Cullen, and Ted<br>
      </div>
      <br>
      <fieldset></fieldset>
      <br>
      </div></div><div class=3D"im"><pre>__________________________________=
_____________
rtcweb mailing list
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </div></blockquote>
    <br>
  </div>

<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>

--047d7b6d8afc91b17104e69d4fe7--

From lgeyser@gmail.com  Wed Sep 18 01:39:20 2013
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Date: Wed, 18 Sep 2013 10:39:17 +0200
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From: Leon Geyser <lgeyser@gmail.com>
To: Eric Rescorla <ekr@rtfm.com>
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Subject: Re: [rtcweb] Video Codec Selection Plan
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Anyone willing to create a draft for H.261?


On 18 September 2013 02:13, Eric Rescorla <ekr@rtfm.com> wrote:

>
>
>
> On Tue, Sep 17, 2013 at 11:54 AM, cowwoc <cowwoc@bbs.darktech.org> wrote:
>
>>  Hi Ted,
>>
>>     Seeing as this discussion stems from licensing concerns, I like to
>> propose the following alternative:
>>
>>    1. Mandate a video codec whose IPR has expired. I agree that video
>>    quality will degrade, which brings me to the next point.
>>    2. Provide a negotiation mechanism which would allow peers to
>>    "upgrade" to a superior (optionally-implemented) video codec.
>>
>> Negotiation has always been part of the design. The sole question is which
> codec is mandatory to implement, not which is the sole codec to be
> mandated.
>
> -Ekr
>
>
>     This will allow us to support VP8, VP9, H264, H265 or whatever other
>> codec people like without the fear of transcoding or IPR. I believe that in
>> most cases negotiation will succeed in upgrading to a superior codec. It
>> will also encourage (as opposed to force) vendors to support each other's
>> codecs, which is the right way to go in light of the political nature of
>> this decision.
>>
>> Gili
>>
>> 1. If you support H.264 as the mandatory to implement codec or are
>> willing to live with it as the MTI, please raise your hand now.
>>
>> 2. If you support VP8 as the mandatory to implement codec or are
>> willing to live with it as the MTI, please raise your hand now.
>>
>>
>> Gili
>>
>>
>> On 13/09/2013 12:52 PM, Ted Hardie wrote:
>>
>>  WG,
>>
>> The chairs have created a plan for how to perform the Video Codec
>> selection in our WG. The chairs are asking for review of our plan on
>> how to undertake the mandatory-to-implement video codec selection.
>> We'd much prefer to have comments on the mechanics before they begin,
>> so please review now.  Proponents of a particular proposal should
>> note both the actions required and the timelines proposed.
>>
>> The main goal of this plan is to hold a consensus call on which of
>> the proposed alternatives we as a WG should select at one of the WG
>> sessions in Vancouver. Such a consensus call will of course be
>> verified on the mailing list for anyone who can't participate. The
>> chairs will recuse themselves from judging this particular
>> consensus.
>>
>> In the WG session each codec proposal will be allowed an equal amount
>> of time to highlight the arguments for their proposal. After that a
>> there will be a slot for discussion and clarifying questions.
>>
>> To enable the WG participants to get answers to any questions, the
>> proposals in draft form and any supporting material MUST be made
>> available by 6th of October. This is to ensure that the WG
>> participants can verify or object to any claims or statements in
>> the proposal material prior to the WG session. We chairs would really
>> not like to see the proponents bring up new arguments at their
>> presentation. Also the WG participants are expected to raise any
>> arguments on the list ahead of time to enable the proponents to
>> respond to such arguments.
>>
>> The proposed consensus questions will be of the following form:
>>
>> 1. If you support H.264 as the mandatory to implement codec or are
>> willing to live with it as the MTI, please raise your hand now.
>>
>> 2. If you support VP8 as the mandatory to implement codec or are
>> willing to live with it as the MTI, please raise your hand now.
>>
>> You may indicate support on both questions and we encourage you to do
>> so if you can live with either, even if you have a preference for one
>> over the other.
>>
>> Additional proposals than the previous ones are welcome, but must be
>> submitted as draft and their proponents must notify the chairs no later
>> than the 6th of October that they also have a candidate proposal.
>>
>> In case the WG fails to reach consensus we chairs propose that we use
>> the alternative decision process as discussed in RFC3929. The method
>> and its usage will be discussed on the list should the WG not
>> establish consensus on a proposal for mandatory to implement video codec.
>>
>>  regards,
>>
>>  Magnus,  Cullen, and Ted
>>
>>
>> _______________________________________________
>> rtcweb mailing listrtcweb@ietf.orghttps://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--001a11336bba7239ba04e6a45f88
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Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Anyone willing to create a draft for H.261?<br></div><div =
class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On 18 September 20=
13 02:13, Eric Rescorla <span dir=3D"ltr">&lt;<a href=3D"mailto:ekr@rtfm.co=
m" target=3D"_blank">ekr@rtfm.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr"><br><div class=3D"gmail_ext=
ra"><br><br><div class=3D"gmail_quote"><div class=3D"im">On Tue, Sep 17, 20=
13 at 11:54 AM, cowwoc <span dir=3D"ltr">&lt;<a href=3D"mailto:cowwoc@bbs.d=
arktech.org" target=3D"_blank">cowwoc@bbs.darktech.org</a>&gt;</span> wrote=
:<br>


</div><div class=3D"im"><blockquote class=3D"gmail_quote" style=3D"margin:0=
 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
 =20
   =20
 =20
  <div text=3D"#000000" bgcolor=3D"#FFFFFF">
    <div>Hi Ted,<br>
      <br>
      =A0=A0=A0 Seeing as this discussion stems from licensing concerns, I
      like to propose the following alternative:<br>
      <ol>
        <li>Mandate a video codec whose IPR has expired. I agree that
          video quality will degrade, which brings me to the next point.</l=
i>
        <li>Provide a negotiation mechanism which would allow peers to
          &quot;upgrade&quot; to a superior (optionally-implemented) video =
codec.</li></ol></div></div></blockquote></div><div>Negotiation has always =
been part of the design. The sole question is which</div><div>codec is mand=
atory to implement, not which is the sole codec to be mandated.</div>


<div><br></div><div>-Ekr</div><div><div class=3D"h5"><div><br></div><div><b=
r></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border=
-left:1px #ccc solid;padding-left:1ex"><div text=3D"#000000" bgcolor=3D"#FF=
FFFF">
<div><p>=A0 =A0 This will allow us to support VP8, VP9, H264, H265 or
        whatever other codec people like without the fear of transcoding
        or IPR. I believe that in most cases negotiation will succeed in
        upgrading to a superior codec. It will also encourage (as
        opposed to force) vendors to support each other&#39;s codecs, which
        is the right way to go in light of the political nature of this
        decision.<br>
      </p>
      <p>Gili<br>
      </p><div>
      <br>
      1. If you support H.264 as the mandatory to implement codec or are<br=
>
      willing to live with it as the MTI, please raise your hand now.<br>
      <br>
      2. If you support VP8 as the mandatory to implement codec or are<br>
      willing to live with it as the MTI, please raise your hand now.<br>
      <br>
      <br></div>
      Gili<div><div><br>
      <br>
      On 13/09/2013 12:52 PM, Ted Hardie wrote:<br>
    </div></div></div>
    <blockquote type=3D"cite"><div><div>
      <div dir=3D"ltr">
        <div>WG,<br>
          <br>
          The chairs have created a plan for how to perform the Video
          Codec<br>
          selection in our WG. The chairs are asking for review of our
          plan on<br>
          how to undertake the mandatory-to-implement video codec
          selection.<br>
          We&#39;d much prefer to have comments on the mechanics before the=
y
          begin,<br>
          so please review now.=A0 Proponents of a particular proposal
          should<br>
          note both the actions required and the timelines proposed.<br>
          <br>
          The main goal of this plan is to hold a consensus call on
          which of<br>
          the proposed alternatives we as a WG should select at one of
          the WG<br>
          sessions in Vancouver. Such a consensus call will of course be<br=
>
          verified on the mailing list for anyone who can&#39;t participate=
.
          The<br>
          chairs will recuse themselves from judging this particular<br>
          consensus.<br>
          <br>
          In the WG session each codec proposal will be allowed an equal
          amount<br>
          of time to highlight the arguments for their proposal. After
          that a<br>
          there will be a slot for discussion and clarifying questions.<br>
          <br>
          To enable the WG participants to get answers to any questions,
          the<br>
          proposals in draft form and any supporting material MUST be
          made<br>
          available by 6th of October. This is to ensure that the WG<br>
          participants can verify or object to any claims or statements
          in<br>
          the proposal material prior to the WG session. We chairs would
          really<br>
          not like to see the proponents bring up new arguments at their<br=
>
          presentation. Also the WG participants are expected to raise
          any<br>
          arguments on the list ahead of time to enable the proponents
          to<br>
          respond to such arguments.<br>
          <br>
          The proposed consensus questions will be of the following
          form:<br>
          <br>
          1. If you support H.264 as the mandatory to implement codec or
          are<br>
          willing to live with it as the MTI, please raise your hand
          now.<br>
          <br>
          2. If you support VP8 as the mandatory to implement codec or
          are<br>
          willing to live with it as the MTI, please raise your hand
          now.<br>
          <br>
          You may indicate support on both questions and we encourage
          you to do<br>
          so if you can live with either, even if you have a preference
          for one<br>
          over the other.<br>
          <br>
          Additional proposals than the previous ones are welcome, but
          must be<br>
          submitted as draft and their proponents must notify the chairs
          no later<br>
          than the 6th of October that they also have a candidate
          proposal.<br>
          <br>
          In case the WG fails to reach consensus we chairs propose that
          we use<br>
          the alternative decision process as discussed in RFC3929. The
          method<br>
          and its usage will be discussed on the list should the WG not<br>
          establish consensus on a proposal for mandatory to implement
          video codec.<br>
          <br>
        </div>
        <div>regards,<br>
          <br>
        </div>
        Magnus,=A0 Cullen, and Ted<br>
      </div>
      <br>
      <fieldset></fieldset>
      <br>
      </div></div><div><pre>_______________________________________________
rtcweb mailing list
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </div></blockquote>
    <br>
  </div>

<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div></div></div><br></div></div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

--001a11336bba7239ba04e6a45f88--

From xiphmont@gmail.com  Wed Sep 18 01:50:03 2013
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From: Monty Montgomery <xiphmont@gmail.com>
To: Leon Geyser <lgeyser@gmail.com>
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Subject: Re: [rtcweb] Video Codec Selection Plan
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I'll see your h.261 and raise one Theora.  At least it has one hell of
an encoder.

Cheers,
Monty

On Wed, Sep 18, 2013 at 4:39 AM, Leon Geyser <lgeyser@gmail.com> wrote:
> Anyone willing to create a draft for H.261?
>
>
> On 18 September 2013 02:13, Eric Rescorla <ekr@rtfm.com> wrote:
>>
>>
>>
>>
>> On Tue, Sep 17, 2013 at 11:54 AM, cowwoc <cowwoc@bbs.darktech.org> wrote:
>>>
>>> Hi Ted,
>>>
>>>     Seeing as this discussion stems from licensing concerns, I like to
>>> propose the following alternative:
>>>
>>> Mandate a video codec whose IPR has expired. I agree that video quality
>>> will degrade, which brings me to the next point.
>>> Provide a negotiation mechanism which would allow peers to "upgrade" to a
>>> superior (optionally-implemented) video codec.
>>
>> Negotiation has always been part of the design. The sole question is which
>> codec is mandatory to implement, not which is the sole codec to be
>> mandated.
>>
>> -Ekr
>>
>>
>>>     This will allow us to support VP8, VP9, H264, H265 or whatever other
>>> codec people like without the fear of transcoding or IPR. I believe that in
>>> most cases negotiation will succeed in upgrading to a superior codec. It
>>> will also encourage (as opposed to force) vendors to support each other's
>>> codecs, which is the right way to go in light of the political nature of
>>> this decision.
>>>
>>> Gili
>>>
>>>
>>> 1. If you support H.264 as the mandatory to implement codec or are
>>> willing to live with it as the MTI, please raise your hand now.
>>>
>>> 2. If you support VP8 as the mandatory to implement codec or are
>>> willing to live with it as the MTI, please raise your hand now.
>>>
>>>
>>> Gili
>>>
>>>
>>> On 13/09/2013 12:52 PM, Ted Hardie wrote:
>>>
>>> WG,
>>>
>>> The chairs have created a plan for how to perform the Video Codec
>>> selection in our WG. The chairs are asking for review of our plan on
>>> how to undertake the mandatory-to-implement video codec selection.
>>> We'd much prefer to have comments on the mechanics before they begin,
>>> so please review now.  Proponents of a particular proposal should
>>> note both the actions required and the timelines proposed.
>>>
>>> The main goal of this plan is to hold a consensus call on which of
>>> the proposed alternatives we as a WG should select at one of the WG
>>> sessions in Vancouver. Such a consensus call will of course be
>>> verified on the mailing list for anyone who can't participate. The
>>> chairs will recuse themselves from judging this particular
>>> consensus.
>>>
>>> In the WG session each codec proposal will be allowed an equal amount
>>> of time to highlight the arguments for their proposal. After that a
>>> there will be a slot for discussion and clarifying questions.
>>>
>>> To enable the WG participants to get answers to any questions, the
>>> proposals in draft form and any supporting material MUST be made
>>> available by 6th of October. This is to ensure that the WG
>>> participants can verify or object to any claims or statements in
>>> the proposal material prior to the WG session. We chairs would really
>>> not like to see the proponents bring up new arguments at their
>>> presentation. Also the WG participants are expected to raise any
>>> arguments on the list ahead of time to enable the proponents to
>>> respond to such arguments.
>>>
>>> The proposed consensus questions will be of the following form:
>>>
>>> 1. If you support H.264 as the mandatory to implement codec or are
>>> willing to live with it as the MTI, please raise your hand now.
>>>
>>> 2. If you support VP8 as the mandatory to implement codec or are
>>> willing to live with it as the MTI, please raise your hand now.
>>>
>>> You may indicate support on both questions and we encourage you to do
>>> so if you can live with either, even if you have a preference for one
>>> over the other.
>>>
>>> Additional proposals than the previous ones are welcome, but must be
>>> submitted as draft and their proponents must notify the chairs no later
>>> than the 6th of October that they also have a candidate proposal.
>>>
>>> In case the WG fails to reach consensus we chairs propose that we use
>>> the alternative decision process as discussed in RFC3929. The method
>>> and its usage will be discussed on the list should the WG not
>>> establish consensus on a proposal for mandatory to implement video codec.
>>>
>>> regards,
>>>
>>> Magnus,  Cullen, and Ted
>>>
>>>
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>
>>>
>>>
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

From christer.holmberg@ericsson.com  Wed Sep 18 03:50:38 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Justin Uberti <juberti@google.com>, Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: [rtcweb] JSEP Walk Through on Wednesday the 18th of September at 17.00 CEST (8 AM PDT)
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From fippo@goodadvice.pages.de  Wed Sep 18 04:46:06 2013
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Subject: Re: [rtcweb] JSEP Walk Through on Wednesday the 18th of September at 17.00 CEST (8 AM PDT)
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On Tue, 17 Sep 2013, Justin Uberti wrote:
> The draft has now been posted atÂ http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-jsep-04.txt, with 13 new pages of text.
> Readers should focus on Section 5, which specifies the behavior of createOffer/createAnswer, as well as Appendix A.2, which provides examples of the generated SDP.

I have a question about the OfferToReceiveAudio / OfferToReceiveVideo
sections (5.2.3.1& 5.2.3.2)...

In createAnswer (I'd note that both section currently only talk about the
offerer), what is the expected behaviour when those constraints
are set to false and a MediaStreamTrack of the respective type has been
attached to the peerconnection?

Is the behaviour different when there is no MST of a certain type attached?

Does this only affect the direction attribute or also whether the m-line 
is rejected by setting the port to zero?
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To: Justin Uberti <juberti@google.com>
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--001a11c2b01c4857ef04e6a77ad1
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I think we should include RFC5104 referenced in the sections corresponding
to RTCP feedback

Thanks
Suhas

On Tuesday, September 17, 2013, Justin Uberti wrote:

> The draft has now been posted at
> http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-jsep-04.txt, with
> 13 new pages of text.
>
> Readers should focus on Section 5, which specifies the behavior of
> createOffer/createAnswer, as well as Appendix A.2, which provides examples
> of the generated SDP.
>
>
> On Tue, Sep 17, 2013 at 1:51 AM, Magnus Westerlund <
> magnus.westerlund@ericsson.com> wrote:
>
> On 2013-09-17 10:47, Harald Alvestrand wrote:
> > Magnus,
> >
> > what is the expected duration of the call?
>
> 2 hour maximum
>
> Magnus
>
> >
> > On 09/17/2013 10:38 AM, Magnus Westerlund wrote:
> >> WG,
> >>
> >> I know that the new JSEP Draft is not yet submitted. I am hoping that it
> >> will show up during the morning hours for the US based people. That way
> >> everyone gets at least some few hours to look at the draft prior to the
> >> call. If it hasn't we will postpone the call, but I really hope we can
> >> avoid this. I will send a notice to the WG if we have to postpone it.
> >>
> >> Cheers
> >>
> >> Magnus
> >>
> >> On 2013-09-11 07:58, Magnus Westerlund wrote:
> >>> WG,
> >>>
> >>> There will be a new version of JSEP draft coming out Monday. As a
> follow
> >>> up to get as much progress on this topic as possible. The chairs and
> >>> authors are offering a informal walk through session on Wednesday the
> >>> 18th. People will be strongly recommended to have reviewed the new
> >>> draft.
> >>>
> >>> The intention of the session is enable the authors to inform you about
> >>> the latest changes, any issues to consider and what needs further
> >>> discussion. The will also enable you as participant after your review
> of
> >>> the document to ask clarifying questions or for motivations behind
> >>> choices, and raise issues you have spotted.
> >>>
> >>> The session is not intended to solve issues or judge consensus on part
> >>> of the draft. Discussions of what is the best solution etc, will to be
> >>> cut off.
> >>>
> >>> This will be a webex conference per details below.
> >>>
> >>> Cheers
> >>>
> >>> Magnus Westerlund
> >>>
> >>> Topic: RTCWeb - JSEP
> >>> Date: Wednesday, September 18, 2013
> >>> Time: 8:00 am, Pacific Daylight Time (San Francisco, GMT-07:00)
> >>> Meeting Number: 302 061 238
> >>> Meeting Password: ietf
> >>>
> >>> -------------------------------------------------------
> >>> To join the online meeting (Now from mobile devices!)
> >>> -------------------------------------------------------
> >>> 1. Go to
> >>>
> https://cisco.webex.com/ciscosales/j.php?ED=206400898&UID=0&PW=NYjk4NmE0ODBj&RT=MiM0
> >>>
> >>>
> >>> 2. Enter your name and email address.
> >>> 3. Enter the meeting password: ietf
> >>> 4. Click "Join Now".
> >>>
> >>> To view in other time zones or languages, please click the link:
> >>>
> https://cisco.webex.com/ciscosales/j.php?ED=206400898&UID=0&PW=NYjk4NmE0ODBj&ORT=MiM0
> >>>
> >>>
> >>>
> >>> -------------------------------------------------------
> >>> To join the teleconference only
> >>> -------------------------------------------------------
> >>> 1. Dial into Cisco WebEx (view all Global Access Numbers at
> >>> http://cisco.com/en/US/about/doing_business/conferencing/index.html
> >>> 2. Follow the prompts to enter the Meeting Number (listed above) or
> >>> Access Code followed by the # sign.
> >>>
> >>> San Jose, CA: +1.408.525.6800 RTP: +1.919.392.3330
> >>>
> >>> US/Canada: +1.866.432.9903 United Kingdom: +44.20.8824.0117
> >>>
> >>> India: +91
>
>

--001a11c2b01c4857ef04e6a77ad1
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

I think we should include RFC5104 referenced in the sections corresponding =
to RTCP feedback<div><br></div><div>Thanks</div><div>Suhas=A0<span></span><=
br><br>On Tuesday, September 17, 2013, Justin Uberti  wrote:<br><blockquote=
 class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc soli=
d;padding-left:1ex">
<div dir=3D"ltr">The draft has now been posted at=A0<a href=3D"http://www.i=
etf.org/internet-drafts/draft-ietf-rtcweb-jsep-04.txt" target=3D"_blank">ht=
tp://www.ietf.org/internet-drafts/draft-ietf-rtcweb-jsep-04.txt</a>, with 1=
3 new pages of text.<div>


<br></div><div>Readers should focus on Section 5, which specifies the behav=
ior of createOffer/createAnswer, as well as Appendix A.2, which provides ex=
amples of the generated SDP.</div></div><div><br><br>

<div>On Tue, Sep 17, 2013 at 1:51 AM, Magnus Westerlund <span dir=3D"ltr">&=
lt;<a>magnus.westerlund@ericsson.com</a>&gt;</span> wrote:<br><blockquote s=
tyle=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

<div>On 2013-09-17 10:47, Harald Alvestrand wrote:<br>
&gt; Magnus,<br>
&gt;<br>
&gt; what is the expected duration of the call?<br>
<br>
</div>2 hour maximum<br>
<span><font color=3D"#888888"><br>
Magnus<br>
</font></span><div><div><br>
&gt;<br>
&gt; On 09/17/2013 10:38 AM, Magnus Westerlund wrote:<br>
&gt;&gt; WG,<br>
&gt;&gt;<br>
&gt;&gt; I know that the new JSEP Draft is not yet submitted. I am hoping t=
hat it<br>
&gt;&gt; will show up during the morning hours for the US based people. Tha=
t way<br>
&gt;&gt; everyone gets at least some few hours to look at the draft prior t=
o the<br>
&gt;&gt; call. If it hasn&#39;t we will postpone the call, but I really hop=
e we can<br>
&gt;&gt; avoid this. I will send a notice to the WG if we have to postpone =
it.<br>
&gt;&gt;<br>
&gt;&gt; Cheers<br>
&gt;&gt;<br>
&gt;&gt; Magnus<br>
&gt;&gt;<br>
&gt;&gt; On 2013-09-11 07:58, Magnus Westerlund wrote:<br>
&gt;&gt;&gt; WG,<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; There will be a new version of JSEP draft coming out Monday. A=
s a follow<br>
&gt;&gt;&gt; up to get as much progress on this topic as possible. The chai=
rs and<br>
&gt;&gt;&gt; authors are offering a informal walk through session on Wednes=
day the<br>
&gt;&gt;&gt; 18th. People will be strongly recommended to have reviewed the=
 new<br>
&gt;&gt;&gt; draft.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; The intention of the session is enable the authors to inform y=
ou about<br>
&gt;&gt;&gt; the latest changes, any issues to consider and what needs furt=
her<br>
&gt;&gt;&gt; discussion. The will also enable you as participant after your=
 review of<br>
&gt;&gt;&gt; the document to ask clarifying questions or for motivations be=
hind<br>
&gt;&gt;&gt; choices, and raise issues you have spotted.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; The session is not intended to solve issues or judge consensus=
 on part<br>
&gt;&gt;&gt; of the draft. Discussions of what is the best solution etc, wi=
ll to be<br>
&gt;&gt;&gt; cut off.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; This will be a webex conference per details below.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Cheers<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Magnus Westerlund<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Topic: RTCWeb - JSEP<br>
&gt;&gt;&gt; Date: Wednesday, September 18, 2013<br>
&gt;&gt;&gt; Time: 8:00 am, Pacific Daylight Time (San Francisco, GMT-07:00=
)<br>
&gt;&gt;&gt; Meeting Number: 302 061 238<br>
&gt;&gt;&gt; Meeting Password: ietf<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; -------------------------------------------------------<br>
&gt;&gt;&gt; To join the online meeting (Now from mobile devices!)<br>
&gt;&gt;&gt; -------------------------------------------------------<br>
&gt;&gt;&gt; 1. Go to<br>
&gt;&gt;&gt; <a href=3D"https://cisco.webex.com/ciscosales/j.php?ED=3D20640=
0898&amp;UID=3D0&amp;PW=3DNYjk4NmE0ODBj&amp;RT=3DMiM0" target=3D"_blank">ht=
tps://cisco.webex.com/ciscosales/j.php?ED=3D206400898&amp;UID=3D0&amp;PW=3D=
NYjk4NmE0ODBj&amp;RT=3DMiM0</a><br>



&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; 2. Enter your name and email address.<br>
&gt;&gt;&gt; 3. Enter the meeting password: ietf<br>
&gt;&gt;&gt; 4. Click &quot;Join Now&quot;.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; To view in other time zones or languages, please click the lin=
k:<br>
&gt;&gt;&gt; <a href=3D"https://cisco.webex.com/ciscosales/j.php?ED=3D20640=
0898&amp;UID=3D0&amp;PW=3DNYjk4NmE0ODBj&amp;ORT=3DMiM0" target=3D"_blank">h=
ttps://cisco.webex.com/ciscosales/j.php?ED=3D206400898&amp;UID=3D0&amp;PW=
=3DNYjk4NmE0ODBj&amp;ORT=3DMiM0</a><br>



&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; -------------------------------------------------------<br>
&gt;&gt;&gt; To join the teleconference only<br>
&gt;&gt;&gt; -------------------------------------------------------<br>
&gt;&gt;&gt; 1. Dial into Cisco WebEx (view all Global Access Numbers at<br=
>
&gt;&gt;&gt; <a href=3D"http://cisco.com/en/US/about/doing_business/confere=
ncing/index.html" target=3D"_blank">http://cisco.com/en/US/about/doing_busi=
ness/conferencing/index.html</a><br>
&gt;&gt;&gt; 2. Follow the prompts to enter the Meeting Number (listed abov=
e) or<br>
&gt;&gt;&gt; Access Code followed by the # sign.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; San Jose, CA: <a value=3D"+14085256800">+1.408.525.6800</a> RT=
P: <a value=3D"+19193923330">+1.919.392.3330</a><br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; US/Canada: <a value=3D"+18664329903">+1.866.432.9903</a> Unite=
d Kingdom: <a value=3D"+442088240117">+44.20.8824.0117</a><br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; India: <a value=3D"+918043501111">+91</a></div></div></blockqu=
ote></div></div></blockquote></div>

--001a11c2b01c4857ef04e6a77ad1--

From magnus.westerlund@ericsson.com  Wed Sep 18 05:46:46 2013
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Subject: Re: [rtcweb] JSEP Walk Through on Wednesday the 18th of September at 17.00 CEST (8 AM PDT)
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On 2013-09-18 12:50, Christer Holmberg wrote:
> Hi,
> 
>  
> 
> I may not be able to attend the phone meeting today, so I send my input
> by the e-mail.
> 
>  
> 
> As requested by the WG chairs, Iâ€™ll focus on *clarification
> issues/questions*, and will for now not repeat the issues I have with
> pranswer and forking (also, we may have found a way forward to solve that).

A point of clarification.

In the phone conference we will focus getting understanding for JSEP for
the participants so that you all can provide comments, feedback and
bring proposals for improvements.

It is also okay to ask about technical issues. But in the interest of
time I think we need to focus on exposing and understanding potential
issues, not solving them.

All type of issues are fine to contribute on the list and towards the
authors. However, I would recommend to use another thread than this one.
Create a new thread or follow up on the I-D announcement for the version
you are commenting on.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
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Ericsson AB                | Phone  +46 10 7148287
FÃ¤rÃ¶gatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
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From christer.holmberg@ericsson.com  Wed Sep 18 05:54:47 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: [rtcweb] JSEP Walk Through on Wednesday the 18th of September at 17.00 CEST (8 AM PDT)
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: how to control which track plays in a video tag
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Subject: [rtcweb] how to control which track plays in a video tag
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The issue got raised if there is multiple video tracks in a media stream, h=
ow do you control which video track gets played in a video stream =

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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: Constraints on a track in a a given PC 
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It would be nice to have a way specify constraints to a given track on a pe=
er connection. For example, like to be able specify if a track should be bu=
ndle only or not.=20



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--f46d04389087f5095904e6aaf47b
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+1


On Wed, Sep 18, 2013 at 9:04 AM, Cullen Jennings (fluffy)
<fluffy@cisco.com>wrote:

>
> It would be nice to have a way specify constraints to a given track on a
> peer connection. For example, like to be able specify if a track should be
> bundle only or not.
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr">+1=A0</div><div class=3D"gmail_extra"><br><br><div class=
=3D"gmail_quote">On Wed, Sep 18, 2013 at 9:04 AM, Cullen Jennings (fluffy) =
<span dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@cisco.com" target=3D"_blank"=
>fluffy@cisco.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><br>
It would be nice to have a way specify constraints to a given track on a pe=
er connection. For example, like to be able specify if a track should be bu=
ndle only or not.<br>
<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div>

--f46d04389087f5095904e6aaf47b--

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To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
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Subject: Re: [rtcweb] how to control which track plays in a video tag
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Shouldn't this be dealt with a W3C API, say videoelem.src =
mediastream.trackAt(0)


On Wed, Sep 18, 2013 at 9:01 AM, Cullen Jennings (fluffy)
<fluffy@cisco.com>wrote:

>
> The issue got raised if there is multiple video tracks in a media stream,
> how do you control which video track gets played in a video stream
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr">Shouldn&#39;t this be dealt with a W3C API, say videoelem.=
src =3D mediastream.trackAt(0)=A0</div><div class=3D"gmail_extra"><br><br><=
div class=3D"gmail_quote">On Wed, Sep 18, 2013 at 9:01 AM, Cullen Jennings =
(fluffy) <span dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@cisco.com" target=
=3D"_blank">fluffy@cisco.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><br>
The issue got raised if there is multiple video tracks in a media stream, h=
ow do you control which video track gets played in a video stream<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div>

--047d7bfced9c6dbb8204e6aafb92--

From suhasietf@gmail.com  Wed Sep 18 09:44:05 2013
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To: "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: [rtcweb] m=section recycling and a=mid
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After thinking about it a bit, I think there are some side effects
associated with changing/not changing mid attribute of a re-cycled m=
section and it being part of a BUNDLE group or not..

example

say , we have these video m=section part of BUNDLE group as below
a=group:BUNDLE 1, 2

m=video .....
a=mid:1
a=inactive

m=video .....
a=mid:2
a=sendrecv

Now If JS user adds a new video track and he wants it to be out of BUNDLE.
If we recycle first video m=section and dont change mid attribute,  we end
up BUNDLing even if the user dint want to

On the other hand, if we allowed mid value to be changed, we might end up
UNBundling a video stream , when the user did want to BUNDLE them

So allowing a mid value to be changed or not should be tied to what the JS
API is asking on behalf of the user


Any thoughts ?

Cheers
Suhas

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<div dir=3D"ltr">After thinking about it a bit, I think there are some side=
 effects associated with changing/not changing mid attribute of a re-cycled=
 m=3D section and it being part of a BUNDLE group or not..<div><br></div><d=
iv>
example</div><div><br></div><div>say , we have these video m=3Dsection part=
 of BUNDLE group as below=A0</div><div>a=3Dgroup:BUNDLE 1, 2</div><div><br>=
</div><div>m=3Dvideo .....</div><div>a=3Dmid:1=A0</div><div>a=3Dinactive</d=
iv><div><br>
</div><div><div>m=3Dvideo .....</div><div>a=3Dmid:2=A0</div><div>a=3Dsendre=
cv</div><div><br></div><div>Now If JS user adds a new video track and he wa=
nts it to be out of BUNDLE. If we recycle first video m=3Dsection and dont =
change mid attribute, =A0we end up BUNDLing even if the user dint want to</=
div>
<div><br></div><div>On the other hand, if we allowed mid value to be change=
d, we might end up UNBundling a video stream , when the user did want to BU=
NDLE them</div><div><br></div><div>So allowing a mid value to be changed or=
 not should be tied to what the JS API is asking on behalf of the user</div=
>
</div><div><br></div><div><br></div><div>Any thoughts ?</div><div><br></div=
><div>Cheers</div><div>Suhas</div><div><br></div><div><br></div></div>

--047d7bfced9c2a08e704e6ab25a3--

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Subject: Re: [rtcweb] how to control which track plays in a video tag
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On Wed, Sep 18, 2013 at 9:01 AM, Cullen Jennings (fluffy)
<fluffy@cisco.com>wrote:
>
> The issue got raised if there is multiple video tracks in a media stream,
> how do you control which video track gets played in a video stream
>

Set the selected<http://www.whatwg.org/specs/web-apps/current-work/multipage/the-video-element.html#dom-videotrack-selected>attribute
on a
VideoTrack<http://www.whatwg.org/specs/web-apps/current-work/multipage/the-video-element.html#audiotracklist-and-videotracklist-objects>from
HTMLMediaElement.videoTracks<http://www.whatwg.org/specs/web-apps/current-work/multipage/the-video-element.html#media-resources-with-multiple-media-tracks>
?
(though I don't know that it's implemented in any shipping browser today;
e.g. crbug.com/249427)

Cheers,
-a

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On W=
ed, Sep 18, 2013 at 9:01 AM, Cullen Jennings (fluffy) <span dir=3D"ltr">&lt=
;<a href=3D"mailto:fluffy@cisco.com" target=3D"_blank" class=3D"cremed">flu=
ffy@cisco.com</a>&gt;</span> wrote:<blockquote class=3D"gmail_quote" style=
=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(20=
4,204,204);border-left-style:solid;padding-left:1ex">

The issue got raised if there is multiple video tracks in a media stream, h=
ow do you control which video track gets played in a video stream<br></bloc=
kquote><div><br></div>Set the <a href=3D"http://www.whatwg.org/specs/web-ap=
ps/current-work/multipage/the-video-element.html#dom-videotrack-selected" c=
lass=3D"cremed">selected</a> attribute on a <a href=3D"http://www.whatwg.or=
g/specs/web-apps/current-work/multipage/the-video-element.html#audiotrackli=
st-and-videotracklist-objects" class=3D"cremed">VideoTrack</a> from=A0<a hr=
ef=3D"http://www.whatwg.org/specs/web-apps/current-work/multipage/the-video=
-element.html#media-resources-with-multiple-media-tracks" class=3D"cremed">=
HTMLMediaElement.videoTracks</a>?<div>
(though I don&#39;t know that it&#39;s implemented in any shipping browser =
today; e.g.=A0<a href=3D"http://crbug.com/249427" class=3D"cremed">crbug.co=
m/249427</a>)</div><div><br></div><div>Cheers,</div><div>-a</div></div></di=
v>
</div>

--001a11c381222964dc04e6ab477f--

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>=20
>=20
> Your recording is now available on the WebEx service site. Click the =
link below to play it:=20
>=20
> =
https://cisco.webex.com/ciscosales/lsr.php?AT=3Dpb&SP=3DMC&rID=3D71503827&=
rKey=3D2cb05592aa7be2c2=20
>=20
> RTCWeb - JSEP-20130918 1508-1=20
> Wednesday, September 18, 2013 8:08 am San Francisco Time=20
> 1 Hour 49 Minutes=20
>=20



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<html><head><meta http-equiv="Content-Type" content="text/html charset=us-ascii"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><br><div><blockquote type="cite"><font face="Tahoma, Arial, sans-serif, Helvetica, Geneva" size="2"><br> <br> Your recording is now available on the WebEx service site. Click the link below to play it: <br> <br> <a href="https://cisco.webex.com/ciscosales/lsr.php?AT=pb&amp;SP=MC&amp;rID=71503827&amp;rKey=2cb05592aa7be2c2" target="_blank">https://cisco.webex.com/ciscosales/lsr.php?AT=pb&amp;SP=MC&amp;rID=71503827&amp;rKey=2cb05592aa7be2c2</a> <br> <br> RTCWeb - JSEP-20130918 1508-1 <br> Wednesday, September 18, 2013 8:08 am San Francisco Time <br> 1 Hour 49 Minutes <br> <br></font></blockquote><div><br></div></div><br></body></html>
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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Cullen Jennings \(fluffy\)'" <fluffy@cisco.com>, <juberti@google.com>
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Cc: rtcweb@ietf.org
Subject: [rtcweb] Missing Offer event handling in Pranswer state [was RE: I-D Action: draft-ietf-rtcweb-jsep-04.txt]
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Hi Cullen/Justin,

Unfortunately, I could not attend today's JSEP Walk through meeting. I had
gone through the draft to understand Offer event handling in Pranswer state
which I raised long back
(http://www.ietf.org/mail-archive/web/rtcweb/current/msg08589.html) and no
resolution in the updated draft as well.

Currently in JSEP state machine, OFFER event is not possible to be handled
at Local-Pranswer, Remote-Pranswer state. OFFER event is required in
local-pranswer for multiple offer/answer handling like BUNDLE, ICE
negotiations. The example callflow for early-dialog with multiple
offer/answer handling (parallel forking) is available at Sec 5.2 of
draft-partha-rtcweb-jsep-sip-00. It is open item in JSEP. Please let me know
your opinion on the same.

Thanks
Partha

> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of internet-drafts@ietf.org
> Sent: Wednesday, September 18, 2013 3:53 AM
> To: i-d-announce@ietf.org
> Cc: rtcweb@ietf.org
> Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-jsep-04.txt
> 
> 
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>  This draft is a work item of the Real-Time Communication in WEB-
> browsers Working Group of the IETF.
> 
> 	Title           : Javascript Session Establishment Protocol
> 	Author(s)       : Justin Uberti
>                           Cullen Jennings
> 	Filename        : draft-ietf-rtcweb-jsep-04.txt
> 	Pages           : 46
> 	Date            : 2013-09-17
> 
> Abstract:
>    This document describes the mechanisms for allowing a Javascript
>    application to control the signaling plane of a multimedia session
>    via the interface specified in the W3C RTCPeerConnection API, and
>    discusses how this relates to existing signaling protocols.
> 
> 
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-jsep
> 
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-rtcweb-jsep-04
> 
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-jsep-04
> 
> 
> Please note that it may take a couple of minutes from the time of
> submission
> until the htmlized version and diff are available at tools.ietf.org.
> 
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From basilgohar@librevideo.org  Wed Sep 18 11:32:23 2013
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Date: Wed, 18 Sep 2013 14:32:13 -0400
From: Basil Mohamed Gohar <basilgohar@librevideo.org>
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Subject: Re: [rtcweb] Video Codec Selection Plan
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I was thinking if we're going to go with an older, non-IPR-encumbered
codec, Theora is the obvious choice, because it's been around about as
long as anything currently in use and has never had it's IPR seriously
challenged.

The encoder is being actively developed and has long since surpassed
pretty-much everything aside from H.264 and VP8 that are in common use
nowadays.

The fact that it takes much less CPU time to decode compared with other
modern codecs is also a plus.

Theora is also used in Elphel camera devices, so it's been used in
hardware as well (granted, those are not "burned-in-silicon" devices).

Efficiency wise, for talking-head scenarios (a common use case), I think
Theora will likely do quite well.

On 09/18/2013 04:49 AM, Monty Montgomery wrote:
> I'll see your h.261 and raise one Theora.  At least it has one hell of
> an encoder.
> 
> Cheers,
> Monty
> 
> On Wed, Sep 18, 2013 at 4:39 AM, Leon Geyser <lgeyser@gmail.com> wrote:
>> Anyone willing to create a draft for H.261?
>>
>>
>> On 18 September 2013 02:13, Eric Rescorla <ekr@rtfm.com> wrote:
>>>
>>>
>>>
>>>
>>> On Tue, Sep 17, 2013 at 11:54 AM, cowwoc <cowwoc@bbs.darktech.org> wrote:
>>>>
>>>> Hi Ted,
>>>>
>>>>     Seeing as this discussion stems from licensing concerns, I like to
>>>> propose the following alternative:
>>>>
>>>> Mandate a video codec whose IPR has expired. I agree that video quality
>>>> will degrade, which brings me to the next point.
>>>> Provide a negotiation mechanism which would allow peers to "upgrade" to a
>>>> superior (optionally-implemented) video codec.
>>>
>>> Negotiation has always been part of the design. The sole question is which
>>> codec is mandatory to implement, not which is the sole codec to be
>>> mandated.
>>>
>>> -Ekr
>>>
>>>
>>>>     This will allow us to support VP8, VP9, H264, H265 or whatever other
>>>> codec people like without the fear of transcoding or IPR. I believe that in
>>>> most cases negotiation will succeed in upgrading to a superior codec. It
>>>> will also encourage (as opposed to force) vendors to support each other's
>>>> codecs, which is the right way to go in light of the political nature of
>>>> this decision.
>>>>
>>>> Gili
>>>>
>>>>
>>>> 1. If you support H.264 as the mandatory to implement codec or are
>>>> willing to live with it as the MTI, please raise your hand now.
>>>>
>>>> 2. If you support VP8 as the mandatory to implement codec or are
>>>> willing to live with it as the MTI, please raise your hand now.
>>>>
>>>>
>>>> Gili
>>>>
>>>>
>>>> On 13/09/2013 12:52 PM, Ted Hardie wrote:
>>>>
>>>> WG,
>>>>
>>>> The chairs have created a plan for how to perform the Video Codec
>>>> selection in our WG. The chairs are asking for review of our plan on
>>>> how to undertake the mandatory-to-implement video codec selection.
>>>> We'd much prefer to have comments on the mechanics before they begin,
>>>> so please review now.  Proponents of a particular proposal should
>>>> note both the actions required and the timelines proposed.
>>>>
>>>> The main goal of this plan is to hold a consensus call on which of
>>>> the proposed alternatives we as a WG should select at one of the WG
>>>> sessions in Vancouver. Such a consensus call will of course be
>>>> verified on the mailing list for anyone who can't participate. The
>>>> chairs will recuse themselves from judging this particular
>>>> consensus.
>>>>
>>>> In the WG session each codec proposal will be allowed an equal amount
>>>> of time to highlight the arguments for their proposal. After that a
>>>> there will be a slot for discussion and clarifying questions.
>>>>
>>>> To enable the WG participants to get answers to any questions, the
>>>> proposals in draft form and any supporting material MUST be made
>>>> available by 6th of October. This is to ensure that the WG
>>>> participants can verify or object to any claims or statements in
>>>> the proposal material prior to the WG session. We chairs would really
>>>> not like to see the proponents bring up new arguments at their
>>>> presentation. Also the WG participants are expected to raise any
>>>> arguments on the list ahead of time to enable the proponents to
>>>> respond to such arguments.
>>>>
>>>> The proposed consensus questions will be of the following form:
>>>>
>>>> 1. If you support H.264 as the mandatory to implement codec or are
>>>> willing to live with it as the MTI, please raise your hand now.
>>>>
>>>> 2. If you support VP8 as the mandatory to implement codec or are
>>>> willing to live with it as the MTI, please raise your hand now.
>>>>
>>>> You may indicate support on both questions and we encourage you to do
>>>> so if you can live with either, even if you have a preference for one
>>>> over the other.
>>>>
>>>> Additional proposals than the previous ones are welcome, but must be
>>>> submitted as draft and their proponents must notify the chairs no later
>>>> than the 6th of October that they also have a candidate proposal.
>>>>
>>>> In case the WG fails to reach consensus we chairs propose that we use
>>>> the alternative decision process as discussed in RFC3929. The method
>>>> and its usage will be discussed on the list should the WG not
>>>> establish consensus on a proposal for mandatory to implement video codec.
>>>>
>>>> regards,
>>>>
>>>> Magnus,  Cullen, and Ted
>>>>
>>>>
>>>> _______________________________________________
>>>> rtcweb mailing list
>>>> rtcweb@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> rtcweb mailing list
>>>> rtcweb@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>
>>>
>>>
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> 


-- 
Libre Video
http://librevideo.org

From christer.holmberg@ericsson.com  Thu Sep 19 03:08:34 2013
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To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: JSEP-04: Some comments on Section 5.2.1. and 5.2.2 (19th september)
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Subject: [rtcweb] JSEP-04: Some comments on Section 5.2.1. and 5.2.2 (19th september)
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Hi,

Section 5.2.1:
-----------------

Q_1:      o- line <sess-version> initial value - deviation

What is the reason for recommending a <sess-version> zero value in the init=
ial Offer, when the RFC says SHOULD use an NTP format timestamp?

I don't have any strong feelings, but in general: whenever we deviate from =
the "base" SDP procedures I think we should justify why.


Q_2:      o- line <sess-version> initial value - forking

When a new PeerConnection is created due to forking, the <sess-version> val=
ue must be based (incremented) on the value associated with the "mother" Pe=
erConnection, for which the initial Offer of the whole communication sessio=
n was created.


Q_3:      RTP

The text says:

"The <proto> field MUST be set to "RTP/SAVPF". "

But, that of course only applies to RTP based streams (not the data channel=
). Also, in general, it needs to be clear what information needs to be in e=
very m- line, and what information is protocol specific. I would suggest to=
 have a "General" sub-section, a "RTP" sub-section, etc.


Q_4:      BUNDLE

The text says:

"If a m=3D section is not being bundled into another m=3D section, it MUST
                generate a unique set of ICE credentials and gather its own=
 set of
                candidates.  Otherwise, it MUST use the same ICE credential=
s and
                candidates that were used in the m=3D section that it is be=
ing bundled
                into."

As, when BUNDLE is used, the initial Offer will contain identical ICE candi=
dates, does that mean that we will also include identical address informati=
on in the initial Offer?

I don't object to that - I just want to clarify, as it has impacts on the t=
ext in the BUNDLE spec :)


Section 5.2.2:
-----------------


Q_5:      Offer when in "local-offer"

The text says:

"If the initial offer was applied using setLocalDescription, but an
                answer from the remote side has not yet been applied, meani=
ng the
                PeerConnection is still in the "local-offer" state, the ste=
ps for
                generating an initial offer should be followed,"

I don't understand this. Why would you create a new Offer while you are wai=
ting for an Answer to the previously sent Offer?

Also, when setRemote() is called, to which Offer does it apply?


Q_6:      BUNDLE

We'll probably also need some text about BUNDLE.


Regards,

Christer


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<div class=3D"WordSection1">
<p class=3D"MsoNormal">Hi,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Section 5.2.1:<o:p></o:p></p>
<p class=3D"MsoNormal">-----------------<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><b>Q_1:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; o- line &lt;se=
ss-version&gt; initial value - deviation<o:p></o:p></b></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">What is the reason for recommending a &lt;sess-versi=
on&gt; zero value in the initial Offer, when the RFC says SHOULD use an NTP=
 format timestamp?
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">I don&#8217;t have any strong feelings, but in gener=
al: whenever we deviate from the &#8220;base&#8221; SDP procedures I think =
we should justify why.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><b>Q_2:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; o- line &lt;se=
ss-version&gt; initial value - forking<o:p></o:p></b></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">When a new PeerConnection is created due to forking,=
 the &lt;sess-version&gt; value must be based (incremented) on the value as=
sociated with the &#8220;mother&#8221; PeerConnection, for which the initia=
l Offer of the whole communication session was created.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><b>Q_3:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; RTP<o:p></o:p>=
</b></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The text says:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">&#8220;The &lt;proto&gt=
; field MUST be set to &quot;RTP/SAVPF&quot;. &#8220;<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">But, that of course only applies to RTP based stream=
s (not the data channel). Also, in general, it needs to be clear what infor=
mation needs to be in every m- line, and what information is protocol speci=
fic. I would suggest to have a &#8220;General&#8221;
 sub-section, a &#8220;RTP&#8221; sub-section, etc.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><b>Q_4: &nbsp;&nbsp;&nbsp;&nbsp; BUNDLE<o:p></o:p></=
b></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The text says:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">&#8220;If a m=3D sectio=
n is not being bundled into another m=3D section, it MUST<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; generate a unique set of ICE credentials =
and gather its own set of<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; candidates.&nbsp; Otherwise, it MUST use =
the same ICE credentials and<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; candidates that were used in the m=3D sec=
tion that it is being bundled<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; into.&#8221;<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">As, when BUNDLE is used, the initial Offer will cont=
ain identical ICE candidates, does that mean that we will also include iden=
tical address information in the initial Offer?<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">I don&#8217;t object to that &#8211; I just want to =
clarify, as it has impacts on the text in the BUNDLE spec :)<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Section 5.2.2:<o:p></o:p></p>
<p class=3D"MsoNormal">-----------------<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><b>Q_5:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Offer when in =
&#8220;local-offer&#8221;<o:p></o:p></b></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The text says:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">&#8220;If the initial o=
ffer was applied using setLocalDescription, but an<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; answer from the remote side has not yet b=
een applied, meaning the<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; PeerConnection is still in the &quot;loca=
l-offer&quot; state, the steps for<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; generating an initial offer should be fol=
lowed,&#8221;<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">I don&#8217;t understand this. Why would you create =
a new Offer while you are waiting for an Answer to the previously sent Offe=
r?
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Also, when setRemote() is called, to which Offer doe=
s it apply?<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><b><o:p>&nbsp;</o:p></b></p>
<p class=3D"MsoNormal"><b>Q_6:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; BUNDLE<o:p></o=
:p></b></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">We&#8217;ll probably also need some text about BUNDL=
E.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Christer<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</body>
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From harald@alvestrand.no  Thu Sep 19 05:34:56 2013
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Subject: Re: [rtcweb] Recording from todays call
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--------------060309000000090802080605
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Thanks - can someone place this recording on a website where I can get 
at it?

Accessing this page requires Java, which only my non-work computer 
supports; my non-work computer is on IPv6, which leads me to a page that 
tells me (presumably) how to configure the Java runtime parameters on 
Windows, which I don't run (it wants 
-Djava.net.preferIPv6Addresses=true). I'm stuck.

Help?

On 09/18/2013 07:49 PM, Cullen Jennings wrote:
>
>>
>>
>> Your recording is now available on the WebEx service site. Click the 
>> link below to play it:
>>
>> https://cisco.webex.com/ciscosales/lsr.php?AT=pb&SP=MC&rID=71503827&rKey=2cb05592aa7be2c2 
>>
>>
>> RTCWeb - JSEP-20130918 1508-1
>> Wednesday, September 18, 2013 8:08 am San Francisco Time
>> 1 Hour 49 Minutes
>>
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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  <body bgcolor="#FFFFFF" text="#000000">
    <div class="moz-cite-prefix">Thanks - can someone place this
      recording on a website where I can get at it?<br>
      <br>
      Accessing this page requires Java, which only my non-work computer
      supports; my non-work computer is on IPv6, which leads me to a
      page that tells me (presumably) how to configure the Java runtime
      parameters on Windows, which I don't run (it wants
      <meta http-equiv="content-type" content="text/html;
        charset=ISO-8859-1">
      <span style="color: rgb(0, 0, 0); font-family: 'Times New Roman';
        font-size: medium; font-style: normal; font-variant: normal;
        font-weight: normal; letter-spacing: normal; line-height:
        normal; orphans: auto; text-align: left; text-indent: 0px;
        text-transform: none; white-space: normal; widows: auto;
        word-spacing: 0px; -webkit-text-stroke-width: 0px; display:
        inline !important; float: none;">
        -Djava.net.preferIPv6Addresses=true).</span> I'm stuck.<br>
      <br>
      Help?<br>
      <br>
      On 09/18/2013 07:49 PM, Cullen Jennings wrote:<br>
    </div>
    <blockquote cite="mid:DB137210-3CE4-4FE5-AEB7-1240071A15DE@iii.ca"
      type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=ISO-8859-1">
      <br>
      <div>
        <blockquote type="cite"><font face="Tahoma, Arial, sans-serif,
            Helvetica, Geneva" size="2"><br>
            <br>
            Your recording is now available on the WebEx service site.
            Click the link below to play it: <br>
            <br>
            <a moz-do-not-send="true"
href="https://cisco.webex.com/ciscosales/lsr.php?AT=pb&amp;SP=MC&amp;rID=71503827&amp;rKey=2cb05592aa7be2c2"
              target="_blank">https://cisco.webex.com/ciscosales/lsr.php?AT=pb&amp;SP=MC&amp;rID=71503827&amp;rKey=2cb05592aa7be2c2</a>
            <br>
            <br>
            RTCWeb - JSEP-20130918 1508-1 <br>
            Wednesday, September 18, 2013 8:08 am San Francisco Time <br>
            1 Hour 49 Minutes <br>
            <br>
          </font></blockquote>
        <div><br>
        </div>
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      <br>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
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From harald@alvestrand.no  Thu Sep 19 05:53:46 2013
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Subject: [rtcweb] JSEP-04: Nits and niggles
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I'm sure we'll have lots of things to discuss about JSEP. I just thought 
I'd get these out of the way.

- Usage of MSID. In addition to the a=msid: lines, the session-level 
generation procedure needs to say that a line "a=msid-semantic:WMS <id>" 
is added to the session-level description, containing the IDs of the 
MediaStreams that are signalled.

- a=ssrc lines. The spec (5.2.1 second list last bullet) simply says "an 
a=ssrc line". But RFC 5576 specifies that all ssrc lines contain an 
attribute. Suggestion: define one that does no harm (role=primary?)

- recycling of m= lines. The section in 5.2.2 (subsequent offers) that 
talks about this doesn't mention that the media in the m-line has to 
match, although section 5.3.1 (initial answer) actually talks about it. 
I'm assuming that's an oversight.

I'd suggest pulling the language about "selecting an appropriate m-line 
for a track" out in a separate subsection, so that we have one place to 
point to for this.

Out of time, let's get these ones out of the way for -05....




From michael@voip.co.uk  Thu Sep 19 08:40:26 2013
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From: Michael Procter <michael@voip.co.uk>
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Subject: [rtcweb] jsep-04 questions about phrasing
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Hi,

Section 5.2.2 Subsequent Offers

This section starts with the sentence "When createOffer is called a
second (or later) time,".  As far as I can tell, the rest of this
section applies both to subsequent offers created by the original
offerer and also offers created by the original answerer.  In this
second case, createOffer has never been called.  I would suggest
something like "When createOffer is called after the initial
offer/answer exchange,".

Section 5.2.2 11th bullet:

   o  The m= line and corresponding "a=rtpmap" and "a=fmtp" lines MUST
      only include codecs present in the remote description.

Is this necessary?  Can't we offer new codecs in a subsequent offer?
RFC3264 Sec 8.3.2 permits new codecs to be offered, old ones to be
removed etc, with the only constraint being to not reuse dynamic
payload types.

Finally, the two example SDPs both contain "s=-" whereas section 5.2.1
encourages "s= " instead. Personally, I prefer the hyphen, but
consistency is probably more important.

Regards,

Michael

From gsalguei@cisco.com  Thu Sep 19 09:20:14 2013
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From: "Gonzalo Salgueiro (gsalguei)" <gsalguei@cisco.com>
To: "clue@ietf.org" <clue@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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[Please forgive the cross-post...]

Significant changes have been made to the draft, please provide feedback or=
 comment on its usefulness @AVTEXT

Cheers,=20

Gonzalo

> On Sep 19, 2013, at 10:55 AM, "internet-drafts@ietf.org" <internet-drafts=
@ietf.org> wrote:
>=20
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts direct=
ories.
>=20
>=20
>    Title           : A Taxonomy of Grouping Semantics and Mechanisms for =
Real-Time Transport Protocol (RTP) Sources
>    Author(s)       : Jonathan Lennox
>                          Kevin Gross
>                          Suhas Nandakumar
>                          Gonzalo Salgueiro
>                          Bo Burman
>    Filename        : draft-lennox-raiarea-rtp-grouping-taxonomy-02.txt
>    Pages           : 40
>    Date            : 2013-09-19
>=20
> Abstract:
>   The terminology about, and associations among, Real-Time Transport
>   Protocol (RTP) sources can be complex and somewhat opaque.  This
>   document describes a number of existing and proposed relationships
>   among RTP sources, and attempts to define common terminology for
>   discussing protocol entities and their relationships.
>=20
>   This document is still very rough, but is submitted in the hopes of
>   making future discussion productive.
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-lennox-raiarea-rtp-grouping-taxono=
my
>=20
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-lennox-raiarea-rtp-grouping-taxonomy-02
>=20
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=3Ddraft-lennox-raiarea-rtp-grouping-taxo=
nomy-02
>=20
>=20
> Please note that it may take a couple of minutes from the time of submiss=
ion
> until the htmlized version and diff are available at tools.ietf.org.
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> _______________________________________________
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> Internet-Draft directories: http://www.ietf.org/shadow.html
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From christer.holmberg@ericsson.com  Thu Sep 19 09:26:27 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Suhas Nandakumar <suhasietf@gmail.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] m=section recycling and a=mid
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Subject: Re: [rtcweb] m=section recycling and a=mid
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Hi Suhas,



The mid attribute itself does not add an m- line to a BUNDLE group. The att=
ribute value also needs to be listed in the group:BUNDLE attribute mid list=
.



So, if you want to re-use an m- line and mid attribute, but you don't want =
the m- line to be in the BUNDLE group, simply remove the mid from the mid l=
ist.



Regards,



Christer



________________________________
From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] on behalf of Suhas =
Nandakumar [suhasietf@gmail.com]
Sent: Wednesday, 18 September 2013 7:44 PM
To: rtcweb@ietf.org
Subject: [rtcweb] m=3Dsection recycling and a=3Dmid

After thinking about it a bit, I think there are some side effects associat=
ed with changing/not changing mid attribute of a re-cycled m=3D section and=
 it being part of a BUNDLE group or not..

example

say , we have these video m=3Dsection part of BUNDLE group as below
a=3Dgroup:BUNDLE 1, 2

m=3Dvideo .....
a=3Dmid:1
a=3Dinactive

m=3Dvideo .....
a=3Dmid:2
a=3Dsendrecv

Now If JS user adds a new video track and he wants it to be out of BUNDLE. =
If we recycle first video m=3Dsection and dont change mid attribute,  we en=
d up BUNDLing even if the user dint want to

On the other hand, if we allowed mid value to be changed, we might end up U=
NBundling a video stream , when the user did want to BUNDLE them

So allowing a mid value to be changed or not should be tied to what the JS =
API is asking on behalf of the user


Any thoughts ?

Cheers
Suhas



--_000_7594FB04B1934943A5C02806D1A2204B1C4A7F29ESESSMB209erics_
Content-Type: text/html; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<html dir=3D"ltr">
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Diso-8859-=
1">
<style id=3D"owaParaStyle">P {
	MARGIN-BOTTOM: 0px; MARGIN-TOP: 0px
}
</style>
</head>
<body fPStyle=3D"1" ocsi=3D"0">
<div style=3D"direction: ltr;font-family: Tahoma;color: #000000;font-size: =
10pt;">
<p>Hi Suhas,</p>
<p>&nbsp;</p>
<p>The mid attribute itself does not add an m- line to a BUNDLE group. The =
attribute value also needs to be listed in the group:BUNDLE attribute mid l=
ist.</p>
<p>&nbsp;</p>
<p>So, if you want to re-use an m- line and mid attribute, but you don't wa=
nt the m- line to be in the BUNDLE group, simply remove the mid from the mi=
d list.</p>
<p>&nbsp;</p>
<p>Regards,</p>
<p>&nbsp;</p>
<p>Christer</p>
<p>&nbsp;</p>
<div style=3D"FONT-SIZE: 16px; FONT-FAMILY: Times New Roman; COLOR: #000000=
">
<hr tabindex=3D"-1">
<div id=3D"divRpF859731" style=3D"DIRECTION: ltr"><font color=3D"#000000" s=
ize=3D"2" face=3D"Tahoma"><b>From:</b> rtcweb-bounces@ietf.org [rtcweb-boun=
ces@ietf.org] on behalf of Suhas Nandakumar [suhasietf@gmail.com]<br>
<b>Sent:</b> Wednesday, 18 September 2013 7:44 PM<br>
<b>To:</b> rtcweb@ietf.org<br>
<b>Subject:</b> [rtcweb] m=3Dsection recycling and a=3Dmid<br>
</font><br>
</div>
<div></div>
<div>
<div dir=3D"ltr">After thinking about it a bit, I think there are some side=
 effects associated with changing/not changing mid attribute of a re-cycled=
 m=3D section and it being part of a BUNDLE group or not..
<div><br>
</div>
<div>example</div>
<div><br>
</div>
<div>say , we have these video m=3Dsection part of BUNDLE group as below&nb=
sp;</div>
<div>a=3Dgroup:BUNDLE 1, 2</div>
<div><br>
</div>
<div>m=3Dvideo .....</div>
<div>a=3Dmid:1&nbsp;</div>
<div>a=3Dinactive</div>
<div><br>
</div>
<div>
<div>m=3Dvideo .....</div>
<div>a=3Dmid:2&nbsp;</div>
<div>a=3Dsendrecv</div>
<div><br>
</div>
<div>Now If JS user adds a new video track and he wants it to be out of BUN=
DLE. If we recycle first video m=3Dsection and dont change mid attribute, &=
nbsp;we end up BUNDLing even if the user dint want to</div>
<div><br>
</div>
<div>On the other hand, if we allowed mid value to be changed, we might end=
 up UNBundling a video stream , when the user did want to BUNDLE them</div>
<div><br>
</div>
<div>So allowing a mid value to be changed or not should be tied to what th=
e JS API is asking on behalf of the user</div>
</div>
<div><br>
</div>
<div><br>
</div>
<div>Any thoughts ?</div>
<div><br>
</div>
<div>Cheers</div>
<div>Suhas</div>
<div><br>
</div>
<div><br>
</div>
</div>
</div>
</div>
</div>
</body>
</html>

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From christer.holmberg@ericsson.com  Thu Sep 19 09:36:02 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Suhas Nandakumar <suhasietf@gmail.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] m=section recycling and a=mid
Thread-Index: AQHOtI5OlzgN5l68MU+Yf86f9mtTsJnNP/0ZgAADHRE=
Date: Thu, 19 Sep 2013 16:35:51 +0000
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Subject: Re: [rtcweb] m=section recycling and a=mid
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Hi,



Below is some suggested BUNDLE text:



6.4.5.  Moving A Media Description Out Of A BUNDLE Group

   When an Offerer generates an Offer, in which an "m=3D" line is moved
   out of a BUNDLE group, the Offerer MUST assign a unique address to
   the moved "m=3D" line.  In addition, the Offerer MUST NOT anymore
   include a mid value, representing the "m=3D" line, in the SDP
   group:BUNDLE attribute mid list associated with the BUNDLE group.

   Section 11.4 shows an example of an Offer used to move an "m=3D" line
   out of a BUNDLE group.



Regards,



Christer



________________________________
From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] on behalf of Christ=
er Holmberg [christer.holmberg@ericsson.com]
Sent: Thursday, 19 September 2013 7:24 PM
To: Suhas Nandakumar; rtcweb@ietf.org
Subject: Re: [rtcweb] m=3Dsection recycling and a=3Dmid


Hi Suhas,



The mid attribute itself does not add an m- line to a BUNDLE group. The att=
ribute value also needs to be listed in the group:BUNDLE attribute mid list=
.



So, if you want to re-use an m- line and mid attribute, but you don't want =
the m- line to be in the BUNDLE group, simply remove the mid from the mid l=
ist.



Regards,



Christer



________________________________
From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] on behalf of Suhas =
Nandakumar [suhasietf@gmail.com]
Sent: Wednesday, 18 September 2013 7:44 PM
To: rtcweb@ietf.org
Subject: [rtcweb] m=3Dsection recycling and a=3Dmid

After thinking about it a bit, I think there are some side effects associat=
ed with changing/not changing mid attribute of a re-cycled m=3D section and=
 it being part of a BUNDLE group or not..

example

say , we have these video m=3Dsection part of BUNDLE group as below
a=3Dgroup:BUNDLE 1, 2

m=3Dvideo .....
a=3Dmid:1
a=3Dinactive

m=3Dvideo .....
a=3Dmid:2
a=3Dsendrecv

Now If JS user adds a new video track and he wants it to be out of BUNDLE. =
If we recycle first video m=3Dsection and dont change mid attribute,  we en=
d up BUNDLing even if the user dint want to

On the other hand, if we allowed mid value to be changed, we might end up U=
NBundling a video stream , when the user did want to BUNDLE them

So allowing a mid value to be changed or not should be tied to what the JS =
API is asking on behalf of the user


Any thoughts ?

Cheers
Suhas



--_000_7594FB04B1934943A5C02806D1A2204B1C4A7F58ESESSMB209erics_
Content-Type: text/html; charset="iso-8859-1"
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<html dir=3D"ltr">
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Diso-8859-=
1">
<style id=3D"owaParaStyle">P {
	MARGIN-BOTTOM: 0px; MARGIN-TOP: 0px
}
P {
	MARGIN-BOTTOM: 0px; MARGIN-TOP: 0px
}
</style>
</head>
<body fPStyle=3D"1" ocsi=3D"0">
<div style=3D"direction: ltr;font-family: Tahoma;color: #000000;font-size: =
10pt;">
<p>Hi,</p>
<p>&nbsp;</p>
<p>Below is some suggested BUNDLE text:</p>
<p>&nbsp;</p>
<div style=3D"MARGIN: 0px"><font size=3D"2" face=3D"Calibri,sans-serif"><sp=
an style=3D"FONT-SIZE: 11pt"><font size=3D"2" face=3D"Courier New"><span st=
yle=3D"FONT-SIZE: 10pt">6.4.5.&nbsp; Moving A Media Description Out Of A BU=
NDLE Group</span></font></span></font></div>
<div style=3D"MARGIN: 0px"><font size=3D"2" face=3D"Calibri,sans-serif"><sp=
an style=3D"FONT-SIZE: 11pt"><font size=3D"2" face=3D"Courier New"><span st=
yle=3D"FONT-SIZE: 10pt"></span></font></span></font>&nbsp;</div>
<div style=3D"MARGIN: 0px"><font size=3D"2" face=3D"Calibri,sans-serif"><sp=
an style=3D"FONT-SIZE: 11pt"><font size=3D"2" face=3D"Courier New"><span st=
yle=3D"FONT-SIZE: 10pt">&nbsp;&nbsp; When an Offerer generates an Offer, in=
 which an &quot;m=3D&quot; line is moved</span></font></span></font></div>
<div style=3D"MARGIN: 0px"><font size=3D"2" face=3D"Calibri,sans-serif"><sp=
an style=3D"FONT-SIZE: 11pt"><font size=3D"2" face=3D"Courier New"><span st=
yle=3D"FONT-SIZE: 10pt">&nbsp;&nbsp; out of a BUNDLE group, the Offerer MUS=
T assign a unique address to</span></font></span></font></div>
<div style=3D"MARGIN: 0px"><font size=3D"2" face=3D"Calibri,sans-serif"><sp=
an style=3D"FONT-SIZE: 11pt"><font size=3D"2" face=3D"Courier New"><span st=
yle=3D"FONT-SIZE: 10pt">&nbsp;&nbsp; the moved &quot;m=3D&quot; line.&nbsp;
<strong><font color=3D"#ff0000">In addition, the Offerer MUST NOT anymore</=
font></strong></span></font></span></font></div>
<div style=3D"MARGIN: 0px"><font size=3D"2" face=3D"Calibri,sans-serif"><sp=
an style=3D"FONT-SIZE: 11pt"><font color=3D"#ff0000" size=3D"2" face=3D"Cou=
rier New"><span style=3D"FONT-SIZE: 10pt"><strong>&nbsp;&nbsp; include a mi=
d value, representing the &quot;m=3D&quot; line, in the SDP</strong></span>=
</font></span></font></div>
<div style=3D"MARGIN: 0px"><font size=3D"2" face=3D"Calibri,sans-serif"><sp=
an style=3D"FONT-SIZE: 11pt"><font size=3D"2" face=3D"Courier New"><span st=
yle=3D"FONT-SIZE: 10pt"><font color=3D"#ff0000"><strong>&nbsp;&nbsp; group:=
BUNDLE attribute mid list associated with the BUNDLE group</strong>.</font>=
</span></font></span></font></div>
<div style=3D"MARGIN: 0px"><font size=3D"2" face=3D"Calibri,sans-serif"><sp=
an style=3D"FONT-SIZE: 11pt"><font size=3D"2" face=3D"Courier New"><span st=
yle=3D"FONT-SIZE: 10pt"></span></font></span></font>&nbsp;</div>
<div style=3D"MARGIN: 0px"><font size=3D"2" face=3D"Calibri,sans-serif"><sp=
an style=3D"FONT-SIZE: 11pt"><font size=3D"2" face=3D"Courier New"><span st=
yle=3D"FONT-SIZE: 10pt">&nbsp;&nbsp; Section 11.4 shows an example of an Of=
fer used to move an &quot;m=3D&quot; line</span></font></span></font></div>
<div style=3D"MARGIN: 0px"><font size=3D"2" face=3D"Calibri,sans-serif"><sp=
an style=3D"FONT-SIZE: 11pt"><font size=3D"2" face=3D"Courier New"><span st=
yle=3D"FONT-SIZE: 10pt">&nbsp;&nbsp; out of a BUNDLE group.</span></font></=
span></font></div>
<p>&nbsp;</p>
<p>Regards,</p>
<p>&nbsp;</p>
<p>Christer</p>
<p>&nbsp;</p>
<div style=3D"FONT-SIZE: 16px; FONT-FAMILY: Times New Roman; COLOR: #000000=
">
<hr tabindex=3D"-1">
<div id=3D"divRpF498818" style=3D"DIRECTION: ltr"><font color=3D"#000000" s=
ize=3D"2" face=3D"Tahoma"><b>From:</b> rtcweb-bounces@ietf.org [rtcweb-boun=
ces@ietf.org] on behalf of Christer Holmberg [christer.holmberg@ericsson.co=
m]<br>
<b>Sent:</b> Thursday, 19 September 2013 7:24 PM<br>
<b>To:</b> Suhas Nandakumar; rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] m=3Dsection recycling and a=3Dmid<br>
</font><br>
</div>
<div></div>
<div>
<div style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Tahoma; COLOR: #000000; DIRECTI=
ON: ltr">
<p>Hi Suhas,</p>
<p>&nbsp;</p>
<p>The mid attribute itself does not add an m- line to a BUNDLE group. The =
attribute value also needs to be listed in the group:BUNDLE attribute mid l=
ist.</p>
<p>&nbsp;</p>
<p>So, if you want to re-use an m- line and mid attribute, but you don't wa=
nt the m- line to be in the BUNDLE group, simply remove the mid from the mi=
d list.</p>
<p>&nbsp;</p>
<p>Regards,</p>
<p>&nbsp;</p>
<p>Christer</p>
<p>&nbsp;</p>
<div style=3D"FONT-SIZE: 16px; FONT-FAMILY: Times New Roman; COLOR: #000000=
">
<hr tabindex=3D"-1">
<div id=3D"divRpF859731" style=3D"DIRECTION: ltr"><font color=3D"#000000" s=
ize=3D"2" face=3D"Tahoma"><b>From:</b> rtcweb-bounces@ietf.org [rtcweb-boun=
ces@ietf.org] on behalf of Suhas Nandakumar [suhasietf@gmail.com]<br>
<b>Sent:</b> Wednesday, 18 September 2013 7:44 PM<br>
<b>To:</b> rtcweb@ietf.org<br>
<b>Subject:</b> [rtcweb] m=3Dsection recycling and a=3Dmid<br>
</font><br>
</div>
<div></div>
<div>
<div dir=3D"ltr">After thinking about it a bit, I think there are some side=
 effects associated with changing/not changing mid attribute of a re-cycled=
 m=3D section and it being part of a BUNDLE group or not..
<div><br>
</div>
<div>example</div>
<div><br>
</div>
<div>say , we have these video m=3Dsection part of BUNDLE group as below&nb=
sp;</div>
<div>a=3Dgroup:BUNDLE 1, 2</div>
<div><br>
</div>
<div>m=3Dvideo .....</div>
<div>a=3Dmid:1&nbsp;</div>
<div>a=3Dinactive</div>
<div><br>
</div>
<div>
<div>m=3Dvideo .....</div>
<div>a=3Dmid:2&nbsp;</div>
<div>a=3Dsendrecv</div>
<div><br>
</div>
<div>Now If JS user adds a new video track and he wants it to be out of BUN=
DLE. If we recycle first video m=3Dsection and dont change mid attribute, &=
nbsp;we end up BUNDLing even if the user dint want to</div>
<div><br>
</div>
<div>On the other hand, if we allowed mid value to be changed, we might end=
 up UNBundling a video stream , when the user did want to BUNDLE them</div>
<div><br>
</div>
<div>So allowing a mid value to be changed or not should be tied to what th=
e JS API is asking on behalf of the user</div>
</div>
<div><br>
</div>
<div><br>
</div>
<div>Any thoughts ?</div>
<div><br>
</div>
<div>Cheers</div>
<div>Suhas</div>
<div><br>
</div>
<div><br>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</body>
</html>

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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Suhas Nandakumar <suhasietf@gmail.com>
Thread-Topic: [rtcweb] how to control which track plays in a video tag
Thread-Index: AQHOtV1qkxTQZpKAjkKP4VCSBKGS4A==
Date: Thu, 19 Sep 2013 17:26:47 +0000
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] how to control which track plays in a video tag
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OMG - that was bad. I meant to send this to webrtc list. I apologize and I =
will have some chair give me a proper lecture on mailing list use. Please s=
end any replies to webrtc list.=20




On Sep 18, 2013, at 9:32 AM, Suhas Nandakumar <suhasietf@gmail.com> wrote:

> Shouldn't this be dealt with a W3C API, say videoelem.src =3D mediastream=
.trackAt(0)=20
>=20
>=20
> On Wed, Sep 18, 2013 at 9:01 AM, Cullen Jennings (fluffy) <fluffy@cisco.c=
om> wrote:
>=20
> The issue got raised if there is multiple video tracks in a media stream,=
 how do you control which video track gets played in a video stream
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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Date: Thu, 19 Sep 2013 11:48:05 -0700
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From: Suhas Nandakumar <suhasietf@gmail.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] m=section recycling and a=mid
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--047d7bb03e1e8706a804e6c0fe88
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Hello Christer

  Completely agree with you

My point with the above examples was

  "recycling of m=line should take into account its existing BUNDLE
membership and JS user's preference on requiring the added media track to
be BUNDLED or not"

This would imply, as suggested by you,

  - remove the mid from a=group:BUNDLE line if the JS user doesn't want the
new track to be BUNDLEd

 - or if the mid value is changed, then add the new mid value to the BUNDLE
group if the JS user want's the new track to be BUNDLED

- or just retain the mid value and its BUNDLE membership for the new track,
if it is going to be BUNDLEd


Cheers
Suhas







On Thu, Sep 19, 2013 at 9:24 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>  Hi Suhas,
>
>
>
> The mid attribute itself does not add an m- line to a BUNDLE group. The
> attribute value also needs to be listed in the group:BUNDLE attribute mid
> list.
>
>
>
> So, if you want to re-use an m- line and mid attribute, but you don't want
> the m- line to be in the BUNDLE group, simply remove the mid from the mid
> list.
>
>
>
> Regards,
>
>
>
> Christer
>
>
>  ------------------------------
> *From:* rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] on behalf of
> Suhas Nandakumar [suhasietf@gmail.com]
> *Sent:* Wednesday, 18 September 2013 7:44 PM
> *To:* rtcweb@ietf.org
> *Subject:* [rtcweb] m=section recycling and a=mid
>
>   After thinking about it a bit, I think there are some side effects
> associated with changing/not changing mid attribute of a re-cycled m=
> section and it being part of a BUNDLE group or not..
>
>  example
>
>  say , we have these video m=section part of BUNDLE group as below
> a=group:BUNDLE 1, 2
>
>  m=video .....
> a=mid:1
> a=inactive
>
>  m=video .....
> a=mid:2
> a=sendrecv
>
>  Now If JS user adds a new video track and he wants it to be out of
> BUNDLE. If we recycle first video m=section and dont change mid attribute,
>  we end up BUNDLing even if the user dint want to
>
>  On the other hand, if we allowed mid value to be changed, we might end
> up UNBundling a video stream , when the user did want to BUNDLE them
>
>  So allowing a mid value to be changed or not should be tied to what the
> JS API is asking on behalf of the user
>
>
>  Any thoughts ?
>
>  Cheers
> Suhas
>
>
>

--047d7bb03e1e8706a804e6c0fe88
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Hello Christer<div><br></div><div>=A0 Completely agree wit=
h you</div><div><br></div><div>My point with the above examples was=A0</div=
><div><br></div><div>=A0 &quot;recycling of m=3Dline should take into accou=
nt its existing BUNDLE membership and JS user&#39;s preference on requiring=
 the added media track to be BUNDLED or not&quot;</div>
<div><br></div><div>This would imply, as suggested by you,</div><div><br></=
div><div>=A0 - remove the mid from a=3Dgroup:BUNDLE line if the JS user doe=
sn&#39;t want the new track to be BUNDLEd</div><div><br></div><div>=A0- or =
if the mid value is changed, then add the new mid value to the BUNDLE group=
 if the JS user want&#39;s the new track to be BUNDLED</div>
<div><br></div><div>- or just retain the mid value and its BUNDLE membershi=
p for the new track, if it is going to be BUNDLEd</div><div><br></div><div>=
<br></div><div>Cheers</div><div>Suhas</div><div><br></div><div><br></div>
<div><br></div><div><br></div><div><br></div></div><div class=3D"gmail_extr=
a"><br><br><div class=3D"gmail_quote">On Thu, Sep 19, 2013 at 9:24 AM, Chri=
ster Holmberg <span dir=3D"ltr">&lt;<a href=3D"mailto:christer.holmberg@eri=
csson.com" target=3D"_blank">christer.holmberg@ericsson.com</a>&gt;</span> =
wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">




<div>
<div style=3D"direction:ltr;font-size:10pt;font-family:Tahoma">
<p>Hi Suhas,</p>
<p>=A0</p>
<p>The mid attribute itself does not add an m- line to a BUNDLE group. The =
attribute value also needs to be listed in the group:BUNDLE attribute mid l=
ist.</p>
<p>=A0</p>
<p>So, if you want to re-use an m- line and mid attribute, but you don&#39;=
t want the m- line to be in the BUNDLE group, simply remove the mid from th=
e mid list.</p>
<p>=A0</p>
<p>Regards,</p>
<p>=A0</p>
<p>Christer</p>
<p>=A0</p>
<div style=3D"font-size:16px;font-family:Times New Roman">
<hr>
<div style=3D"DIRECTION:ltr"><font color=3D"#000000" face=3D"Tahoma"><b>Fro=
m:</b> <a href=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_blank">rtcweb-=
bounces@ietf.org</a> [<a href=3D"mailto:rtcweb-bounces@ietf.org" target=3D"=
_blank">rtcweb-bounces@ietf.org</a>] on behalf of Suhas Nandakumar [<a href=
=3D"mailto:suhasietf@gmail.com" target=3D"_blank">suhasietf@gmail.com</a>]<=
br>

<b>Sent:</b> Wednesday, 18 September 2013 7:44 PM<br>
<b>To:</b> <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a><br>
<b>Subject:</b> [rtcweb] m=3Dsection recycling and a=3Dmid<br>
</font><br>
</div><div><div class=3D"h5">
<div></div>
<div>
<div dir=3D"ltr">After thinking about it a bit, I think there are some side=
 effects associated with changing/not changing mid attribute of a re-cycled=
 m=3D section and it being part of a BUNDLE group or not..
<div><br>
</div>
<div>example</div>
<div><br>
</div>
<div>say , we have these video m=3Dsection part of BUNDLE group as below=A0=
</div>
<div>a=3Dgroup:BUNDLE 1, 2</div>
<div><br>
</div>
<div>m=3Dvideo .....</div>
<div>a=3Dmid:1=A0</div>
<div>a=3Dinactive</div>
<div><br>
</div>
<div>
<div>m=3Dvideo .....</div>
<div>a=3Dmid:2=A0</div>
<div>a=3Dsendrecv</div>
<div><br>
</div>
<div>Now If JS user adds a new video track and he wants it to be out of BUN=
DLE. If we recycle first video m=3Dsection and dont change mid attribute, =
=A0we end up BUNDLing even if the user dint want to</div>
<div><br>
</div>
<div>On the other hand, if we allowed mid value to be changed, we might end=
 up UNBundling a video stream , when the user did want to BUNDLE them</div>
<div><br>
</div>
<div>So allowing a mid value to be changed or not should be tied to what th=
e JS API is asking on behalf of the user</div>
</div>
<div><br>
</div>
<div><br>
</div>
<div>Any thoughts ?</div>
<div><br>
</div>
<div>Cheers</div>
<div>Suhas</div>
<div><br>
</div>
<div><br>
</div>
</div>
</div>
</div></div></div>
</div>
</div>

</blockquote></div><br></div>

--047d7bb03e1e8706a804e6c0fe88--

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Suhas Nandakumar <suhasietf@gmail.com>
Thread-Topic: [rtcweb] m=section recycling and a=mid
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Subject: Re: [rtcweb] m=section recycling and a=mid
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Correct.



Regards,



Christer

________________________________
From: Suhas Nandakumar [suhasietf@gmail.com]
Sent: Thursday, 19 September 2013 9:48 PM
To: Christer Holmberg
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] m=3Dsection recycling and a=3Dmid

Hello Christer

  Completely agree with you

My point with the above examples was

  "recycling of m=3Dline should take into account its existing BUNDLE membe=
rship and JS user's preference on requiring the added media track to be BUN=
DLED or not"

This would imply, as suggested by you,

  - remove the mid from a=3Dgroup:BUNDLE line if the JS user doesn't want t=
he new track to be BUNDLEd

 - or if the mid value is changed, then add the new mid value to the BUNDLE=
 group if the JS user want's the new track to be BUNDLED

- or just retain the mid value and its BUNDLE membership for the new track,=
 if it is going to be BUNDLEd


Cheers
Suhas







On Thu, Sep 19, 2013 at 9:24 AM, Christer Holmberg <christer.holmberg@erics=
son.com<mailto:christer.holmberg@ericsson.com>> wrote:

Hi Suhas,



The mid attribute itself does not add an m- line to a BUNDLE group. The att=
ribute value also needs to be listed in the group:BUNDLE attribute mid list=
.



So, if you want to re-use an m- line and mid attribute, but you don't want =
the m- line to be in the BUNDLE group, simply remove the mid from the mid l=
ist.



Regards,



Christer



________________________________
From: rtcweb-bounces@ietf.org<mailto:rtcweb-bounces@ietf.org> [rtcweb-bounc=
es@ietf.org<mailto:rtcweb-bounces@ietf.org>] on behalf of Suhas Nandakumar =
[suhasietf@gmail.com<mailto:suhasietf@gmail.com>]
Sent: Wednesday, 18 September 2013 7:44 PM
To: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
Subject: [rtcweb] m=3Dsection recycling and a=3Dmid

After thinking about it a bit, I think there are some side effects associat=
ed with changing/not changing mid attribute of a re-cycled m=3D section and=
 it being part of a BUNDLE group or not..

example

say , we have these video m=3Dsection part of BUNDLE group as below
a=3Dgroup:BUNDLE 1, 2

m=3Dvideo .....
a=3Dmid:1
a=3Dinactive

m=3Dvideo .....
a=3Dmid:2
a=3Dsendrecv

Now If JS user adds a new video track and he wants it to be out of BUNDLE. =
If we recycle first video m=3Dsection and dont change mid attribute,  we en=
d up BUNDLing even if the user dint want to

On the other hand, if we allowed mid value to be changed, we might end up U=
NBundling a video stream , when the user did want to BUNDLE them

So allowing a mid value to be changed or not should be tied to what the JS =
API is asking on behalf of the user


Any thoughts ?

Cheers
Suhas




--_000_7594FB04B1934943A5C02806D1A2204B1C4A800EESESSMB209erics_
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<html dir=3D"ltr">
<head>
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1">
<style id=3D"owaParaStyle">P {
	MARGIN-BOTTOM: 0px; MARGIN-TOP: 0px
}
</style>
</head>
<body fPStyle=3D"1" ocsi=3D"0">
<div style=3D"direction: ltr;font-family: Tahoma;color: #000000;font-size: =
10pt;">
<p>Correct.</p>
<p>&nbsp;</p>
<p>Regards,</p>
<p>&nbsp;</p>
<p>Christer</p>
<div style=3D"FONT-SIZE: 16px; FONT-FAMILY: Times New Roman; COLOR: #000000=
">
<hr tabindex=3D"-1">
<div id=3D"divRpF925856" style=3D"DIRECTION: ltr"><font color=3D"#000000" s=
ize=3D"2" face=3D"Tahoma"><b>From:</b> Suhas Nandakumar [suhasietf@gmail.co=
m]<br>
<b>Sent:</b> Thursday, 19 September 2013 9:48 PM<br>
<b>To:</b> Christer Holmberg<br>
<b>Cc:</b> rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] m=3Dsection recycling and a=3Dmid<br>
</font><br>
</div>
<div></div>
<div>
<div dir=3D"ltr">Hello Christer
<div><br>
</div>
<div>&nbsp; Completely agree with you</div>
<div><br>
</div>
<div>My point with the above examples was&nbsp;</div>
<div><br>
</div>
<div>&nbsp; &quot;recycling of m=3Dline should take into account its existi=
ng BUNDLE membership and JS user's preference on requiring the added media =
track to be BUNDLED or not&quot;</div>
<div><br>
</div>
<div>This would imply, as suggested by you,</div>
<div><br>
</div>
<div>&nbsp; - remove the mid from a=3Dgroup:BUNDLE line if the JS user does=
n't want the new track to be BUNDLEd</div>
<div><br>
</div>
<div>&nbsp;- or if the mid value is changed, then add the new mid value to =
the BUNDLE group if the JS user want's the new track to be BUNDLED</div>
<div><br>
</div>
<div>- or just retain the mid value and its BUNDLE membership for the new t=
rack, if it is going to be BUNDLEd</div>
<div><br>
</div>
<div><br>
</div>
<div>Cheers</div>
<div>Suhas</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
</div>
<div class=3D"gmail_extra"><br>
<br>
<div class=3D"gmail_quote">On Thu, Sep 19, 2013 at 9:24 AM, Christer Holmbe=
rg <span dir=3D"ltr">
&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chr=
ister.holmberg@ericsson.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"PADDING-LEFT: 1ex; MARGIN: 0px 0=
px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">
<div>
<div style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Tahoma; DIRECTION: ltr">
<p>Hi Suhas,</p>
<p>&nbsp;</p>
<p>The mid attribute itself does not add an m- line to a BUNDLE group. The =
attribute value also needs to be listed in the group:BUNDLE attribute mid l=
ist.</p>
<p>&nbsp;</p>
<p>So, if you want to re-use an m- line and mid attribute, but you don't wa=
nt the m- line to be in the BUNDLE group, simply remove the mid from the mi=
d list.</p>
<p>&nbsp;</p>
<p>Regards,</p>
<p>&nbsp;</p>
<p>Christer</p>
<p>&nbsp;</p>
<div style=3D"FONT-SIZE: 16px; FONT-FAMILY: Times New Roman">
<hr>
<div style=3D"DIRECTION: ltr"><font color=3D"#000000" face=3D"Tahoma"><b>Fr=
om:</b> <a href=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_blank">
rtcweb-bounces@ietf.org</a> [<a href=3D"mailto:rtcweb-bounces@ietf.org" tar=
get=3D"_blank">rtcweb-bounces@ietf.org</a>] on behalf of Suhas Nandakumar [=
<a href=3D"mailto:suhasietf@gmail.com" target=3D"_blank">suhasietf@gmail.co=
m</a>]<br>
<b>Sent:</b> Wednesday, 18 September 2013 7:44 PM<br>
<b>To:</b> <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a><br>
<b>Subject:</b> [rtcweb] m=3Dsection recycling and a=3Dmid<br>
</font><br>
</div>
<div>
<div class=3D"h5">
<div></div>
<div>
<div dir=3D"ltr">After thinking about it a bit, I think there are some side=
 effects associated with changing/not changing mid attribute of a re-cycled=
 m=3D section and it being part of a BUNDLE group or not..
<div><br>
</div>
<div>example</div>
<div><br>
</div>
<div>say , we have these video m=3Dsection part of BUNDLE group as below&nb=
sp;</div>
<div>a=3Dgroup:BUNDLE 1, 2</div>
<div><br>
</div>
<div>m=3Dvideo .....</div>
<div>a=3Dmid:1&nbsp;</div>
<div>a=3Dinactive</div>
<div><br>
</div>
<div>
<div>m=3Dvideo .....</div>
<div>a=3Dmid:2&nbsp;</div>
<div>a=3Dsendrecv</div>
<div><br>
</div>
<div>Now If JS user adds a new video track and he wants it to be out of BUN=
DLE. If we recycle first video m=3Dsection and dont change mid attribute, &=
nbsp;we end up BUNDLing even if the user dint want to</div>
<div><br>
</div>
<div>On the other hand, if we allowed mid value to be changed, we might end=
 up UNBundling a video stream , when the user did want to BUNDLE them</div>
<div><br>
</div>
<div>So allowing a mid value to be changed or not should be tied to what th=
e JS API is asking on behalf of the user</div>
</div>
<div><br>
</div>
<div><br>
</div>
<div>Any thoughts ?</div>
<div><br>
</div>
<div>Cheers</div>
<div>Suhas</div>
<div><br>
</div>
<div><br>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</div>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1C4A800EESESSMB209erics_--

From suhasietf@gmail.com  Thu Sep 19 13:55:41 2013
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From: Suhas Nandakumar <suhasietf@gmail.com>
To: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] JSEP-04: Nits and niggles
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Regarding a=ssrc comment above,
The text from RFC5576 says this

Each "ssrc" media attribute specifies a single source-level attribute
   for the given <ssrc-id>.  For each source mentioned in SDP, the
   source-level attribute "cname", defined in Section 6.1
<http://tools.ietf.org/html/rfc5576#section-6.1>, MUST be
   provided.  Any number of other source-level attributes for the source
   MAY also be provided.



given the above, would it be cname attribute for a given ssrc that
should be mandated at the minimum ?



Cheers

Suhas





On Thu, Sep 19, 2013 at 5:54 AM, Harald Alvestrand <harald@alvestrand.no>wrote:

> I'm sure we'll have lots of things to discuss about JSEP. I just thought
> I'd get these out of the way.
>
> - Usage of MSID. In addition to the a=msid: lines, the session-level
> generation procedure needs to say that a line "a=msid-semantic:WMS <id>" is
> added to the session-level description, containing the IDs of the
> MediaStreams that are signalled.
>
> - a=ssrc lines. The spec (5.2.1 second list last bullet) simply says "an
> a=ssrc line". But RFC 5576 specifies that all ssrc lines contain an
> attribute. Suggestion: define one that does no harm (role=primary?)
>
> - recycling of m= lines. The section in 5.2.2 (subsequent offers) that
> talks about this doesn't mention that the media in the m-line has to match,
> although section 5.3.1 (initial answer) actually talks about it. I'm
> assuming that's an oversight.
>
> I'd suggest pulling the language about "selecting an appropriate m-line
> for a track" out in a separate subsection, so that we have one place to
> point to for this.
>
> Out of time, let's get these ones out of the way for -05....
>
>
>
> ______________________________**_________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/mailman/listinfo/rtcweb>
>

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<div dir=3D"ltr"><div>Regarding a=3Dssrc comment above,</div>The text from =
RFC5576 says this=A0<div><br></div><div><pre class=3D"" style=3D"font-size:=
1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">Each &quot;ssrc&quot=
; media attribute specifies a single source-level attribute
   for the given &lt;ssrc-id&gt;.  For each source mentioned in SDP, the
   source-level attribute &quot;cname&quot;, defined in <a href=3D"http://t=
ools.ietf.org/html/rfc5576#section-6.1">Section 6.1</a>, MUST be
   provided.  Any number of other source-level attributes for the source
   MAY also be provided.</pre><pre class=3D"" style=3D"font-size:1em;margin=
-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><br></pre><pre class=3D"" styl=
e=3D"font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><br><=
/pre>
<pre class=3D"" style=3D"margin-top:0px;margin-bottom:0px"><font face=3D"ar=
ial"><span style=3D"white-space:normal">given the above, would it be cname =
attribute for a given ssrc that should be mandated at the minimum ?</span><=
/font></pre>
<pre class=3D"" style=3D"margin-top:0px;margin-bottom:0px"><font face=3D"ar=
ial"><span style=3D"white-space:normal"><br></span></font></pre><pre class=
=3D"" style=3D"margin-top:0px;margin-bottom:0px"><font face=3D"arial"><span=
 style=3D"white-space:normal"><br>
</span></font></pre><pre class=3D"" style=3D"margin-top:0px;margin-bottom:0=
px"><font face=3D"arial"><span style=3D"white-space:normal">Cheers</span></=
font></pre><pre class=3D"" style=3D"margin-top:0px;margin-bottom:0px"><font=
 face=3D"arial"><span style=3D"white-space:normal">Suhas</span></font></pre=
>
<pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;col=
or:rgb(0,0,0)"><br></pre><pre class=3D"" style=3D"font-size:1em;margin-top:=
0px;margin-bottom:0px;color:rgb(0,0,0)"><br></pre></div></div><div class=3D=
"gmail_extra">
<br><br><div class=3D"gmail_quote">On Thu, Sep 19, 2013 at 5:54 AM, Harald =
Alvestrand <span dir=3D"ltr">&lt;<a href=3D"mailto:harald@alvestrand.no" ta=
rget=3D"_blank">harald@alvestrand.no</a>&gt;</span> wrote:<br><blockquote c=
lass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;=
padding-left:1ex">
I&#39;m sure we&#39;ll have lots of things to discuss about JSEP. I just th=
ought I&#39;d get these out of the way.<br>
<br>
- Usage of MSID. In addition to the a=3Dmsid: lines, the session-level gene=
ration procedure needs to say that a line &quot;a=3Dmsid-semantic:WMS &lt;i=
d&gt;&quot; is added to the session-level description, containing the IDs o=
f the MediaStreams that are signalled.<br>

<br>
- a=3Dssrc lines. The spec (5.2.1 second list last bullet) simply says &quo=
t;an a=3Dssrc line&quot;. But RFC 5576 specifies that all ssrc lines contai=
n an attribute. Suggestion: define one that does no harm (role=3Dprimary?)<=
br>

<br>
- recycling of m=3D lines. The section in 5.2.2 (subsequent offers) that ta=
lks about this doesn&#39;t mention that the media in the m-line has to matc=
h, although section 5.3.1 (initial answer) actually talks about it. I&#39;m=
 assuming that&#39;s an oversight.<br>

<br>
I&#39;d suggest pulling the language about &quot;selecting an appropriate m=
-line for a track&quot; out in a separate subsection, so that we have one p=
lace to point to for this.<br>
<br>
Out of time, let&#39;s get these ones out of the way for -05....<br>
<br>
<br>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</blockquote></div><br></div>

--e89a8f2353a72d50c904e6c2c64e--

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From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] JSEP-04: Nits and niggles
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On 09/19/2013 10:55 PM, Suhas Nandakumar wrote:
> Regarding a=ssrc comment above,
> The text from RFC5576 says this
>
> Each "ssrc" media attribute specifies a single source-level attribute
>     for the given <ssrc-id>.  For each source mentioned in SDP, the
>     source-level attribute "cname", defined inSection 6.1  <http://tools.ietf.org/html/rfc5576#section-6.1>, MUST be
>     provided.  Any number of other source-level attributes for the source
>     MAY also be provided.
> given the above, would it be cname attribute for a given ssrc that should be mandated at the minimum ?

Thanks, that makes sense to me. Updating JSEP to say "add a line with 
a=ssrc cname=<cname>" sounds like something that does what we want and 
is consistent with 5576.
>
>
>
> Cheers
> Suhas
>
>
> On Thu, Sep 19, 2013 at 5:54 AM, Harald Alvestrand 
> <harald@alvestrand.no <mailto:harald@alvestrand.no>> wrote:
>
>     I'm sure we'll have lots of things to discuss about JSEP. I just
>     thought I'd get these out of the way.
>
>     - Usage of MSID. In addition to the a=msid: lines, the
>     session-level generation procedure needs to say that a line
>     "a=msid-semantic:WMS <id>" is added to the session-level
>     description, containing the IDs of the MediaStreams that are
>     signalled.
>
>     - a=ssrc lines. The spec (5.2.1 second list last bullet) simply
>     says "an a=ssrc line". But RFC 5576 specifies that all ssrc lines
>     contain an attribute. Suggestion: define one that does no harm
>     (role=primary?)
>
>     - recycling of m= lines. The section in 5.2.2 (subsequent offers)
>     that talks about this doesn't mention that the media in the m-line
>     has to match, although section 5.3.1 (initial answer) actually
>     talks about it. I'm assuming that's an oversight.
>
>     I'd suggest pulling the language about "selecting an appropriate
>     m-line for a track" out in a separate subsection, so that we have
>     one place to point to for this.
>
>     Out of time, let's get these ones out of the way for -05....
>
>
>
>     _______________________________________________
>     rtcweb mailing list
>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     https://www.ietf.org/mailman/listinfo/rtcweb
>
>


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<html>
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    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    <div class="moz-cite-prefix">On 09/19/2013 10:55 PM, Suhas
      Nandakumar wrote:<br>
    </div>
    <blockquote
cite="mid:CAMRcRGQXeH7D+u2g-jdoWfHJ+FmE+mgpAztmsyArZNS0KAz=Fw@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div>Regarding a=ssrc comment above,</div>
        The text from RFC5576 says this&nbsp;
        <div><br>
        </div>
        <div>
          <pre class="" style="font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">Each "ssrc" media attribute specifies a single source-level attribute
   for the given &lt;ssrc-id&gt;.  For each source mentioned in SDP, the
   source-level attribute "cname", defined in <a moz-do-not-send="true" href="http://tools.ietf.org/html/rfc5576#section-6.1">Section 6.1</a>, MUST be
   provided.  Any number of other source-level attributes for the source
   MAY also be provided.</pre>
          <pre class="" style="font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">
</pre>
          <pre class="" style="font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">
</pre>
          <pre class="" style="margin-top:0px;margin-bottom:0px"><font face="arial"><span style="white-space:normal">given the above, would it be cname attribute for a given ssrc that should be mandated at the minimum ?</span></font></pre>
        </div>
      </div>
    </blockquote>
    <br>
    <font face="arial">Thanks, that makes sense to me. Updating JSEP to
      say "add a line with a=ssrc cname=&lt;cname&gt;" sounds like
      something that does what we want and is consistent with 5576.</font><br>
    <blockquote
cite="mid:CAMRcRGQXeH7D+u2g-jdoWfHJ+FmE+mgpAztmsyArZNS0KAz=Fw@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div>
          <pre class="" style="margin-top:0px;margin-bottom:0px"><font face="arial"><span style="white-space:normal">
</span></font></pre>
          <pre class="" style="margin-top:0px;margin-bottom:0px"><font face="arial"><span style="white-space:normal">

</span></font></pre>
          <pre class="" style="margin-top:0px;margin-bottom:0px"><font face="arial"><span style="white-space:normal">Cheers</span></font></pre>
          <pre class="" style="margin-top:0px;margin-bottom:0px"><font face="arial"><span style="white-space:normal">Suhas</span></font></pre>
          <pre class="" style="font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">
</pre>
          <pre class="" style="font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">
</pre>
        </div>
      </div>
      <div class="gmail_extra">
        <br>
        <br>
        <div class="gmail_quote">On Thu, Sep 19, 2013 at 5:54 AM, Harald
          Alvestrand <span dir="ltr">&lt;<a moz-do-not-send="true"
              href="mailto:harald@alvestrand.no" target="_blank">harald@alvestrand.no</a>&gt;</span>
          wrote:<br>
          <blockquote class="gmail_quote" style="margin:0 0 0
            .8ex;border-left:1px #ccc solid;padding-left:1ex">
            I'm sure we'll have lots of things to discuss about JSEP. I
            just thought I'd get these out of the way.<br>
            <br>
            - Usage of MSID. In addition to the a=msid: lines, the
            session-level generation procedure needs to say that a line
            "a=msid-semantic:WMS &lt;id&gt;" is added to the
            session-level description, containing the IDs of the
            MediaStreams that are signalled.<br>
            <br>
            - a=ssrc lines. The spec (5.2.1 second list last bullet)
            simply says "an a=ssrc line". But RFC 5576 specifies that
            all ssrc lines contain an attribute. Suggestion: define one
            that does no harm (role=primary?)<br>
            <br>
            - recycling of m= lines. The section in 5.2.2 (subsequent
            offers) that talks about this doesn't mention that the media
            in the m-line has to match, although section 5.3.1 (initial
            answer) actually talks about it. I'm assuming that's an
            oversight.<br>
            <br>
            I'd suggest pulling the language about "selecting an
            appropriate m-line for a track" out in a separate
            subsection, so that we have one place to point to for this.<br>
            <br>
            Out of time, let's get these ones out of the way for -05....<br>
            <br>
            <br>
            <br>
            _______________________________________________<br>
            rtcweb mailing list<br>
            <a moz-do-not-send="true" href="mailto:rtcweb@ietf.org"
              target="_blank">rtcweb@ietf.org</a><br>
            <a moz-do-not-send="true"
              href="https://www.ietf.org/mailman/listinfo/rtcweb"
              target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
          </blockquote>
        </div>
        <br>
      </div>
    </blockquote>
    <br>
  </body>
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Date: Fri, 20 Sep 2013 00:23:56 +0200
From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] m=section recycling and a=mid
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I suggest that changing the MID value can be very painful, since it 
modifies the mapping between m-line index and MID.

Is there any other spec that describes changing the MID value?

If not, I'd suggest that we explicitly forbid changing the MID value. If 
we want to add or remove an M-line from a bundle, we should update the 
bundles instead.


On 09/19/2013 08:52 PM, Christer Holmberg wrote:
>
> Correct.
>
> Regards,
>
> Christer
>
> ------------------------------------------------------------------------
> *From:* Suhas Nandakumar [suhasietf@gmail.com]
> *Sent:* Thursday, 19 September 2013 9:48 PM
> *To:* Christer Holmberg
> *Cc:* rtcweb@ietf.org
> *Subject:* Re: [rtcweb] m=section recycling and a=mid
>
> Hello Christer
>
>   Completely agree with you
>
> My point with the above examples was
>
>   "recycling of m=line should take into account its existing BUNDLE 
> membership and JS user's preference on requiring the added media track 
> to be BUNDLED or not"
>
> This would imply, as suggested by you,
>
>   - remove the mid from a=group:BUNDLE line if the JS user doesn't 
> want the new track to be BUNDLEd
>
>  - or if the mid value is changed, then add the new mid value to the 
> BUNDLE group if the JS user want's the new track to be BUNDLED
>
> - or just retain the mid value and its BUNDLE membership for the new 
> track, if it is going to be BUNDLEd
>
>
> Cheers
> Suhas
>
>
>
>
>
>
>
> On Thu, Sep 19, 2013 at 9:24 AM, Christer Holmberg 
> <christer.holmberg@ericsson.com 
> <mailto:christer.holmberg@ericsson.com>> wrote:
>
>     Hi Suhas,
>
>     The mid attribute itself does not add an m- line to a BUNDLE
>     group. The attribute value also needs to be listed in the
>     group:BUNDLE attribute mid list.
>
>     So, if you want to re-use an m- line and mid attribute, but you
>     don't want the m- line to be in the BUNDLE group, simply remove
>     the mid from the mid list.
>
>     Regards,
>
>     Christer
>
>     ------------------------------------------------------------------------
>     *From:* rtcweb-bounces@ietf.org <mailto:rtcweb-bounces@ietf.org>
>     [rtcweb-bounces@ietf.org <mailto:rtcweb-bounces@ietf.org>] on
>     behalf of Suhas Nandakumar [suhasietf@gmail.com
>     <mailto:suhasietf@gmail.com>]
>     *Sent:* Wednesday, 18 September 2013 7:44 PM
>     *To:* rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     *Subject:* [rtcweb] m=section recycling and a=mid
>
>     After thinking about it a bit, I think there are some side effects
>     associated with changing/not changing mid attribute of a re-cycled
>     m= section and it being part of a BUNDLE group or not..
>
>     example
>
>     say , we have these video m=section part of BUNDLE group as below
>     a=group:BUNDLE 1, 2
>
>     m=video .....
>     a=mid:1
>     a=inactive
>
>     m=video .....
>     a=mid:2
>     a=sendrecv
>
>     Now If JS user adds a new video track and he wants it to be out of
>     BUNDLE. If we recycle first video m=section and dont change mid
>     attribute,  we end up BUNDLing even if the user dint want to
>
>     On the other hand, if we allowed mid value to be changed, we might
>     end up UNBundling a video stream , when the user did want to
>     BUNDLE them
>
>     So allowing a mid value to be changed or not should be tied to
>     what the JS API is asking on behalf of the user
>
>
>     Any thoughts ?
>
>     Cheers
>     Suhas
>
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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    <div class="moz-cite-prefix">I suggest that changing the MID value
      can be very painful, since it modifies the mapping between m-line
      index and MID.<br>
      <br>
      Is there any other spec that describes changing the MID value?<br>
      <br>
      If not, I'd suggest that we explicitly forbid changing the MID
      value. If we want to add or remove an M-line from a bundle, we
      should update the bundles instead.<br>
      <br>
      <br>
      On 09/19/2013 08:52 PM, Christer Holmberg wrote:<br>
    </div>
    <blockquote
cite="mid:7594FB04B1934943A5C02806D1A2204B1C4A800E@ESESSMB209.ericsson.se"
      type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=ISO-8859-1">
      <style id="owaParaStyle">P {
	MARGIN-BOTTOM: 0px; MARGIN-TOP: 0px
}
</style>
      <div style="direction: ltr;font-family: Tahoma;color:
        #000000;font-size: 10pt;">
        <p>Correct.</p>
        <p>&nbsp;</p>
        <p>Regards,</p>
        <p>&nbsp;</p>
        <p>Christer</p>
        <div style="FONT-SIZE: 16px; FONT-FAMILY: Times New Roman;
          COLOR: #000000">
          <hr tabindex="-1">
          <div id="divRpF925856" style="DIRECTION: ltr"><font
              color="#000000" face="Tahoma" size="2"><b>From:</b> Suhas
              Nandakumar [<a class="moz-txt-link-abbreviated" href="mailto:suhasietf@gmail.com">suhasietf@gmail.com</a>]<br>
              <b>Sent:</b> Thursday, 19 September 2013 9:48 PM<br>
              <b>To:</b> Christer Holmberg<br>
              <b>Cc:</b> <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
              <b>Subject:</b> Re: [rtcweb] m=section recycling and a=mid<br>
            </font><br>
          </div>
          <div>
            <div dir="ltr">Hello Christer
              <div><br>
              </div>
              <div>&nbsp; Completely agree with you</div>
              <div><br>
              </div>
              <div>My point with the above examples was&nbsp;</div>
              <div><br>
              </div>
              <div>&nbsp; "recycling of m=line should take into account its
                existing BUNDLE membership and JS user's preference on
                requiring the added media track to be BUNDLED or not"</div>
              <div><br>
              </div>
              <div>This would imply, as suggested by you,</div>
              <div><br>
              </div>
              <div>&nbsp; - remove the mid from a=group:BUNDLE line if the JS
                user doesn't want the new track to be BUNDLEd</div>
              <div><br>
              </div>
              <div>&nbsp;- or if the mid value is changed, then add the new
                mid value to the BUNDLE group if the JS user want's the
                new track to be BUNDLED</div>
              <div><br>
              </div>
              <div>- or just retain the mid value and its BUNDLE
                membership for the new track, if it is going to be
                BUNDLEd</div>
              <div><br>
              </div>
              <div><br>
              </div>
              <div>Cheers</div>
              <div>Suhas</div>
              <div><br>
              </div>
              <div><br>
              </div>
              <div><br>
              </div>
              <div><br>
              </div>
              <div><br>
              </div>
            </div>
            <div class="gmail_extra"><br>
              <br>
              <div class="gmail_quote">On Thu, Sep 19, 2013 at 9:24 AM,
                Christer Holmberg <span dir="ltr">
                  &lt;<a moz-do-not-send="true"
                    href="mailto:christer.holmberg@ericsson.com"
                    target="_blank">christer.holmberg@ericsson.com</a>&gt;</span>
                wrote:<br>
                <blockquote class="gmail_quote" style="PADDING-LEFT:
                  1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px
                  solid">
                  <div>
                    <div style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma;
                      DIRECTION: ltr">
                      <p>Hi Suhas,</p>
                      <p>&nbsp;</p>
                      <p>The mid attribute itself does not add an m-
                        line to a BUNDLE group. The attribute value also
                        needs to be listed in the group:BUNDLE attribute
                        mid list.</p>
                      <p>&nbsp;</p>
                      <p>So, if you want to re-use an m- line and mid
                        attribute, but you don't want the m- line to be
                        in the BUNDLE group, simply remove the mid from
                        the mid list.</p>
                      <p>&nbsp;</p>
                      <p>Regards,</p>
                      <p>&nbsp;</p>
                      <p>Christer</p>
                      <p>&nbsp;</p>
                      <div style="FONT-SIZE: 16px; FONT-FAMILY: Times
                        New Roman">
                        <hr>
                        <div style="DIRECTION: ltr"><font
                            color="#000000" face="Tahoma"><b>From:</b> <a
                              moz-do-not-send="true"
                              href="mailto:rtcweb-bounces@ietf.org"
                              target="_blank">
                              rtcweb-bounces@ietf.org</a> [<a
                              moz-do-not-send="true"
                              href="mailto:rtcweb-bounces@ietf.org"
                              target="_blank">rtcweb-bounces@ietf.org</a>]
                            on behalf of Suhas Nandakumar [<a
                              moz-do-not-send="true"
                              href="mailto:suhasietf@gmail.com"
                              target="_blank">suhasietf@gmail.com</a>]<br>
                            <b>Sent:</b> Wednesday, 18 September 2013
                            7:44 PM<br>
                            <b>To:</b> <a moz-do-not-send="true"
                              href="mailto:rtcweb@ietf.org"
                              target="_blank">rtcweb@ietf.org</a><br>
                            <b>Subject:</b> [rtcweb] m=section recycling
                            and a=mid<br>
                          </font><br>
                        </div>
                        <div>
                          <div class="h5">
                            <div>
                              <div dir="ltr">After thinking about it a
                                bit, I think there are some side effects
                                associated with changing/not changing
                                mid attribute of a re-cycled m= section
                                and it being part of a BUNDLE group or
                                not..
                                <div><br>
                                </div>
                                <div>example</div>
                                <div><br>
                                </div>
                                <div>say , we have these video m=section
                                  part of BUNDLE group as below&nbsp;</div>
                                <div>a=group:BUNDLE 1, 2</div>
                                <div><br>
                                </div>
                                <div>m=video .....</div>
                                <div>a=mid:1&nbsp;</div>
                                <div>a=inactive</div>
                                <div><br>
                                </div>
                                <div>
                                  <div>m=video .....</div>
                                  <div>a=mid:2&nbsp;</div>
                                  <div>a=sendrecv</div>
                                  <div><br>
                                  </div>
                                  <div>Now If JS user adds a new video
                                    track and he wants it to be out of
                                    BUNDLE. If we recycle first video
                                    m=section and dont change mid
                                    attribute, &nbsp;we end up BUNDLing even
                                    if the user dint want to</div>
                                  <div><br>
                                  </div>
                                  <div>On the other hand, if we allowed
                                    mid value to be changed, we might
                                    end up UNBundling a video stream ,
                                    when the user did want to BUNDLE
                                    them</div>
                                  <div><br>
                                  </div>
                                  <div>So allowing a mid value to be
                                    changed or not should be tied to
                                    what the JS API is asking on behalf
                                    of the user</div>
                                </div>
                                <div><br>
                                </div>
                                <div><br>
                                </div>
                                <div>Any thoughts ?</div>
                                <div><br>
                                </div>
                                <div>Cheers</div>
                                <div>Suhas</div>
                                <div><br>
                                </div>
                                <div><br>
                                </div>
                              </div>
                            </div>
                          </div>
                        </div>
                      </div>
                    </div>
                  </div>
                </blockquote>
              </div>
              <br>
            </div>
          </div>
        </div>
      </div>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
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References: <CAMRcRGRUufCBRg77DEEcqSvJ8uP2o2HU2GoJ23xu6wKPc9G3zA@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1C4A7F29@ESESSMB209.ericsson.se> <CAMRcRGT5T6FQSxgot4HObMW2r+z9id+BsXd0O8ZEauZunNmtzw@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1C4A800E@ESESSMB209.ericsson.se> <523B797C.7040603@alvestrand.no>
Date: Thu, 19 Sep 2013 15:25:09 -0700
Message-ID: <CAMRcRGSVwTfCK_znni5K29e3UEP_r7COPTyB23mSJ5vzWhLOAw@mail.gmail.com>
From: Suhas Nandakumar <suhasietf@gmail.com>
To: Harald Alvestrand <harald@alvestrand.no>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] m=section recycling and a=mid
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+1


On Thu, Sep 19, 2013 at 3:23 PM, Harald Alvestrand <harald@alvestrand.no>wrote:

>  I suggest that changing the MID value can be very painful, since it
> modifies the mapping between m-line index and MID.
>
> Is there any other spec that describes changing the MID value?
>
> If not, I'd suggest that we explicitly forbid changing the MID value. If
> we want to add or remove an M-line from a bundle, we should update the
> bundles instead.
>
>
>
> On 09/19/2013 08:52 PM, Christer Holmberg wrote:
>
>  Correct.
>
>
>
> Regards,
>
>
>
> Christer
>  ------------------------------
> *From:* Suhas Nandakumar [suhasietf@gmail.com]
> *Sent:* Thursday, 19 September 2013 9:48 PM
> *To:* Christer Holmberg
> *Cc:* rtcweb@ietf.org
> *Subject:* Re: [rtcweb] m=section recycling and a=mid
>
>  Hello Christer
>
>    Completely agree with you
>
>  My point with the above examples was
>
>    "recycling of m=line should take into account its existing BUNDLE
> membership and JS user's preference on requiring the added media track to
> be BUNDLED or not"
>
>  This would imply, as suggested by you,
>
>    - remove the mid from a=group:BUNDLE line if the JS user doesn't want
> the new track to be BUNDLEd
>
>   - or if the mid value is changed, then add the new mid value to the
> BUNDLE group if the JS user want's the new track to be BUNDLED
>
>  - or just retain the mid value and its BUNDLE membership for the new
> track, if it is going to be BUNDLEd
>
>
>  Cheers
> Suhas
>
>
>
>
>
>
>
> On Thu, Sep 19, 2013 at 9:24 AM, Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
>
>>  Hi Suhas,
>>
>>
>>
>> The mid attribute itself does not add an m- line to a BUNDLE group. The
>> attribute value also needs to be listed in the group:BUNDLE attribute mid
>> list.
>>
>>
>>
>> So, if you want to re-use an m- line and mid attribute, but you don't
>> want the m- line to be in the BUNDLE group, simply remove the mid from the
>> mid list.
>>
>>
>>
>> Regards,
>>
>>
>>
>> Christer
>>
>>
>>  ------------------------------
>> *From:* rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] on behalf of
>> Suhas Nandakumar [suhasietf@gmail.com]
>> *Sent:* Wednesday, 18 September 2013 7:44 PM
>> *To:* rtcweb@ietf.org
>> *Subject:* [rtcweb] m=section recycling and a=mid
>>
>>   After thinking about it a bit, I think there are some side effects
>> associated with changing/not changing mid attribute of a re-cycled m=
>> section and it being part of a BUNDLE group or not..
>>
>>  example
>>
>>  say , we have these video m=section part of BUNDLE group as below
>> a=group:BUNDLE 1, 2
>>
>>  m=video .....
>> a=mid:1
>> a=inactive
>>
>>  m=video .....
>> a=mid:2
>> a=sendrecv
>>
>>  Now If JS user adds a new video track and he wants it to be out of
>> BUNDLE. If we recycle first video m=section and dont change mid attribute,
>>  we end up BUNDLing even if the user dint want to
>>
>>  On the other hand, if we allowed mid value to be changed, we might end
>> up UNBundling a video stream , when the user did want to BUNDLE them
>>
>>  So allowing a mid value to be changed or not should be tied to what the
>> JS API is asking on behalf of the user
>>
>>
>>  Any thoughts ?
>>
>>  Cheers
>> Suhas
>>
>>
>>
>
>
> _______________________________________________
> rtcweb mailing listrtcweb@ietf.orghttps://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--f46d04389087d5f9c504e6c406be
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">+1</div><div class=3D"gmail_extra"><br><br><div class=3D"g=
mail_quote">On Thu, Sep 19, 2013 at 3:23 PM, Harald Alvestrand <span dir=3D=
"ltr">&lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@=
alvestrand.no</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
 =20
   =20
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000">
    <div>I suggest that changing the MID value
      can be very painful, since it modifies the mapping between m-line
      index and MID.<br>
      <br>
      Is there any other spec that describes changing the MID value?<br>
      <br>
      If not, I&#39;d suggest that we explicitly forbid changing the MID
      value. If we want to add or remove an M-line from a bundle, we
      should update the bundles instead.<div><div class=3D"h5"><br>
      <br>
      <br>
      On 09/19/2013 08:52 PM, Christer Holmberg wrote:<br>
    </div></div></div>
    <blockquote type=3D"cite"><div><div class=3D"h5">
     =20
     =20
      <div style=3D"direction:ltr;font-size:10pt;font-family:Tahoma">
        <p>Correct.</p>
        <p>=A0</p>
        <p>Regards,</p>
        <p>=A0</p>
        <p>Christer</p>
        <div style=3D"font-size:16px;font-family:Times New Roman">
          <hr>
          <div style=3D"DIRECTION:ltr"><font color=3D"#000000" face=3D"Taho=
ma"><b>From:</b> Suhas
              Nandakumar [<a href=3D"mailto:suhasietf@gmail.com" target=3D"=
_blank">suhasietf@gmail.com</a>]<br>
              <b>Sent:</b> Thursday, 19 September 2013 9:48 PM<br>
              <b>To:</b> Christer Holmberg<br>
              <b>Cc:</b> <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blan=
k">rtcweb@ietf.org</a><br>
              <b>Subject:</b> Re: [rtcweb] m=3Dsection recycling and a=3Dmi=
d<br>
            </font><br>
          </div>
          <div>
            <div dir=3D"ltr">Hello Christer
              <div><br>
              </div>
              <div>=A0 Completely agree with you</div>
              <div><br>
              </div>
              <div>My point with the above examples was=A0</div>
              <div><br>
              </div>
              <div>=A0 &quot;recycling of m=3Dline should take into account=
 its
                existing BUNDLE membership and JS user&#39;s preference on
                requiring the added media track to be BUNDLED or not&quot;<=
/div>
              <div><br>
              </div>
              <div>This would imply, as suggested by you,</div>
              <div><br>
              </div>
              <div>=A0 - remove the mid from a=3Dgroup:BUNDLE line if the J=
S
                user doesn&#39;t want the new track to be BUNDLEd</div>
              <div><br>
              </div>
              <div>=A0- or if the mid value is changed, then add the new
                mid value to the BUNDLE group if the JS user want&#39;s the
                new track to be BUNDLED</div>
              <div><br>
              </div>
              <div>- or just retain the mid value and its BUNDLE
                membership for the new track, if it is going to be
                BUNDLEd</div>
              <div><br>
              </div>
              <div><br>
              </div>
              <div>Cheers</div>
              <div>Suhas</div>
              <div><br>
              </div>
              <div><br>
              </div>
              <div><br>
              </div>
              <div><br>
              </div>
              <div><br>
              </div>
            </div>
            <div class=3D"gmail_extra"><br>
              <br>
              <div class=3D"gmail_quote">On Thu, Sep 19, 2013 at 9:24 AM,
                Christer Holmberg <span dir=3D"ltr">
                  &lt;<a href=3D"mailto:christer.holmberg@ericsson.com" tar=
get=3D"_blank">christer.holmberg@ericsson.com</a>&gt;</span>
                wrote:<br>
                <blockquote class=3D"gmail_quote" style=3D"PADDING-LEFT:1ex=
;MARGIN:0px 0px 0px 0.8ex;BORDER-LEFT:#ccc 1px solid">
                  <div>
                    <div style=3D"FONT-SIZE:10pt;FONT-FAMILY:Tahoma;DIRECTI=
ON:ltr">
                      <p>Hi Suhas,</p>
                      <p>=A0</p>
                      <p>The mid attribute itself does not add an m-
                        line to a BUNDLE group. The attribute value also
                        needs to be listed in the group:BUNDLE attribute
                        mid list.</p>
                      <p>=A0</p>
                      <p>So, if you want to re-use an m- line and mid
                        attribute, but you don&#39;t want the m- line to be
                        in the BUNDLE group, simply remove the mid from
                        the mid list.</p>
                      <p>=A0</p>
                      <p>Regards,</p>
                      <p>=A0</p>
                      <p>Christer</p>
                      <p>=A0</p>
                      <div style=3D"FONT-SIZE:16px;FONT-FAMILY:Times New Ro=
man">
                        <hr>
                        <div style=3D"DIRECTION:ltr"><font color=3D"#000000=
" face=3D"Tahoma"><b>From:</b> <a href=3D"mailto:rtcweb-bounces@ietf.org" t=
arget=3D"_blank">
                              rtcweb-bounces@ietf.org</a> [<a href=3D"mailt=
o:rtcweb-bounces@ietf.org" target=3D"_blank">rtcweb-bounces@ietf.org</a>]
                            on behalf of Suhas Nandakumar [<a href=3D"mailt=
o:suhasietf@gmail.com" target=3D"_blank">suhasietf@gmail.com</a>]<br>
                            <b>Sent:</b> Wednesday, 18 September 2013
                            7:44 PM<br>
                            <b>To:</b> <a href=3D"mailto:rtcweb@ietf.org" t=
arget=3D"_blank">rtcweb@ietf.org</a><br>
                            <b>Subject:</b> [rtcweb] m=3Dsection recycling
                            and a=3Dmid<br>
                          </font><br>
                        </div>
                        <div>
                          <div>
                            <div>
                              <div dir=3D"ltr">After thinking about it a
                                bit, I think there are some side effects
                                associated with changing/not changing
                                mid attribute of a re-cycled m=3D section
                                and it being part of a BUNDLE group or
                                not..
                                <div><br>
                                </div>
                                <div>example</div>
                                <div><br>
                                </div>
                                <div>say , we have these video m=3Dsection
                                  part of BUNDLE group as below=A0</div>
                                <div>a=3Dgroup:BUNDLE 1, 2</div>
                                <div><br>
                                </div>
                                <div>m=3Dvideo .....</div>
                                <div>a=3Dmid:1=A0</div>
                                <div>a=3Dinactive</div>
                                <div><br>
                                </div>
                                <div>
                                  <div>m=3Dvideo .....</div>
                                  <div>a=3Dmid:2=A0</div>
                                  <div>a=3Dsendrecv</div>
                                  <div><br>
                                  </div>
                                  <div>Now If JS user adds a new video
                                    track and he wants it to be out of
                                    BUNDLE. If we recycle first video
                                    m=3Dsection and dont change mid
                                    attribute, =A0we end up BUNDLing even
                                    if the user dint want to</div>
                                  <div><br>
                                  </div>
                                  <div>On the other hand, if we allowed
                                    mid value to be changed, we might
                                    end up UNBundling a video stream ,
                                    when the user did want to BUNDLE
                                    them</div>
                                  <div><br>
                                  </div>
                                  <div>So allowing a mid value to be
                                    changed or not should be tied to
                                    what the JS API is asking on behalf
                                    of the user</div>
                                </div>
                                <div><br>
                                </div>
                                <div><br>
                                </div>
                                <div>Any thoughts ?</div>
                                <div><br>
                                </div>
                                <div>Cheers</div>
                                <div>Suhas</div>
                                <div><br>
                                </div>
                                <div><br>
                                </div>
                              </div>
                            </div>
                          </div>
                        </div>
                      </div>
                    </div>
                  </div>
                </blockquote>
              </div>
              <br>
            </div>
          </div>
        </div>
      </div>
      <br>
      <fieldset></fieldset>
      <br>
      </div></div><pre>_______________________________________________
rtcweb mailing list
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
  </div>

<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

--f46d04389087d5f9c504e6c406be--

From magnus.westerlund@ericsson.com  Thu Sep 19 23:01:50 2013
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Date: Fri, 20 Sep 2013 08:02:03 +0200
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: [rtcweb] JSEP and video stream properties selection (a=imageattr usage)
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WG,

In the JSEP call I commented regarding a=imageattr open issue that we
had an original agreement from over a year ago (Vancouver) to go with
Harald's proposal in then
http://tools.ietf.org/id/draft-alvestrand-rtcweb-resolution-00.txt

This is documented in the minutes:
http://www.ietf.org/proceedings/84/minutes/minutes-84-rtcweb


However, I did forget to note that there has been discussion on this
issue after that between Harald Alvestrand and Stefan Håkansson
primarily on the W3C list

http://lists.w3.org/Archives/Public/public-webrtc/2013Aug/0051.html

Where they are seriously discussing only using WebRTC API for this.

It is in the context of
https://datatracker.ietf.org/doc/draft-alvestrand-constraints-resolution/

Thus I would like to point out that there appear to be some indication
of desire move away from the consensus last year.

Thus it appears that this topic may require some further discussion.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From christer.holmberg@ericsson.com  Thu Sep 19 23:47:47 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Suhas Nandakumar <suhasietf@gmail.com>, Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: [rtcweb] m=section recycling and a=mid
Thread-Index: AQHOtI5OlzgN5l68MU+Yf86f9mtTsJnNP/0ZgAAG7ICAACKcC4AAGbIAgAAAV4CAAK3l0A==
Date: Fri, 20 Sep 2013 06:47:37 +0000
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] m=section recycling and a=mid
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+1

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Suhas Nandakumar
Sent: 20. syyskuuta 2013 1:25
To: Harald Alvestrand
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] m=3Dsection recycling and a=3Dmid

+1

On Thu, Sep 19, 2013 at 3:23 PM, Harald Alvestrand <harald@alvestrand.no<ma=
ilto:harald@alvestrand.no>> wrote:
I suggest that changing the MID value can be very painful, since it modifie=
s the mapping between m-line index and MID.

Is there any other spec that describes changing the MID value?

If not, I'd suggest that we explicitly forbid changing the MID value. If we=
 want to add or remove an M-line from a bundle, we should update the bundle=
s instead.



On 09/19/2013 08:52 PM, Christer Holmberg wrote:

Correct.



Regards,



Christer

________________________________
From: Suhas Nandakumar [suhasietf@gmail.com<mailto:suhasietf@gmail.com>]
Sent: Thursday, 19 September 2013 9:48 PM
To: Christer Holmberg
Cc: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
Subject: Re: [rtcweb] m=3Dsection recycling and a=3Dmid
Hello Christer

  Completely agree with you

My point with the above examples was

  "recycling of m=3Dline should take into account its existing BUNDLE membe=
rship and JS user's preference on requiring the added media track to be BUN=
DLED or not"

This would imply, as suggested by you,

  - remove the mid from a=3Dgroup:BUNDLE line if the JS user doesn't want t=
he new track to be BUNDLEd

 - or if the mid value is changed, then add the new mid value to the BUNDLE=
 group if the JS user want's the new track to be BUNDLED

- or just retain the mid value and its BUNDLE membership for the new track,=
 if it is going to be BUNDLEd


Cheers
Suhas






On Thu, Sep 19, 2013 at 9:24 AM, Christer Holmberg <christer.holmberg@erics=
son.com<mailto:christer.holmberg@ericsson.com>> wrote:

Hi Suhas,



The mid attribute itself does not add an m- line to a BUNDLE group. The att=
ribute value also needs to be listed in the group:BUNDLE attribute mid list=
.



So, if you want to re-use an m- line and mid attribute, but you don't want =
the m- line to be in the BUNDLE group, simply remove the mid from the mid l=
ist.



Regards,



Christer



________________________________
From: rtcweb-bounces@ietf.org<mailto:rtcweb-bounces@ietf.org> [rtcweb-bounc=
es@ietf.org<mailto:rtcweb-bounces@ietf.org>] on behalf of Suhas Nandakumar =
[suhasietf@gmail.com<mailto:suhasietf@gmail.com>]
Sent: Wednesday, 18 September 2013 7:44 PM
To: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
Subject: [rtcweb] m=3Dsection recycling and a=3Dmid
After thinking about it a bit, I think there are some side effects associat=
ed with changing/not changing mid attribute of a re-cycled m=3D section and=
 it being part of a BUNDLE group or not..

example

say , we have these video m=3Dsection part of BUNDLE group as below
a=3Dgroup:BUNDLE 1, 2

m=3Dvideo .....
a=3Dmid:1
a=3Dinactive

m=3Dvideo .....
a=3Dmid:2
a=3Dsendrecv

Now If JS user adds a new video track and he wants it to be out of BUNDLE. =
If we recycle first video m=3Dsection and dont change mid attribute,  we en=
d up BUNDLing even if the user dint want to

On the other hand, if we allowed mid value to be changed, we might end up U=
NBundling a video stream , when the user did want to BUNDLE them

So allowing a mid value to be changed or not should be tied to what the JS =
API is asking on behalf of the user


Any thoughts ?

Cheers
Suhas





_______________________________________________

rtcweb mailing list

rtcweb@ietf.org<mailto:rtcweb@ietf.org>

https://www.ietf.org/mailman/listinfo/rtcweb


_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org>
https://www.ietf.org/mailman/listinfo/rtcweb


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</head>
<body lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&#43;1<o:p></o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> rtcweb-b=
ounces@ietf.org [mailto:rtcweb-bounces@ietf.org]
<b>On Behalf Of </b>Suhas Nandakumar<br>
<b>Sent:</b> 20. syyskuuta 2013 1:25<br>
<b>To:</b> Harald Alvestrand<br>
<b>Cc:</b> rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] m=3Dsection recycling and a=3Dmid<o:p></o:p></=
span></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">&#43;1<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On Thu, Sep 19, 2013 at 3:23 PM, Harald Alvestrand &=
lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestr=
and.no</a>&gt; wrote:<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal">I suggest that changing the MID value can be very pa=
inful, since it modifies the mapping between m-line index and MID.<br>
<br>
Is there any other spec that describes changing the MID value?<br>
<br>
If not, I'd suggest that we explicitly forbid changing the MID value. If we=
 want to add or remove an M-line from a bundle, we should update the bundle=
s instead.<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal"><br>
<br>
<br>
On 09/19/2013 08:52 PM, Christer Holmberg wrote:<o:p></o:p></p>
</div>
</div>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<div>
<div>
<p><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;san=
s-serif&quot;">Correct.<o:p></o:p></span></p>
<p><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;san=
s-serif&quot;">&nbsp;<o:p></o:p></span></p>
<p><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;san=
s-serif&quot;">Regards,<o:p></o:p></span></p>
<p><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;san=
s-serif&quot;">&nbsp;<o:p></o:p></span></p>
<p><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;san=
s-serif&quot;">Christer<o:p></o:p></span></p>
<div>
<div class=3D"MsoNormal" align=3D"center" style=3D"text-align:center">
<hr size=3D"2" width=3D"100%" align=3D"center">
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><b><span style=3D"fon=
t-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:black">From:</span=
></b><span style=3D"font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;c=
olor:black"> Suhas Nandakumar [<a href=3D"mailto:suhasietf@gmail.com" targe=
t=3D"_blank">suhasietf@gmail.com</a>]<br>
<b>Sent:</b> Thursday, 19 September 2013 9:48 PM<br>
<b>To:</b> Christer Holmberg<br>
<b>Cc:</b> <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a><br>
<b>Subject:</b> Re: [rtcweb] m=3Dsection recycling and a=3Dmid</span><o:p><=
/o:p></p>
</div>
<div>
<div>
<p class=3D"MsoNormal">Hello Christer <o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp; Completely agree with you<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">My point with the above examples was&nbsp;<o:p></o:p=
></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp; &quot;recycling of m=3Dline should take into =
account its existing BUNDLE membership and JS user's preference on requirin=
g the added media track to be BUNDLED or not&quot;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">This would imply, as suggested by you,<o:p></o:p></p=
>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp; - remove the mid from a=3Dgroup:BUNDLE line i=
f the JS user doesn't want the new track to be BUNDLEd<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;- or if the mid value is changed, then add the=
 new mid value to the BUNDLE group if the JS user want's the new track to b=
e BUNDLED<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">- or just retain the mid value and its BUNDLE member=
ship for the new track, if it is going to be BUNDLEd<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Cheers<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Suhas<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On Thu, Sep 19, 2013 at 9:24 AM, Christer Holmberg &=
lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chri=
ster.holmberg@ericsson.com</a>&gt; wrote:<o:p></o:p></p>
<div>
<div>
<p><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;san=
s-serif&quot;">Hi Suhas,<o:p></o:p></span></p>
<p><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;san=
s-serif&quot;">&nbsp;<o:p></o:p></span></p>
<p><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;san=
s-serif&quot;">The mid attribute itself does not add an m- line to a BUNDLE=
 group. The attribute value also needs to be listed in the group:BUNDLE att=
ribute mid list.<o:p></o:p></span></p>
<p><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;san=
s-serif&quot;">&nbsp;<o:p></o:p></span></p>
<p><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;san=
s-serif&quot;">So, if you want to re-use an m- line and mid attribute, but =
you don't want the m- line to be in the BUNDLE group, simply remove the mid=
 from the mid list.<o:p></o:p></span></p>
<p><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;san=
s-serif&quot;">&nbsp;<o:p></o:p></span></p>
<p><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;san=
s-serif&quot;">Regards,<o:p></o:p></span></p>
<p><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;san=
s-serif&quot;">&nbsp;<o:p></o:p></span></p>
<p><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;san=
s-serif&quot;">Christer<o:p></o:p></span></p>
<p><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;san=
s-serif&quot;">&nbsp;<o:p></o:p></span></p>
<div>
<div class=3D"MsoNormal" align=3D"center" style=3D"text-align:center">
<hr size=3D"2" width=3D"100%" align=3D"center">
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><b><span style=3D"fon=
t-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:black">From:</span=
></b><span style=3D"font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;c=
olor:black">
<a href=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_blank">rtcweb-bounces=
@ietf.org</a> [<a href=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_blank"=
>rtcweb-bounces@ietf.org</a>] on behalf of Suhas Nandakumar [<a href=3D"mai=
lto:suhasietf@gmail.com" target=3D"_blank">suhasietf@gmail.com</a>]<br>
<b>Sent:</b> Wednesday, 18 September 2013 7:44 PM<br>
<b>To:</b> <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a><br>
<b>Subject:</b> [rtcweb] m=3Dsection recycling and a=3Dmid</span><o:p></o:p=
></p>
</div>
<div>
<div>
<div>
<div>
<p class=3D"MsoNormal">After thinking about it a bit, I think there are som=
e side effects associated with changing/not changing mid attribute of a re-=
cycled m=3D section and it being part of a BUNDLE group or not..
<o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">example<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">say , we have these video m=3Dsection part of BUNDLE=
 group as below&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">a=3Dgroup:BUNDLE 1, 2<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">m=3Dvideo .....<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">a=3Dmid:1&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">a=3Dinactive<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<div>
<p class=3D"MsoNormal">m=3Dvideo .....<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">a=3Dmid:2&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">a=3Dsendrecv<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Now If JS user adds a new video track and he wants i=
t to be out of BUNDLE. If we recycle first video m=3Dsection and dont chang=
e mid attribute, &nbsp;we end up BUNDLing even if the user dint want to<o:p=
></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">On the other hand, if we allowed mid value to be cha=
nged, we might end up UNBundling a video stream , when the user did want to=
 BUNDLE them<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">So allowing a mid value to be changed or not should =
be tied to what the JS API is asking on behalf of the user<o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Any thoughts ?<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Cheers<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Suhas<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><o:p>&nbsp;</o:p></p>
</div>
</div>
<pre>_______________________________________________<o:p></o:p></pre>
<pre>rtcweb mailing list<o:p></o:p></pre>
<pre><a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</=
a><o:p></o:p></pre>
<pre><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></o:p></pre>
</blockquote>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1C4A82DFESESSMB209erics_--

From harald@alvestrand.no  Fri Sep 20 00:25:28 2013
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Date: Fri, 20 Sep 2013 09:25:20 +0200
From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] JSEP and video stream properties selection (a=imageattr usage)
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On 09/20/2013 08:02 AM, Magnus Westerlund wrote:
> WG,
>
> In the JSEP call I commented regarding a=imageattr open issue that we
> had an original agreement from over a year ago (Vancouver) to go with
> Harald's proposal in then
> http://tools.ietf.org/id/draft-alvestrand-rtcweb-resolution-00.txt
>
> This is documented in the minutes:
> http://www.ietf.org/proceedings/84/minutes/minutes-84-rtcweb
>
>
> However, I did forget to note that there has been discussion on this
> issue after that between Harald Alvestrand and Stefan Håkansson
> primarily on the W3C list
>
> http://lists.w3.org/Archives/Public/public-webrtc/2013Aug/0051.html
>
> Where they are seriously discussing only using WebRTC API for this.
>
> It is in the context of
> https://datatracker.ietf.org/doc/draft-alvestrand-constraints-resolution/
>
> Thus I would like to point out that there appear to be some indication
> of desire move away from the consensus last year.

My thinking is that there are 3 points on the spectrum of solutions:

- Constraints at source only (the approach most thoroughly described in
draft-alvestrand-constraints-resolution).

- Constraints at destination can be sigalled in SDP, and acted on at
source (described in draft-alvestrand-rtcweb-resolution). This spec
could not be pursued as long as the "Plan X" discussions were still
unsettled, but now it should really be "a piece of cake".

- Defining a new way of signalling. This was the approach rejected in
Vancouver.

Formalistically, if the IETF decides to offer a well defined mechanism
for negotiating resolution through SDP, the W3C will certainly take up
the discussion on whether or not we should connect the API to that
functionality.

Happy to discuss more.


From internet-drafts@ietf.org  Fri Sep 20 00:33:31 2013
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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-00.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Real-Time Communication in WEB-browsers W=
orking Group of the IETF.

	Title           : STUN Usage for Consent Freshness
	Author(s)       : Muthu Arul Mozhi Perumal
                          Dan Wing
                          Ram Mohan Ravindranath
                          Tirumaleswar Reddy
	Filename        : draft-ietf-rtcweb-stun-consent-freshness-00.txt
	Pages           : 7
	Date            : 2013-09-19

Abstract:
   Verification of peer consent before sending traffic is necessary in
   WebRTC deployments to ensure that a malicious JavaScript cannot use
   the browser as a platform for launching attacks.  A related problem
   is session liveness.  WebRTC application may want to detect
   connection failure and take appropriate action.

   This document describes how a WebRTC browser can verify peer consent
   to continue sending traffic and detect connection failure.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshness

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-stun-consent-freshness-00


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


From ietf-secretariat-reply@ietf.org  Fri Sep 20 00:48:27 2013
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Subject: [rtcweb] Milestones changed for rtcweb WG
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Changed milestone "Send STUN Usage for Consent Freshness to IESG for
publication as proposed standard", added
draft-ietf-rtcweb-stun-consent-freshness to milestone.

URL: http://datatracker.ietf.org/wg/rtcweb/charter/

From christer.holmberg@ericsson.com  Fri Sep 20 00:51:53 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-00.txt
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-00.txt
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Hi,

The draft doesn't mention the case when one entity (e.g. a gateway) is ICE =
lite. Eventhough ICE already specifies that an ICE lite entity does not sen=
d STUN requests, I think it would be good to cover it also for consent refr=
eshness.

Regards,

Christer

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 internet-drafts@ietf.org
Sent: 20. syyskuuta 2013 10:34
To: i-d-announce@ietf.org
Cc: rtcweb@ietf.org
Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-00.t=
xt


A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Real-Time Communication in WEB-browsers W=
orking Group of the IETF.

	Title           : STUN Usage for Consent Freshness
	Author(s)       : Muthu Arul Mozhi Perumal
                          Dan Wing
                          Ram Mohan Ravindranath
                          Tirumaleswar Reddy
	Filename        : draft-ietf-rtcweb-stun-consent-freshness-00.txt
	Pages           : 7
	Date            : 2013-09-19

Abstract:
   Verification of peer consent before sending traffic is necessary in
   WebRTC deployments to ensure that a malicious JavaScript cannot use
   the browser as a platform for launching attacks.  A related problem
   is session liveness.  WebRTC application may want to detect
   connection failure and take appropriate action.

   This document describes how a WebRTC browser can verify peer consent
   to continue sending traffic and detect connection failure.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshness

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-stun-consent-freshness-00


Please note that it may take a couple of minutes from the time of submissio=
n until the htmlized version and diff are available at tools.ietf.org.

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

From mperumal@cisco.com  Fri Sep 20 01:24:46 2013
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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-00.txt
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Date: Fri, 20 Sep 2013 08:24:23 +0000
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-00.txt
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Sure, noted.

This was a resubmit of the -04 individual draft as a WG draft, with minor c=
hanges. We'll cover the ICE-lite scenario in the next revision..

Muthu

|-----Original Message-----
|From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf O=
f Christer Holmberg
|Sent: Friday, September 20, 2013 1:22 PM
|To: rtcweb@ietf.org
|Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness=
-00.txt
|
|Hi,
|
|The draft doesn't mention the case when one entity (e.g. a gateway) is ICE=
 lite. Eventhough ICE
|already specifies that an ICE lite entity does not send STUN requests, I t=
hink it would be good to
|cover it also for consent refreshness.
|
|Regards,
|
|Christer
|
|-----Original Message-----
|From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf O=
f internet-drafts@ietf.org
|Sent: 20. syyskuuta 2013 10:34
|To: i-d-announce@ietf.org
|Cc: rtcweb@ietf.org
|Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-00.=
txt
|
|
|A New Internet-Draft is available from the on-line Internet-Drafts directo=
ries.
| This draft is a work item of the Real-Time Communication in WEB-browsers =
Working Group of the IETF.
|
|	Title           : STUN Usage for Consent Freshness
|	Author(s)       : Muthu Arul Mozhi Perumal
|                          Dan Wing
|                          Ram Mohan Ravindranath
|                          Tirumaleswar Reddy
|	Filename        : draft-ietf-rtcweb-stun-consent-freshness-00.txt
|	Pages           : 7
|	Date            : 2013-09-19
|
|Abstract:
|   Verification of peer consent before sending traffic is necessary in
|   WebRTC deployments to ensure that a malicious JavaScript cannot use
|   the browser as a platform for launching attacks.  A related problem
|   is session liveness.  WebRTC application may want to detect
|   connection failure and take appropriate action.
|
|   This document describes how a WebRTC browser can verify peer consent
|   to continue sending traffic and detect connection failure.
|
|
|The IETF datatracker status page for this draft is:
|https://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshness
|
|There's also a htmlized version available at:
|http://tools.ietf.org/html/draft-ietf-rtcweb-stun-consent-freshness-00
|
|
|Please note that it may take a couple of minutes from the time of submissi=
on until the htmlized
|version and diff are available at tools.ietf.org.
|
|Internet-Drafts are also available by anonymous FTP at:
|ftp://ftp.ietf.org/internet-drafts/
|
|_______________________________________________
|rtcweb mailing list
|rtcweb@ietf.org
|https://www.ietf.org/mailman/listinfo/rtcweb
|_______________________________________________
|rtcweb mailing list
|rtcweb@ietf.org
|https://www.ietf.org/mailman/listinfo/rtcweb

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Subject: Re: [rtcweb] JSEP-04: Some comments on Section 5.2.1. and 5.2.2 (19th september)
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For Q.3, doesn't RFC 5764 say the proto field for DTLS-SRTP should be "
UDP/TLS/RTP/SAVPF"


On 19 September 2013 11:08, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>  Hi,****
>
> ** **
>
> Section 5.2.1:****
>
> -----------------****
>
> ** **
>
> *Q_1:      o- line <sess-version> initial value - deviation*
>
> ** **
>
> What is the reason for recommending a <sess-version> zero value in the
> initial Offer, when the RFC says SHOULD use an NTP format timestamp? ****
>
> ** **
>
> I don=92t have any strong feelings, but in general: whenever we deviate f=
rom
> the =93base=94 SDP procedures I think we should justify why.****
>
> ** **
>
> ** **
>
> *Q_2:      o- line <sess-version> initial value - forking*
>
> ** **
>
> When a new PeerConnection is created due to forking, the <sess-version>
> value must be based (incremented) on the value associated with the =93mot=
her=94
> PeerConnection, for which the initial Offer of the whole communication
> session was created.****
>
> ** **
>
> ** **
>
> *Q_3:      RTP*
>
> ** **
>
> The text says:****
>
> ** **
>
> =93The <proto> field MUST be set to "RTP/SAVPF". =93****
>
> ** **
>
> But, that of course only applies to RTP based streams (not the data
> channel). Also, in general, it needs to be clear what information needs t=
o
> be in every m- line, and what information is protocol specific. I would
> suggest to have a =93General=94 sub-section, a =93RTP=94 sub-section, etc=
.****
>
> ** **
>
> ** **
>
> *Q_4:      BUNDLE*
>
> ** **
>
> The text says:****
>
> ** **
>
> =93If a m=3D section is not being bundled into another m=3D section, it M=
UST****
>
>                 generate a unique set of ICE credentials and gather its
> own set of****
>
>                 candidates.  Otherwise, it MUST use the same ICE
> credentials and****
>
>                 candidates that were used in the m=3D section that it is
> being bundled****
>
>                 into.=94****
>
> ** **
>
> As, when BUNDLE is used, the initial Offer will contain identical ICE
> candidates, does that mean that we will also include identical address
> information in the initial Offer?****
>
> ** **
>
> I don=92t object to that =96 I just want to clarify, as it has impacts on=
 the
> text in the BUNDLE spec :)****
>
> ** **
>
> ** **
>
> Section 5.2.2:****
>
> -----------------****
>
> ** **
>
> ** **
>
> *Q_5:      Offer when in =93local-offer=94*
>
> ** **
>
> The text says:****
>
> ** **
>
> =93If the initial offer was applied using setLocalDescription, but an****
>
>                 answer from the remote side has not yet been applied,
> meaning the****
>
>                 PeerConnection is still in the "local-offer" state, the
> steps for****
>
>                 generating an initial offer should be followed,=94****
>
> ** **
>
> I don=92t understand this. Why would you create a new Offer while you are
> waiting for an Answer to the previously sent Offer? ****
>
> ** **
>
> Also, when setRemote() is called, to which Offer does it apply?****
>
> ** **
>
> * *
>
> *Q_6:      BUNDLE*
>
> ** **
>
> We=92ll probably also need some text about BUNDLE.****
>
> ** **
>
> ** **
>
> Regards,****
>
> ** **
>
> Christer****
>
> ** **
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr">For Q.3, doesn&#39;t RFC 5764 say the proto field for DTLS=
-SRTP should be &quot;<span style=3D"color:rgb(0,0,0);font-size:1em">UDP/TL=
S/RTP/SAVPF&quot;</span></div><div class=3D"gmail_extra"><br><br><div class=
=3D"gmail_quote">
On 19 September 2013 11:08, Christer Holmberg <span dir=3D"ltr">&lt;<a href=
=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.holmb=
erg@ericsson.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote"=
 style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">






<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal">Hi,<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal">Section 5.2.1:<u></u><u></u></p>
<p class=3D"MsoNormal">-----------------<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal"><b>Q_1:=A0=A0=A0=A0=A0 o- line &lt;sess-version&gt; =
initial value - deviation<u></u><u></u></b></p>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal">What is the reason for recommending a &lt;sess-versi=
on&gt; zero value in the initial Offer, when the RFC says SHOULD use an NTP=
 format timestamp?
<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal">I don=92t have any strong feelings, but in general: =
whenever we deviate from the =93base=94 SDP procedures I think we should ju=
stify why.<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal"><b>Q_2:=A0=A0=A0=A0=A0 o- line &lt;sess-version&gt; =
initial value - forking<u></u><u></u></b></p>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal">When a new PeerConnection is created due to forking,=
 the &lt;sess-version&gt; value must be based (incremented) on the value as=
sociated with the =93mother=94 PeerConnection, for which the initial Offer =
of the whole communication session was created.<u></u><u></u></p>

<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal"><b>Q_3:=A0=A0=A0=A0=A0 RTP<u></u><u></u></b></p>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal">The text says:<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">=93The &lt;proto&gt; fi=
eld MUST be set to &quot;RTP/SAVPF&quot;. =93<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal">But, that of course only applies to RTP based stream=
s (not the data channel). Also, in general, it needs to be clear what infor=
mation needs to be in every m- line, and what information is protocol speci=
fic. I would suggest to have a =93General=94
 sub-section, a =93RTP=94 sub-section, etc.<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal"><b>Q_4: =A0=A0=A0=A0 BUNDLE<u></u><u></u></b></p>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal">The text says:<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">=93If a m=3D section is=
 not being bundled into another m=3D section, it MUST<u></u><u></u></p>
<p class=3D"MsoNormal">=A0=A0 =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 generate=
 a unique set of ICE credentials and gather its own set of<u></u><u></u></p=
>
<p class=3D"MsoNormal">=A0=A0 =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 candidat=
es.=A0 Otherwise, it MUST use the same ICE credentials and<u></u><u></u></p=
>
<p class=3D"MsoNormal">=A0=A0 =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 candidat=
es that were used in the m=3D section that it is being bundled<u></u><u></u=
></p>
<p class=3D"MsoNormal">=A0=A0 =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 into.=94=
<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal">As, when BUNDLE is used, the initial Offer will cont=
ain identical ICE candidates, does that mean that we will also include iden=
tical address information in the initial Offer?<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal">I don=92t object to that =96 I just want to clarify,=
 as it has impacts on the text in the BUNDLE spec :)<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal">Section 5.2.2:<u></u><u></u></p>
<p class=3D"MsoNormal">-----------------<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal"><b>Q_5:=A0=A0=A0=A0=A0 Offer when in =93local-offer=
=94<u></u><u></u></b></p>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal">The text says:<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">=93If the initial offer=
 was applied using setLocalDescription, but an<u></u><u></u></p>
<p class=3D"MsoNormal">=A0=A0 =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 answer f=
rom the remote side has not yet been applied, meaning the<u></u><u></u></p>
<p class=3D"MsoNormal">=A0=A0 =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 PeerConn=
ection is still in the &quot;local-offer&quot; state, the steps for<u></u><=
u></u></p>
<p class=3D"MsoNormal">=A0=A0 =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 generati=
ng an initial offer should be followed,=94<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal">I don=92t understand this. Why would you create a ne=
w Offer while you are waiting for an Answer to the previously sent Offer?
<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal">Also, when setRemote() is called, to which Offer doe=
s it apply?<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal"><b><u></u>=A0<u></u></b></p>
<p class=3D"MsoNormal"><b>Q_6:=A0=A0=A0=A0=A0 BUNDLE<u></u><u></u></b></p>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal">We=92ll probably also need some text about BUNDLE.<u=
></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal">Regards,<span class=3D"HOEnZb"><font color=3D"#88888=
8"><u></u><u></u></font></span></p><span class=3D"HOEnZb"><font color=3D"#8=
88888">
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal">Christer<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
</font></span></div>
</div>

<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

--001a11c3dfbae6326004e6cc706e--

From christer.holmberg@ericsson.com  Fri Sep 20 01:31:30 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-00.txt
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-00.txt
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Thanks!

-----Original Message-----
From: Muthu Arul Mozhi Perumal (mperumal) [mailto:mperumal@cisco.com]=20
Sent: 20. syyskuuta 2013 11:24
To: Christer Holmberg; rtcweb@ietf.org
Subject: RE: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-=
00.txt

Sure, noted.

This was a resubmit of the -04 individual draft as a WG draft, with minor c=
hanges. We'll cover the ICE-lite scenario in the next revision..

Muthu

|-----Original Message-----
|From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On=20
|Behalf Of Christer Holmberg
|Sent: Friday, September 20, 2013 1:22 PM
|To: rtcweb@ietf.org
|Subject: Re: [rtcweb] I-D Action:=20
|draft-ietf-rtcweb-stun-consent-freshness-00.txt
|
|Hi,
|
|The draft doesn't mention the case when one entity (e.g. a gateway) is=20
|ICE lite. Eventhough ICE already specifies that an ICE lite entity does=20
|not send STUN requests, I think it would be good to cover it also for cons=
ent refreshness.
|
|Regards,
|
|Christer
|
|-----Original Message-----
|From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On=20
|Behalf Of internet-drafts@ietf.org
|Sent: 20. syyskuuta 2013 10:34
|To: i-d-announce@ietf.org
|Cc: rtcweb@ietf.org
|Subject: [rtcweb] I-D Action:=20
|draft-ietf-rtcweb-stun-consent-freshness-00.txt
|
|
|A New Internet-Draft is available from the on-line Internet-Drafts directo=
ries.
| This draft is a work item of the Real-Time Communication in WEB-browsers =
Working Group of the IETF.
|
|	Title           : STUN Usage for Consent Freshness
|	Author(s)       : Muthu Arul Mozhi Perumal
|                          Dan Wing
|                          Ram Mohan Ravindranath
|                          Tirumaleswar Reddy
|	Filename        : draft-ietf-rtcweb-stun-consent-freshness-00.txt
|	Pages           : 7
|	Date            : 2013-09-19
|
|Abstract:
|   Verification of peer consent before sending traffic is necessary in
|   WebRTC deployments to ensure that a malicious JavaScript cannot use
|   the browser as a platform for launching attacks.  A related problem
|   is session liveness.  WebRTC application may want to detect
|   connection failure and take appropriate action.
|
|   This document describes how a WebRTC browser can verify peer consent
|   to continue sending traffic and detect connection failure.
|
|
|The IETF datatracker status page for this draft is:
|https://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshne
|ss
|
|There's also a htmlized version available at:
|http://tools.ietf.org/html/draft-ietf-rtcweb-stun-consent-freshness-00
|
|
|Please note that it may take a couple of minutes from the time of=20
|submission until the htmlized version and diff are available at tools.ietf=
.org.
|
|Internet-Drafts are also available by anonymous FTP at:
|ftp://ftp.ietf.org/internet-drafts/
|
|_______________________________________________
|rtcweb mailing list
|rtcweb@ietf.org
|https://www.ietf.org/mailman/listinfo/rtcweb
|_______________________________________________
|rtcweb mailing list
|rtcweb@ietf.org
|https://www.ietf.org/mailman/listinfo/rtcweb

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Subject: Re: [rtcweb] JSEP-04: Some comments on Section 5.2.1. and 5.2.2 (19th september)
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This is a multi-part message in MIME format.
--------------030400040603030202080209
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On 09/20/2013 10:27 AM, Kevin Dempsey wrote:
>
>     *Q_4:      BUNDLE*
>
>     The text says:
>
>     "If a m= section is not being bundled into another m= section, it MUST
>
>                     generate a unique set of ICE credentials and
>     gather its own set of
>
>                     candidates. Otherwise, it MUST use the same ICE
>     credentials and
>
>                     candidates that were used in the m= section that
>     it is being bundled
>
>                     into."
>
>     As, when BUNDLE is used, the initial Offer will contain identical
>     ICE candidates, does that mean that we will also include identical
>     address information in the initial Offer?
>
>     I don't object to that -- I just want to clarify, as it has
>     impacts on the text in the BUNDLE spec :)
>
>

I think we should be careful here to specify that BUNDLE is used 
according to the BUNDLE spec, and that any description given here is 
non-normative.

Over to MMUSIC....

--------------030400040603030202080209
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<html>
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    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    <div class="moz-cite-prefix">On 09/20/2013 10:27 AM, Kevin Dempsey
      wrote:<br>
    </div>
    <blockquote
cite="mid:CAMvTgcfevD4FQDqmVccF0UMZ-tSOtt1Fvjof_gkwvoNFUoBeQA@mail.gmail.com"
      type="cite">&nbsp;
      <div class="gmail_extra">
        <div class="gmail_quote">
          <blockquote class="gmail_quote" style="margin:0 0 0
            .8ex;border-left:1px #ccc solid;padding-left:1ex">
            <div link="blue" vlink="purple" lang="EN-US">
              <div>
                <p class="MsoNormal"><b>Q_4: &nbsp;&nbsp;&nbsp;&nbsp; BUNDLE</b></p>
                <p class="MsoNormal">&nbsp;</p>
                <p class="MsoNormal">The text says:</p>
                <p class="MsoNormal">&nbsp;</p>
                <p class="MsoNormal" style="text-indent:36.0pt">&#8220;If a m=
                  section is not being bundled into another m= section,
                  it MUST</p>
                <p class="MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; generate a unique
                  set of ICE credentials and gather its own set of</p>
                <p class="MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; candidates.&nbsp;
                  Otherwise, it MUST use the same ICE credentials and</p>
                <p class="MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; candidates that
                  were used in the m= section that it is being bundled</p>
                <p class="MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; into.&#8221;</p>
                <p class="MsoNormal">&nbsp;</p>
                <p class="MsoNormal">As, when BUNDLE is used, the
                  initial Offer will contain identical ICE candidates,
                  does that mean that we will also include identical
                  address information in the initial Offer?</p>
                <p class="MsoNormal">&nbsp;</p>
                <p class="MsoNormal">I don&#8217;t object to that &#8211; I just
                  want to clarify, as it has impacts on the text in the
                  BUNDLE spec :)</p>
                <br>
              </div>
            </div>
          </blockquote>
        </div>
      </div>
    </blockquote>
    <br>
    I think we should be careful here to specify that BUNDLE is used
    according to the BUNDLE spec, and that any description given here is
    non-normative.<br>
    <br>
    Over to MMUSIC....<br>
  </body>
</html>

--------------030400040603030202080209--

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] JSEP-04: Some comments on Section 5.2.1. and 5.2.2 (19th september)
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Subject: Re: [rtcweb] JSEP-04: Some comments on Section 5.2.1. and 5.2.2 (19th september)
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I really hope BUNDLE will be used as described in BUNDLE. At least that sho=
uld be our starting point. If we want to change something, it shall be just=
ified and discussed, before put into a document.

In addition, in the example flow in JSEP, BUNDLE IS used with different add=
resses, so an alignment will be needed in any case.

Regards,

Christer

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Harald Alvestrand
Sent: 20. syyskuuta 2013 13:12
To: rtcweb@ietf.org
Subject: Re: [rtcweb] JSEP-04: Some comments on Section 5.2.1. and 5.2.2 (1=
9th september)

On 09/20/2013 10:27 AM, Kevin Dempsey wrote:

Q_4:      BUNDLE

The text says:

"If a m=3D section is not being bundled into another m=3D section, it MUST
                generate a unique set of ICE credentials and gather its own=
 set of
                candidates.  Otherwise, it MUST use the same ICE credential=
s and
                candidates that were used in the m=3D section that it is be=
ing bundled
                into."

As, when BUNDLE is used, the initial Offer will contain identical ICE candi=
dates, does that mean that we will also include identical address informati=
on in the initial Offer?

I don't object to that - I just want to clarify, as it has impacts on the t=
ext in the BUNDLE spec :)


I think we should be careful here to specify that BUNDLE is used according =
to the BUNDLE spec, and that any description given here is non-normative.

Over to MMUSIC....

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<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">I really hope BUNDLE will=
 be used as described in BUNDLE. At least that should be our starting point=
. If we want to change something, it shall be justified
 and discussed, before put into a document.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">In addition, in the examp=
le flow in JSEP, BUNDLE IS used with different addresses, so an alignment w=
ill be needed in any case.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Regards,<o:p></o:p></span=
></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Christer<o:p></o:p></span=
></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">From:</span></b><spa=
n style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif=
&quot;;color:windowtext"> rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ie=
tf.org]
<b>On Behalf Of </b>Harald Alvestrand<br>
<b>Sent:</b> 20. syyskuuta 2013 13:12<br>
<b>To:</b> rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] JSEP-04: Some comments on Section 5.2.1. and 5=
.2.2 (19th september)<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On 09/20/2013 10:27 AM, Kevin Dempsey wrote:<o:p></o=
:p></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<p class=3D"MsoNormal">&nbsp; <o:p></o:p></p>
<div>
<div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><b>Q_4: &nbsp;&nbsp;&nbsp;&nbsp; BUNDLE</b><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">The text says:<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;text-indent:36.0pt">
&#8220;If a m=3D section is not being bundled into another m=3D section, it=
 MUST<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp; generate a unique set of ICE credentials and gather its=
 own set of<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp; candidates.&nbsp; Otherwise, it MUST use the same ICE c=
redentials and<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp; candidates that were used in the m=3D section that it i=
s being bundled<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp; into.&#8221;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">As, when BUNDLE is used, the initial Offer will contain identical =
ICE candidates, does that mean that we will also include identical address =
information in the initial Offer?<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">I don&#8217;t object to that &#8211; I just want to clarify, as it=
 has impacts on the text in the BUNDLE spec :)<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</blockquote>
</div>
</div>
</blockquote>
<p class=3D"MsoNormal"><br>
I think we should be careful here to specify that BUNDLE is used according =
to the BUNDLE spec, and that any description given here is non-normative.<b=
r>
<br>
Over to MMUSIC....<o:p></o:p></p>
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Subject: Re: [rtcweb] JSEP-04: Some comments on Section 5.2.1. and 5.2.2 (19th september)
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On 09/20/2013 12:20 PM, Christer Holmberg wrote:
>
> I really hope BUNDLE will be used as described in BUNDLE. At least 
> that should be our starting point. If we want to change something, it 
> shall be justified and discussed, before put into a document.
>
> In addition, in the example flow in JSEP, BUNDLE IS used with 
> different addresses, so an alignment will be needed in any case.
>
>

Since we're all tracking moving objects, I think we have to write text 
as best we can, and just point out which parts are normative and which 
parts are not.

In this particular case, using MUST language for whether or not ICE 
credentials are the same or different is inappropriate for this spec; 
instead it should say something like "The BUNDLE spec says that it MUST 
generate".

Then we just make sure it gets right before BUNDLE is published.


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    <div class="moz-cite-prefix">On 09/20/2013 12:20 PM, Christer
      Holmberg wrote:<br>
    </div>
    <blockquote
cite="mid:7594FB04B1934943A5C02806D1A2204B1C4A8710@ESESSMB209.ericsson.se"
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      <div class="WordSection1">
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">I
            really hope BUNDLE will be used as described in BUNDLE. At
            least that should be our starting point. If we want to
            change something, it shall be justified and discussed,
            before put into a document.<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">In
            addition, in the example flow in JSEP, BUNDLE IS used with
            different addresses, so an alignment will be needed in any
            case.<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><br>
        </p>
      </div>
    </blockquote>
    <br>
    Since we're all tracking moving objects, I think we have to write
    text as best we can, and just point out which parts are normative
    and which parts are not.<br>
    <br>
    In this particular case, using MUST language for whether or not ICE
    credentials are the same or different is inappropriate for this
    spec; instead it should say something like "The BUNDLE spec says
    that it MUST generate".<br>
    <br>
    Then we just make sure it gets right before BUNDLE is published.<br>
    <br>
  </body>
</html>

--------------040600030805050300090102--

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: [rtcweb] JSEP-04: Some comments on Section 5.2.1. and 5.2.2 (19th september)
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] JSEP-04: Some comments on Section 5.2.1. and 5.2.2 (19th september)
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Hi,

The reason I asked is because one of the open issues in BUNDLE is whether i=
dentical ICE credentials (and identical address properties) are allowed in =
the initial Offer, and one of the JSEP authors has been against that :)

But, in general I do agree with you. Whenever we reference procedures defin=
ed elsewhere, we shall reference those.

And, in this case, BUNDLE has a section on ICE, so we should reference thos=
e. And, if we have any issues with the BUNDLE ICE procedures, we should bri=
ng it to MMUSIC, before making deviations "on the fly"...

Regards,

Christer

From: Harald Alvestrand [mailto:harald@alvestrand.no]
Sent: 20. syyskuuta 2013 13:38
To: Christer Holmberg
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] JSEP-04: Some comments on Section 5.2.1. and 5.2.2 (1=
9th september)

On 09/20/2013 12:20 PM, Christer Holmberg wrote:
I really hope BUNDLE will be used as described in BUNDLE. At least that sho=
uld be our starting point. If we want to change something, it shall be just=
ified and discussed, before put into a document.

In addition, in the example flow in JSEP, BUNDLE IS used with different add=
resses, so an alignment will be needed in any case.


Since we're all tracking moving objects, I think we have to write text as b=
est we can, and just point out which parts are normative and which parts ar=
e not.

In this particular case, using MUST language for whether or not ICE credent=
ials are the same or different is inappropriate for this spec; instead it s=
hould say something like "The BUNDLE spec says that it MUST generate".

Then we just make sure it gets right before BUNDLE is published.

--_000_7594FB04B1934943A5C02806D1A2204B1C4A8798ESESSMB209erics_
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<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">The reason I asked is bec=
ause one of the open issues in BUNDLE is whether identical ICE credentials =
(and identical address properties) are allowed in the initial
 Offer, and one of the JSEP authors has been against that :)<o:p></o:p></sp=
an></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">But, in general I do agre=
e with you. Whenever we reference procedures defined elsewhere, we shall re=
ference those.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">And, in this case, BUNDLE=
 has a section on ICE, so we should reference those. And, if we have any is=
sues with the BUNDLE ICE procedures, we should bring it
 to MMUSIC, before making deviations &#8220;on the fly&#8221;&#8230;<o:p></=
o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Regards,<o:p></o:p></span=
></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Christer<o:p></o:p></span=
></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">From:</span></b><spa=
n style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif=
&quot;;color:windowtext"> Harald Alvestrand [mailto:harald@alvestrand.no]
<br>
<b>Sent:</b> 20. syyskuuta 2013 13:38<br>
<b>To:</b> Christer Holmberg<br>
<b>Cc:</b> rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] JSEP-04: Some comments on Section 5.2.1. and 5=
.2.2 (19th september)<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On 09/20/2013 12:20 PM, Christer Holmberg wrote:<o:p=
></o:p></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">I really hope BUNDLE will=
 be used as described in BUNDLE. At least that should be our starting point=
. If we want to change something, it shall be justified
 and discussed, before put into a document.</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">In addition, in the examp=
le flow in JSEP, BUNDLE IS used with different addresses, so an alignment w=
ill be needed in any case.</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p><=
/p>
</blockquote>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
Since we're all tracking moving objects, I think we have to write text as b=
est we can, and just point out which parts are normative and which parts ar=
e not.<br>
<br>
In this particular case, using MUST language for whether or not ICE credent=
ials are the same or different is inappropriate for this spec; instead it s=
hould say something like &quot;The BUNDLE spec says that it MUST generate&q=
uot;.<br>
<br>
Then we just make sure it gets right before BUNDLE is published.<o:p></o:p>=
</p>
</div>
</body>
</html>

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Subject: Re: [rtcweb] JSEP and video stream properties selection (a=imageattr usage)
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On 2013-09-20 09:25, Harald Alvestrand wrote:=0A=
> On 09/20/2013 08:02 AM, Magnus Westerlund wrote:=0A=
>> WG,=0A=
>>=0A=
>> In the JSEP call I commented regarding a=3Dimageattr open issue that we=
=0A=
>> had an original agreement from over a year ago (Vancouver) to go with=0A=
>> Harald's proposal in then=0A=
>> http://tools.ietf.org/id/draft-alvestrand-rtcweb-resolution-00.txt=0A=
>>=0A=
>> This is documented in the minutes:=0A=
>> http://www.ietf.org/proceedings/84/minutes/minutes-84-rtcweb=0A=
>>=0A=
>>=0A=
>> However, I did forget to note that there has been discussion on this=0A=
>> issue after that between Harald Alvestrand and Stefan H=E5kansson=0A=
>> primarily on the W3C list=0A=
>>=0A=
>> http://lists.w3.org/Archives/Public/public-webrtc/2013Aug/0051.html=0A=
>>=0A=
>> Where they are seriously discussing only using WebRTC API for this.=0A=
>>=0A=
>> It is in the context of=0A=
>> https://datatracker.ietf.org/doc/draft-alvestrand-constraints-resolution=
/=0A=
>>=0A=
>> Thus I would like to point out that there appear to be some indication=
=0A=
>> of desire move away from the consensus last year.=0A=
>=0A=
> My thinking is that there are 3 points on the spectrum of solutions:=0A=
>=0A=
> - Constraints at source only (the approach most thoroughly described in=
=0A=
> draft-alvestrand-constraints-resolution).=0A=
>=0A=
> - Constraints at destination can be sigalled in SDP, and acted on at=0A=
> source (described in draft-alvestrand-rtcweb-resolution). This spec=0A=
> could not be pursued as long as the "Plan X" discussions were still=0A=
> unsettled, but now it should really be "a piece of cake".=0A=
=0A=
My personal view is still that SDP is the wrong level for carrying this =0A=
kind of signaling (resolution, and perhaps further down the road =0A=
pause/resume related signaling). Each update requires an O/A, and locks =0A=
out other changes to the session. And since the SDPs must be handled by =0A=
the application (that is responsible for sending them over and applying =0A=
them) I wonder if there is any advantage compared to just having an API =0A=
on the sending side. If the receiving application wants another =0A=
resolution sent, it could just tell the sending application so, rather =0A=
than going through a createOffer/setLocal/ send-offer =0A=
/receive-answer/setRemote cycle.=0A=
=0A=
>=0A=
> - Defining a new way of signalling. This was the approach rejected in=0A=
> Vancouver.=0A=
>=0A=
> Formalistically, if the IETF decides to offer a well defined mechanism=0A=
> for negotiating resolution through SDP, the W3C will certainly take up=0A=
> the discussion on whether or not we should connect the API to that=0A=
> functionality.=0A=
>=0A=
> Happy to discuss more.=0A=
>=0A=
> _______________________________________________=0A=
> rtcweb mailing list=0A=
> rtcweb@ietf.org=0A=
> https://www.ietf.org/mailman/listinfo/rtcweb=0A=
>=0A=
=0A=

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To: "'Cullen Jennings \(fluffy\)'" <fluffy@cisco.com>, <rtcweb@ietf.org>
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For NAT/Firewall traversal WebRTC uses ICE, where the browser today gets =
the
TURN server address from the web application, or gets it configured by a =
LAN
admin via an =93admin policy template=94. =20

However, there is a need (motivations below) for the TURN server address =
to
be provided via DHCP (and maybe RA - Router Advertisement - for IPv6, =
and
the OTT channel in mobiles have their own method I=92ve been told). =
Simply
put, just as you often get your IP-address, DNS address, etc. =
automatically
from the network access, you should also get the TURN server address.

I suggest the following is added to the end of use case (which today is
about multiple TURN servers)
3.2.4.  Simple Video Communication Service, global service provider
3.2.4.1.  Description
...
"A network service provider must be able to automatically supply a TURN
server addresses to the browser when accessing a network. The address =
may
come via DHCP or a similar mechanism (maybe RA - Router Advertisement - =
for
IPv6, an addition to the IPCP protocol like RFC1877 for PPPoE, or =
whatever
method the mobile OTT channels use). The mechanism should be similar to
automatically getting the IP-address, DNS address, etc.. and need =
extensions
to current recommendations/standards.=20

There are several reasons for a network service provider to supply a =
TURN
server as part of his offered access:
- to keep media paths short, specifically not sending media outside its =
own
network to some distant application provided TURN server
- to support mobility, i.e. you may want to move from a LAN with a
configured TURN server to accessing via WiFi or 3G/4G OTT channels=20
- to offer a media path with better quality (than best effort data =
traffic).
Getting =93WebRTC-ready=94 access and we look forward to telepresence =
for
everyone.

An enterprise network that want to keep a restrictive firewall not =
allowing
UDP traffic, could provide a real-time path using a TURN server =
paralleling
the firewall, instead of tunneling RTP through always open http or https
ports resulting in RTP media over TCP =96 with severe quality problems =
from
TCP retransmissions of dropped packets. The TURN server address is most
easily provided in the same way as the IP address and DNS address. (That
would also put the right party in control =96 The network provider =
decides
what is allowed on his network.)

This browser should select which available TURN server address to use in =
the
following priority order, where ICE could be used to try several:

1) TURN server address configured in the browser by the user (special =
cases,
normally not used)=20
2) TURN server address configured by the network administrator via an =
=93admin
policy template=94
3) TURN server address supplied by DHCP or similar automatic network =
method=20
4) TURN server address being supplied by the web application"

Two new requirements can be extracted:
"   ----------------------------------------------------------------
   F40     The browser must support retrieving TURN server addresses via
DHCP or a similar mechanism (maybe RA - Router Advertisement - for IPv6, =
an
addition to the IPCP protocol like RFC1877 for PPPoE, or whatever method =
the
mobile OTT channels use). The mechanism should be similar to =
automatically
getting the IP-address, DNS address, etc..
   ----------------------------------------------------------------
   F42     This browser should select which available TURN server =
address to
use in the following priority order, where ICE could be used to try =
several:

1) TURN server address configured in the browser by the user (special =
cases,
normally not used)=20
2) TURN server address configured by the network administrator via an =
=93admin
policy template=94
3) TURN server address supplied by DHCP or similar automatic network =
method=20
4) TURN server address being supplied by the web application  =20
----------------------------------------------------------------"

I suppose this also will add to:
5.  IANA Considerations
   TBD

/Karl


-----Ursprungligt meddelande-----
Fr=E5n: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] F=F6r =
Cullen
Jennings (fluffy)
Skickat: den 4 september 2013 18:24
Till: rtcweb@ietf.org
=C4mne: [rtcweb] WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11


We would like to start a working group last call of
draft-ietf-rtcweb-use-cases-and-requirements-11.

Please send comments by the end of the day on September 21.=20

Thank you,=20

The chairs =85.




_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb


From andrew.hutton@siemens-enterprise.com  Fri Sep 20 10:09:14 2013
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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [pntaw] New version of draft-hutton-rtcweb-nat-firewall-considerations
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Subject: [rtcweb] FW: [pntaw] New version of	draft-hutton-rtcweb-nat-firewall-considerations
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Hi All,

We have submitted draft-hutton-rtcweb-nat-firewall-considerations-02 in whi=
ch we have tried to take account of the feedback we have received over the =
last couple of months.

Comments should be sent to the PNTAW list.

Regards
Andy



-----Original Message-----
From: i-d-announce-bounces@ietf.org [mailto:i-d-announce-bounces@ietf.org] =
On Behalf Of internet-drafts@ietf.org
Sent: 20 September 2013 15:33
To: i-d-announce@ietf.org
Subject: I-D Action: draft-hutton-rtcweb-nat-firewall-considerations-02.txt


A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.


	Title           : RTCWEB Considerations for NATs, Firewalls and HTTP proxi=
es
	Author(s)       : Thomas Stach
                          Andrew Hutton
                          Justin Uberti
	Filename        : draft-hutton-rtcweb-nat-firewall-considerations-02.txt
	Pages           : 12
	Date            : 2013-09-20

Abstract:
   This document describes mechanism to enable media stream
   establishment for Real-Time Communication in WEB-browsers (WebRTC) in
   the presence of network address translators, firewalls and HTTP
   proxies.  HTTP proxy and firewall deployed in many private network
   domains introduce obstacles to the successful establishment of media
   stream via WebRTC.  This document examines some of these deployment
   scenarios and develops requirements on the web browsers designed to
   provide the best possible chance of media connectivity between WebRTC
   peers.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-hutton-rtcweb-nat-firewall-considera=
tions

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-hutton-rtcweb-nat-firewall-considerations-=
02

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-hutton-rtcweb-nat-firewall-conside=
rations-02


Please note that it may take a couple of minutes from the time of submissio=
n
until the htmlized version and diff are available at tools.ietf.org.

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

_______________________________________________
I-D-Announce mailing list
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From andrew.hutton@siemens-enterprise.com  Fri Sep 20 10:58:07 2013
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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>, "Cullen Jennings (fluffy)" <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org" <draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org>
Thread-Topic: [rtcweb] WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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On: 17 September 2013 13:01 Magnus Westerlund Wrote:
> 3.2.3.1.  Description
>=20
>    This use-case is almost identical to the Simple Video Communication
>    Service use-case (Section 3.2.1).  The difference is that one of the
>    users is behind a FW that only allows http traffic.
>=20
> If a firewall only allows HTTP traffic, then can we really assume that
> the firewall administrator per default will accept WebRTC Media and
> Data
> traffic?
>=20
> I am far from certain of this, and think on a requirement level needs
> to
> express a situation where the firewall administrator allows WebRTC
> across its FW, or at least can easily configure a rule to block it.
> Thus
> resulting in that any solution for this needs to be easily identifiable
> and possible to block.
>=20

Stefan already stated that this use case will be changed to take account of=
 my comments in http://www.ietf.org/mail-archive/web/rtcweb/current/msg0826=
4.html. So the "only allows HTTP traffic" is going to be removed and change=
d to describe the use case when the firewall requires traffic to flow via a=
 HTTP Proxy.=20

We also have a use case when a network specific TURN server is deployed (3.=
2.5.1).

In both cases I agree that a network administrator must have a way to block=
 webrtc but there are various ways of achieving this including just configu=
ring the browser not to do webrtc, normal HTTP black/white lists, and also =
I can imagine solutions based on putting something in the HTTP Connect sent=
 to the proxy so the proxy can block it.=20

So I agree that it must be possible to block webrtc traffic but we need to =
careful not to specify the solution in the use case. =20

Andy



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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
Thread-Topic: [rtcweb] JSEP-04: Some comments on Section 5.2.1. and 5.2.2 (19th september)
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Good points - proposed resolutions inline =85

On Sep 19, 2013, at 3:08 AM, Christer Holmberg <christer.holmberg@ericsson.=
com> wrote:

> Hi,
>=20
> Section 5.2.1:
> -----------------
>=20
> Q_1:      o- line <sess-version> initial value - deviation
>=20
> What is the reason for recommending a <sess-version> zero value in the in=
itial Offer, when the RFC says SHOULD use an NTP format timestamp?
>=20
> I don=92t have any strong feelings, but in general: whenever we deviate f=
rom the =93base=94 SDP procedures I think we should justify why.

Lets use NTP=20

>=20
>=20
> Q_2:      o- line <sess-version> initial value - forking
>=20
> When a new PeerConnection is created due to forking, the <sess-version> v=
alue must be based (incremented) on the value associated with the =93mother=
=94 PeerConnection, for which the initial Offer of the whole communication =
session was created.


uh, that is old text sorry. I'm proposing the principal that any time the S=
DP changes (including new candidate lines) the version gets larger.=20

>=20
>=20
> Q_3:      RTP
>=20
> The text says:
>=20
> =93The <proto> field MUST be set to "RTP/SAVPF". =93
>=20
> But, that of course only applies to RTP based streams (not the data chann=
el). Also, in general, it needs to be clear what information needs to be in=
 every m- line, and what information is protocol specific. I would suggest =
to have a =93General=94 sub-section, a =93RTP=94 sub-section, etc.

Yep - I think this should be offers are created with RTP/SAVPF for audio an=
d video and system can reeve offers with SAVPF or SAVP
>=20
>=20
> Q_4:      BUNDLE
>=20
> The text says:
>=20
> =93If a m=3D section is not being bundled into another m=3D section, it M=
UST
>                generate a unique set of ICE credentials and gather its ow=
n set of
>                candidates.  Otherwise, it MUST use the same ICE credentia=
ls and
>                candidates that were used in the m=3D section that it is b=
eing bundled
>                into.=94
>=20
> As, when BUNDLE is used, the initial Offer will contain identical ICE can=
didates, does that mean that we will also include identical address informa=
tion in the initial Offer?

no the initial offer will not have identical stuff other than on lines that=
 "bundle-only" and we still need to add more text on this but idea is to ma=
ke just like what is in united-plan

>=20
> I don=92t object to that =96 I just want to clarify, as it has impacts on=
 the text in the BUNDLE spec :)
>=20
>=20
> Section 5.2.2:
> -----------------
>=20
>=20
> Q_5:      Offer when in =93local-offer=94
>=20
> The text says:
>=20
> =93If the initial offer was applied using setLocalDescription, but an
>                answer from the remote side has not yet been applied, mean=
ing the
>                PeerConnection is still in the "local-offer" state, the st=
eps for
>                generating an initial offer should be followed,=94
>=20
> I don=92t understand this. Why would you create a new Offer while you are=
 waiting for an Answer to the previously sent Offer?

Justin has asked for this in the case when the offer has not been sent to a=
nyone else. This is not going to be useful for typical systems trying to ha=
ve SIP or Jingle compatible flow but who knows =85

>=20
> Also, when setRemote() is called, to which Offer does it apply?

The most recent one. And the JS app would have to make sure it passed the w=
rite one or you run high odds of getting completely broken behavior.=20

>=20
>=20
> Q_6:      BUNDLE
>=20
> We=92ll probably also need some text about BUNDLE.
>=20

yep - but plan is to match up with unified plan=20

>=20
> Regards,
>=20
> Christer
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From harald@alvestrand.no  Fri Sep 20 14:16:02 2013
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Subject: Re: [rtcweb] TURN server address via DHCP, WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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I'm very hesitant to add more requirements that translate to "new 
protocol needs to be developed, implemented and deployed across the 
network before the requirement is useful".

In this case, it's "only" a DHCP option, but what new DHCP options do we 
have that have become widely used in the last 10 years?

On 09/20/2013 05:43 PM, Karl Stahl wrote:
> For NAT/Firewall traversal WebRTC uses ICE, where the browser today gets the
> TURN server address from the web application, or gets it configured by a LAN
> admin via an “admin policy template”.
>
> However, there is a need (motivations below) for the TURN server address to
> be provided via DHCP (and maybe RA - Router Advertisement - for IPv6, and
> the OTT channel in mobiles have their own method I’ve been told). Simply
> put, just as you often get your IP-address, DNS address, etc. automatically
> from the network access, you should also get the TURN server address.
>
> I suggest the following is added to the end of use case (which today is
> about multiple TURN servers)
> 3.2.4.  Simple Video Communication Service, global service provider
> 3.2.4.1.  Description
> ...
> "A network service provider must be able to automatically supply a TURN
> server addresses to the browser when accessing a network. The address may
> come via DHCP or a similar mechanism (maybe RA - Router Advertisement - for
> IPv6, an addition to the IPCP protocol like RFC1877 for PPPoE, or whatever
> method the mobile OTT channels use). The mechanism should be similar to
> automatically getting the IP-address, DNS address, etc.. and need extensions
> to current recommendations/standards.
>
> There are several reasons for a network service provider to supply a TURN
> server as part of his offered access:
> - to keep media paths short, specifically not sending media outside its own
> network to some distant application provided TURN server
> - to support mobility, i.e. you may want to move from a LAN with a
> configured TURN server to accessing via WiFi or 3G/4G OTT channels
> - to offer a media path with better quality (than best effort data traffic).
> Getting “WebRTC-ready” access and we look forward to telepresence for
> everyone.
>
> An enterprise network that want to keep a restrictive firewall not allowing
> UDP traffic, could provide a real-time path using a TURN server paralleling
> the firewall, instead of tunneling RTP through always open http or https
> ports resulting in RTP media over TCP – with severe quality problems from
> TCP retransmissions of dropped packets. The TURN server address is most
> easily provided in the same way as the IP address and DNS address. (That
> would also put the right party in control – The network provider decides
> what is allowed on his network.)
>
> This browser should select which available TURN server address to use in the
> following priority order, where ICE could be used to try several:
>
> 1) TURN server address configured in the browser by the user (special cases,
> normally not used)
> 2) TURN server address configured by the network administrator via an “admin
> policy template”
> 3) TURN server address supplied by DHCP or similar automatic network method
> 4) TURN server address being supplied by the web application"
>
> Two new requirements can be extracted:
> "   ----------------------------------------------------------------
>     F40     The browser must support retrieving TURN server addresses via
> DHCP or a similar mechanism (maybe RA - Router Advertisement - for IPv6, an
> addition to the IPCP protocol like RFC1877 for PPPoE, or whatever method the
> mobile OTT channels use). The mechanism should be similar to automatically
> getting the IP-address, DNS address, etc..
>     ----------------------------------------------------------------
>     F42     This browser should select which available TURN server address to
> use in the following priority order, where ICE could be used to try several:
>
> 1) TURN server address configured in the browser by the user (special cases,
> normally not used)
> 2) TURN server address configured by the network administrator via an “admin
> policy template”
> 3) TURN server address supplied by DHCP or similar automatic network method
> 4) TURN server address being supplied by the web application
> ----------------------------------------------------------------"
>
> I suppose this also will add to:
> 5.  IANA Considerations
>     TBD
>
> /Karl
>
>
> -----Ursprungligt meddelande-----
> Från: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] För Cullen
> Jennings (fluffy)
> Skickat: den 4 september 2013 18:24
> Till: rtcweb@ietf.org
> Ämne: [rtcweb] WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
>
>
> We would like to start a working group last call of
> draft-ietf-rtcweb-use-cases-and-requirements-11.
>
> Please send comments by the end of the day on September 21.
>
> Thank you,
>
> The chairs ….
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From karl.stahl@intertex.se  Fri Sep 20 15:11:40 2013
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From: "Karl Stahl" <karl.stahl@intertex.se>
To: "'Hutton, Andrew'" <andrew.hutton@siemens-enterprise.com>, "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>, "'Cullen Jennings	\(fluffy\)'" <fluffy@cisco.com>, <rtcweb@ietf.org>, <draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org>
References: <C5E08FE080ACFD4DAE31E4BDBF944EB11667BBA0@xmb-aln-x02.cisco.com>	<5238446D.8050700@ericsson.com> <9F33F40F6F2CD847824537F3C4E37DDF17BCF581@MCHP04MSX.global-ad.net>
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Subject: Re: [rtcweb] WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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I want to point out that all methods suggested for RTP media through
firewalls that do not allow UDP traffic, by using "always open" https =
port
443 (or http port 80) result in tunneling RTP media over TCP. That gives
with severe quality problems from TCP retransmissions of dropped media
packet packets. Do we really want that for WebRTC?

A quality wise better solution is already pointed out in use case =
3.2.5.1
mentioned below:
"An enterprise that uses a RTCWEB based web application for =
communication
desires to audit all RTCWEB based application session used from inside =
the
company towards any external peer. To be able to do this they deploy a =
TURN
server that straddle the boundary
between the internal network and the external"
=20
I suggest the following is added to that use case:
"Such TURN server allows a media path, separated from the usually data
crowded enterprise Internet access, offering superior quality. This is a
better way to enable WebRTC on a network behind a restrictive firewall =
not
allowing UDP traffic, instead of tunneling RTP through always open http =
or
https ports resulting in RTP media over TCP =96 with severe quality =
problems
from TCP retransmissions of dropped packets.=20

Such network administrator provided auditing TURN server would also put =
the
right party in control =96 The network administrator decides what is =
allowed
on his network."

Why should a network administrator that wants to block webrtc media, =
have to
do more than just closing his firewall for UDP, like below suggested
"configuring the browser not to do webrtc, normal HTTP black/white =
lists,
and solutions based on putting something in the HTTP Connect sent to the
proxy so the proxy can block it"?

Another thing in the second half of the same use case 3.2.5.1:=20
Are NAT/firewalls inside the enterprise network assumed? If not, I think =
the
second half can be deleted.

" In cases where employees are using RTCWEB applications provided by an
   external service provider they still want to have the traffic to stay
   inside their internal network and in addition not load the straddling
   TURN server, "

ICE establishes a direct connection in this case, not using any TURN =
server
at all, doesn=92t it? This is then not a problem.

/Karl

-----Ursprungligt meddelande-----
Fr=E5n: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] F=F6r =
Hutton,
Andrew
Skickat: den 20 september 2013 19:58
Till: Magnus Westerlund; Cullen Jennings (fluffy); rtcweb@ietf.org;
draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org
=C4mne: Re: [rtcweb] WGLC of =
draft-ietf-rtcweb-use-cases-and-requirements-11

On: 17 September 2013 13:01 Magnus Westerlund Wrote:
> 3.2.3.1.  Description
>=20
>    This use-case is almost identical to the Simple Video Communication
>    Service use-case (Section 3.2.1).  The difference is that one of =
the
>    users is behind a FW that only allows http traffic.
>=20
> If a firewall only allows HTTP traffic, then can we really assume that =

> the firewall administrator per default will accept WebRTC Media and=20
> Data traffic?
>=20
> I am far from certain of this, and think on a requirement level needs=20
> to express a situation where the firewall administrator allows WebRTC=20
> across its FW, or at least can easily configure a rule to block it.
> Thus
> resulting in that any solution for this needs to be easily=20
> identifiable and possible to block.
>=20

Stefan already stated that this use case will be changed to take account =
of
my comments in
http://www.ietf.org/mail-archive/web/rtcweb/current/msg08264.html. So =
the
"only allows HTTP traffic" is going to be removed and changed to =
describe
the use case when the firewall requires traffic to flow via a HTTP =
Proxy.=20

We also have a use case when a network specific TURN server is deployed
(3.2.5.1).

In both cases I agree that a network administrator must have a way to =
block
webrtc but there are various ways of achieving this including just
configuring the browser not to do webrtc, normal HTTP black/white lists, =
and
also I can imagine solutions based on putting something in the HTTP =
Connect
sent to the proxy so the proxy can block it.=20

So I agree that it must be possible to block webrtc traffic but we need =
to
careful not to specify the solution in the use case. =20

Andy


_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb


From christer.holmberg@ericsson.com  Fri Sep 20 15:17:36 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
Thread-Topic: [rtcweb] JSEP-04: Some comments on Section 5.2.1. and 5.2.2 (19th september)
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Subject: Re: [rtcweb] JSEP-04: Some comments on Section 5.2.1. and 5.2.2 (19th september)
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Hi Cullen,

>> Q_2:      o- line <sess-version> initial value - forking
>>=20
>> When a new PeerConnection is created due to forking, the <sess-version> =
value must be based (incremented) on the value associated with the "mother"=
 PeerConnection, for which the initial Offer of the whole communication ses=
sion was created.
>
> uh, that is old text sorry. I'm proposing the principal that any time the=
 SDP changes (including new candidate lines) the version gets larger.=20

Yes. But keep in mind that this is an Offer created for a new PeerConnectio=
n.=20


----------------

=20
>> Q_4:      BUNDLE
>>=20
>> The text says:
>>=20
>> "If a m=3D section is not being bundled into another m=3D section, it MU=
ST
>>                generate a unique set of ICE credentials and gather its o=
wn set of
>>                candidates.  Otherwise, it MUST use the same ICE credenti=
als and
>>                candidates that were used in the m=3D section that it is =
being bundled
>>                into."
>>=20
>> As, when BUNDLE is used, the initial Offer will contain identical ICE ca=
ndidates, does that mean that we will also include identical address inform=
ation in the initial Offer?
>
> no the initial offer will not have identical stuff other than on lines th=
at "bundle-only" and we still need to add more text on this but idea is to =
make just like what is in united-plan

Regarding the details (usage of port zero etc) of "bundle-only", that is st=
ill something we have to agree upon. But, that discussion belongs to MMUSIC=
 (and, I believe there is a thread about that already :)

But again, in the case of forking, when a new PeerConnection is created, I =
think we should allow identical stuff in the initial Offer of that PeerConn=
ection. Because, the remote party has already indicated support of BUNDLE, =
in the Answer that was sent for the "mother" PeerConnection, so the initial=
 Offer for the new PeerConnection will be the second Offer for the remote p=
arty.

At some point, we are going to have to add much more details about forking,=
 in the case multiple PeerConnections are used. It is not enough to only sa=
y "create a new PeerConnection" :)


----------------


>> Section 5.2.2:
>> -----------------
>>=20
>>=20
>> Q_5:      Offer when in "local-offer"
>>=20
>> The text says:
>>=20
>> "If the initial offer was applied using setLocalDescription, but an
>>                answer from the remote side has not yet been applied, mea=
ning the
>>                PeerConnection is still in the "local-offer" state, the s=
teps for
>>                generating an initial offer should be followed,"
>>=20
>> I don't understand this. Why would you create a new Offer while you are =
waiting for an Answer to the previously sent Offer?
>
> Justin has asked for this in the case when the offer has not been sent to=
 anyone else. This is not going to be useful for typical systems trying to =
have SIP or Jingle compatible flow but who knows ...

Again, we have consensus to base JSEP on 3264. And, if we are going to devi=
ate from 3264, I'd like to have some discussion and justification for that.=
=20

I'd also have a section somewhere which summarizes *all* 3264 deviations th=
at JSEP makes.


Regards,

Christer

From bernard_aboba@hotmail.com  Fri Sep 20 17:09:50 2013
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Harald said:=20
 =20
> In this case=2C it's "only" a DHCP option=2C but what new DHCP options do=
 we=20
> have that have become widely used in the last 10 years?

[BA] And of those=2C which ones are supported by *browsers*?
=20
I do think it is useful for a browser to be able to locate a local TURN ser=
ver=2C if this is to happen it will need to be done at the application laye=
r=2C not via DHCP.=20
 		 	   		  =

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<html>
<head>
<style><!--
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font-size: 12pt=3B
font-family:Calibri
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<body class=3D'hmmessage'><div dir=3D'ltr'>Harald&nbsp=3Bsaid:&nbsp=3B<br>&=
nbsp=3B&nbsp=3B<br>&gt=3B In this case=2C it's "only" a DHCP option=2C but =
what new DHCP options do we <br>&gt=3B have that have become widely used in=
 the last 10 years?<br><BR>[BA] And of those=2C which ones are supported by=
 *browsers*?<BR>&nbsp=3B<BR>I do think it is useful for a browser to be abl=
e to locate a local TURN server=2C if this is to happen it will need to be =
done at the application layer=2C not via DHCP. <BR> 		 	   		  </div></body=
>
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From karl.stahl@intertex.se  Fri Sep 20 17:16:34 2013
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From: "Karl Stahl" <karl.stahl@intertex.se>
To: <rtcweb@ietf.org>, <draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org>
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Subject: [rtcweb]  WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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While reading the draft-ietf-rtcweb-use-cases-and-requirements-11, here are
a few "telephony related" WebRTC things I think should be clarified in the
use cases. 


3.2.1.  Simple Video Communication Service
3.2.1.1.  Description
...  
The invited user might accept or reject the session. 
[Suggest adding] The invited user might accept only audio, rejecting video
(even if a camera is enabled). A user may also select to initiate an audio
session, without video.

And in API requirements:
   ----------------------------------------------------------------
   A1      The Web API must provide means for the application to ask the
browser for permission to use cameras and microphones, individually as input
devices. (One must be able to answer with voice only - declining video.)
   ----------------------------------------------------------------
Same under
6.2.  Browser Considerations
...
The browser is expected to provide mechanisms for users to revise and even
completely revoke consent to use device resources such as camera and
microphone. [Suggest adding] Specifically, a user must be given the
opportunity to only accept audio in a video call invitation.



3.2.12.  Multiparty video communication
3.2.12.1.  Description
...
[Suggest adding] It is essential that automatic adjustments of microphone
volume is disabled, or microphones not spoken into are muted. (This is a
serious problem with most soft clients (SIP clients) of today, plaguing
conferences with ever increasing noise from silent participants.)

And in API requirements:
   ----------------------------------------------------------------
   A15     The Web API must provide means for the web application to adjust
the level in audio streams.
  ----------------------------------------------------------------
   Axx     The Web API must provide means to disable any automatic volume
adjustment in the sent audio streams. (To avoid disturbing noise in
conferences - making many softclients unusable). 
   ----------------------------------------------------------------



3.2.6.  Simple Video Communication Service, access change
3.2.6.1.  Description 
...
the user has to start a trip during the session. The communication device
automatically changes to use WiFi when the Ethernet cable is removed and
then moves to cellular access to the Internet when moving out of WiFi
coverage.  The session continues even though the access method changes.

[Question] Is this some sort of roaming without network support (please
clarify)? Getting a new access will also give the client a new IP address,
won't it? How could then the session continue? The browsers will have no
signaling connection and cannot renegotiate a media connection, can they?


From marc@mocet.com  Fri Sep 20 18:33:50 2013
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Hi, Bernard:

Even hard phones can use "WebRTC" and DHCP options for programming so
having a DHCP option for pointing the phone at a local TURN server makes
sense.

Marc.


On Fri, Sep 20, 2013 at 5:09 PM, Bernard Aboba <bernard_aboba@hotmail.com>wrote:

> Harald said:
>
> > In this case, it's "only" a DHCP option, but what new DHCP options do we
> > have that have become widely used in the last 10 years?
>
> [BA] And of those, which ones are supported by *browsers*?
>
> I do think it is useful for a browser to be able to locate a local TURN
> server, if this is to happen it will need to be done at the application
> layer, not via DHCP.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>


-- 

____________________
+1-949-270-0935

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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:trebuche=
t ms,sans-serif;color:#000000">Hi, Bernard:</div><div class=3D"gmail_defaul=
t" style=3D"font-family:trebuchet ms,sans-serif;color:#000000"><br></div><d=
iv class=3D"gmail_default" style=3D"font-family:trebuchet ms,sans-serif;col=
or:#000000">

Even hard phones can use &quot;WebRTC&quot; and DHCP options for programmin=
g so having a DHCP option for pointing the phone at a local TURN server mak=
es sense.</div><div class=3D"gmail_default" style=3D"font-family:trebuchet =
ms,sans-serif;color:#000000">

<br></div><div class=3D"gmail_default" style=3D"font-family:trebuchet ms,sa=
ns-serif;color:#000000">Marc.</div></div><div class=3D"gmail_extra"><br><br=
><div class=3D"gmail_quote">On Fri, Sep 20, 2013 at 5:09 PM, Bernard Aboba =
<span dir=3D"ltr">&lt;<a href=3D"mailto:bernard_aboba@hotmail.com" target=
=3D"_blank">bernard_aboba@hotmail.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">


<div><div dir=3D"ltr">Harald=A0said:=A0<br>=A0=A0<br>&gt; In this case, it&=
#39;s &quot;only&quot; a DHCP option, but what new DHCP options do we <br>&=
gt; have that have become widely used in the last 10 years?<br><br>[BA] And=
 of those, which ones are supported by *browsers*?<br>

=A0<br>I do think it is useful for a browser to be able to locate a local T=
URN server, if this is to happen it will need to be done at the application=
 layer, not via DHCP. <br> 		 	   		  </div></div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br><br><div=
><font face=3D"trebuchet ms, sans-serif" color=3D"#000000">________________=
____</font></div><div><font face=3D"trebuchet ms, sans-serif" color=3D"#000=
000">+1-949-270-0935</font></div>


</div>

--047d7b624cbeb5470704e6dac620--

From cowwoc@bbs.darktech.org  Sat Sep 21 10:43:20 2013
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On 9/17/2013 4:10 PM, Ted Hardie wrote:
> On Tue, Sep 17, 2013 at 11:54 AM, cowwoc <cowwoc@bbs.darktech.org 
> <mailto:cowwoc@bbs.darktech.org>> wrote:
>
>     Hi Ted,
>
>         Seeing as this discussion stems from licensing concerns, I
>     like to propose the following alternative:
>
>      1. Mandate a video codec whose IPR has expired. I agree that
>         video quality will degrade, which brings me to the next point.
>
>
> To propose a video codec, please write a draft before the deadline and 
> put forward the technical details you're proposing (if there are 
> profiles, for example, which profile).  Saying you want this on the 
> list isn't enough information for folks to judge the proposal.  This 
> is why we are insisting on drafts at a deadline that gives folks the 
> opportunity to review the proposal.

Ted,

     I understand, but unfortunately I don't have the bandwidth to 
author such a draft so I'm out. However, I will support anyone who 
decides go ahead (with H261, Theora, etc).

Gili

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    <meta content="text/html; charset=ISO-8859-1"
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  <body text="#000000" bgcolor="#FFFFFF">
    <div class="moz-cite-prefix">On 9/17/2013 4:10 PM, Ted Hardie wrote:<br>
    </div>
    <blockquote
cite="mid:CA+9kkMC9pTRvNsSeO5CzxsiL7k3ab3MDcceyECawXrpz33tGQQ@mail.gmail.com"
      type="cite">
      <div dir="ltr">On Tue, Sep 17, 2013 at 11:54 AM, cowwoc <span
          dir="ltr">&lt;<a moz-do-not-send="true"
            href="mailto:cowwoc@bbs.darktech.org" target="_blank">cowwoc@bbs.darktech.org</a>&gt;</span>
        wrote:<br>
        <div class="gmail_extra">
          <div class="gmail_quote">
            <blockquote class="gmail_quote" style="margin:0 0 0
              .8ex;border-left:1px #ccc solid;padding-left:1ex">
              <div text="#000000" bgcolor="#FFFFFF">
                <div>Hi Ted,<br>
                  <br>
                  &nbsp;&nbsp;&nbsp; Seeing as this discussion stems from licensing
                  concerns, I like to propose the following alternative:<br>
                  <ol>
                    <li>Mandate a video codec whose IPR has expired. I
                      agree that video quality will degrade, which
                      brings me to the next point.</li>
                  </ol>
                </div>
              </div>
            </blockquote>
            <div><br>
            </div>
            <div>To propose a video codec, please write a draft before
              the deadline and put forward the technical details you're
              proposing (if there are profiles, for example, which
              profile).&nbsp; Saying you want this on the list isn't enough
              information for folks to judge the proposal.&nbsp; This is why
              we are insisting on drafts at a deadline that gives folks
              the opportunity to review the proposal.<br>
            </div>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
    Ted,<br>
    <br>
    &nbsp;&nbsp;&nbsp; I understand, but unfortunately I don't have the bandwidth to
    author such a draft so I'm out. However, I will support anyone who
    decides go ahead (with H261, Theora, etc).<br>
    <br>
    Gili<br>
  </body>
</html>

--------------000101090908020904000101--

From karl.stahl@intertex.se  Sat Sep 21 14:44:45 2013
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From: "Karl Stahl" <karl.stahl@intertex.se>
To: <rtcweb@ietf.org>, <draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org>
References: <C5E08FE080ACFD4DAE31E4BDBF944EB11667BBA0@xmb-aln-x02.cisco.com>	<5238446D.8050700@ericsson.com>	<9F33F40F6F2CD847824537F3C4E37DDF17BCF581@MCHP04MSX.global-ad.net>	<07a601ceb64e$5caaba00$16002e00$@stahl@intertex.se> <07b001ceb65f$ce3f0cf0$6abd26d0$@stahl@intertex.se>
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Subject: [rtcweb] [mmusic] WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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Yet another thing related to
draft-ietf-rtcweb-use-cases-and-requirements-11:
It is about payload type, PT=3D, in SDP and RTP, so I am copying MMUSIC

Network service providers have expressed an interest to know whether =
packets
carry audio or video, to be able to handle them differently in the =
network
(e.g. quality wise). PT is visible outside the encrypted payload in RTP,
however if dynamic payload types PT:96-127 are used, you cannot know =
what
the payload is without knowledge of the SDP (which we for WebRTC must =
assume
the network provider has no knowledge of).

In http://www.ietf.org/assignments/rtp-parameters/rtp-parameters.xml I =
see
no PTs defined for Opus, VP8, H.264 etc. considered for WebRTC.

So, can we have payload types assigned to codecs that will be =
recommended
for WebRTC (PT:35-71 are unassigned)?
Or can we at least split dynamic payload types PT:96-127 into groups for
audio and video codecs?


I relation to that simple request, one may wonder how the network anyway =
can
know what is carried in an UPD packet (the RTP header is no reserved =
field -
it could be the payload of something else).

Quality related requirements F38, A23 and A26 in the use case draft,
nowadays only seem to relate to the browsers, not assuming that =
diffserve
bits or similar are conveyed to the network. That is realistic, since =
most
operating systems don't allow quality markings (diffserve, TOS) of =
packets.
However, 3.2.1.  Simple Video Communication Service, mentions "The web
service monitors the quality of the service (focus on quality of audio =
and
video) the end-users experience.". I don't understand how "The *web =
service*
monitors" based on the listed requirements. Should it be "The *browsers*
monitors"?

What are then the possibilities for a network to classify traffic for
quality or other purposes?

3G/4G networks have DPIs (Deep Packet Inspection) - such box may guess =
what
encrypted RTP traffic is... or may not...

Real time communication protocols using ICE as a pre-protocol to =
establish
media paths give a possibility though. If the network provider offers a =
TURN
server at his access, and enforces the TURN server to be used (by eating
STUN packets), then the RTP flows set up through the TURN server could
classify and mark packets. Then it is useful to know whether it is an =
audio
or video packet by looking at the payload type.

A LAN firewall, can also include a TURN-server, that in addition to
classifying and marking packets for the transport network (if honored -
which rarely is the case on the Internet today), it can also prioritize =
and
traffic shape so the RTP traffic at least is undisturbed through a data
crowded Internet access.

A more difficult request comes from networks using bandwidth reservation
(RSVP) for quality, like mobile and cable networks. Such networks would
benefit from knowing the bandwidth used, but that is not fixes for =
advanced
codecs like the ones considered for WebRTC. One way would be to reserve
maximum bandwidth, and possibly repeat the reservation if less is =
actually
used.

/Karl


-----Ursprungligt meddelande-----
Fr=E5n: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] F=F6r =
Karl
Stahl
Skickat: den 21 september 2013 02:16
Till: rtcweb@ietf.org;
draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org
=C4mne: [rtcweb] WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11

While reading the draft-ietf-rtcweb-use-cases-and-requirements-11, here =
are
a few "telephony related" WebRTC things I think should be clarified in =
the
use cases.=20


3.2.1.  Simple Video Communication Service 3.2.1.1.  Description ... =20
The invited user might accept or reject the session.=20
[Suggest adding] The invited user might accept only audio, rejecting =
video
(even if a camera is enabled). A user may also select to initiate an =
audio
session, without video.

And in API requirements:
   ----------------------------------------------------------------
   A1      The Web API must provide means for the application to ask the
browser for permission to use cameras and microphones, individually as =
input
devices. (One must be able to answer with voice only - declining video.)
   ----------------------------------------------------------------
Same under
6.2.  Browser Considerations
...
The browser is expected to provide mechanisms for users to revise and =
even
completely revoke consent to use device resources such as camera and
microphone. [Suggest adding] Specifically, a user must be given the
opportunity to only accept audio in a video call invitation.



3.2.12.  Multiparty video communication
3.2.12.1.  Description
...
[Suggest adding] It is essential that automatic adjustments of =
microphone
volume is disabled, or microphones not spoken into are muted. (This is a
serious problem with most soft clients (SIP clients) of today, plaguing
conferences with ever increasing noise from silent participants.)

And in API requirements:
   ----------------------------------------------------------------
   A15     The Web API must provide means for the web application to =
adjust
the level in audio streams.
  ----------------------------------------------------------------
   Axx     The Web API must provide means to disable any automatic =
volume
adjustment in the sent audio streams. (To avoid disturbing noise in
conferences - making many softclients unusable).=20
   ----------------------------------------------------------------



3.2.6.  Simple Video Communication Service, access change 3.2.6.1.
Description ...
the user has to start a trip during the session. The communication =
device
automatically changes to use WiFi when the Ethernet cable is removed and
then moves to cellular access to the Internet when moving out of WiFi
coverage.  The session continues even though the access method changes.

[Question] Is this some sort of roaming without network support (please
clarify)? Getting a new access will also give the client a new IP =
address,
won't it? How could then the session continue? The browsers will have no
signaling connection and cannot renegotiate a media connection, can =
they?

_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb


From karl.stahl@intertex.se  Sat Sep 21 15:29:35 2013
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From: "Karl Stahl" <karl.stahl@intertex.se>
To: <rtcweb@ietf.org>, <draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org>
References: <C5E08FE080ACFD4DAE31E4BDBF944EB11667BBA0@xmb-aln-x02.cisco.com> <079001ceb618$2f551ae0$8dff50a0$@stahl@intertex.se>
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Subject: [rtcweb]   [mmusic] TURN server address via DHCP, WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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This is an ICE network infrastructure request (rather than WebRTC =
specific)
so I am copying MMUSIC.

Harald Alvestrand said:=20
> I'm very hesitant to add more requirements that translate to "new =
protocol
needs to be developed, implemented and deployed across the network =
before
the requirement is useful".
--- Who isn't? Still the need is there, especially for network providers =
to
offer TURN servers with their accesses (see below). And worse, it is not =

> "only" a DHCP option
--- it is: "DHCP or a similar mechanism (maybe RA - Router Advertisement =
-
for IPv6, an addition to the IPCP protocol like RFC1877 for PPPoE, or
whatever method the mobile OTT channels use)"
--- But for the browser it is a trivial thing to implement - Just use =
the
DHCP provided TURN server address suggested.=20

The burden to get the standards done fall on others...

There were some comments seeing the need for local TURN-servers (where =
there
are other possibilities than the "automatic DHCP", see below), but I =
want to
stress the importance of allowing network providers to offer TURN =
servers
with their accesses. That is why I call it a "WebRTC-ready" broadband
access.=20

I've heard several carries expressing that they want to provide their =
own
TURN servers with their accesses. The reason has both been to avoid that
media unnecessarily is sent outside their own network to some remote =
TURN
server and for quality reasons. The quality reason is especially =
important
e.g. in cable operator networks, using bandwidth reservation (RSVP) type =
of
QoS. They need to get hold of the RTP media streams and reserve =
bandwidth
for them and can do that by enforcing a TURN server at the access to be
used, and that TURN server can classify and reserve the bandwidth. (They =
do
these things for POTS 3.5kHz voice - They certainly need it for =
telepresence
capable WebRTC.)=20

I think we should be happy and encourage carrier's interest to supply a =
high
quality path for RTC traffic, here WebRTC. (That is a different attitude
than e.g. trying to block Skype.)

/Karl


-----Ursprungligt meddelande-----
Fr=E5n: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] F=F6r =
Karl
Stahl
Skickat: den 20 september 2013 17:44
Till: 'Cullen Jennings (fluffy)'; rtcweb@ietf.org
=C4mne: [rtcweb] TURN server address via DHCP, WGLC of
draft-ietf-rtcweb-use-cases-and-requirements-11

For NAT/Firewall traversal WebRTC uses ICE, where the browser today gets =
the
TURN server address from the web application, or gets it configured by a =
LAN
admin via an =93admin policy template=94. =20

However, there is a need (motivations below) for the TURN server address =
to
be provided via DHCP (and maybe RA - Router Advertisement - for IPv6, =
and
the OTT channel in mobiles have their own method I=92ve been told). =
Simply
put, just as you often get your IP-address, DNS address, etc. =
automatically
from the network access, you should also get the TURN server address.

I suggest the following is added to the end of use case (which today is
about multiple TURN servers) 3.2.4.  Simple Video Communication Service,
global service provider 3.2.4.1.  Description ...
"A network service provider must be able to automatically supply a TURN
server addresses to the browser when accessing a network. The address =
may
come via DHCP or a similar mechanism (maybe RA - Router Advertisement - =
for
IPv6, an addition to the IPCP protocol like RFC1877 for PPPoE, or =
whatever
method the mobile OTT channels use). The mechanism should be similar to
automatically getting the IP-address, DNS address, etc.. and need =
extensions
to current recommendations/standards.=20

There are several reasons for a network service provider to supply a =
TURN
server as part of his offered access:
- to keep media paths short, specifically not sending media outside its =
own
network to some distant application provided TURN server
- to support mobility, i.e. you may want to move from a LAN with a
configured TURN server to accessing via WiFi or 3G/4G OTT channels
- to offer a media path with better quality (than best effort data =
traffic).
Getting =93WebRTC-ready=94 access and we look forward to telepresence =
for
everyone.

An enterprise network that want to keep a restrictive firewall not =
allowing
UDP traffic, could provide a real-time path using a TURN server =
paralleling
the firewall, instead of tunneling RTP through always open http or https
ports resulting in RTP media over TCP =96 with severe quality problems =
from
TCP retransmissions of dropped packets. The TURN server address is most
easily provided in the same way as the IP address and DNS address. (That
would also put the right party in control =96 The network provider =
decides
what is allowed on his network.)

This browser should select which available TURN server address to use in =
the
following priority order, where ICE could be used to try several:

1) TURN server address configured in the browser by the user (special =
cases,
normally not used)
2) TURN server address configured by the network administrator via an =
=93admin
policy template=94
3) TURN server address supplied by DHCP or similar automatic network =
method
4) TURN server address being supplied by the web application"

Two new requirements can be extracted:
"   ----------------------------------------------------------------
   F40     The browser must support retrieving TURN server addresses via
DHCP or a similar mechanism (maybe RA - Router Advertisement - for IPv6, =
an
addition to the IPCP protocol like RFC1877 for PPPoE, or whatever method =
the
mobile OTT channels use). The mechanism should be similar to =
automatically
getting the IP-address, DNS address, etc..
   ----------------------------------------------------------------
   F42     This browser should select which available TURN server =
address to
use in the following priority order, where ICE could be used to try =
several:

1) TURN server address configured in the browser by the user (special =
cases,
normally not used)
2) TURN server address configured by the network administrator via an =
=93admin
policy template=94
3) TURN server address supplied by DHCP or similar automatic network =
method=20
4) TURN server address being supplied by the web application  =20
----------------------------------------------------------------"

I suppose this also will add to:
5.  IANA Considerations
   TBD

/Karl


-----Ursprungligt meddelande-----
Fr=E5n: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] F=F6r =
Cullen
Jennings (fluffy)
Skickat: den 4 september 2013 18:24
Till: rtcweb@ietf.org
=C4mne: [rtcweb] WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11


We would like to start a working group last call of
draft-ietf-rtcweb-use-cases-and-requirements-11.

Please send comments by the end of the day on September 21.=20

Thank you,=20

The chairs =85.




_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb


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To: Bernard Aboba <bernard_aboba@hotmail.com>, Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: Re: [rtcweb] TURN server address via DHCP, WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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=0A=
The use case draft section 3.2.5.1 states=0A=
=0A=
"It mustbe possibele to configure the browsers used in the enterprise with =
 network specific STUN and TURN servers.  This should be possible to achiev=
e by autoconfiguration methods."=0A=
=0A=
I had assumed that this meant using the browser configuration scripts in th=
e same way as the browser would discover the HTTP Proxy I can see that DHCP=
 is also an option which might be useful but I am not sure whether browsers=
 would use it.=0A=
=0A=
Andy=0A=
=0A=
=0A=
From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] on behalf of Bernar=
d Aboba [bernard_aboba@hotmail.com]=0A=
=0A=
Sent: Saturday, September 21, 2013 1:09 AM=0A=
=0A=
To: Harald Alvestrand; rtcweb@ietf.org=0A=
=0A=
Subject: Re: [rtcweb] TURN server address via DHCP, WGLC of draft-ietf-rtcw=
eb-use-cases-and-requirements-11=0A=
=0A=
=0A=
=0A=
=0A=
Harald said: =0A=
=0A=
  =0A=
=0A=
> In this case, it's "only" a DHCP option, but what new DHCP options do we =
=0A=
=0A=
> have that have become widely used in the last 10 years?=0A=
=0A=
=0A=
=0A=
[BA] And of those, which ones are supported by *browsers*?=0A=
=0A=
 =0A=
=0A=
I do think it is useful for a browser to be able to locate a local TURN ser=
ver, if this is to happen it will need to be done at the application layer,=
 not via DHCP.=0A=
=0A=
=0A=
=0A=
=0A=
=0A=
=0A=

From marc@mocet.com  Sat Sep 21 16:26:50 2013
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From: "Marc Abrams" <marc@mocet.com>
To: "Andrew Hutton" <andrew.hutton@siemens-enterprise.com>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] TURN server address via DHCP, WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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But phones and ATAs and other IoT devices would.




Marc=C2=A0

=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F
+1-949-270-0935

On Sat, Sep 21, 2013 at 4:14 PM, Hutton, Andrew
<andrew.hutton@siemens-enterprise.com> wrote:

> The use case draft section 3.2.5.1 states
> =22It mustbe possibele to configure the browsers used in the enterprise =
with  network specific STUN and TURN servers.  This should be possible to =
achieve by autoconfiguration methods.=22
> I had assumed that this meant using the browser configuration scripts in =
the same way as the browser would discover the HTTP Proxy I can see that =
DHCP is also an option which might be useful but I am not sure whether =
browsers would use it.
> Andy
> From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] on behalf of =
Bernard Aboba [bernard=5Faboba@hotmail.com]
> Sent: Saturday, September 21, 2013 1:09 AM
> To: Harald Alvestrand; rtcweb@ietf.org
> Subject: Re: [rtcweb] TURN server address via DHCP, WGLC of =
draft-ietf-rtcweb-use-cases-and-requirements-11
> Harald said:=20
>=20=20=20
>> In this case, it's =22only=22 a DHCP option, but what new DHCP options =
do we=20
>> have that have become widely used in the last 10 years=3F
> [BA] And of those, which ones are supported by *browsers*=3F
>=20=20
> I do think it is useful for a browser to be able to locate a local TURN =
server, if this is to happen it will need to be done at the application =
layer, not via DHCP.
> =5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=
=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
------Nodemailer-0.5.0-?=_1-1379806004426
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<div>But phones and ATAs and other IoT devices would.</div>
<div><br></div>
<div>Marc=C2=A0</div>
<div class=3D=22mailbox=5Fsignature=22>=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=
=5F=5F=5F=5F<br>+1-949-270-0935</div>
<br><br><div class=3D=22gmail=5Fquote=22><p>On Sat, Sep 21, 2013 at 4:14 PM=
, Hutton, Andrew <span dir=3D=22ltr=22>&lt;<a href=3D=22mailto:andrew.=
hutton@siemens-enterprise.com=22 target=3D=22=5Fblank=22>andrew.=
hutton@siemens-enterprise.com</a>&gt;</span> wrote:<br></p><blockquote =
class=3D=22gmail=5Fquote=22 style=3D=22margin:0 0 0 .8ex;border-left:1px =
#ccc solid;padding-left:1ex;=22><br>The use case draft section 3.2.5.1 =
states
<br><br>=22It mustbe possibele to configure the browsers used in the =
enterprise with  network specific STUN and TURN servers.  This should be =
possible to achieve by autoconfiguration methods.=22
<br><br>I had assumed that this meant using the browser configuration =
scripts in the same way as the browser would discover the HTTP Proxy I can =
see that DHCP is also an option which might be useful but I am not sure =
whether browsers would use it.
<br><br>Andy
<br><br><br>From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] on =
behalf of Bernard Aboba [bernard=5Faboba@hotmail.com]
<br><br>Sent: Saturday, September 21, 2013 1:09 AM
<br><br>To: Harald Alvestrand; rtcweb@ietf.org
<br><br>Subject: Re: [rtcweb] TURN server address via DHCP, WGLC of =
draft-ietf-rtcweb-use-cases-and-requirements-11
<br><br><br><br><br>Harald said:=20
<br><br><br><br>&gt; In this case, it's =22only=22 a DHCP option, but what =
new DHCP options do we=20
<br><br>&gt; have that have become widely used in the last 10 years=3F
<br><br><br><br>[BA] And of those, which ones are supported by =
*browsers*=3F
<br><br><br><br>I do think it is useful for a browser to be able to locate =
a local TURN server, if this is to happen it will need to be done at the =
application layer, not via DHCP.
<br><br><br><br><br><br><br>=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=
=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=
=5F=5F=5F=5F=5F=5F=5F
<br>rtcweb mailing list
<br>rtcweb@ietf.org
<br>https://www.ietf.org/mailman/listinfo/rtcweb
<br></blockquote></div><br>
------Nodemailer-0.5.0-?=_1-1379806004426--

From andrew.hutton@siemens-enterprise.com  Sat Sep 21 16:40:58 2013
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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: Karl Stahl <karl.stahl@intertex.se>, 'Magnus Westerlund' <magnus.westerlund@ericsson.com>, "'Cullen Jennings	(fluffy)'" <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org" <draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org>
Thread-Topic: [rtcweb] WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
Thread-Index: AQHOs51tjte/V8mikk+NN5idi26Ro5nO5Y8ggABAlmCAAbVQhg==
Date: Sat, 21 Sep 2013 23:40:37 +0000
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References: <C5E08FE080ACFD4DAE31E4BDBF944EB11667BBA0@xmb-aln-x02.cisco.com> <5238446D.8050700@ericsson.com> <9F33F40F6F2CD847824537F3C4E37DDF17BCF581@MCHP04MSX.global-ad.net>, <07a601ceb64e$5caaba00$16002e00$@stahl@intertex.se>
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Subject: Re: [rtcweb] WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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=0A=
See below.=0A=
=0A=
=0A=
________________________________________=0A=
From: Karl Stahl [karl.stahl@intertex.se]=0A=
Sent: Friday, September 20, 2013 11:11 PM=0A=
To: Hutton, Andrew; 'Magnus Westerlund'; 'Cullen Jennings       (fluffy)'; =
rtcweb@ietf.org; draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.or=
g=0A=
Subject: SV: [rtcweb] WGLC of draft-ietf-rtcweb-use-cases-and-requirements-=
11=0A=
=0A=
I want to point out that all methods suggested for RTP media through=0A=
firewalls that do not allow UDP traffic, by using "always open" https port=
=0A=
443 (or http port 80) result in tunneling RTP media over TCP. That gives=0A=
with severe quality problems from TCP retransmissions of dropped media=0A=
packet packets. Do we really want that for WebRTC?=0A=
=0A=
A quality wise better solution is already pointed out in use case 3.2.5.1=
=0A=
mentioned below:=0A=
"An enterprise that uses a RTCWEB based web application for communication=
=0A=
desires to audit all RTCWEB based application session used from inside the=
=0A=
company towards any external peer. To be able to do this they deploy a TURN=
=0A=
server that straddle the boundary=0A=
between the internal network and the external"=0A=
=0A=
[AndyH] This is not about which is the better solution it is simply describ=
ing use cases. One in which the network administrator is aware of WebRTC an=
d deploys mechanisms to deal with it and one in which there is no administr=
ator or network specific mechanism for handling WebRTC media.  I think the =
use case in which there is no network specific TURN server will be at least=
 in the near term by far the most common use case and therefore we need mec=
hanisms to deal with it and it is a valid use case.=0A=
=0A=
I suggest the following is added to that use case:=0A=
"Such TURN server allows a media path, separated from the usually data=0A=
crowded enterprise Internet access, offering superior quality. This is a=0A=
better way to enable WebRTC on a network behind a restrictive firewall not=
=0A=
allowing UDP traffic, instead of tunneling RTP through always open http or=
=0A=
https ports resulting in RTP media over TCP =96 with severe quality problem=
s=0A=
from TCP retransmissions of dropped packets.=0A=
=0A=
[AndyH] This does not sound like use case text to me it is comparing soluti=
ons to different use cases.=0A=
=0A=
Such network administrator provided auditing TURN server would also put the=
=0A=
right party in control =96 The network administrator decides what is allowe=
d=0A=
on his network."=0A=
=0A=
Why should a network administrator that wants to block webrtc media, have t=
o=0A=
do more than just closing his firewall for UDP, like below suggested=0A=
"configuring the browser not to do webrtc, normal HTTP black/white lists,=
=0A=
and solutions based on putting something in the HTTP Connect sent to the=0A=
proxy so the proxy can block it"?=0A=
=0A=
[AndyH] The use case of most concern is the one where there is no network a=
dministrator and we need WebRTC to work in those environments by default if=
 that means an administrator who is WebRTC aware has to go to a little effo=
rt to block it then I think that is ok.=0A=
=0A=
Another thing in the second half of the same use case 3.2.5.1:=0A=
Are NAT/firewalls inside the enterprise network assumed? If not, I think th=
e=0A=
second half can be deleted.=0A=
=0A=
" In cases where employees are using RTCWEB applications provided by an=0A=
   external service provider they still want to have the traffic to stay=0A=
   inside their internal network and in addition not load the straddling=0A=
   TURN server, "=0A=
=0A=
ICE establishes a direct connection in this case, not using any TURN server=
=0A=
at all, doesn=92t it? This is then not a problem.=0A=
=0A=
/Karl=0A=
=0A=
-----Ursprungligt meddelande-----=0A=
Fr=E5n: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] F=F6r Hutt=
on,=0A=
Andrew=0A=
Skickat: den 20 september 2013 19:58=0A=
Till: Magnus Westerlund; Cullen Jennings (fluffy); rtcweb@ietf.org;=0A=
draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org=0A=
=C4mne: Re: [rtcweb] WGLC of draft-ietf-rtcweb-use-cases-and-requirements-1=
1=0A=
=0A=
On: 17 September 2013 13:01 Magnus Westerlund Wrote:=0A=
> 3.2.3.1.  Description=0A=
>=0A=
>    This use-case is almost identical to the Simple Video Communication=0A=
>    Service use-case (Section 3.2.1).  The difference is that one of the=
=0A=
>    users is behind a FW that only allows http traffic.=0A=
>=0A=
> If a firewall only allows HTTP traffic, then can we really assume that=0A=
> the firewall administrator per default will accept WebRTC Media and=0A=
> Data traffic?=0A=
>=0A=
> I am far from certain of this, and think on a requirement level needs=0A=
> to express a situation where the firewall administrator allows WebRTC=0A=
> across its FW, or at least can easily configure a rule to block it.=0A=
> Thus=0A=
> resulting in that any solution for this needs to be easily=0A=
> identifiable and possible to block.=0A=
>=0A=
=0A=
Stefan already stated that this use case will be changed to take account of=
=0A=
my comments in=0A=
http://www.ietf.org/mail-archive/web/rtcweb/current/msg08264.html. So the=
=0A=
"only allows HTTP traffic" is going to be removed and changed to describe=
=0A=
the use case when the firewall requires traffic to flow via a HTTP Proxy.=
=0A=
=0A=
We also have a use case when a network specific TURN server is deployed=0A=
(3.2.5.1).=0A=
=0A=
In both cases I agree that a network administrator must have a way to block=
=0A=
webrtc but there are various ways of achieving this including just=0A=
configuring the browser not to do webrtc, normal HTTP black/white lists, an=
d=0A=
also I can imagine solutions based on putting something in the HTTP Connect=
=0A=
sent to the proxy so the proxy can block it.=0A=
=0A=
So I agree that it must be possible to block webrtc traffic but we need to=
=0A=
careful not to specify the solution in the use case.=0A=
=0A=
Andy=0A=
=0A=
=0A=
_______________________________________________=0A=
rtcweb mailing list=0A=
rtcweb@ietf.org=0A=
https://www.ietf.org/mailman/listinfo/rtcweb=0A=
=0A=

From andrew.hutton@siemens-enterprise.com  Sat Sep 21 16:44:33 2013
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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: Marc Abrams <marc@mocet.com>
Thread-Topic: [rtcweb] TURN server address via DHCP, WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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Agreed.

So it may be needed but the rtcweb use case draft is about browser requirem=
ents.

Andy

________________________________
From: Marc Abrams [marc@mocet.com]
Sent: Sunday, September 22, 2013 12:26 AM
To: Hutton, Andrew
Cc: rtcweb@ietf.org; Harald Alvestrand; Bernard Aboba
Subject: Re: [rtcweb] TURN server address via DHCP, WGLC of draft-ietf-rtcw=
eb-use-cases-and-requirements-11

But phones and ATAs and other IoT devices would.

Marc
________________
+1-949-270-0935



On Sat, Sep 21, 2013 at 4:14 PM, Hutton, Andrew <andrew.hutton@siemens-ente=
rprise.com<mailto:andrew.hutton@siemens-enterprise.com>> wrote:

The use case draft section 3.2.5.1 states

"It mustbe possibele to configure the browsers used in the enterprise with =
network specific STUN and TURN servers. This should be possible to achieve =
by autoconfiguration methods."

I had assumed that this meant using the browser configuration scripts in th=
e same way as the browser would discover the HTTP Proxy I can see that DHCP=
 is also an option which might be useful but I am not sure whether browsers=
 would use it.

Andy


From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] on behalf of Bernar=
d Aboba [bernard_aboba@hotmail.com]

Sent: Saturday, September 21, 2013 1:09 AM

To: Harald Alvestrand; rtcweb@ietf.org

Subject: Re: [rtcweb] TURN server address via DHCP, WGLC of draft-ietf-rtcw=
eb-use-cases-and-requirements-11




Harald said:



> In this case, it's "only" a DHCP option, but what new DHCP options do we

> have that have become widely used in the last 10 years?



[BA] And of those, which ones are supported by *browsers*?



I do think it is useful for a browser to be able to locate a local TURN ser=
ver, if this is to happen it will need to be done at the application layer,=
 not via DHCP.






_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb


--_000_9F33F40F6F2CD847824537F3C4E37DDF17BD02E8MCHP04MSXglobal_
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<html dir=3D"ltr">
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Diso-8859-=
1">
<style type=3D"text/css" id=3D"owaParaStyle"></style>
</head>
<body fpstyle=3D"1" ocsi=3D"0">
<div style=3D"direction: ltr;font-family: Tahoma;color: #000000;font-size: =
10pt;">
<div><br>
</div>
<div>Agreed. &nbsp;</div>
<div><br>
</div>
<div>So it may be needed but the rtcweb use case draft is about browser req=
uirements.</div>
<div><br>
</div>
<div>Andy</div>
<br>
<div style=3D"font-family: Times New Roman; color: #000000; font-size: 16px=
">
<hr tabindex=3D"-1">
<div id=3D"divRpF503469" style=3D"direction: ltr;"><font face=3D"Tahoma" si=
ze=3D"2" color=3D"#000000"><b>From:</b> Marc Abrams [marc@mocet.com]<br>
<b>Sent:</b> Sunday, September 22, 2013 12:26 AM<br>
<b>To:</b> Hutton, Andrew<br>
<b>Cc:</b> rtcweb@ietf.org; Harald Alvestrand; Bernard Aboba<br>
<b>Subject:</b> Re: [rtcweb] TURN server address via DHCP, WGLC of draft-ie=
tf-rtcweb-use-cases-and-requirements-11<br>
</font><br>
</div>
<div></div>
<div>
<div>But phones and ATAs and other IoT devices would.</div>
<div><br>
</div>
<div>Marc&nbsp;</div>
<div class=3D"mailbox_signature">________________<br>
&#43;1-949-270-0935</div>
<br>
<br>
<div class=3D"gmail_quote">
<p>On Sat, Sep 21, 2013 at 4:14 PM, Hutton, Andrew <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:andrew.hutton@siemens-enterprise.com" target=3D"_blank">and=
rew.hutton@siemens-enterprise.com</a>&gt;</span> wrote:<br>
</p>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex; border-left:1=
px #ccc solid; padding-left:1ex">
<br>
The use case draft section 3.2.5.1 states <br>
<br>
&quot;It mustbe possibele to configure the browsers used in the enterprise =
with network specific STUN and TURN servers. This should be possible to ach=
ieve by autoconfiguration methods.&quot;
<br>
<br>
I had assumed that this meant using the browser configuration scripts in th=
e same way as the browser would discover the HTTP Proxy I can see that DHCP=
 is also an option which might be useful but I am not sure whether browsers=
 would use it.
<br>
<br>
Andy <br>
<br>
<br>
From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] on behalf of Bernar=
d Aboba [bernard_aboba@hotmail.com]
<br>
<br>
Sent: Saturday, September 21, 2013 1:09 AM <br>
<br>
To: Harald Alvestrand; rtcweb@ietf.org <br>
<br>
Subject: Re: [rtcweb] TURN server address via DHCP, WGLC of draft-ietf-rtcw=
eb-use-cases-and-requirements-11
<br>
<br>
<br>
<br>
<br>
Harald said: <br>
<br>
<br>
<br>
&gt; In this case, it's &quot;only&quot; a DHCP option, but what new DHCP o=
ptions do we <br>
<br>
&gt; have that have become widely used in the last 10 years? <br>
<br>
<br>
<br>
[BA] And of those, which ones are supported by *browsers*? <br>
<br>
<br>
<br>
I do think it is useful for a browser to be able to locate a local TURN ser=
ver, if this is to happen it will need to be done at the application layer,=
 not via DHCP.
<br>
<br>
<br>
<br>
<br>
<br>
<br>
_______________________________________________ <br>
rtcweb mailing list <br>
rtcweb@ietf.org <br>
https://www.ietf.org/mailman/listinfo/rtcweb <br>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</body>
</html>

--_000_9F33F40F6F2CD847824537F3C4E37DDF17BD02E8MCHP04MSXglobal_--

From daniel@pocock.com.au  Sun Sep 22 11:16:51 2013
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Subject: [rtcweb] determining mic/webcam availability on page load
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Hi,

I'm not sure if this has been discussed before, but I think it would be
very useful for some applications to determine what hardware is
available very early, well before trying to do things like SIP
registration or making a call.

However, if I use the code below, the result is that the user is
prompted twice for permissions, once during the initial hardware
detection and later when they try to make or receive a call.

Regards,

Daniel



navigator.webkitGetUserMedia(

 {audio: true, video: true},

  function (stream) {
          var has_audio = false;
          var has_video = false;
	  if(stream.getAudioTracks().length > 0)
		  has_audio = true;
	  if(stream.getVideoTracks().length > 0)
		  has_video = true;
	  console.log("closing unused stream");
	  stream.stop();
	  delete stream;
          checksDone(has_audio, has_video);
  },

  function(err) {
	  console.log(err.name + ": " + err.message);
          checksDone(false, false);
  }

);


From harald@alvestrand.no  Sun Sep 22 22:06:06 2013
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On 09/22/2013 08:16 PM, Daniel Pocock wrote:
>
>
> Hi,
>
> I'm not sure if this has been discussed before, but I think it would be
> very useful for some applications to determine what hardware is
> available very early, well before trying to do things like SIP
> registration or making a call.
>
> However, if I use the code below, the result is that the user is
> prompted twice for permissions, once during the initial hardware
> detection and later when they try to make or receive a call.

This is an API issue, so it should go to public-webrtc@w3.org

But it's a solved issue in the spec, even though the implementations may 
be lagging: Use the getSourceInfos call to probe, not the getUserMedia call.


http://dev.w3.org/2011/webrtc/editor/getusermedia.html#methods-1
>
> Regards,
>
> Daniel
>
>
>
> navigator.webkitGetUserMedia(
>
>   {audio: true, video: true},
>
>    function (stream) {
>            var has_audio = false;
>            var has_video = false;
> 	  if(stream.getAudioTracks().length > 0)
> 		  has_audio = true;
> 	  if(stream.getVideoTracks().length > 0)
> 		  has_video = true;
> 	  console.log("closing unused stream");
> 	  stream.stop();
> 	  delete stream;
>            checksDone(has_audio, has_video);
>    },
>
>    function(err) {
> 	  console.log(err.name + ": " + err.message);
>            checksDone(false, false);
>    }
>
> );
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From stefan.lk.hakansson@ericsson.com  Sun Sep 22 22:46:23 2013
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] determining mic/webcam availability on page load
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On 2013-09-23 07:06, Harald Alvestrand wrote:=0A=
> On 09/22/2013 08:16 PM, Daniel Pocock wrote:=0A=
>>=0A=
>>=0A=
>> Hi,=0A=
>>=0A=
>> I'm not sure if this has been discussed before, but I think it would be=
=0A=
>> very useful for some applications to determine what hardware is=0A=
>> available very early, well before trying to do things like SIP=0A=
>> registration or making a call.=0A=
>>=0A=
>> However, if I use the code below, the result is that the user is=0A=
>> prompted twice for permissions, once during the initial hardware=0A=
>> detection and later when they try to make or receive a call.=0A=
>=0A=
> This is an API issue, so it should go to public-webrtc@w3.org=0A=
=0A=
As this deals with access to input devices, I think the correct place is =
=0A=
actually public-media-capture@w3.org=0A=
=0A=
>=0A=
> But it's a solved issue in the spec, even though the implementations may=
=0A=
> be lagging: Use the getSourceInfos call to probe, not the getUserMedia ca=
ll.=0A=
>=0A=
>=0A=
> http://dev.w3.org/2011/webrtc/editor/getusermedia.html#methods-1=0A=
>>=0A=
>> Regards,=0A=
>>=0A=
>> Daniel=0A=
>>=0A=
>>=0A=
>>=0A=
>> navigator.webkitGetUserMedia(=0A=
>>=0A=
>>    {audio: true, video: true},=0A=
>>=0A=
>>     function (stream) {=0A=
>>             var has_audio =3D false;=0A=
>>             var has_video =3D false;=0A=
>> 	  if(stream.getAudioTracks().length > 0)=0A=
>> 		  has_audio =3D true;=0A=
>> 	  if(stream.getVideoTracks().length > 0)=0A=
>> 		  has_video =3D true;=0A=
>> 	  console.log("closing unused stream");=0A=
>> 	  stream.stop();=0A=
>> 	  delete stream;=0A=
>>             checksDone(has_audio, has_video);=0A=
>>     },=0A=
>>=0A=
>>     function(err) {=0A=
>> 	  console.log(err.name + ": " + err.message);=0A=
>>             checksDone(false, false);=0A=
>>     }=0A=
>>=0A=
>> );=0A=
>>=0A=
>> _______________________________________________=0A=
>> rtcweb mailing list=0A=
>> rtcweb@ietf.org=0A=
>> https://www.ietf.org/mailman/listinfo/rtcweb=0A=
>=0A=
> _______________________________________________=0A=
> rtcweb mailing list=0A=
> rtcweb@ietf.org=0A=
> https://www.ietf.org/mailman/listinfo/rtcweb=0A=
>=0A=
=0A=

From sergio.garcia.murillo@gmail.com  Mon Sep 23 02:13:43 2013
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Date: Mon, 23 Sep 2013 11:13:36 +0200
From: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
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Subject: Re: [rtcweb] WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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+1

Shouldn't we add it as a requirement also?

Best regards
Sergio
El 17/09/2013 14:00, Magnus Westerlund escribió:
> 3.2.3.1.  Description
>
>     This use-case is almost identical to the Simple Video Communication
>     Service use-case (Section 3.2.1).  The difference is that one of the
>     users is behind a FW that only allows http traffic.
>
> If a firewall only allows HTTP traffic, then can we really assume that
> the firewall administrator per default will accept WebRTC Media and Data
> traffic?
>
> I am far from certain of this, and think on a requirement level needs to
> express a situation where the firewall administrator allows WebRTC
> across its FW, or at least can easily configure a rule to block it. Thus
> resulting in that any solution for this needs to be easily identifiable
> and possible to block.


From tireddy@cisco.com  Mon Sep 23 02:28:45 2013
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From: "Tirumaleswar Reddy (tireddy)" <tireddy@cisco.com>
To: Bernard Aboba <bernard_aboba@hotmail.com>, Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] TURN server address via DHCP, WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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Subject: Re: [rtcweb] TURN server address via DHCP, WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Bernard Aboba
Sent: Saturday, September 21, 2013 5:40 AM
To: Harald Alvestrand; rtcweb@ietf.org
Subject: Re: [rtcweb] TURN server address via DHCP, WGLC of draft-ietf-rtcw=
eb-use-cases-and-requirements-11

Harald said:

> In this case, it's "only" a DHCP option, but what new DHCP options do we
> have that have become widely used in the last 10 years?

[BA] And of those, which ones are supported by *browsers*?

I do think it is useful for a browser to be able to locate a local TURN ser=
ver, if this is to happen it will need to be done at the application layer,=
 not via DHCP.

 [TR] The other way to locate local TURN servers could be to use DNS-Based =
Service Discovery.

-Tiru.

--_000_913383AAA69FF945B8F946018B75898A1907D036xmbrcdx10ciscoc_
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<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> rtcweb-b=
ounces@ietf.org [mailto:rtcweb-bounces@ietf.org]
<b>On Behalf Of </b>Bernard Aboba<br>
<b>Sent:</b> Saturday, September 21, 2013 5:40 AM<br>
<b>To:</b> Harald Alvestrand; rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] TURN server address via DHCP, WGLC of draft-ie=
tf-rtcweb-use-cases-and-requirements-11<o:p></o:p></span></p>
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<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,&quot=
;sans-serif&quot;">Harald&nbsp;said:&nbsp;<br>
&nbsp;&nbsp;<br>
&gt; In this case, it's &quot;only&quot; a DHCP option, but what new DHCP o=
ptions do we <br>
&gt; have that have become widely used in the last 10 years?<br>
<br>
[BA] And of those, which ones are supported by *browsers*?<br>
&nbsp;<br>
I do think it is useful for a browser to be able to locate a local TURN ser=
ver, if this is to happen it will need to be done at the application layer,=
 not via DHCP.
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;
color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;
color:#1F497D">&nbsp;[TR] The other way to locate local TURN servers could =
be to use DNS-Based Service Discovery.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;
color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;
color:#1F497D">-Tiru.<o:p></o:p></span></p>
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</div>
</div>
</body>
</html>

--_000_913383AAA69FF945B8F946018B75898A1907D036xmbrcdx10ciscoc_--

From cb.list6@gmail.com  Mon Sep 23 09:58:06 2013
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Subject: Re: [rtcweb] TURN server address via DHCP, WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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On Sep 20, 2013 8:43 AM, "Karl Stahl" <karl.stahl@intertex.se> wrote:
>
> For NAT/Firewall traversal WebRTC uses ICE, where the browser today gets
the
> TURN server address from the web application, or gets it configured by a
LAN
> admin via an =93admin policy template=94.
>
> However, there is a need (motivations below) for the TURN server address
to
> be provided via DHCP (and maybe RA - Router Advertisement - for IPv6, and
> the OTT channel in mobiles have their own method I=92ve been told). Simpl=
y
> put, just as you often get your IP-address, DNS address, etc.
automatically
> from the network access, you should also get the TURN server address.
>
> I suggest the following is added to the end of use case (which today is
> about multiple TURN servers)
> 3.2.4.  Simple Video Communication Service, global service provider
> 3.2.4.1.  Description
> ...
> "A network service provider must be able to automatically supply a TURN
> server addresses to the browser when accessing a network. The address may
> come via DHCP or a similar mechanism (maybe RA - Router Advertisement -
for
> IPv6, an addition to the IPCP protocol like RFC1877 for PPPoE, or whateve=
r
> method the mobile OTT channels use). The mechanism should be similar to
> automatically getting the IP-address, DNS address, etc.. and need
extensions
> to current recommendations/standards.
>
> There are several reasons for a network service provider to supply a TURN
> server as part of his offered access:
> - to keep media paths short, specifically not sending media outside its
own
> network to some distant application provided TURN server
> - to support mobility, i.e. you may want to move from a LAN with a
> configured TURN server to accessing via WiFi or 3G/4G OTT channels

I have not read the draft but would like to make it clear that no 2g/3g/4g
provider uses dhcp over the mobile network so this dhcp solution would not
apply.

CB

> - to offer a media path with better quality (than best effort data
traffic).
> Getting =93WebRTC-ready=94 access and we look forward to telepresence for
> everyone.
>
> An enterprise network that want to keep a restrictive firewall not
allowing
> UDP traffic, could provide a real-time path using a TURN server
paralleling
> the firewall, instead of tunneling RTP through always open http or https
> ports resulting in RTP media over TCP =96 with severe quality problems fr=
om
> TCP retransmissions of dropped packets. The TURN server address is most
> easily provided in the same way as the IP address and DNS address. (That
> would also put the right party in control =96 The network provider decide=
s
> what is allowed on his network.)
>
> This browser should select which available TURN server address to use in
the
> following priority order, where ICE could be used to try several:
>
> 1) TURN server address configured in the browser by the user (special
cases,
> normally not used)
> 2) TURN server address configured by the network administrator via an
=93admin
> policy template=94
> 3) TURN server address supplied by DHCP or similar automatic network
method
> 4) TURN server address being supplied by the web application"
>
> Two new requirements can be extracted:
> "   ----------------------------------------------------------------
>    F40     The browser must support retrieving TURN server addresses via
> DHCP or a similar mechanism (maybe RA - Router Advertisement - for IPv6,
an
> addition to the IPCP protocol like RFC1877 for PPPoE, or whatever method
the
> mobile OTT channels use). The mechanism should be similar to automaticall=
y
> getting the IP-address, DNS address, etc..
>    ----------------------------------------------------------------
>    F42     This browser should select which available TURN server address
to
> use in the following priority order, where ICE could be used to try
several:
>
> 1) TURN server address configured in the browser by the user (special
cases,
> normally not used)
> 2) TURN server address configured by the network administrator via an
=93admin
> policy template=94
> 3) TURN server address supplied by DHCP or similar automatic network
method
> 4) TURN server address being supplied by the web application
> ----------------------------------------------------------------"
>
> I suppose this also will add to:
> 5.  IANA Considerations
>    TBD
>
> /Karl
>
>
> -----Ursprungligt meddelande-----
> Fr=E5n: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] F=F6r Cu=
llen
> Jennings (fluffy)
> Skickat: den 4 september 2013 18:24
> Till: rtcweb@ietf.org
> =C4mne: [rtcweb] WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
>
>
> We would like to start a working group last call of
> draft-ietf-rtcweb-use-cases-and-requirements-11.
>
> Please send comments by the end of the day on September 21.
>
> Thank you,
>
> The chairs =85.
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

--001a11c227e8416da104e70feca1
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<p dir=3D"ltr"><br>
On Sep 20, 2013 8:43 AM, &quot;Karl Stahl&quot; &lt;<a href=3D"mailto:karl.=
stahl@intertex.se">karl.stahl@intertex.se</a>&gt; wrote:<br>
&gt;<br>
&gt; For NAT/Firewall traversal WebRTC uses ICE, where the browser today ge=
ts the<br>
&gt; TURN server address from the web application, or gets it configured by=
 a LAN<br>
&gt; admin via an =93admin policy template=94.<br>
&gt;<br>
&gt; However, there is a need (motivations below) for the TURN server addre=
ss to<br>
&gt; be provided via DHCP (and maybe RA - Router Advertisement - for IPv6, =
and<br>
&gt; the OTT channel in mobiles have their own method I=92ve been told). Si=
mply<br>
&gt; put, just as you often get your IP-address, DNS address, etc. automati=
cally<br>
&gt; from the network access, you should also get the TURN server address.<=
br>
&gt;<br>
&gt; I suggest the following is added to the end of use case (which today i=
s<br>
&gt; about multiple TURN servers)<br>
&gt; 3.2.4. =A0Simple Video Communication Service, global service provider<=
br>
&gt; 3.2.4.1. =A0Description<br>
&gt; ...<br>
&gt; &quot;A network service provider must be able to automatically supply =
a TURN<br>
&gt; server addresses to the browser when accessing a network. The address =
may<br>
&gt; come via DHCP or a similar mechanism (maybe RA - Router Advertisement =
- for<br>
&gt; IPv6, an addition to the IPCP protocol like RFC1877 for PPPoE, or what=
ever<br>
&gt; method the mobile OTT channels use). The mechanism should be similar t=
o<br>
&gt; automatically getting the IP-address, DNS address, etc.. and need exte=
nsions<br>
&gt; to current recommendations/standards.<br>
&gt;<br>
&gt; There are several reasons for a network service provider to supply a T=
URN<br>
&gt; server as part of his offered access:<br>
&gt; - to keep media paths short, specifically not sending media outside it=
s own<br>
&gt; network to some distant application provided TURN server<br>
&gt; - to support mobility, i.e. you may want to move from a LAN with a<br>
&gt; configured TURN server to accessing via WiFi or 3G/4G OTT channels</p>
<p dir=3D"ltr">I have not read the draft but would like to make it clear th=
at no 2g/3g/4g provider uses dhcp over the mobile network so this dhcp solu=
tion would not apply. </p>
<p dir=3D"ltr">CB</p>
<p dir=3D"ltr">&gt; - to offer a media path with better quality (than best =
effort data traffic).<br>
&gt; Getting =93WebRTC-ready=94 access and we look forward to telepresence =
for<br>
&gt; everyone.<br>
&gt;<br>
&gt; An enterprise network that want to keep a restrictive firewall not all=
owing<br>
&gt; UDP traffic, could provide a real-time path using a TURN server parall=
eling<br>
&gt; the firewall, instead of tunneling RTP through always open http or htt=
ps<br>
&gt; ports resulting in RTP media over TCP =96 with severe quality problems=
 from<br>
&gt; TCP retransmissions of dropped packets. The TURN server address is mos=
t<br>
&gt; easily provided in the same way as the IP address and DNS address. (Th=
at<br>
&gt; would also put the right party in control =96 The network provider dec=
ides<br>
&gt; what is allowed on his network.)<br>
&gt;<br>
&gt; This browser should select which available TURN server address to use =
in the<br>
&gt; following priority order, where ICE could be used to try several:<br>
&gt;<br>
&gt; 1) TURN server address configured in the browser by the user (special =
cases,<br>
&gt; normally not used)<br>
&gt; 2) TURN server address configured by the network administrator via an =
=93admin<br>
&gt; policy template=94<br>
&gt; 3) TURN server address supplied by DHCP or similar automatic network m=
ethod<br>
&gt; 4) TURN server address being supplied by the web application&quot;<br>
&gt;<br>
&gt; Two new requirements can be extracted:<br>
&gt; &quot; =A0 -----------------------------------------------------------=
-----<br>
&gt; =A0 =A0F40 =A0 =A0 The browser must support retrieving TURN server add=
resses via<br>
&gt; DHCP or a similar mechanism (maybe RA - Router Advertisement - for IPv=
6, an<br>
&gt; addition to the IPCP protocol like RFC1877 for PPPoE, or whatever meth=
od the<br>
&gt; mobile OTT channels use). The mechanism should be similar to automatic=
ally<br>
&gt; getting the IP-address, DNS address, etc..<br>
&gt; =A0 =A0---------------------------------------------------------------=
-<br>
&gt; =A0 =A0F42 =A0 =A0 This browser should select which available TURN ser=
ver address to<br>
&gt; use in the following priority order, where ICE could be used to try se=
veral:<br>
&gt;<br>
&gt; 1) TURN server address configured in the browser by the user (special =
cases,<br>
&gt; normally not used)<br>
&gt; 2) TURN server address configured by the network administrator via an =
=93admin<br>
&gt; policy template=94<br>
&gt; 3) TURN server address supplied by DHCP or similar automatic network m=
ethod<br>
&gt; 4) TURN server address being supplied by the web application<br>
&gt; ----------------------------------------------------------------&quot;=
<br>
&gt;<br>
&gt; I suppose this also will add to:<br>
&gt; 5. =A0IANA Considerations<br>
&gt; =A0 =A0TBD<br>
&gt;<br>
&gt; /Karl<br>
&gt;<br>
&gt;<br>
&gt; -----Ursprungligt meddelande-----<br>
&gt; Fr=E5n: <a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf=
.org</a> [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@=
ietf.org</a>] F=F6r Cullen<br>
&gt; Jennings (fluffy)<br>
&gt; Skickat: den 4 september 2013 18:24<br>
&gt; Till: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; =C4mne: [rtcweb] WGLC of draft-ietf-rtcweb-use-cases-and-requirements-=
11<br>
&gt;<br>
&gt;<br>
&gt; We would like to start a working group last call of<br>
&gt; draft-ietf-rtcweb-use-cases-and-requirements-11.<br>
&gt;<br>
&gt; Please send comments by the end of the day on September 21.<br>
&gt;<br>
&gt; Thank you,<br>
&gt;<br>
&gt; The chairs =85.<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.i=
etf.org/mailman/listinfo/rtcweb</a><br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.i=
etf.org/mailman/listinfo/rtcweb</a><br>
</p>

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From: Dan Wing <dwing@cisco.com>
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On Sep 21, 2013, at 2:44 PM, Karl Stahl <karl.stahl@intertex.se> wrote:

> Yet another thing related to
> draft-ietf-rtcweb-use-cases-and-requirements-11:
> It is about payload type, PT=3D, in SDP and RTP, so I am copying =
MMUSIC
>=20
> Network service providers have expressed an interest to know whether =
packets
> carry audio or video, to be able to handle them differently in the =
network
> (e.g. quality wise). PT is visible outside the encrypted payload in =
RTP,
> however if dynamic payload types PT:96-127 are used, you cannot know =
what
> the payload is without knowledge of the SDP (which we for WebRTC must =
assume
> the network provider has no knowledge of).
>=20
> In http://www.ietf.org/assignments/rtp-parameters/rtp-parameters.xml I =
see
> no PTs defined for Opus, VP8, H.264 etc. considered for WebRTC.
>=20
> So, can we have payload types assigned to codecs that will be =
recommended
> for WebRTC (PT:35-71 are unassigned)?
> Or can we at least split dynamic payload types PT:96-127 into groups =
for
> audio and video codecs?
>=20
>=20
> I relation to that simple request, one may wonder how the network =
anyway can
> know what is carried in an UPD packet (the RTP header is no reserved =
field -
> it could be the payload of something else).
>=20
> Quality related requirements F38, A23 and A26 in the use case draft,
> nowadays only seem to relate to the browsers, not assuming that =
diffserve
> bits or similar are conveyed to the network. That is realistic, since =
most
> operating systems don't allow quality markings (diffserve, TOS) of =
packets.
> However, 3.2.1.  Simple Video Communication Service, mentions "The web
> service monitors the quality of the service (focus on quality of audio =
and
> video) the end-users experience.". I don't understand how "The *web =
service*
> monitors" based on the listed requirements. Should it be "The =
*browsers*
> monitors"?
>=20
> What are then the possibilities for a network to classify traffic for
> quality or other purposes?

I agree there is a problem here, and have been trying to convince others =
there is a problem that needs to be solved.  So far, there appears to be =
scant agreement there is a problem.  See thread on TSVWG starting at =
http://www.ietf.org/mail-archive/web/tsvwg/current/msg12182.html.  The =
solution I am pitching is an extension to PCP, draft-wing-pcp-flowdata.  =
Another solution is MALICE which adds information to the ICE =
connectivity checks (bandwidth, drop preference, etc.) which can be =
DPI'd by network devices.

But before getting deep on solutions, we first we need some consensus =
that some flows need different handling than other flows, and then =
acquiescence that existing techniques do not solve the problem (RSVP, =
NSIS, Diffserv).  Unfortunately the industry doesn't yet seem to agree =
there is even a problem.

-d


>=20
> 3G/4G networks have DPIs (Deep Packet Inspection) - such box may guess =
what
> encrypted RTP traffic is... or may not...
>=20
> Real time communication protocols using ICE as a pre-protocol to =
establish
> media paths give a possibility though. If the network provider offers =
a TURN
> server at his access, and enforces the TURN server to be used (by =
eating
> STUN packets), then the RTP flows set up through the TURN server could
> classify and mark packets. Then it is useful to know whether it is an =
audio
> or video packet by looking at the payload type.
>=20
> A LAN firewall, can also include a TURN-server, that in addition to
> classifying and marking packets for the transport network (if honored =
-
> which rarely is the case on the Internet today), it can also =
prioritize and
> traffic shape so the RTP traffic at least is undisturbed through a =
data
> crowded Internet access.
>=20
> A more difficult request comes from networks using bandwidth =
reservation
> (RSVP) for quality, like mobile and cable networks. Such networks =
would
> benefit from knowing the bandwidth used, but that is not fixes for =
advanced
> codecs like the ones considered for WebRTC. One way would be to =
reserve
> maximum bandwidth, and possibly repeat the reservation if less is =
actually
> used.
>=20
> /Karl
>=20
>=20
> -----Ursprungligt meddelande-----
> Fr=E5n: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] F=F6r =
Karl
> Stahl
> Skickat: den 21 september 2013 02:16
> Till: rtcweb@ietf.org;
> draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org
> =C4mne: [rtcweb] WGLC of =
draft-ietf-rtcweb-use-cases-and-requirements-11
>=20
> While reading the draft-ietf-rtcweb-use-cases-and-requirements-11, =
here are
> a few "telephony related" WebRTC things I think should be clarified in =
the
> use cases.=20
>=20
>=20
> 3.2.1.  Simple Video Communication Service 3.2.1.1.  Description ... =20=

> The invited user might accept or reject the session.=20
> [Suggest adding] The invited user might accept only audio, rejecting =
video
> (even if a camera is enabled). A user may also select to initiate an =
audio
> session, without video.
>=20
> And in API requirements:
>   ----------------------------------------------------------------
>   A1      The Web API must provide means for the application to ask =
the
> browser for permission to use cameras and microphones, individually as =
input
> devices. (One must be able to answer with voice only - declining =
video.)
>   ----------------------------------------------------------------
> Same under
> 6.2.  Browser Considerations
> ...
> The browser is expected to provide mechanisms for users to revise and =
even
> completely revoke consent to use device resources such as camera and
> microphone. [Suggest adding] Specifically, a user must be given the
> opportunity to only accept audio in a video call invitation.
>=20
>=20
>=20
> 3.2.12.  Multiparty video communication
> 3.2.12.1.  Description
> ...
> [Suggest adding] It is essential that automatic adjustments of =
microphone
> volume is disabled, or microphones not spoken into are muted. (This is =
a
> serious problem with most soft clients (SIP clients) of today, =
plaguing
> conferences with ever increasing noise from silent participants.)
>=20
> And in API requirements:
>   ----------------------------------------------------------------
>   A15     The Web API must provide means for the web application to =
adjust
> the level in audio streams.
>  ----------------------------------------------------------------
>   Axx     The Web API must provide means to disable any automatic =
volume
> adjustment in the sent audio streams. (To avoid disturbing noise in
> conferences - making many softclients unusable).=20
>   ----------------------------------------------------------------
>=20
>=20
>=20
> 3.2.6.  Simple Video Communication Service, access change 3.2.6.1.
> Description ...
> the user has to start a trip during the session. The communication =
device
> automatically changes to use WiFi when the Ethernet cable is removed =
and
> then moves to cellular access to the Internet when moving out of WiFi
> coverage.  The session continues even though the access method =
changes.
>=20
> [Question] Is this some sort of roaming without network support =
(please
> clarify)? Getting a new access will also give the client a new IP =
address,
> won't it? How could then the session continue? The browsers will have =
no
> signaling connection and cannot renegotiate a media connection, can =
they?
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From karl.stahl@intertex.se  Mon Sep 23 12:57:58 2013
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To: "'cb.list6'" <cb.list6@gmail.com>
References: <C5E08FE080ACFD4DAE31E4BDBF944EB11667BBA0@xmb-aln-x02.cisco.com>	<523c6d3d.c9d1440a.3b96.7499SMTPIN_ADDED_BROKEN@mx.google.com> <CAD6AjGRXr5kPRQdN+4jkgXHciN3NE7HiRmsb7kaYuzwHEPa7ZA@mail.gmail.com>
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Date: Mon, 23 Sep 2013 21:57:47 +0200
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Cc: "'Cullen Jennings \(fluffy\)'" <fluffy@cisco.com>, rtcweb@ietf.org
Subject: Re: [rtcweb] TURN server address via DHCP, WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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CB > I have not read the draft but would like to make it clear that no
2g/3g/4g provider uses dhcp over the mobile network so this dhcp =
solution
would not apply.=20

[Karl] True, my proposed requirement was:=20

> The browser must support retrieving TURN server addresses via DHCP or =
a
similar mechanism (maybe RA - Router Advertisement - for IPv6, an
addition to the IPCP protocol like RFC1877 for PPPoE, or whatever method =
the
mobile OTT channels use).

=20

=93or whatever method the  mobile OTT channels use=94

is of course not proper language =96 3G/4G do provide IP and DNS =
addresses by
some method and the same is proposed for a network provided TURN server
address.

=20

Actually, it is up to the OS to retrieve, and the browser should only =
use it
for its ICE implementation.

=20

If (hopefully) such network provided TURN server addresses becomes
available, the next suggested browser requirement becomes important:

=20

> The browser should select which available TURN server address to
> use in the following priority order, where ICE could be used to try
several:
>
> 1) TURN server address configured in the browser by the user (special
cases,
> normally not used)
> 2) TURN server address configured by the network administrator via an
=93admin
> policy template=94
> 3) TURN server address supplied by DHCP or similar automatic network
method
> 4) TURN server address being supplied by the web application=20

=20

I believe step 2) and 4) is implemented in Chrome today

/Karl



=20

=20

=20

Fr=E5n: cb.list6 [mailto:cb.list6@gmail.com]=20
Skickat: den 23 september 2013 18:58
Till: Karl Stahl
Kopia: rtcweb@ietf.org; Cullen Jennings (fluffy)
=C4mne: Re: [rtcweb] TURN server address via DHCP, WGLC of
draft-ietf-rtcweb-use-cases-and-requirements-11

=20


On Sep 20, 2013 8:43 AM, "Karl Stahl" <karl.stahl@intertex.se> wrote:
>
> For NAT/Firewall traversal WebRTC uses ICE, where the browser today =
gets
the
> TURN server address from the web application, or gets it configured by =
a
LAN
> admin via an =93admin policy template=94.
>
> However, there is a need (motivations below) for the TURN server =
address
to
> be provided via DHCP (and maybe RA - Router Advertisement - for IPv6, =
and
> the OTT channel in mobiles have their own method I=92ve been told). =
Simply
> put, just as you often get your IP-address, DNS address, etc.
automatically
> from the network access, you should also get the TURN server address.
>
> I suggest the following is added to the end of use case (which today =
is
> about multiple TURN servers)
> 3.2.4.  Simple Video Communication Service, global service provider
> 3.2.4.1.  Description
> ...
> "A network service provider must be able to automatically supply a =
TURN
> server addresses to the browser when accessing a network. The address =
may
> come via DHCP or a similar mechanism (maybe RA - Router Advertisement =
-
for
> IPv6, an addition to the IPCP protocol like RFC1877 for PPPoE, or =
whatever
> method the mobile OTT channels use). The mechanism should be similar =
to
> automatically getting the IP-address, DNS address, etc.. and need
extensions
> to current recommendations/standards.
>
> There are several reasons for a network service provider to supply a =
TURN
> server as part of his offered access:
> - to keep media paths short, specifically not sending media outside =
its
own
> network to some distant application provided TURN server
> - to support mobility, i.e. you may want to move from a LAN with a
> configured TURN server to accessing via WiFi or 3G/4G OTT channels

I have not read the draft but would like to make it clear that no =
2g/3g/4g
provider uses dhcp over the mobile network so this dhcp solution would =
not
apply.=20

CB

> - to offer a media path with better quality (than best effort data
traffic).
> Getting =93WebRTC-ready=94 access and we look forward to telepresence =
for
> everyone.
>
> An enterprise network that want to keep a restrictive firewall not
allowing
> UDP traffic, could provide a real-time path using a TURN server
paralleling
> the firewall, instead of tunneling RTP through always open http or =
https
> ports resulting in RTP media over TCP =96 with severe quality problems =
from
> TCP retransmissions of dropped packets. The TURN server address is =
most
> easily provided in the same way as the IP address and DNS address. =
(That
> would also put the right party in control =96 The network provider =
decides
> what is allowed on his network.)
>
> This browser should select which available TURN server address to use =
in
the
> following priority order, where ICE could be used to try several:
>
> 1) TURN server address configured in the browser by the user (special
cases,
> normally not used)
> 2) TURN server address configured by the network administrator via an
=93admin
> policy template=94
> 3) TURN server address supplied by DHCP or similar automatic network
method
> 4) TURN server address being supplied by the web application"
>
> Two new requirements can be extracted:
> "   ----------------------------------------------------------------
>    F40     The browser must support retrieving TURN server addresses =
via
> DHCP or a similar mechanism (maybe RA - Router Advertisement - for =
IPv6,
an
> addition to the IPCP protocol like RFC1877 for PPPoE, or whatever =
method
the
> mobile OTT channels use). The mechanism should be similar to =
automatically
> getting the IP-address, DNS address, etc..
>    ----------------------------------------------------------------
>    F42     This browser should select which available TURN server =
address
to
> use in the following priority order, where ICE could be used to try
several:
>
> 1) TURN server address configured in the browser by the user (special
cases,
> normally not used)
> 2) TURN server address configured by the network administrator via an
=93admin
> policy template=94
> 3) TURN server address supplied by DHCP or similar automatic network
method
> 4) TURN server address being supplied by the web application
> ----------------------------------------------------------------"
>
> I suppose this also will add to:
> 5.  IANA Considerations
>    TBD
>
> /Karl
>
>
> -----Ursprungligt meddelande-----
> Fr=E5n: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] F=F6r =
Cullen
> Jennings (fluffy)
> Skickat: den 4 september 2013 18:24
> Till: rtcweb@ietf.org
> =C4mne: [rtcweb] WGLC of =
draft-ietf-rtcweb-use-cases-and-requirements-11
>
>
> We would like to start a working group last call of
> draft-ietf-rtcweb-use-cases-and-requirements-11.
>
> Please send comments by the end of the day on September 21.
>
> Thank you,
>
> The chairs =85.
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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</o:shapelayout></xml><![endif]--></head><body lang=3DSV link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p><span lang=3DEN-US>CB =
</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&g=
t; </span><span lang=3DEN-US>I have not read the draft but would like to =
make it clear that no 2g/3g/4g provider uses dhcp over the mobile =
network so this dhcp solution would not apply. =
<o:p></o:p></span></p><p><span lang=3DEN-US>[Karl] True, my proposed =
requirement was: <o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>&gt; The browser must support retrieving TURN server =
addresses via DHCP or a similar mechanism (maybe RA - Router =
Advertisement - for IPv6, an<br>addition to the IPCP protocol like =
RFC1877 for PPPoE, or whatever method the =A0mobile OTT channels =
use).<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>&#8220;or whatever method the =A0mobile OTT channels =
use&#8221;<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>is of course not proper language &#8211; 3G/4G do provide =
IP and DNS addresses by some method and the same is proposed for a =
network provided TURN server address.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>Actually, it is up to the OS to =
retrieve, and the browser should only use it for its ICE =
implementation.<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>If (hopefully) such network provided TURN server addresses =
becomes available, the next suggested browser requirement becomes =
important:<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>&gt; The browser should select which available TURN server =
address to<br>&gt; use in the following priority order, where ICE could =
be used to try several:<br>&gt;<br>&gt; 1) TURN server address =
configured in the browser by the user (special cases,<br>&gt; normally =
not used)<br>&gt; 2) TURN server address configured by the network =
administrator via an &#8220;admin<br>&gt; policy template&#8221;<br>&gt; =
3) TURN server address supplied by DHCP or similar automatic network =
method<br>&gt; 4) TURN server address being supplied by the web =
application <o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>I believe step 2) and 4) is implemented in Chrome =
today<o:p></o:p></span></p><p =
class=3DMsoNormal>/Karl<br><br><o:p></o:p></p><p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
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n></b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> cb.list6 =
[mailto:cb.list6@gmail.com] <br><b>Skickat:</b> den 23 september 2013 =
18:58<br><b>Till:</b> Karl Stahl<br><b>Kopia:</b> rtcweb@ietf.org; =
Cullen Jennings (fluffy)<br><b>=C4mne:</b> Re: [rtcweb] TURN server =
address via DHCP, WGLC of =
draft-ietf-rtcweb-use-cases-and-requirements-11<o:p></o:p></span></p></di=
v><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p><br>On Sep 20, 2013 8:43 =
AM, &quot;Karl Stahl&quot; &lt;<a =
href=3D"mailto:karl.stahl@intertex.se">karl.stahl@intertex.se</a>&gt; =
wrote:<br>&gt;<br>&gt; For NAT/Firewall traversal WebRTC uses ICE, where =
the browser today gets the<br>&gt; TURN server address from the web =
application, or gets it configured by a LAN<br>&gt; admin via an =
&#8220;admin policy template&#8221;.<br>&gt;<br>&gt; However, there is a =
need (motivations below) for the TURN server address to<br>&gt; be =
provided via DHCP (and maybe RA - Router Advertisement - for IPv6, =
and<br>&gt; the OTT channel in mobiles have their own method I&#8217;ve =
been told). Simply<br>&gt; put, just as you often get your IP-address, =
DNS address, etc. automatically<br>&gt; from the network access, you =
should also get the TURN server address.<br>&gt;<br>&gt; I suggest the =
following is added to the end of use case (which today is<br>&gt; about =
multiple TURN servers)<br>&gt; 3.2.4. &nbsp;Simple Video Communication =
Service, global service provider<br>&gt; 3.2.4.1. =
&nbsp;Description<br>&gt; ...<br>&gt; &quot;A network service provider =
must be able to automatically supply a TURN<br>&gt; server addresses to =
the browser when accessing a network. The address may<br>&gt; come via =
DHCP or a similar mechanism (maybe RA - Router Advertisement - =
for<br>&gt; IPv6, an addition to the IPCP protocol like RFC1877 for =
PPPoE, or whatever<br>&gt; method the mobile OTT channels use). The =
mechanism should be similar to<br>&gt; automatically getting the =
IP-address, DNS address, etc.. and need extensions<br>&gt; to current =
recommendations/standards.<br>&gt;<br>&gt; There are several reasons for =
a network service provider to supply a TURN<br>&gt; server as part of =
his offered access:<br>&gt; - to keep media paths short, specifically =
not sending media outside its own<br>&gt; network to some distant =
application provided TURN server<br>&gt; - to support mobility, i.e. you =
may want to move from a LAN with a<br>&gt; configured TURN server to =
accessing via WiFi or 3G/4G OTT channels<o:p></o:p></p><p>I have not =
read the draft but would like to make it clear that no 2g/3g/4g provider =
uses dhcp over the mobile network so this dhcp solution would not apply. =
<o:p></o:p></p><p>CB<o:p></o:p></p><p>&gt; - to offer a media path with =
better quality (than best effort data traffic).<br>&gt; Getting =
&#8220;WebRTC-ready&#8221; access and we look forward to telepresence =
for<br>&gt; everyone.<br>&gt;<br>&gt; An enterprise network that want to =
keep a restrictive firewall not allowing<br>&gt; UDP traffic, could =
provide a real-time path using a TURN server paralleling<br>&gt; the =
firewall, instead of tunneling RTP through always open http or =
https<br>&gt; ports resulting in RTP media over TCP &#8211; with severe =
quality problems from<br>&gt; TCP retransmissions of dropped packets. =
The TURN server address is most<br>&gt; easily provided in the same way =
as the IP address and DNS address. (That<br>&gt; would also put the =
right party in control &#8211; The network provider decides<br>&gt; what =
is allowed on his network.)<br>&gt;<br>&gt; This browser should select =
which available TURN server address to use in the<br>&gt; following =
priority order, where ICE could be used to try several:<br>&gt;<br>&gt; =
1) TURN server address configured in the browser by the user (special =
cases,<br>&gt; normally not used)<br>&gt; 2) TURN server address =
configured by the network administrator via an &#8220;admin<br>&gt; =
policy template&#8221;<br>&gt; 3) TURN server address supplied by DHCP =
or similar automatic network method<br>&gt; 4) TURN server address being =
supplied by the web application&quot;<br>&gt;<br>&gt; Two new =
requirements can be extracted:<br>&gt; &quot; &nbsp; =
----------------------------------------------------------------<br>&gt; =
&nbsp; &nbsp;F40 &nbsp; &nbsp; The browser must support retrieving TURN =
server addresses via<br>&gt; DHCP or a similar mechanism (maybe RA - =
Router Advertisement - for IPv6, an<br>&gt; addition to the IPCP =
protocol like RFC1877 for PPPoE, or whatever method the<br>&gt; mobile =
OTT channels use). The mechanism should be similar to =
automatically<br>&gt; getting the IP-address, DNS address, etc..<br>&gt; =
&nbsp; =
&nbsp;----------------------------------------------------------------<br=
>&gt; &nbsp; &nbsp;F42 &nbsp; &nbsp; This browser should select which =
available TURN server address to<br>&gt; use in the following priority =
order, where ICE could be used to try several:<br>&gt;<br>&gt; 1) TURN =
server address configured in the browser by the user (special =
cases,<br>&gt; normally not used)<br>&gt; 2) TURN server address =
configured by the network administrator via an &#8220;admin<br>&gt; =
policy template&#8221;<br>&gt; 3) TURN server address supplied by DHCP =
or similar automatic network method<br>&gt; 4) TURN server address being =
supplied by the web application<br>&gt; =
----------------------------------------------------------------&quot;<br=
>&gt;<br>&gt; I suppose this also will add to:<br>&gt; 5. &nbsp;IANA =
Considerations<br>&gt; &nbsp; &nbsp;TBD<br>&gt;<br>&gt; =
/Karl<br>&gt;<br>&gt;<br>&gt; -----Ursprungligt meddelande-----<br>&gt; =
Fr=E5n: <a =
href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a> =
[mailto:<a =
href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a>] =
F=F6r Cullen<br>&gt; Jennings (fluffy)<br>&gt; Skickat: den 4 september =
2013 18:24<br>&gt; Till: <a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>&gt; =C4mne: =
[rtcweb] WGLC of =
draft-ietf-rtcweb-use-cases-and-requirements-11<br>&gt;<br>&gt;<br>&gt; =
We would like to start a working group last call of<br>&gt; =
draft-ietf-rtcweb-use-cases-and-requirements-11.<br>&gt;<br>&gt; Please =
send comments by the end of the day on September 21.<br>&gt;<br>&gt; =
Thank you,<br>&gt;<br>&gt; The chairs =
&#8230;.<br>&gt;<br>&gt;<br>&gt;<br>&gt;<br>&gt; =
_______________________________________________<br>&gt; rtcweb mailing =
list<br>&gt; <a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>&gt; <a =
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.or=
g/mailman/listinfo/rtcweb</a><br>&gt;<br>&gt; =
_______________________________________________<br>&gt; rtcweb mailing =
list<br>&gt; <a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>&gt; <a =
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.or=
g/mailman/listinfo/rtcweb</a><o:p></o:p></p></div></body></html>
------=_NextPart_000_0981_01CEB8A7.F0A53090--


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From: "Chenxin (Xin)" <hangzhou.chenxin@huawei.com>
To: Karl Stahl <karl.stahl@intertex.se>, "rtcweb@ietf.org" <rtcweb@ietf.org>,  "draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org" <draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org>
Thread-Topic: [rtcweb]  WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
Thread-Index: AQHOtl/hYQwJQvwb8keiBCP8AmFFOZnUHYEw
Date: Tue, 24 Sep 2013 02:14:26 +0000
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References: <C5E08FE080ACFD4DAE31E4BDBF944EB11667BBA0@xmb-aln-x02.cisco.com> <5238446D.8050700@ericsson.com> <9F33F40F6F2CD847824537F3C4E37DDF17BCF581@MCHP04MSX.global-ad.net> <07a601ceb64e$5caaba00$16002e00$@stahl@intertex.se> <07b001ceb65f$ce3f0cf0$6abd26d0$@stahl@intertex.se>
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Subject: Re: [rtcweb] WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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Hi Karl,=20
>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf O=
f
>Karl Stahl
>Sent: Saturday, September 21, 2013 8:16 AM
>To: rtcweb@ietf.org;
>draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org
>Subject: [rtcweb] WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
>
>While reading the draft-ietf-rtcweb-use-cases-and-requirements-11, here ar=
e
>a few "telephony related" WebRTC things I think should be clarified in the
>use cases.
>
>
>3.2.1.  Simple Video Communication Service
>3.2.1.1.  Description
>...
>The invited user might accept or reject the session.
>[Suggest adding] The invited user might accept only audio, rejecting video
>(even if a camera is enabled). A user may also select to initiate an audio
>session, without video.
>
>And in API requirements:
>   ----------------------------------------------------------------
>   A1      The Web API must provide means for the application to ask the
>browser for permission to use cameras and microphones, individually as inp=
ut
>devices. (One must be able to answer with voice only - declining video.)
>   ----------------------------------------------------------------

>Same under
>6.2.  Browser Considerations
>...
>The browser is expected to provide mechanisms for users to revise and even
>completely revoke consent to use device resources such as camera and
>microphone. [Suggest adding] Specifically, a user must be given the
>opportunity to only accept audio in a video call invitation.
>
[Xin] it is a common use case to accept only audio call and reject the vide=
o and quite useful. But I am doubt that this function should be mixed with =
video or audio device access permission . Do I misunderstand your proposal?=
  I think we could just disable the video stream when signaling. So we coul=
d make video call with one and reject it with other in the same web-service=
. I think the audio and video device access permission is not for each call=
(peer connection).=20

Thanks,
   Xin

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Date: Mon, 23 Sep 2013 22:00:18 -0700
Importance: Normal
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Subject: Re: [rtcweb] additional ICE info
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--_55af0b3f-ccb6-4b86-950e-4c3df1150277_
Content-Type: text/plain; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

Dan Wing said:=20
"Another solution is MALICE which adds information to the ICE connectivity =
checks (bandwidth=2C drop preference=2C etc.) which can be DPI'd by network=
 devices."
[BA]  I am fairly keen on adding some of this information to ICE=2C though =
not to assist DPI.   Rather=2C my concern is that with SDP being optional i=
n WebRTC=2C some of the info that would otherwise be exchanged in the signa=
ling channel may not be present=2C and if present=2C cannot necessarily be =
trusted=2C so that putting this into the ICE exchange has potential securit=
y value.  Also=2C in a situation where TURN servers are rented in the cloud=
=2C it can be useful to signal the maximum bandwidth that could be used for=
 a given allocation.  To my mind=2C the issue to be fixed in the TURN REST =
API wasn't necessarily the ability to steal credentials=2C but the ability =
to misuse those credentials in a way that could impose significant cost on =
the victim.  The act of mis-using a TURN credential does not by itself impo=
se that cost -- but pumping huge amounts of traffic through it would.  		 	=
   		  =

--_55af0b3f-ccb6-4b86-950e-4c3df1150277_
Content-Type: text/html; charset="iso-8859-1"
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<html>
<head>
<style><!--
.hmmessage P
{
margin:0px=3B
padding:0px
}
body.hmmessage
{
font-size: 12pt=3B
font-family:Calibri
}
--></style></head>
<body class=3D'hmmessage'><div dir=3D'ltr'><span style=3D"font-size: 12pt=
=3B">Dan Wing said:&nbsp=3B</span><div><span style=3D"font-size: 12pt=3B"><=
br></span></div><div><span style=3D"font-size: 12pt=3B">"Another solution i=
s MALICE which adds information to the ICE connectivity checks (bandwidth=
=2C drop preference=2C etc.) which can be DPI'd by network devices."</span>=
</div><div><span style=3D"font-size: 12pt=3B"><br></span></div><div><span s=
tyle=3D"font-size: 12pt=3B">[BA] &nbsp=3BI am fairly keen on adding some of=
 this information to ICE=2C though not to assist DPI. &nbsp=3B Rather=2C my=
 concern is that with SDP being optional in WebRTC=2C some of the info that=
 would otherwise be exchanged in the signaling channel may not be present=
=2C and if present=2C cannot necessarily be trusted=2C so that putting this=
 into the ICE exchange has potential security value. &nbsp=3BAlso=2C in a s=
ituation where TURN servers are rented in the cloud=2C it can be useful to =
signal the maximum bandwidth that could be used for a given allocation. &nb=
sp=3BTo my mind=2C the issue to be fixed in the TURN REST API wasn't necess=
arily the ability to steal credentials=2C but the ability to misuse those c=
redentials in a way that could impose significant cost on the victim. &nbsp=
=3BThe act of mis-using a TURN credential does not by itself impose that co=
st -- but pumping huge amounts of traffic through it would.&nbsp=3B</span><=
/div> 		 	   		  </div></body>
</html>=

--_55af0b3f-ccb6-4b86-950e-4c3df1150277_--

From karl.stahl@intertex.se  Tue Sep 24 00:04:03 2013
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From: "Karl Stahl" <karl.stahl@intertex.se>
To: "'Chenxin \(Xin\)'" <hangzhou.chenxin@huawei.com>, <rtcweb@ietf.org>, <draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org>
References: <C5E08FE080ACFD4DAE31E4BDBF944EB11667BBA0@xmb-aln-x02.cisco.com>	<5238446D.8050700@ericsson.com>	<9F33F40F6F2CD847824537F3C4E37DDF17BCF581@MCHP04MSX.global-ad.net>	<07a601ceb64e$5caaba00$16002e00$@stahl@intertex.se> <07b001ceb65f$ce3f0cf0$6abd26d0$@stahl@intertex.se> <9E34D50A21D1D1489134B4D770CE0397680804B3@SZXEMA504-MBX.china.huawei.com>
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-----Ursprungligt meddelande-----
Fr=E5n: Chenxin (Xin) [mailto:hangzhou.chenxin@huawei.com]=20
Skickat: den 24 september 2013 04:14
Till: Karl Stahl; rtcweb@ietf.org;
draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org
=C4mne: RE: [rtcweb] WGLC of =
draft-ietf-rtcweb-use-cases-and-requirements-11

Hi Karl,=20
>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On=20
>Behalf Of Karl Stahl
>Sent: Saturday, September 21, 2013 8:16 AM
>To: rtcweb@ietf.org;
>draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org
>Subject: [rtcweb] WGLC of=20
>draft-ietf-rtcweb-use-cases-and-requirements-11
>
>While reading the draft-ietf-rtcweb-use-cases-and-requirements-11, here =

>are a few "telephony related" WebRTC things I think should be clarified =

>in the use cases.
>
>
>3.2.1.  Simple Video Communication Service 3.2.1.1.  Description ...
>The invited user might accept or reject the session.
>[Suggest adding] The invited user might accept only audio, rejecting=20
>video (even if a camera is enabled). A user may also select to initiate =

>an audio session, without video.
>
>And in API requirements:
>   ----------------------------------------------------------------
>   A1      The Web API must provide means for the application to ask =
the
>browser for permission to use cameras and microphones, individually as=20
>input devices. (One must be able to answer with voice only - declining
video.)
>   ----------------------------------------------------------------

>Same under
>6.2.  Browser Considerations
>...
>The browser is expected to provide mechanisms for users to revise and=20
>even completely revoke consent to use device resources such as camera=20
>and microphone. [Suggest adding] Specifically, a user must be given the =

>opportunity to only accept audio in a video call invitation.
>
[Xin] it is a common use case to accept only audio call and reject the =
video
and quite useful. But I am doubt that this function should be mixed with
video or audio device access permission . Do I misunderstand your =
proposal?
I think we could just disable the video stream when signaling. So we =
could
make video call with one and reject it with other in the same =
web-service. I
think the audio and video device access permission is not for each =
call(peer
connection).=20

Thanks,
   Xin
[Karl] Try using a WebRTC application with Chrome and you will see: The
Permission/Allowance to use Camera and Microphone comes up at a bar at =
the
top of the browser window and is the actual answering of a call.


From hangzhou.chenxin@huawei.com  Tue Sep 24 04:56:16 2013
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From: "Chenxin (Xin)" <hangzhou.chenxin@huawei.com>
To: Karl Stahl <karl.stahl@intertex.se>, "rtcweb@ietf.org" <rtcweb@ietf.org>,  "draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org" <draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org>
Thread-Topic: [rtcweb]  WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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Date: Tue, 24 Sep 2013 11:55:34 +0000
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Hi Karl,
>>While reading the draft-ietf-rtcweb-use-cases-and-requirements-11, here
>>are a few "telephony related" WebRTC things I think should be clarified
>>in the use cases.
>>
>>
>>3.2.1.  Simple Video Communication Service 3.2.1.1.  Description ...
>>The invited user might accept or reject the session.
>>[Suggest adding] The invited user might accept only audio, rejecting
>>video (even if a camera is enabled). A user may also select to initiate
>>an audio session, without video.
>>
>>And in API requirements:
>>   ----------------------------------------------------------------
>>   A1      The Web API must provide means for the application to ask the
>>browser for permission to use cameras and microphones, individually as
>>input devices. (One must be able to answer with voice only - declining
>video.)
>>   ----------------------------------------------------------------
>
>>Same under
>>6.2.  Browser Considerations
>>...
>>The browser is expected to provide mechanisms for users to revise and
>>even completely revoke consent to use device resources such as camera
>>and microphone. [Suggest adding] Specifically, a user must be given the
>>opportunity to only accept audio in a video call invitation.
>>
>[Xin] it is a common use case to accept only audio call and reject the vid=
eo
>and quite useful. But I am doubt that this function should be mixed with
>video or audio device access permission . Do I misunderstand your proposal=
?
>I think we could just disable the video stream when signaling. So we could
>make video call with one and reject it with other in the same web-service.=
 I
>think the audio and video device access permission is not for each call(pe=
er
>connection).
>
>Thanks,
>   Xin
>[Karl] Try using a WebRTC application with Chrome and you will see: The
>Permission/Allowance to use Camera and Microphone comes up at a bar at the
>top of the browser window and is the actual answering of a call.

[Xin] yes, it come up because invoking the getUserMedia API. When we write =
the webrtc app, we could call getUserMedia API just once and use the same m=
ediaStream later. The webrtc app could decide to send the video stream to t=
he other side or not by configure the signaling(SDP or other).  It is trivi=
al to click the Permission bar at the top every time when I get a call, eve=
n terrible when join a p2p conference.=20
That is the reason I think the use case you mentioned should not mix with p=
ermission, which should be a signaling configuration problem. Now in Chrome=
,we could control it by using creatOffer or createAnswer and setting the Of=
ferToReceiveVideo constraint.=20

Thanks,
   Xin

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From: Matt Fredrickson <creslin@digium.com>
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I tend also to agree with this.  Maybe I'm completely off, but it would
seem that pause and resume might be done using some semi-inband media
related path, perhaps like RTCP or something like that.  I'm not sure where
to fit resolution in though...

Matthew Fredrickson
Digium, Inc.


On Fri, Sep 20, 2013 at 5:57 AM, Stefan H=E5kansson LK <
stefan.lk.hakansson@ericsson.com> wrote:

> On 2013-09-20 09:25, Harald Alvestrand wrote:
> > On 09/20/2013 08:02 AM, Magnus Westerlund wrote:
> >> WG,
> >>
> >> In the JSEP call I commented regarding a=3Dimageattr open issue that w=
e
> >> had an original agreement from over a year ago (Vancouver) to go with
> >> Harald's proposal in then
> >> http://tools.ietf.org/id/draft-alvestrand-rtcweb-resolution-00.txt
> >>
> >> This is documented in the minutes:
> >> http://www.ietf.org/proceedings/84/minutes/minutes-84-rtcweb
> >>
> >>
> >> However, I did forget to note that there has been discussion on this
> >> issue after that between Harald Alvestrand and Stefan H=E5kansson
> >> primarily on the W3C list
> >>
> >> http://lists.w3.org/Archives/Public/public-webrtc/2013Aug/0051.html
> >>
> >> Where they are seriously discussing only using WebRTC API for this.
> >>
> >> It is in the context of
> >>
> https://datatracker.ietf.org/doc/draft-alvestrand-constraints-resolution/
> >>
> >> Thus I would like to point out that there appear to be some indication
> >> of desire move away from the consensus last year.
> >
> > My thinking is that there are 3 points on the spectrum of solutions:
> >
> > - Constraints at source only (the approach most thoroughly described in
> > draft-alvestrand-constraints-resolution).
> >
> > - Constraints at destination can be sigalled in SDP, and acted on at
> > source (described in draft-alvestrand-rtcweb-resolution). This spec
> > could not be pursued as long as the "Plan X" discussions were still
> > unsettled, but now it should really be "a piece of cake".
>
> My personal view is still that SDP is the wrong level for carrying this
> kind of signaling (resolution, and perhaps further down the road
> pause/resume related signaling). Each update requires an O/A, and locks
> out other changes to the session. And since the SDPs must be handled by
> the application (that is responsible for sending them over and applying
> them) I wonder if there is any advantage compared to just having an API
> on the sending side. If the receiving application wants another
> resolution sent, it could just tell the sending application so, rather
> than going through a createOffer/setLocal/ send-offer
> /receive-answer/setRemote cycle.
>
> >
> > - Defining a new way of signalling. This was the approach rejected in
> > Vancouver.
> >
> > Formalistically, if the IETF decides to offer a well defined mechanism
> > for negotiating resolution through SDP, the W3C will certainly take up
> > the discussion on whether or not we should connect the API to that
> > functionality.
> >
> > Happy to discuss more.
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> >
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr">I tend also to agree with this. =A0Maybe I&#39;m completel=
y off, but it would seem that pause and resume might be done using some sem=
i-inband media related path, perhaps like RTCP or something like that. =A0I=
&#39;m not sure where to fit resolution in though...<div>
<br></div><div>Matthew Fredrickson</div><div>Digium, Inc.</div></div><div c=
lass=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Fri, Sep 20, 201=
3 at 5:57 AM, Stefan H=E5kansson LK <span dir=3D"ltr">&lt;<a href=3D"mailto=
:stefan.lk.hakansson@ericsson.com" target=3D"_blank">stefan.lk.hakansson@er=
icsson.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"HOEnZb"><div class=3D"h5">On 2=
013-09-20 09:25, Harald Alvestrand wrote:<br>
&gt; On 09/20/2013 08:02 AM, Magnus Westerlund wrote:<br>
&gt;&gt; WG,<br>
&gt;&gt;<br>
&gt;&gt; In the JSEP call I commented regarding a=3Dimageattr open issue th=
at we<br>
&gt;&gt; had an original agreement from over a year ago (Vancouver) to go w=
ith<br>
&gt;&gt; Harald&#39;s proposal in then<br>
&gt;&gt; <a href=3D"http://tools.ietf.org/id/draft-alvestrand-rtcweb-resolu=
tion-00.txt" target=3D"_blank">http://tools.ietf.org/id/draft-alvestrand-rt=
cweb-resolution-00.txt</a><br>
&gt;&gt;<br>
&gt;&gt; This is documented in the minutes:<br>
&gt;&gt; <a href=3D"http://www.ietf.org/proceedings/84/minutes/minutes-84-r=
tcweb" target=3D"_blank">http://www.ietf.org/proceedings/84/minutes/minutes=
-84-rtcweb</a><br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; However, I did forget to note that there has been discussion on th=
is<br>
&gt;&gt; issue after that between Harald Alvestrand and Stefan H=E5kansson<=
br>
&gt;&gt; primarily on the W3C list<br>
&gt;&gt;<br>
&gt;&gt; <a href=3D"http://lists.w3.org/Archives/Public/public-webrtc/2013A=
ug/0051.html" target=3D"_blank">http://lists.w3.org/Archives/Public/public-=
webrtc/2013Aug/0051.html</a><br>
&gt;&gt;<br>
&gt;&gt; Where they are seriously discussing only using WebRTC API for this=
.<br>
&gt;&gt;<br>
&gt;&gt; It is in the context of<br>
&gt;&gt; <a href=3D"https://datatracker.ietf.org/doc/draft-alvestrand-const=
raints-resolution/" target=3D"_blank">https://datatracker.ietf.org/doc/draf=
t-alvestrand-constraints-resolution/</a><br>
&gt;&gt;<br>
&gt;&gt; Thus I would like to point out that there appear to be some indica=
tion<br>
&gt;&gt; of desire move away from the consensus last year.<br>
&gt;<br>
&gt; My thinking is that there are 3 points on the spectrum of solutions:<b=
r>
&gt;<br>
&gt; - Constraints at source only (the approach most thoroughly described i=
n<br>
&gt; draft-alvestrand-constraints-resolution).<br>
&gt;<br>
&gt; - Constraints at destination can be sigalled in SDP, and acted on at<b=
r>
&gt; source (described in draft-alvestrand-rtcweb-resolution). This spec<br=
>
&gt; could not be pursued as long as the &quot;Plan X&quot; discussions wer=
e still<br>
&gt; unsettled, but now it should really be &quot;a piece of cake&quot;.<br=
>
<br>
</div></div>My personal view is still that SDP is the wrong level for carry=
ing this<br>
kind of signaling (resolution, and perhaps further down the road<br>
pause/resume related signaling). Each update requires an O/A, and locks<br>
out other changes to the session. And since the SDPs must be handled by<br>
the application (that is responsible for sending them over and applying<br>
them) I wonder if there is any advantage compared to just having an API<br>
on the sending side. If the receiving application wants another<br>
resolution sent, it could just tell the sending application so, rather<br>
than going through a createOffer/setLocal/ send-offer<br>
/receive-answer/setRemote cycle.<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
&gt;<br>
&gt; - Defining a new way of signalling. This was the approach rejected in<=
br>
&gt; Vancouver.<br>
&gt;<br>
&gt; Formalistically, if the IETF decides to offer a well defined mechanism=
<br>
&gt; for negotiating resolution through SDP, the W3C will certainly take up=
<br>
&gt; the discussion on whether or not we should connect the API to that<br>
&gt; functionality.<br>
&gt;<br>
&gt; Happy to discuss more.<br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt;<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>

--089e0158c41032b41704e7233600--

From karl.stahl@intertex.se  Tue Sep 24 13:08:57 2013
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From: "Karl Stahl" <karl.stahl@intertex.se>
To: "'Chenxin \(Xin\)'" <hangzhou.chenxin@huawei.com>, <rtcweb@ietf.org>, <draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org>
References: <C5E08FE080ACFD4DAE31E4BDBF944EB11667BBA0@xmb-aln-x02.cisco.com>	<5238446D.8050700@ericsson.com>	<9F33F40F6F2CD847824537F3C4E37DDF17BCF581@MCHP04MSX.global-ad.net>	<07a601ceb64e$5caaba00$16002e00$@stahl@intertex.se> <07b001ceb65f$ce3f0cf0$6abd26d0$@stahl@intertex.se> <9E34D50A21D1D1489134B4D770CE0397680804B3@SZXEMA504-MBX.china.huawei.com> <09d801ceb8f4$3b50dfd0$b1f29f70$@stahl@intertex.se> <9E34D50A21D1D1489134B4D770CE0397680805AC@SZXEMA504-MBX.china.huawei.com>
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Date: Tue, 24 Sep 2013 22:08:38 +0200
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Subject: Re: [rtcweb] WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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-----Ursprungligt meddelande-----
Fr=E5n: Chenxin (Xin) [mailto:hangzhou.chenxin@huawei.com]=20
Skickat: den 24 september 2013 13:56
Till: Karl Stahl; rtcweb@ietf.org;
draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org
=C4mne: RE: [rtcweb] WGLC of =
draft-ietf-rtcweb-use-cases-and-requirements-11

Hi Karl,
>>While reading the draft-ietf-rtcweb-use-cases-and-requirements-11,=20
>>here are a few "telephony related" WebRTC things I think should be=20
>>clarified in the use cases.
>>
>>
>>3.2.1.  Simple Video Communication Service 3.2.1.1.  Description ...
>>The invited user might accept or reject the session.
>>[Suggest adding] The invited user might accept only audio, rejecting=20
>>video (even if a camera is enabled). A user may also select to=20
>>initiate an audio session, without video.
>>
>>And in API requirements:
>>   ----------------------------------------------------------------
>>   A1      The Web API must provide means for the application to ask =
the
>>browser for permission to use cameras and microphones, individually as =

>>input devices. (One must be able to answer with voice only - declining
>video.)
>>   ----------------------------------------------------------------
>
>>Same under
>>6.2.  Browser Considerations
>>...
>>The browser is expected to provide mechanisms for users to revise and=20
>>even completely revoke consent to use device resources such as camera=20
>>and microphone. [Suggest adding] Specifically, a user must be given=20
>>the opportunity to only accept audio in a video call invitation.
>>
>[Xin] it is a common use case to accept only audio call and reject the=20
>video and quite useful. But I am doubt that this function should be=20
>mixed with video or audio device access permission . Do I misunderstand
your proposal?
>I think we could just disable the video stream when signaling. So we=20
>could make video call with one and reject it with other in the same=20
>web-service. I think the audio and video device access permission is=20
>not for each call(peer connection).
>
>Thanks,
>   Xin
>[Karl] Try using a WebRTC application with Chrome and you will see: The =

>Permission/Allowance to use Camera and Microphone comes up at a bar at=20
>the top of the browser window and is the actual answering of a call.

[Xin] yes, it come up because invoking the getUserMedia API. When we =
write
the webrtc app, we could call getUserMedia API just once and use the =
same
mediaStream later. The webrtc app could decide to send the video stream =
to
the other side or not by configure the signaling(SDP or other).  It is
trivial to click the Permission bar at the top every time when I get a =
call,
even terrible when join a p2p conference.=20
That is the reason I think the use case you mentioned should not mix =
with
permission, which should be a signaling configuration problem. Now in
Chrome,we could control it by using creatOffer or createAnswer and =
setting
the OfferToReceiveVideo constraint.=20

Thanks,
   Xin
[Karl] If you permit the Browser to use the camera, then later "answer =
with
only audio in the application" (as I understand you suggest) - Can we =
trust
that the application isn't sending our video anyway - not even informing =
us?

I don't think we can trust any video WebRTC application, but will learn =
to
only trust well known browsers not to view us, when we don't want to be
seen.

Isn't this a necessary security thing?

/Karl=20




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From: Dan Wing <dwing@cisco.com>
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References: <C5E08FE080ACFD4DAE31E4BDBF944EB11667BBA0@xmb-aln-x02.cisco.com> <5238446D.8050700@ericsson.com> <9F33F40F6F2CD847824537F3C4E37DDF17BCF581@MCHP04MSX.global-ad.net> <07a601ceb64e$5caaba00$16002e00$@stahl@intertex.se>, <07b001ceb65f$ce3f0cf0$6abd26d0$@stahl@intertex.se>, <07e401ceb713$bef87a60$3ce96f20$@stahl@intertex.se>, <98A3CE39-1BC6-4A36-8DF0-A3932DCDA9AC@cisco.com> <BLU169-W76F559C10477A2CDBB9D2F932E0@phx.gbl>
To: Bernard Aboba <bernard_aboba@hotmail.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] additional ICE info
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On Sep 23, 2013, at 10:00 PM, Bernard Aboba <bernard_aboba@hotmail.com> =
wrote:

> Dan Wing said:=20
>=20
> "Another solution is MALICE which adds information to the ICE =
connectivity checks (bandwidth, drop preference, etc.) which can be =
DPI'd by network devices."
>=20
> [BA]  I am fairly keen on adding some of this information to ICE, =
though not to assist DPI.   Rather, my concern is that with SDP being =
optional in WebRTC, some of the info that would otherwise be exchanged =
in the signaling channel may not be present, and if present, cannot =
necessarily be trusted, so that putting this into the ICE exchange has =
potential security value.  Also, in a situation where TURN servers are =
rented in the cloud, it can be useful to signal the maximum bandwidth =
that could be used for a given allocation. =20

Yes, bandwidth is part of Microsoft's extensions to TURN as I'm sure you =
are aware (but others are probably not aware), =
http://msdn.microsoft.com/en-us/library/cc431507(v=3Doffice.12).aspx

The advantage of allowing routers along the path to see the bandwidth is =
they can influence their routing decisions to put the traffic on a =
better path (e.g., 3G versus microwave link versus satellite), =
prioritize the traffic, and do other things.  If the network always has =
sufficient bandwidth and thus no need to differentiate traffic, I agree =
there is no value and no purpose to tell the network anything.

> To my mind, the issue to be fixed in the TURN REST API wasn't =
necessarily the ability to steal credentials, but the ability to misuse =
those credentials in a way that could impose significant cost on the =
victim.  The act of mis-using a TURN credential does not by itself =
impose that cost -- but pumping huge amounts of traffic through it =
would.=20

-d


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "cb.list6" <cb.list6@gmail.com>
Thread-Topic: [rtcweb] TURN server address via DHCP, WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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Subject: Re: [rtcweb] TURN server address via DHCP, WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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On desktop OS, it is very hard for an application like a browser to learn a=
ny information that was passed via DHCP to the host. This has long been a p=
roblem for things like geopriv.=20

I will note that browsers have many ways to learn about HTTP proxies from t=
he network and it seems to me that using some of theses same technique migh=
t also be a good way to learn about TURN servers.=20


On Sep 23, 2013, at 9:58 AM, cb.list6 <cb.list6@gmail.com> wrote:

>=20
> On Sep 20, 2013 8:43 AM, "Karl Stahl" <karl.stahl@intertex.se> wrote:
> >
> > For NAT/Firewall traversal WebRTC uses ICE, where the browser today get=
s the
> > TURN server address from the web application, or gets it configured by =
a LAN
> > admin via an =93admin policy template=94.
> >
> > However, there is a need (motivations below) for the TURN server addres=
s to
> > be provided via DHCP (and maybe RA - Router Advertisement - for IPv6, a=
nd
> > the OTT channel in mobiles have their own method I=92ve been told). Sim=
ply
> > put, just as you often get your IP-address, DNS address, etc. automatic=
ally
> > from the network access, you should also get the TURN server address.
> >
> > I suggest the following is added to the end of use case (which today is
> > about multiple TURN servers)
> > 3.2.4.  Simple Video Communication Service, global service provider
> > 3.2.4.1.  Description
> > ...
> > "A network service provider must be able to automatically supply a TURN
> > server addresses to the browser when accessing a network. The address m=
ay
> > come via DHCP or a similar mechanism (maybe RA - Router Advertisement -=
 for
> > IPv6, an addition to the IPCP protocol like RFC1877 for PPPoE, or whate=
ver
> > method the mobile OTT channels use). The mechanism should be similar to
> > automatically getting the IP-address, DNS address, etc.. and need exten=
sions
> > to current recommendations/standards.
> >
> > There are several reasons for a network service provider to supply a TU=
RN
> > server as part of his offered access:
> > - to keep media paths short, specifically not sending media outside its=
 own
> > network to some distant application provided TURN server
> > - to support mobility, i.e. you may want to move from a LAN with a
> > configured TURN server to accessing via WiFi or 3G/4G OTT channels
>=20
> I have not read the draft but would like to make it clear that no 2g/3g/4=
g provider uses dhcp over the mobile network so this dhcp solution would no=
t apply.
>=20
> CB
>=20
> > - to offer a media path with better quality (than best effort data traf=
fic).
> > Getting =93WebRTC-ready=94 access and we look forward to telepresence f=
or
> > everyone.
> >
> > An enterprise network that want to keep a restrictive firewall not allo=
wing
> > UDP traffic, could provide a real-time path using a TURN server paralle=
ling
> > the firewall, instead of tunneling RTP through always open http or http=
s
> > ports resulting in RTP media over TCP =96 with severe quality problems =
from
> > TCP retransmissions of dropped packets. The TURN server address is most
> > easily provided in the same way as the IP address and DNS address. (Tha=
t
> > would also put the right party in control =96 The network provider deci=
des
> > what is allowed on his network.)
> >
> > This browser should select which available TURN server address to use i=
n the
> > following priority order, where ICE could be used to try several:
> >
> > 1) TURN server address configured in the browser by the user (special c=
ases,
> > normally not used)
> > 2) TURN server address configured by the network administrator via an =
=93admin
> > policy template=94
> > 3) TURN server address supplied by DHCP or similar automatic network me=
thod
> > 4) TURN server address being supplied by the web application"
> >
> > Two new requirements can be extracted:
> > "   ----------------------------------------------------------------
> >    F40     The browser must support retrieving TURN server addresses vi=
a
> > DHCP or a similar mechanism (maybe RA - Router Advertisement - for IPv6=
, an
> > addition to the IPCP protocol like RFC1877 for PPPoE, or whatever metho=
d the
> > mobile OTT channels use). The mechanism should be similar to automatica=
lly
> > getting the IP-address, DNS address, etc..
> >    ----------------------------------------------------------------
> >    F42     This browser should select which available TURN server addre=
ss to
> > use in the following priority order, where ICE could be used to try sev=
eral:
> >
> > 1) TURN server address configured in the browser by the user (special c=
ases,
> > normally not used)
> > 2) TURN server address configured by the network administrator via an =
=93admin
> > policy template=94
> > 3) TURN server address supplied by DHCP or similar automatic network me=
thod
> > 4) TURN server address being supplied by the web application
> > ----------------------------------------------------------------"
> >
> > I suppose this also will add to:
> > 5.  IANA Considerations
> >    TBD
> >
> > /Karl
> >
> >
> > -----Ursprungligt meddelande-----
> > Fr=E5n: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] F=F6r =
Cullen
> > Jennings (fluffy)
> > Skickat: den 4 september 2013 18:24
> > Till: rtcweb@ietf.org
> > =C4mne: [rtcweb] WGLC of draft-ietf-rtcweb-use-cases-and-requirements-1=
1
> >
> >
> > We would like to start a working group last call of
> > draft-ietf-rtcweb-use-cases-and-requirements-11.
> >
> > Please send comments by the end of the day on September 21.
> >
> > Thank you,
> >
> > The chairs =85.
> >
> >
> >
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb


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Thanks for the pointer!

The existence of that (public) document explains some strange features
of some earlier Microsoft proposals that I was never able to get an
explanation for.

This approach (which seems to have many of the properties of RSVP) seems
to offer a solution to some problems that people have been solving by
snooping SIP (which is impossible if SIP isn't used, and very hard when
all the SIP channels are encrypted). Is there a chance that we could get
public-stable specification of the required pieces, so that other
companies can dare to depend on it?

(public-stable, a term I just invented, includes independent RFC, info
RFC and standards-track)

On 09/25/2013 02:10 AM, Dan Wing wrote:
> On Sep 23, 2013, at 10:00 PM, Bernard Aboba <bernard_aboba@hotmail.com> wrote:
>
>> Dan Wing said: 
>>
>> "Another solution is MALICE which adds information to the ICE connectivity checks (bandwidth, drop preference, etc.) which can be DPI'd by network devices."
>>
>> [BA]  I am fairly keen on adding some of this information to ICE, though not to assist DPI.   Rather, my concern is that with SDP being optional in WebRTC, some of the info that would otherwise be exchanged in the signaling channel may not be present, and if present, cannot necessarily be trusted, so that putting this into the ICE exchange has potential security value.  Also, in a situation where TURN servers are rented in the cloud, it can be useful to signal the maximum bandwidth that could be used for a given allocation.  
> Yes, bandwidth is part of Microsoft's extensions to TURN as I'm sure you are aware (but others are probably not aware), http://msdn.microsoft.com/en-us/library/cc431507(v=office.12).aspx
>
> The advantage of allowing routers along the path to see the bandwidth is they can influence their routing decisions to put the traffic on a better path (e.g., 3G versus microwave link versus satellite), prioritize the traffic, and do other things.  If the network always has sufficient bandwidth and thus no need to differentiate traffic, I agree there is no value and no purpose to tell the network anything.
>
>> To my mind, the issue to be fixed in the TURN REST API wasn't necessarily the ability to steal credentials, but the ability to misuse those credentials in a way that could impose significant cost on the victim.  The act of mis-using a TURN credential does not by itself impose that cost -- but pumping huge amounts of traffic through it would. 
> -d
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


-- 
Surveillance is pervasive. Go Dark.


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On Sep 24, 2013, at 11:15 PM, Harald Alvestrand <harald@alvestrand.no> =
wrote:

> Thanks for the pointer!
>=20
> The existence of that (public) document explains some strange features
> of some earlier Microsoft proposals that I was never able to get an
> explanation for.
>=20
> This approach (which seems to have many of the properties of RSVP) =
seems
> to offer a solution to some problems that people have been solving by
> snooping SIP (which is impossible if SIP isn't used, and very hard =
when
> all the SIP channels are encrypted).

An additional difficulty, even if the SIP signaling is un-encrypted, is =
on complex (enterprise) networks the SIP signaling path and media path =
may go over different network equipment.  Our existing customers solve =
that problem by ensuring their network topology forces the paths to be =
congruent, but forcing such a network topology has other disadvantages =
(link costs go up, and becomes difficult to design around link failure).

> Is there a chance that we could get
> public-stable specification of the required pieces, so that other
> companies can dare to depend on it?

We recently published =
http://tools.ietf.org/html/draft-martinsen-mmusic-malice which does =
something similar.  We are not married to those exact bits-on-the-wire, =
but we see value in the signaling for both the network and the end hosts =
(such as the TURN server).

-d


> (public-stable, a term I just invented, includes independent RFC, info
> RFC and standards-track)
>=20
> On 09/25/2013 02:10 AM, Dan Wing wrote:
>> On Sep 23, 2013, at 10:00 PM, Bernard Aboba =
<bernard_aboba@hotmail.com> wrote:
>>=20
>>> Dan Wing said:=20
>>>=20
>>> "Another solution is MALICE which adds information to the ICE =
connectivity checks (bandwidth, drop preference, etc.) which can be =
DPI'd by network devices."
>>>=20
>>> [BA]  I am fairly keen on adding some of this information to ICE, =
though not to assist DPI.   Rather, my concern is that with SDP being =
optional in WebRTC, some of the info that would otherwise be exchanged =
in the signaling channel may not be present, and if present, cannot =
necessarily be trusted, so that putting this into the ICE exchange has =
potential security value.  Also, in a situation where TURN servers are =
rented in the cloud, it can be useful to signal the maximum bandwidth =
that could be used for a given allocation. =20
>> Yes, bandwidth is part of Microsoft's extensions to TURN as I'm sure =
you are aware (but others are probably not aware), =
http://msdn.microsoft.com/en-us/library/cc431507(v=3Doffice.12).aspx
>>=20
>> The advantage of allowing routers along the path to see the bandwidth =
is they can influence their routing decisions to put the traffic on a =
better path (e.g., 3G versus microwave link versus satellite), =
prioritize the traffic, and do other things.  If the network always has =
sufficient bandwidth and thus no need to differentiate traffic, I agree =
there is no value and no purpose to tell the network anything.
>>=20
>>> To my mind, the issue to be fixed in the TURN REST API wasn't =
necessarily the ability to steal credentials, but the ability to misuse =
those credentials in a way that could impose significant cost on the =
victim.  The act of mis-using a TURN credential does not by itself =
impose that cost -- but pumping huge amounts of traffic through it =
would.=20
>> -d
>>=20
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
>=20
> --=20
> Surveillance is pervasive. Go Dark.
>=20


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To: "'Cullen Jennings \(fluffy\)'" <fluffy@cisco.com>, "'cb.list6'" <cb.list6@gmail.com>
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Subject: Re: [rtcweb] TURN server address via DHCP, WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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-----Ursprungligt meddelande-----
Fr=E5n: Cullen Jennings (fluffy) [mailto:fluffy@cisco.com]=20
Skickat: den 25 september 2013 06:56
Till: cb.list6
Kopia: Karl Stahl; <rtcweb@ietf.org>
=C4mne: Re: [rtcweb] TURN server address via DHCP, WGLC of
draft-ietf-rtcweb-use-cases-and-requirements-11


> On desktop OS, it is very hard for an application like a browser to =
learn
any information that was passed via DHCP to the host. This has long been =
a
problem for things like geopriv.=20
[Karl] A WebRTC browser must be far beyond "a simple application". How =
can
it otherwise see and hear us, send RTP over UDP, do ICE and other =
advanced
stuff? It must already hook deep into the OS, so that in itself cannot =
be a
problem if the OS would know about such a handy thing as a network =
provided
TURN server...

[Karl] The problem is how will Network Provider can tell the OS about =
the
TURN server he provides together with his access: One of our developers
actually came up with a way of doing that WITHOUT my "provided via DHCP =
(and
maybe RA - Router Advertisement - for IPv6, and the OTT channel in =
mobiles
have their own method I=92ve been told)"! I'll come back tomorrow to =
explain
(asked him to write a few line so I can convey correctly).

>I will note that browsers have many ways to learn about HTTP proxies =
from
the network and it seems to me that using some of theses same technique
might also be a good way to learn about TURN servers.=20


On Sep 23, 2013, at 9:58 AM, cb.list6 <cb.list6@gmail.com> wrote:

>=20
> On Sep 20, 2013 8:43 AM, "Karl Stahl" <karl.stahl@intertex.se> wrote:
> >
> > For NAT/Firewall traversal WebRTC uses ICE, where the browser today=20
> > gets the TURN server address from the web application, or gets it=20
> > configured by a LAN admin via an =93admin policy template=94.
> >
> > However, there is a need (motivations below) for the TURN server=20
> > address to be provided via DHCP (and maybe RA - Router Advertisement =

> > - for IPv6, and the OTT channel in mobiles have their own method=20
> > I=92ve been told). Simply put, just as you often get your =
IP-address,=20
> > DNS address, etc. automatically from the network access, you should =
also
get the TURN server address.
> >
> > I suggest the following is added to the end of use case (which today =

> > is about multiple TURN servers) 3.2.4.  Simple Video Communication=20
> > Service, global service provider 3.2.4.1.  Description ...
> > "A network service provider must be able to automatically supply a=20
> > TURN server addresses to the browser when accessing a network. The=20
> > address may come via DHCP or a similar mechanism (maybe RA - Router=20
> > Advertisement - for IPv6, an addition to the IPCP protocol like=20
> > RFC1877 for PPPoE, or whatever method the mobile OTT channels use).=20
> > The mechanism should be similar to automatically getting the=20
> > IP-address, DNS address, etc.. and need extensions to current
recommendations/standards.
> >
> > There are several reasons for a network service provider to supply a =

> > TURN server as part of his offered access:
> > - to keep media paths short, specifically not sending media outside=20
> > its own network to some distant application provided TURN server
> > - to support mobility, i.e. you may want to move from a LAN with a=20
> > configured TURN server to accessing via WiFi or 3G/4G OTT channels
>=20
> I have not read the draft but would like to make it clear that no =
2g/3g/4g
provider uses dhcp over the mobile network so this dhcp solution would =
not
apply.
>=20
> CB
>=20
> > - to offer a media path with better quality (than best effort data
traffic).
> > Getting =93WebRTC-ready=94 access and we look forward to =
telepresence=20
> > for everyone.
> >
> > An enterprise network that want to keep a restrictive firewall not=20
> > allowing UDP traffic, could provide a real-time path using a TURN=20
> > server paralleling the firewall, instead of tunneling RTP through=20
> > always open http or https ports resulting in RTP media over TCP =96=20
> > with severe quality problems from TCP retransmissions of dropped=20
> > packets. The TURN server address is most easily provided in the same =

> > way as the IP address and DNS address. (That would also put the=20
> > right party in control =96 The network provider decides what is=20
> > allowed on his network.)
> >
> > This browser should select which available TURN server address to=20
> > use in the following priority order, where ICE could be used to try
several:
> >
> > 1) TURN server address configured in the browser by the user=20
> > (special cases, normally not used)
> > 2) TURN server address configured by the network administrator via=20
> > an =93admin policy template=94
> > 3) TURN server address supplied by DHCP or similar automatic network =

> > method
> > 4) TURN server address being supplied by the web application"
> >
> > Two new requirements can be extracted:
> > "   ----------------------------------------------------------------
> >    F40     The browser must support retrieving TURN server addresses =
via
> > DHCP or a similar mechanism (maybe RA - Router Advertisement - for=20
> > IPv6, an addition to the IPCP protocol like RFC1877 for PPPoE, or=20
> > whatever method the mobile OTT channels use). The mechanism should=20
> > be similar to automatically getting the IP-address, DNS address, =
etc..
> >    ----------------------------------------------------------------
> >    F42     This browser should select which available TURN server
address to
> > use in the following priority order, where ICE could be used to try
several:
> >
> > 1) TURN server address configured in the browser by the user=20
> > (special cases, normally not used)
> > 2) TURN server address configured by the network administrator via=20
> > an =93admin policy template=94
> > 3) TURN server address supplied by DHCP or similar automatic network =

> > method
> > 4) TURN server address being supplied by the web application=20
> > ----------------------------------------------------------------"
> >
> > I suppose this also will add to:
> > 5.  IANA Considerations
> >    TBD
> >
> > /Karl
> >
> >
> > -----Ursprungligt meddelande-----
> > Fr=E5n: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] =
F=F6r=20
> > Cullen Jennings (fluffy)
> > Skickat: den 4 september 2013 18:24
> > Till: rtcweb@ietf.org
> > =C4mne: [rtcweb] WGLC of=20
> > draft-ietf-rtcweb-use-cases-and-requirements-11
> >
> >
> > We would like to start a working group last call of=20
> > draft-ietf-rtcweb-use-cases-and-requirements-11.
> >
> > Please send comments by the end of the day on September 21.
> >
> > Thank you,
> >
> > The chairs =85.
> >
> >
> >
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb


From junluan.xia@intel.com  Thu Sep 26 00:55:21 2013
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regards,
Andrew


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From paalh@ifi.uio.no  Thu Sep 26 02:11:41 2013
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Subject: [rtcweb] [Packet Video 2013] Call for posters and demos, deadline 11 October
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-=
--------------------------------------------------------------------------=
---------------------
Packet Video 2013
December 12 and 13th =96 San Jose, CA USA
http://pv2013.itec.aau.at/
http://pv2013.itec.aau.at/2013/09/call-for-posters-and-demos/

CALL FOR POSTERS AND DEMOS
Deadline: Friday, October 11th
=
--------------------------------------------------------------------------=
----------------------

We invite Packet Video attendees to participate to a special poster and =
demo session=20
in the Packet Video technical program that will foster lively, informal, =
and in-depth=20
discussions. Topics of interest include, but are not limited to:

- Cloud and peer-to-peer system architectures
- Media streaming, distribution and storage support
- Multimedia communications and system security
- Multi-core and many-core architecture support
-  Networked GPUs, graphics and virtual environments
-  Networked games, real-time immersive systems
- Operating system, middleware and network support
- Web 2.0 systems and social networks
- Next-generation video like multi-view, panorama and 3D
- Wireless networks and embedded systems for multimedia applications
- Content-centric networking

What and How to Submit

Packet Video posters and demos will be selected on the basis of two-page =
PDF abstracts,=20
with fonts no smaller than 10 point, using the same format as for =
regular papers. You=20
can download templates for your abstract on the IEEE website. Abstracts =
must be submitted=20
through the submission system at

http://pv2013.itec.aau.at/submission-poster/

Posters will be reviewed and selected on the following basis:

- Submissions must describe new, interesting work, not previously =
presented. Posters may=20
  be accompanied by demos (subject to limited space). Preference will be =
given toward=20
  posters accompanied by demos.
- Student submissions meeting the above criteria will be given =
preference; however,=20
  non-students may also submit abstracts.=20

Please provide the following information in your PDF file in addition to =
presenting=20
your research:

- Poster title
- Author names, affiliations and email addresses
- Mark which authors, if any, are students
- Indicate if you plan to set up a demo with your poster. If so, the =
submission must include=20
  the requirements for the demo setup and presentation. Note that the =
authors will be responsible=20
  to bring and set up any equipment they will need.

No reviews will be provided.

Posters will be published online on the workshop Web site. Authors of =
the best posters will be=20
given three minutes each to present their work before the poster =
session. At least one author=20
should register and present the poster throughout the entire session.

Important Dates

- Submission: Friday, October 11th
- Notifications: Friday, October 18th
- Workshop

Best regards,
John Apostolopoulos and P=E5l Halvorsen.
(TPC co-chairs)=

From harald@alvestrand.no  Thu Sep 26 03:45:59 2013
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Subject: Re: [rtcweb] TURN server address via DHCP, WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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On 09/26/2013 01:38 AM, Karl Stahl wrote:
>
> -----Ursprungligt meddelande-----
> Från: Cullen Jennings (fluffy) [mailto:fluffy@cisco.com]
> Skickat: den 25 september 2013 06:56
> Till: cb.list6
> Kopia: Karl Stahl; <rtcweb@ietf.org>
> Ämne: Re: [rtcweb] TURN server address via DHCP, WGLC of
> draft-ietf-rtcweb-use-cases-and-requirements-11
>
>
>> On desktop OS, it is very hard for an application like a browser to learn
> any information that was passed via DHCP to the host. This has long been a
> problem for things like geopriv.
> [Karl] A WebRTC browser must be far beyond "a simple application". How can
> it otherwise see and hear us, send RTP over UDP, do ICE and other advanced
> stuff? It must already hook deep into the OS, so that in itself cannot be a
> problem if the OS would know about such a handy thing as a network provided
> TURN server...

All OSes have APIs. These APIs are controlled by the OS vendor.
Most of the information that's common across the relevant platforms is 
the information encapsulated in the POSIX standard (which is the base 
for both *BSD and Linux, which again form the basis for Android, MacOS 
and Linux).

So far, neither the POSIX standard nor any OS vendor has offered a 
generic facility to access information made available in DHCP packets.

Please don't talk about what "must be". Talk about what is. And if you 
don't know what is, ask.


From Markus.Isomaki@nokia.com  Thu Sep 26 05:13:53 2013
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Thread-Topic: [rtcweb] TURN server address via DHCP, WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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Subject: Re: [rtcweb] TURN server address via DHCP, WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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Hi,

Cullen Jennings wrote:
>=20
> I will note that browsers have many ways to learn about HTTP proxies from
> the network and it seems to me that using some of theses same technique
> might also be a good way to learn about TURN servers.
>=20

I agree. I assume that in enterprises it would be the same people managing =
both of these.

Is there something the IETF can or should do about this, or do we just assu=
me it will happen? A new DHCP option is something the IETF could easily do,=
 but I also doubt how usable that would be.=20

Markus=20

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From: "Chenxin (Xin)" <hangzhou.chenxin@huawei.com>
To: Karl Stahl <karl.stahl@intertex.se>, "rtcweb@ietf.org" <rtcweb@ietf.org>,  "draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org" <draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org>
Thread-Topic: [rtcweb]  WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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Subject: Re: [rtcweb] WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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Hi karl,

>-----Original Message-----
>From: Karl Stahl [mailto:karl.stahl@intertex.se]
>Sent: Wednesday, September 25, 2013 4:09 AM
>To: Chenxin (Xin); rtcweb@ietf.org;
>draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org
>Subject: SV: [rtcweb] WGLC of draft-ietf-rtcweb-use-cases-and-requirements=
-11
>
>
>
>-----Ursprungligt meddelande-----
>Fr=E5n: Chenxin (Xin) [mailto:hangzhou.chenxin@huawei.com]
>Skickat: den 24 september 2013 13:56
>Till: Karl Stahl; rtcweb@ietf.org;
>draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org
>=C4mne: RE: [rtcweb] WGLC of draft-ietf-rtcweb-use-cases-and-requirements-=
11
>
>Hi Karl,
>>>While reading the draft-ietf-rtcweb-use-cases-and-requirements-11,
>>>here are a few "telephony related" WebRTC things I think should be
>>>clarified in the use cases.
>>>
>>>
>>>3.2.1.  Simple Video Communication Service 3.2.1.1.  Description ...
>>>The invited user might accept or reject the session.
>>>[Suggest adding] The invited user might accept only audio, rejecting
>>>video (even if a camera is enabled). A user may also select to
>>>initiate an audio session, without video.
>>>
>>>And in API requirements:
>>>   ----------------------------------------------------------------
>>>   A1      The Web API must provide means for the application to ask the
>>>browser for permission to use cameras and microphones, individually as
>>>input devices. (One must be able to answer with voice only - declining
>>video.)
>>>   ----------------------------------------------------------------
>>
>>>Same under
>>>6.2.  Browser Considerations
>>>...
>>>The browser is expected to provide mechanisms for users to revise and
>>>even completely revoke consent to use device resources such as camera
>>>and microphone. [Suggest adding] Specifically, a user must be given
>>>the opportunity to only accept audio in a video call invitation.
>>>
>>[Xin] it is a common use case to accept only audio call and reject the
>>video and quite useful. But I am doubt that this function should be
>>mixed with video or audio device access permission . Do I misunderstand
>your proposal?
>>I think we could just disable the video stream when signaling. So we
>>could make video call with one and reject it with other in the same
>>web-service. I think the audio and video device access permission is
>>not for each call(peer connection).
>>
>>Thanks,
>>   Xin
>>[Karl] Try using a WebRTC application with Chrome and you will see: The
>>Permission/Allowance to use Camera and Microphone comes up at a bar at
>>the top of the browser window and is the actual answering of a call.
>
>[Xin] yes, it come up because invoking the getUserMedia API. When we write
>the webrtc app, we could call getUserMedia API just once and use the same
>mediaStream later. The webrtc app could decide to send the video stream to
>the other side or not by configure the signaling(SDP or other).  It is
>trivial to click the Permission bar at the top every time when I get a cal=
l,
>even terrible when join a p2p conference.
>That is the reason I think the use case you mentioned should not mix with
>permission, which should be a signaling configuration problem. Now in
>Chrome,we could control it by using creatOffer or createAnswer and setting
>the OfferToReceiveVideo constraint.
>
>Thanks,
>   Xin
>[Karl] If you permit the Browser to use the camera, then later "answer wit=
h
>only audio in the application" (as I understand you suggest)=20
[Xin] this is the function of PeerConnection API in W3C.=20

- Can we trust
>that the application isn't sending our video anyway - not even informing u=
s?
>
>I don't think we can trust any video WebRTC application, but will learn to
>only trust well known browsers not to view us, when we don't want to be
>seen.
>
>Isn't this a necessary security thing?
>
>/Karl
>
[Xin] I agree with you that the app is possible to steal user's video or au=
dio stream.=20
But it is still hard to controI if the stream from getUserMedia could be us=
ed by other call(peerconnection) as designed in API now.

What I want to say is that the permission to access the device is orthogona=
l to consent to transmit the media to whom. The first one is for a site. Th=
e second one is for a user, it could be guaranteed by other ways, such as i=
dentity, media-encryption and so on, which need other consent method.=20

 There are some details in http://tools.ietf.org/html/draft-ietf-rtcweb-sec=
urity-05#section-4.1.2.1


From karl.stahl@intertex.se  Thu Sep 26 08:27:26 2013
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From: "Karl Stahl" <karl.stahl@intertex.se>
To: <Markus.Isomaki@nokia.com>, <fluffy@cisco.com>, <cb.list6@gmail.com>
References: <C5E08FE080ACFD4DAE31E4BDBF944EB11667BBA0@xmb-aln-x02.cisco.com>	<523c6d3d.c9d1440a.3b96.7499SMTPIN_ADDED_BROKEN@mx.google.com>	<CAD6AjGRXr5kPRQdN+4jkgXHciN3NE7HiRmsb7kaYuzwHEPa7ZA@mail.gmail.com>	<C5E08FE080ACFD4DAE31E4BDBF944EB1166CC702@xmb-aln-x02.cisco.com> <E44893DD4E290745BB608EB23FDDB7620A0CDABB@008-AM1MPN1-042.mgdnok.nokia.com>
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Cc: rtcweb@ietf.org, draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org
Subject: Re: [rtcweb] TURN server address via DHCP, WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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Here comes the suggestion I got from my developer that would allow a =
network
provider to offer his TURN server for the WebRTC browser to use. This =
would
require NO new DHCP-options or similar, and NO OS changes (I do realize
those should be avoided if possible...).

The idea is still, that whoever is responsible for giving a device an IP
address (the network provider or a LAN administrator) can also announce =
a
TURN server for the WebRTC browser to use.

The suggestion is to use RFC6763 (DNS-Based Service Discovery, see =
chapter
11) where the network provider (the owner of the IP address) has set up =
a
DNS PTR record for the TURN server in the in-addr.arpa domain.=20

If the device got IP 173.164.252.149, then make a query for the PTR =
record
for:
_turn._udp.149.252.164.173.in-addr.arpa.
Then the SRV record would return the actual address to the TURN server =
(and
you may find several for load balancing and failover I guess)

If the device is on a LAN, the IP 173.164.252.149 to query would be the =
WAN
IP you get via STUN in the ICE process. But if the LAN administrator
provides a local TURN server, then he also should have blocked STUN in =
the
firewall, and the browser should query the device's local host address =
(as
in the ICE procedure) and the local LAN DNS server should answer the =
query
to give the local TURN server address on the LAN.

Shouldn't this work to allow network provider to offer his TURN server
"automatically and generally"?

And wouldn't this work nicely for mobile devices, whenever the device is =
on
a "WebRTC-ready access" (actually good for everything using ICE), =
whether
fixed, WiFi or 3G/4G OTT, you get also a TURN server (and the network
provider hopefully offers a prioritized pipe there as one usage of this
mechanism).

Not to slow down every call setup by doing this in the ICE process, I =
guess
this could be done when starting the browser and when the device gets a =
new
IP address, to have the TURN server address ready for later use.

/Karl


-----Ursprungligt meddelande-----
Fr=E5n: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] F=F6r
Markus.Isomaki@nokia.com
Skickat: den 26 september 2013 14:06
Till: fluffy@cisco.com; cb.list6@gmail.com
Kopia: rtcweb@ietf.org
=C4mne: Re: [rtcweb] TURN server address via DHCP, WGLC of
draft-ietf-rtcweb-use-cases-and-requirements-11

Hi,

Cullen Jennings wrote:
>=20
> I will note that browsers have many ways to learn about HTTP proxies=20
> from the network and it seems to me that using some of theses same=20
> technique might also be a good way to learn about TURN servers.
>=20

I agree. I assume that in enterprises it would be the same people =
managing
both of these.

Is there something the IETF can or should do about this, or do we just
assume it will happen? A new DHCP option is something the IETF could =
easily
do, but I also doubt how usable that would be.=20

Markus
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb


From karl.stahl@intertex.se  Thu Sep 26 08:49:27 2013
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From: "Karl Stahl" <karl.stahl@intertex.se>
To: "'Chenxin \(Xin\)'" <hangzhou.chenxin@huawei.com>, <rtcweb@ietf.org>, <draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org>
References: <C5E08FE080ACFD4DAE31E4BDBF944EB11667BBA0@xmb-aln-x02.cisco.com>	<5238446D.8050700@ericsson.com>	<9F33F40F6F2CD847824537F3C4E37DDF17BCF581@MCHP04MSX.global-ad.net>	<07a601ceb64e$5caaba00$16002e00$@stahl@intertex.se> <07b001ceb65f$ce3f0cf0$6abd26d0$@stahl@intertex.se> <9E34D50A21D1D1489134B4D770CE0397680804B3@SZXEMA504-MBX.china.huawei.com> <09d801ceb8f4$3b50dfd0$b1f29f70$@stahl@intertex.se> <9E34D50A21D1D1489134B4D770CE0397680805AC@SZXEMA504-MBX.china.huawei.com> <0b5b01ceb961$db8cff20$92a6fd60$@stahl@intertex.se> <9E34D50A21D1D1489134B4D770CE03976808095D@SZXEMA504-MBX.china.huawei.com>
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Subject: Re: [rtcweb] WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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Hi Xin,
> permission to access the device is orthogonal to consent to transmit =
the
media to whom....

Thanks for the draft pointer, but what about the common usage that I =
pass
you a link: "Click here so we can discuss further", You click and run =
_my_
application och you are prompted by your browser to permit Mic&Camera, =
and
get no further option at all to restrict to only voice?

Doesn't _the browser_ really have to show an "Only Mic"-permission also,
i.e. three buttons "Deny", "Video Call", "Voice Call"?

/Karl

-----Ursprungligt meddelande-----
Fr=E5n: Chenxin (Xin) [mailto:hangzhou.chenxin@huawei.com]=20
Skickat: den 26 september 2013 14:59
Till: Karl Stahl; rtcweb@ietf.org;
draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org
=C4mne: RE: [rtcweb] WGLC of =
draft-ietf-rtcweb-use-cases-and-requirements-11

Hi karl,

>-----Original Message-----
>From: Karl Stahl [mailto:karl.stahl@intertex.se]
>Sent: Wednesday, September 25, 2013 4:09 AM
>To: Chenxin (Xin); rtcweb@ietf.org;
>draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org
>Subject: SV: [rtcweb] WGLC of=20
>draft-ietf-rtcweb-use-cases-and-requirements-11
>
>
>
>-----Ursprungligt meddelande-----
>Fr=E5n: Chenxin (Xin) [mailto:hangzhou.chenxin@huawei.com]
>Skickat: den 24 september 2013 13:56
>Till: Karl Stahl; rtcweb@ietf.org;
>draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org
>=C4mne: RE: [rtcweb] WGLC of=20
>draft-ietf-rtcweb-use-cases-and-requirements-11
>
>Hi Karl,
>>>While reading the draft-ietf-rtcweb-use-cases-and-requirements-11,
>>>here are a few "telephony related" WebRTC things I think should be=20
>>>clarified in the use cases.
>>>
>>>
>>>3.2.1.  Simple Video Communication Service 3.2.1.1.  Description ...
>>>The invited user might accept or reject the session.
>>>[Suggest adding] The invited user might accept only audio, rejecting=20
>>>video (even if a camera is enabled). A user may also select to=20
>>>initiate an audio session, without video.
>>>
>>>And in API requirements:
>>>   ----------------------------------------------------------------
>>>   A1      The Web API must provide means for the application to ask =
the
>>>browser for permission to use cameras and microphones, individually=20
>>>as input devices. (One must be able to answer with voice only -=20
>>>declining
>>video.)
>>>   ----------------------------------------------------------------
>>
>>>Same under
>>>6.2.  Browser Considerations
>>>...
>>>The browser is expected to provide mechanisms for users to revise and =

>>>even completely revoke consent to use device resources such as camera =

>>>and microphone. [Suggest adding] Specifically, a user must be given=20
>>>the opportunity to only accept audio in a video call invitation.
>>>
>>[Xin] it is a common use case to accept only audio call and reject the =

>>video and quite useful. But I am doubt that this function should be=20
>>mixed with video or audio device access permission . Do I=20
>>misunderstand
>your proposal?
>>I think we could just disable the video stream when signaling. So we=20
>>could make video call with one and reject it with other in the same=20
>>web-service. I think the audio and video device access permission is=20
>>not for each call(peer connection).
>>
>>Thanks,
>>   Xin
>>[Karl] Try using a WebRTC application with Chrome and you will see:=20
>>The Permission/Allowance to use Camera and Microphone comes up at a=20
>>bar at the top of the browser window and is the actual answering of a
call.
>
>[Xin] yes, it come up because invoking the getUserMedia API. When we=20
>write the webrtc app, we could call getUserMedia API just once and use=20
>the same mediaStream later. The webrtc app could decide to send the=20
>video stream to the other side or not by configure the signaling(SDP or =

>other).  It is trivial to click the Permission bar at the top every=20
>time when I get a call, even terrible when join a p2p conference.
>That is the reason I think the use case you mentioned should not mix=20
>with permission, which should be a signaling configuration problem. Now =

>in Chrome,we could control it by using creatOffer or createAnswer and=20
>setting the OfferToReceiveVideo constraint.
>
>Thanks,
>   Xin
>[Karl] If you permit the Browser to use the camera, then later "answer=20
>with only audio in the application" (as I understand you suggest)
[Xin] this is the function of PeerConnection API in W3C.=20

- Can we trust
>that the application isn't sending our video anyway - not even =
informing
us?
>
>I don't think we can trust any video WebRTC application, but will learn =

>to only trust well known browsers not to view us, when we don't want to =

>be seen.
>
>Isn't this a necessary security thing?
>
>/Karl
>
[Xin] I agree with you that the app is possible to steal user's video or
audio stream.=20
But it is still hard to controI if the stream from getUserMedia could be
used by other call(peerconnection) as designed in API now.

What I want to say is that the permission to access the device is =
orthogonal
to consent to transmit the media to whom. The first one is for a site. =
The
second one is for a user, it could be guaranteed by other ways, such as
identity, media-encryption and so on, which need other consent method.=20

 There are some details in
http://tools.ietf.org/html/draft-ietf-rtcweb-security-05#section-4.1.2.1


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Subject: Re: [rtcweb] TURN server address via DHCP, WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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On Sep 26, 2013 8:27 AM, "Karl Stahl" <karl.stahl@intertex.se> wrote:
>
> Here comes the suggestion I got from my developer that would allow a
network
> provider to offer his TURN server for the WebRTC browser to use. This
would
> require NO new DHCP-options or similar, and NO OS changes (I do realize
> those should be avoided if possible...).
>
> The idea is still, that whoever is responsible for giving a device an IP
> address (the network provider or a LAN administrator) can also announce a
> TURN server for the WebRTC browser to use.
>
> The suggestion is to use RFC6763 (DNS-Based Service Discovery, see chapte=
r
> 11) where the network provider (the owner of the IP address) has set up a
> DNS PTR record for the TURN server in the in-addr.arpa domain.
>
> If the device got IP 173.164.252.149, then make a query for the PTR recor=
d
> for:
> _turn._udp.149.252.164.173.in-addr.arpa.
> Then the SRV record would return the actual address to the TURN server
(and
> you may find several for load balancing and failover I guess)
>
> If the device is on a LAN, the IP 173.164.252.149 to query would be the
WAN
> IP you get via STUN in the ICE process. But if the LAN administrator
> provides a local TURN server, then he also should have blocked STUN in th=
e
> firewall, and the browser should query the device's local host address (a=
s
> in the ICE procedure) and the local LAN DNS server should answer the quer=
y
> to give the local TURN server address on the LAN.
>
> Shouldn't this work to allow network provider to offer his TURN server
> "automatically and generally"?
>
> And wouldn't this work nicely for mobile devices, whenever the device is
on
> a "WebRTC-ready access" (actually good for everything using ICE), whether
> fixed, WiFi or 3G/4G OTT, you get also a TURN server (and the network
> provider hopefully offers a prioritized pipe there as one usage of this
> mechanism).
>

Does your mobile provider currently do reverse dns for your mobile's real
ip address?

CB

> Not to slow down every call setup by doing this in the ICE process, I
guess
> this could be done when starting the browser and when the device gets a
new
> IP address, to have the TURN server address ready for later use.
>
> /Karl
>
>
> -----Ursprungligt meddelande-----
> Fr=E5n: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] F=F6r
> Markus.Isomaki@nokia.com
> Skickat: den 26 september 2013 14:06
> Till: fluffy@cisco.com; cb.list6@gmail.com
> Kopia: rtcweb@ietf.org
> =C4mne: Re: [rtcweb] TURN server address via DHCP, WGLC of
> draft-ietf-rtcweb-use-cases-and-requirements-11
>
> Hi,
>
> Cullen Jennings wrote:
> >
> > I will note that browsers have many ways to learn about HTTP proxies
> > from the network and it seems to me that using some of theses same
> > technique might also be a good way to learn about TURN servers.
> >
>
> I agree. I assume that in enterprises it would be the same people managin=
g
> both of these.
>
> Is there something the IETF can or should do about this, or do we just
> assume it will happen? A new DHCP option is something the IETF could
easily
> do, but I also doubt how usable that would be.
>
> Markus
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<p dir=3D"ltr"><br>
On Sep 26, 2013 8:27 AM, &quot;Karl Stahl&quot; &lt;<a href=3D"mailto:karl.=
stahl@intertex.se">karl.stahl@intertex.se</a>&gt; wrote:<br>
&gt;<br>
&gt; Here comes the suggestion I got from my developer that would allow a n=
etwork<br>
&gt; provider to offer his TURN server for the WebRTC browser to use. This =
would<br>
&gt; require NO new DHCP-options or similar, and NO OS changes (I do realiz=
e<br>
&gt; those should be avoided if possible...).<br>
&gt;<br>
&gt; The idea is still, that whoever is responsible for giving a device an =
IP<br>
&gt; address (the network provider or a LAN administrator) can also announc=
e a<br>
&gt; TURN server for the WebRTC browser to use.<br>
&gt;<br>
&gt; The suggestion is to use RFC6763 (DNS-Based Service Discovery, see cha=
pter<br>
&gt; 11) where the network provider (the owner of the IP address) has set u=
p a<br>
&gt; DNS PTR record for the TURN server in the in-addr.arpa domain.<br>
&gt;<br>
&gt; If the device got IP 173.164.252.149, then make a query for the PTR re=
cord<br>
&gt; for:<br>
&gt; _turn._udp.149.252.164.173.in-addr.arpa.<br>
&gt; Then the SRV record would return the actual address to the TURN server=
 (and<br>
&gt; you may find several for load balancing and failover I guess)<br>
&gt;<br>
&gt; If the device is on a LAN, the IP 173.164.252.149 to query would be th=
e WAN<br>
&gt; IP you get via STUN in the ICE process. But if the LAN administrator<b=
r>
&gt; provides a local TURN server, then he also should have blocked STUN in=
 the<br>
&gt; firewall, and the browser should query the device&#39;s local host add=
ress (as<br>
&gt; in the ICE procedure) and the local LAN DNS server should answer the q=
uery<br>
&gt; to give the local TURN server address on the LAN.<br>
&gt;<br>
&gt; Shouldn&#39;t this work to allow network provider to offer his TURN se=
rver<br>
&gt; &quot;automatically and generally&quot;?<br>
&gt;<br>
&gt; And wouldn&#39;t this work nicely for mobile devices, whenever the dev=
ice is on<br>
&gt; a &quot;WebRTC-ready access&quot; (actually good for everything using =
ICE), whether<br>
&gt; fixed, WiFi or 3G/4G OTT, you get also a TURN server (and the network<=
br>
&gt; provider hopefully offers a prioritized pipe there as one usage of thi=
s<br>
&gt; mechanism).<br>
&gt;</p>
<p dir=3D"ltr">Does your mobile provider currently do reverse dns for your =
mobile&#39;s real ip address?</p>
<p dir=3D"ltr">CB</p>
<p dir=3D"ltr">&gt; Not to slow down every call setup by doing this in the =
ICE process, I guess<br>
&gt; this could be done when starting the browser and when the device gets =
a new<br>
&gt; IP address, to have the TURN server address ready for later use.<br>
&gt;<br>
&gt; /Karl<br>
&gt;<br>
&gt;<br>
&gt; -----Ursprungligt meddelande-----<br>
&gt; Fr=E5n: <a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf=
.org</a> [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@=
ietf.org</a>] F=F6r<br>
&gt; <a href=3D"mailto:Markus.Isomaki@nokia.com">Markus.Isomaki@nokia.com</=
a><br>
&gt; Skickat: den 26 september 2013 14:06<br>
&gt; Till: <a href=3D"mailto:fluffy@cisco.com">fluffy@cisco.com</a>; <a hre=
f=3D"mailto:cb.list6@gmail.com">cb.list6@gmail.com</a><br>
&gt; Kopia: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; =C4mne: Re: [rtcweb] TURN server address via DHCP, WGLC of<br>
&gt; draft-ietf-rtcweb-use-cases-and-requirements-11<br>
&gt;<br>
&gt; Hi,<br>
&gt;<br>
&gt; Cullen Jennings wrote:<br>
&gt; &gt;<br>
&gt; &gt; I will note that browsers have many ways to learn about HTTP prox=
ies<br>
&gt; &gt; from the network and it seems to me that using some of theses sam=
e<br>
&gt; &gt; technique might also be a good way to learn about TURN servers.<b=
r>
&gt; &gt;<br>
&gt;<br>
&gt; I agree. I assume that in enterprises it would be the same people mana=
ging<br>
&gt; both of these.<br>
&gt;<br>
&gt; Is there something the IETF can or should do about this, or do we just=
<br>
&gt; assume it will happen? A new DHCP option is something the IETF could e=
asily<br>
&gt; do, but I also doubt how usable that would be.<br>
&gt;<br>
&gt; Markus<br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.i=
etf.org/mailman/listinfo/rtcweb</a><br>
&gt;<br>
</p>

--f46d043c7c5691387604e751693e--

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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "Markus.Isomaki@nokia.com" <Markus.Isomaki@nokia.com>
Thread-Topic: [rtcweb] TURN server address via DHCP, WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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Subject: Re: [rtcweb] TURN server address via DHCP, WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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I think we need to give some advise to the browsers vendors on what they sh=
ould implement to find turn servers=20

On Sep 26, 2013, at 5:13 AM, "Markus.Isomaki@nokia.com" <Markus.Isomaki@nok=
ia.com> wrote:

> Hi,
>=20
> Cullen Jennings wrote:
>>=20
>> I will note that browsers have many ways to learn about HTTP proxies fro=
m
>> the network and it seems to me that using some of theses same technique
>> might also be a good way to learn about TURN servers.
>=20
> I agree. I assume that in enterprises it would be the same people managin=
g both of these.
>=20
> Is there something the IETF can or should do about this, or do we just as=
sume it will happen? A new DHCP option is something the IETF could easily d=
o, but I also doubt how usable that would be.=20
>=20
> Markus=20

From christer.holmberg@ericsson.com  Fri Sep 27 06:59:23 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: Unified Plan: 'bundle-only' documentation
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Subject: [rtcweb] Unified Plan: 'bundle-only' documentation
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Hi,



On the MMUSIC list I have suggested updated SDP Offer/Answer text for BUNDL=
E.



Part of that text is the definition, and usage, of the SDP 'bundle-only' at=
tribute.



However, eventhough I personally think it fits well there (and I believe Ad=
am had the same opinion, when I talked with him off-line in Berlin), afaik =
there has never been any formal WG agreement that the 'bundle-only' attribu=
te will be defined in the BUNDLE draft.



So, without going into the details (there are some issues, and if you are i=
nterested there are mail threads on the MMUSIC list dealing with those) do =
people have any issues with defining the SDP 'bundle-only' attribute, and t=
he usage of it, in BUNDLE?



If not, can we make a WG decision on the matter?



Regards,



Christer

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1">
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rection: ltr;">
<p>Hi,</p>
<p>&nbsp;</p>
<p>On the MMUSIC list I have suggested updated SDP Offer/Answer text for BU=
NDLE.</p>
<p>&nbsp;</p>
<p>Part of that text is the definition, and usage, of the SDP 'bundle-only'=
 attribute.</p>
<p>&nbsp;</p>
<p>However, eventhough I personally think it fits well there (and I believe=
 Adam had the same opinion, when I talked with him off-line in Berlin), afa=
ik there has never been any formal WG agreement that the 'bundle-only' attr=
ibute will be defined in the BUNDLE
 draft.</p>
<p>&nbsp;</p>
<p>So, without going into the details (there are some&nbsp;issues, and&nbsp=
;if you are interested there are mail threads on the MMUSIC list dealing wi=
th those)&nbsp;do people have any issues with defining the SDP 'bundle-only=
' attribute, and the usage of it, in BUNDLE?</p>
<p>&nbsp;</p>
<p>If not, can we make a WG decision on the matter?</p>
<p>&nbsp;</p>
<p>Regards,</p>
<p>&nbsp;</p>
<p>Christer</p>
</div>
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From juberti@google.com  Fri Sep 27 08:58:47 2013
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From: Justin Uberti <juberti@google.com>
Date: Fri, 27 Sep 2013 08:58:25 -0700
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Subject: Re: [rtcweb] TURN server address via DHCP, WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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Agree. I still think that extending PAC files to include TURN information
is the right way to go.

PAC files can already be discovered via DHCP or DNS, there is no need to
reinvent this wheel.


On Thu, Sep 26, 2013 at 4:55 PM, Cullen Jennings (fluffy)
<fluffy@cisco.com>wrote:

>
> I think we need to give some advise to the browsers vendors on what they
> should implement to find turn servers
>
> On Sep 26, 2013, at 5:13 AM, "Markus.Isomaki@nokia.com" <
> Markus.Isomaki@nokia.com> wrote:
>
> > Hi,
> >
> > Cullen Jennings wrote:
> >>
> >> I will note that browsers have many ways to learn about HTTP proxies
> from
> >> the network and it seems to me that using some of theses same technique
> >> might also be a good way to learn about TURN servers.
> >
> > I agree. I assume that in enterprises it would be the same people
> managing both of these.
> >
> > Is there something the IETF can or should do about this, or do we just
> assume it will happen? A new DHCP option is something the IETF could easily
> do, but I also doubt how usable that would be.
> >
> > Markus
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir="ltr">Agree. I still think that extending PAC files to include TURN information is the right way to go.<div><br></div><div>PAC files can already be discovered via DHCP or DNS, there is no need to reinvent this wheel.</div>

</div><div class="gmail_extra"><br><br><div class="gmail_quote">On Thu, Sep 26, 2013 at 4:55 PM, Cullen Jennings (fluffy) <span dir="ltr">&lt;<a href="mailto:fluffy@cisco.com" target="_blank">fluffy@cisco.com</a>&gt;</span> wrote:<br>

<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><br>
I think we need to give some advise to the browsers vendors on what they should implement to find turn servers<br>
<div class="HOEnZb"><div class="h5"><br>
On Sep 26, 2013, at 5:13 AM, &quot;<a href="mailto:Markus.Isomaki@nokia.com">Markus.Isomaki@nokia.com</a>&quot; &lt;<a href="mailto:Markus.Isomaki@nokia.com">Markus.Isomaki@nokia.com</a>&gt; wrote:<br>
<br>
&gt; Hi,<br>
&gt;<br>
&gt; Cullen Jennings wrote:<br>
&gt;&gt;<br>
&gt;&gt; I will note that browsers have many ways to learn about HTTP proxies from<br>
&gt;&gt; the network and it seems to me that using some of theses same technique<br>
&gt;&gt; might also be a good way to learn about TURN servers.<br>
&gt;<br>
&gt; I agree. I assume that in enterprises it would be the same people managing both of these.<br>
&gt;<br>
&gt; Is there something the IETF can or should do about this, or do we just assume it will happen? A new DHCP option is something the IETF could easily do, but I also doubt how usable that would be.<br>
&gt;<br>
&gt; Markus<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href="https://www.ietf.org/mailman/listinfo/rtcweb" target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>

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From: Justin Uberti <juberti@google.com>
Date: Fri, 27 Sep 2013 10:34:05 -0700
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Subject: Re: [rtcweb] JSEP-04: Nits and niggles
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Thanks for these comments. Agree with your suggestions, will fix in the
next version.


On Thu, Sep 19, 2013 at 3:19 PM, Harald Alvestrand <harald@alvestrand.no>wrote:

>  On 09/19/2013 10:55 PM, Suhas Nandakumar wrote:
>
>  Regarding a=ssrc comment above,
> The text from RFC5576 says this
>
>  Each "ssrc" media attribute specifies a single source-level attribute
>    for the given <ssrc-id>.  For each source mentioned in SDP, the
>    source-level attribute "cname", defined in Section 6.1 <http://tools.ietf.org/html/rfc5576#section-6.1>, MUST be
>    provided.  Any number of other source-level attributes for the source
>    MAY also be provided.
>
>  given the above, would it be cname attribute for a given ssrc that should be mandated at the minimum ?
>
>
> Thanks, that makes sense to me. Updating JSEP to say "add a line with
> a=ssrc cname=<cname>" sounds like something that does what we want and is
> consistent with 5576.
>
>
>  Cheers
>
> Suhas
>
>
>
> On Thu, Sep 19, 2013 at 5:54 AM, Harald Alvestrand <harald@alvestrand.no>wrote:
>
>> I'm sure we'll have lots of things to discuss about JSEP. I just thought
>> I'd get these out of the way.
>>
>> - Usage of MSID. In addition to the a=msid: lines, the session-level
>> generation procedure needs to say that a line "a=msid-semantic:WMS <id>" is
>> added to the session-level description, containing the IDs of the
>> MediaStreams that are signalled.
>>
>> - a=ssrc lines. The spec (5.2.1 second list last bullet) simply says "an
>> a=ssrc line". But RFC 5576 specifies that all ssrc lines contain an
>> attribute. Suggestion: define one that does no harm (role=primary?)
>>
>> - recycling of m= lines. The section in 5.2.2 (subsequent offers) that
>> talks about this doesn't mention that the media in the m-line has to match,
>> although section 5.3.1 (initial answer) actually talks about it. I'm
>> assuming that's an oversight.
>>
>> I'd suggest pulling the language about "selecting an appropriate m-line
>> for a track" out in a separate subsection, so that we have one place to
>> point to for this.
>>
>> Out of time, let's get these ones out of the way for -05....
>>
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--20cf3010eb8fd077fb04e760e50c
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Thanks for these comments. Agree with your suggestions, wi=
ll fix in the next version.</div><div class=3D"gmail_extra"><br><br><div cl=
ass=3D"gmail_quote">On Thu, Sep 19, 2013 at 3:19 PM, Harald Alvestrand <spa=
n dir=3D"ltr">&lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank"=
>harald@alvestrand.no</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
 =20
   =20
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000"><div class=3D"im">
    <div>On 09/19/2013 10:55 PM, Suhas
      Nandakumar wrote:<br>
    </div>
    <blockquote type=3D"cite">
      <div dir=3D"ltr">
        <div>Regarding a=3Dssrc comment above,</div>
        The text from RFC5576 says this=C2=A0
        <div><br>
        </div>
        <div>
          <pre style=3D"font-size:1em;margin-bottom:0px;margin-top:0px">Eac=
h &quot;ssrc&quot; media attribute specifies a single source-level attribut=
e
   for the given &lt;ssrc-id&gt;.  For each source mentioned in SDP, the
   source-level attribute &quot;cname&quot;, defined in <a href=3D"http://t=
ools.ietf.org/html/rfc5576#section-6.1" target=3D"_blank">Section 6.1</a>, =
MUST be
   provided.  Any number of other source-level attributes for the source
   MAY also be provided.</pre>
          <pre style=3D"font-size:1em;margin-bottom:0px;margin-top:0px"></p=
re>
          <pre style=3D"font-size:1em;margin-bottom:0px;margin-top:0px"></p=
re>
          <pre style=3D"margin-top:0px;margin-bottom:0px"><font face=3D"ari=
al"><span style=3D"white-space:normal">given the above, would it be cname a=
ttribute for a given ssrc that should be mandated at the minimum ?</span></=
font></pre>


        </div>
      </div>
    </blockquote>
    <br>
    </div><font face=3D"arial">Thanks, that makes sense to me. Updating JSE=
P to
      say &quot;add a line with a=3Dssrc cname=3D&lt;cname&gt;&quot; sounds=
 like
      something that does what we want and is consistent with 5576.</font><=
div class=3D"im"><br>
    <blockquote type=3D"cite">
      <div dir=3D"ltr">
        <div>
          <pre style=3D"margin-top:0px;margin-bottom:0px"><font face=3D"ari=
al"><span style=3D"white-space:normal">
</span></font></pre>
          <pre style=3D"margin-top:0px;margin-bottom:0px"><font face=3D"ari=
al"><span style=3D"white-space:normal">

</span></font></pre>
          <pre style=3D"margin-top:0px;margin-bottom:0px"><font face=3D"ari=
al"><span style=3D"white-space:normal">Cheers</span></font></pre>
          <pre style=3D"margin-top:0px;margin-bottom:0px"><font face=3D"ari=
al"><span style=3D"white-space:normal">Suhas</span></font></pre>
          <pre style=3D"font-size:1em;margin-bottom:0px;margin-top:0px"></p=
re>
          <pre style=3D"font-size:1em;margin-bottom:0px;margin-top:0px"></p=
re>
        </div>
      </div>
      <div class=3D"gmail_extra">
        <br>
        <br>
        <div class=3D"gmail_quote">On Thu, Sep 19, 2013 at 5:54 AM, Harald
          Alvestrand <span dir=3D"ltr">&lt;<a href=3D"mailto:harald@alvestr=
and.no" target=3D"_blank">harald@alvestrand.no</a>&gt;</span>
          wrote:<br>
          <blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bord=
er-left:1px #ccc solid;padding-left:1ex">
            I&#39;m sure we&#39;ll have lots of things to discuss about JSE=
P. I
            just thought I&#39;d get these out of the way.<br>
            <br>
            - Usage of MSID. In addition to the a=3Dmsid: lines, the
            session-level generation procedure needs to say that a line
            &quot;a=3Dmsid-semantic:WMS &lt;id&gt;&quot; is added to the
            session-level description, containing the IDs of the
            MediaStreams that are signalled.<br>
            <br>
            - a=3Dssrc lines. The spec (5.2.1 second list last bullet)
            simply says &quot;an a=3Dssrc line&quot;. But RFC 5576 specifie=
s that
            all ssrc lines contain an attribute. Suggestion: define one
            that does no harm (role=3Dprimary?)<br>
            <br>
            - recycling of m=3D lines. The section in 5.2.2 (subsequent
            offers) that talks about this doesn&#39;t mention that the medi=
a
            in the m-line has to match, although section 5.3.1 (initial
            answer) actually talks about it. I&#39;m assuming that&#39;s an
            oversight.<br>
            <br>
            I&#39;d suggest pulling the language about &quot;selecting an
            appropriate m-line for a track&quot; out in a separate
            subsection, so that we have one place to point to for this.<br>
            <br>
            Out of time, let&#39;s get these ones out of the way for -05...=
.<br>
            <br>
            <br>
            <br>
            _______________________________________________<br>
            rtcweb mailing list<br>
            <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@iet=
f.org</a><br>
            <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=
=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
          </blockquote>
        </div>
        <br>
      </div>
    </blockquote>
    <br>
  </div></div>

<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

--20cf3010eb8fd077fb04e760e50c--

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From: Justin Uberti <juberti@google.com>
Date: Fri, 27 Sep 2013 10:43:02 -0700
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On Thu, Sep 19, 2013 at 8:40 AM, Michael Procter <michael@voip.co.uk> wrote:

> Hi,
>
> Section 5.2.2 Subsequent Offers
>
> This section starts with the sentence "When createOffer is called a
> second (or later) time,".  As far as I can tell, the rest of this
> section applies both to subsequent offers created by the original
> offerer and also offers created by the original answerer.  In this
> second case, createOffer has never been called.  I would suggest
> something like "When createOffer is called after the initial
> offer/answer exchange,".
>

Good catch. There are states between the initial state and the completion
of the o/a exchange, so the text probably needs to be more along the lines
of
"When createOffer is called a second (or later) time, or called after a
local or remote description has been installed..."

Will try to wordsmith this a bit.


> Section 5.2.2 11th bullet:
>
>    o  The m= line and corresponding "a=rtpmap" and "a=fmtp" lines MUST
>       only include codecs present in the remote description.
>
> Is this necessary?  Can't we offer new codecs in a subsequent offer?
> RFC3264 Sec 8.3.2 permits new codecs to be offered, old ones to be
> removed etc, with the only constraint being to not reuse dynamic
> payload types.
>

Certainly, this is possible, but we probably want this to only happen when
explicitly requested by JS. For example, if a track is put on hold, I don't
think we want to renegotiate all codecs at that time. Hence the current
text.

If we do provide APIs for this, this behavior change would be covered in
the Constraints Handling section.


> Finally, the two example SDPs both contain "s=-" whereas section 5.2.1
> encourages "s= " instead. Personally, I prefer the hyphen, but
> consistency is probably more important.
>
> I prefer the hyphen too, since it better matches the o= line. Does anyone
feel strongly about this?

--047d7b2e0abbce98b204e7610585
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Thu, Sep 19, 2013 at 8:40 AM, Michael Procter <span dir=3D"ltr">=
&lt;<a href=3D"mailto:michael@voip.co.uk" target=3D"_blank">michael@voip.co=
.uk</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi,<br>
<br>
Section 5.2.2 Subsequent Offers<br>
<br>
This section starts with the sentence &quot;When createOffer is called a<br=
>
second (or later) time,&quot;. =C2=A0As far as I can tell, the rest of this=
<br>
section applies both to subsequent offers created by the original<br>
offerer and also offers created by the original answerer. =C2=A0In this<br>
second case, createOffer has never been called. =C2=A0I would suggest<br>
something like &quot;When createOffer is called after the initial<br>
offer/answer exchange,&quot;.<br></blockquote><div><br></div><div>Good catc=
h. There are states between the initial state and the completion of the o/a=
 exchange, so the text probably needs to be more along the lines of</div>

<div>&quot;When createOffer is called a second (or later) time, or called a=
fter a local or remote description has been installed...&quot;=C2=A0</div><=
div><br>Will try to wordsmith this a bit.</div><div><br></div><blockquote c=
lass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;=
padding-left:1ex">


<br>
Section 5.2.2 11th bullet:<br>
<br>
=C2=A0 =C2=A0o =C2=A0The m=3D line and corresponding &quot;a=3Drtpmap&quot;=
 and &quot;a=3Dfmtp&quot; lines MUST<br>
=C2=A0 =C2=A0 =C2=A0 only include codecs present in the remote description.=
<br>
<br>
Is this necessary? =C2=A0Can&#39;t we offer new codecs in a subsequent offe=
r?<br>
RFC3264 Sec 8.3.2 permits new codecs to be offered, old ones to be<br>
removed etc, with the only constraint being to not reuse dynamic<br>
payload types.<br></blockquote><div><br></div><div>Certainly, this is possi=
ble, but we probably want this to only happen when explicitly requested by =
JS. For example, if a track is put on hold, I don&#39;t think we want to re=
negotiate all codecs at that time. Hence the current text.</div>

<div><br></div><div>If we do provide APIs for this, this behavior change wo=
uld be covered in the Constraints Handling section.</div><div><br></div><bl=
ockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #=
ccc solid;padding-left:1ex">


<br>
Finally, the two example SDPs both contain &quot;s=3D-&quot; whereas sectio=
n 5.2.1<br>
encourages &quot;s=3D &quot; instead. Personally, I prefer the hyphen, but<=
br>
consistency is probably more important.<br>
<br></blockquote><div>I prefer the hyphen too, since it better matches the =
o=3D line. Does anyone feel strongly about this?</div><div>=C2=A0</div></di=
v></div></div>

--047d7b2e0abbce98b204e7610585--

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From: "Karl Stahl" <karl.stahl@intertex.se>
To: "'cb.list6'" <cb.list6@gmail.com>
References: <C5E08FE080ACFD4DAE31E4BDBF944EB11667BBA0@xmb-aln-x02.cisco.com>	<523c6d3d.c9d1440a.3b96.7499SMTPIN_ADDED_BROKEN@mx.google.com>	<CAD6AjGRXr5kPRQdN+4jkgXHciN3NE7HiRmsb7kaYuzwHEPa7ZA@mail.gmail.com>	<C5E08FE080ACFD4DAE31E4BDBF944EB1166CC702@xmb-aln-x02.cisco.com>	<E44893DD4E290745BB608EB23FDDB7620A0CDABB@008-AM1MPN1-042.mgdnok.nokia.com>	<52445257.8581700a.36b0.fffff687SMTPIN_ADDED_BROKEN@mx.google.com> <CAD6AjGTZyCRd9QKniO1D99tSQQddyipKKdqPj2zVnzxB1-c5Kg@mail.gmail.com>
In-Reply-To: <CAD6AjGTZyCRd9QKniO1D99tSQQddyipKKdqPj2zVnzxB1-c5Kg@mail.gmail.com>
Date: Fri, 27 Sep 2013 19:47:08 +0200
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Subject: Re: [rtcweb] TURN server address via DHCP, WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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> Does your mobile provider currently do reverse dns for your mobile's =
real
ip address?

>CB

Good question and the answer is good: Yes

We just checked both major 3G mobile operators TeleaSonera and Tele2 in
Sweden.

E.g. just surfing  to http://whatismyipaddress.com=20

(which tells me it is TeliaSonera, Wireless Broadband)

Then clicking the =93Additional IP Details=94 button, I get the PTR =
record:

host-95-199-196-65.mobileonline.telia.com

Tele2 gave: m83-186-15-99.cust.tele2.se

=20

So, obviously mobile providers have ways of provisioning these kind of
things in DNS.

=20

How does it look in other places of the world?

=20

/Karl

=20

=20

Fr=E5n: cb.list6 [mailto:cb.list6@gmail.com]=20
Skickat: den 27 september 2013 01:06
Till: Karl Stahl
Kopia: fluffy@cisco.com; rtcweb@ietf.org;
draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org;
Markus.Isomaki@nokia.com
=C4mne: Re: SV: [rtcweb] TURN server address via DHCP, WGLC of
draft-ietf-rtcweb-use-cases-and-requirements-11

=20


On Sep 26, 2013 8:27 AM, "Karl Stahl" <karl.stahl@intertex.se> wrote:
>
> Here comes the suggestion I got from my developer that would allow a
network
> provider to offer his TURN server for the WebRTC browser to use. This
would
> require NO new DHCP-options or similar, and NO OS changes (I do =
realize
> those should be avoided if possible...).
>
> The idea is still, that whoever is responsible for giving a device an =
IP
> address (the network provider or a LAN administrator) can also =
announce a
> TURN server for the WebRTC browser to use.
>
> The suggestion is to use RFC6763 (DNS-Based Service Discovery, see =
chapter
> 11) where the network provider (the owner of the IP address) has set =
up a
> DNS PTR record for the TURN server in the in-addr.arpa domain.
>
> If the device got IP 173.164.252.149, then make a query for the PTR =
record
> for:
> _turn._udp.149.252.164.173.in-addr.arpa.
> Then the SRV record would return the actual address to the TURN server
(and
> you may find several for load balancing and failover I guess)
>
> If the device is on a LAN, the IP 173.164.252.149 to query would be =
the
WAN
> IP you get via STUN in the ICE process. But if the LAN administrator
> provides a local TURN server, then he also should have blocked STUN in =
the
> firewall, and the browser should query the device's local host address =
(as
> in the ICE procedure) and the local LAN DNS server should answer the =
query
> to give the local TURN server address on the LAN.
>
> Shouldn't this work to allow network provider to offer his TURN server
> "automatically and generally"?
>
> And wouldn't this work nicely for mobile devices, whenever the device =
is
on
> a "WebRTC-ready access" (actually good for everything using ICE), =
whether
> fixed, WiFi or 3G/4G OTT, you get also a TURN server (and the network
> provider hopefully offers a prioritized pipe there as one usage of =
this
> mechanism).
>

Does your mobile provider currently do reverse dns for your mobile's =
real ip
address?

CB

> Not to slow down every call setup by doing this in the ICE process, I
guess
> this could be done when starting the browser and when the device gets =
a
new
> IP address, to have the TURN server address ready for later use.
>
> /Karl
>
>
> -----Ursprungligt meddelande-----
> Fr=E5n: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] F=F6r
> Markus.Isomaki@nokia.com
> Skickat: den 26 september 2013 14:06
> Till: fluffy@cisco.com; cb.list6@gmail.com
> Kopia: rtcweb@ietf.org
> =C4mne: Re: [rtcweb] TURN server address via DHCP, WGLC of
> draft-ietf-rtcweb-use-cases-and-requirements-11
>
> Hi,
>
> Cullen Jennings wrote:
> >
> > I will note that browsers have many ways to learn about HTTP proxies
> > from the network and it seems to me that using some of theses same
> > technique might also be a good way to learn about TURN servers.
> >
>
> I agree. I assume that in enterprises it would be the same people =
managing
> both of these.
>
> Is there something the IETF can or should do about this, or do we just
> assume it will happen? A new DHCP option is something the IETF could
easily
> do, but I also doubt how usable that would be.
>
> Markus
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


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<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]--></head><body lang=3DSV link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&g=
t; </span><span lang=3DEN-US>Does your mobile provider currently do =
reverse dns for your mobile's real ip =
address?<o:p></o:p></span></p><p><span =
lang=3DEN-US>&gt;CB<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Go=
od question and the answer is good: Yes<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>We=
 just checked both major 3G mobile operators TeleaSonera and Tele2 in =
Sweden.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>E.=
g. just surfing =A0to <a =
href=3D"http://whatismyipaddress.com">http://whatismyipaddress.com</a> =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(w=
hich tells me it is TeliaSonera, Wireless =
Broadband)<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
en clicking the &#8220;Additional IP Details&#8221; button, I get the =
PTR record:<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>ho=
st-95-199-196-65.mobileonline.telia.com<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Te=
le2 gave: m83-186-15-99.cust.tele2.se<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>So=
, obviously mobile providers have ways of provisioning these kind of =
things in DNS.<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Ho=
w does it look in other places of the world?<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>/K=
arl<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><div style=3D'border:none;border-top:solid =
#B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'><p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Fr=E5n:</spa=
n></b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> cb.list6 =
[mailto:cb.list6@gmail.com] <br><b>Skickat:</b> den 27 september 2013 =
01:06<br><b>Till:</b> Karl Stahl<br><b>Kopia:</b> fluffy@cisco.com; =
rtcweb@ietf.org; =
draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org; =
Markus.Isomaki@nokia.com<br><b>=C4mne:</b> Re: SV: [rtcweb] TURN server =
address via DHCP, WGLC of =
draft-ietf-rtcweb-use-cases-and-requirements-11<o:p></o:p></span></p></di=
v><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p><br>On Sep 26, 2013 8:27 =
AM, &quot;Karl Stahl&quot; &lt;<a =
href=3D"mailto:karl.stahl@intertex.se">karl.stahl@intertex.se</a>&gt; =
wrote:<br>&gt;<br>&gt; Here comes the suggestion I got from my developer =
that would allow a network<br>&gt; provider to offer his TURN server for =
the WebRTC browser to use. This would<br>&gt; require NO new =
DHCP-options or similar, and NO OS changes (I do realize<br>&gt; those =
should be avoided if possible...).<br>&gt;<br>&gt; The idea is still, =
that whoever is responsible for giving a device an IP<br>&gt; address =
(the network provider or a LAN administrator) can also announce =
a<br>&gt; TURN server for the WebRTC browser to use.<br>&gt;<br>&gt; The =
suggestion is to use RFC6763 (DNS-Based Service Discovery, see =
chapter<br>&gt; 11) where the network provider (the owner of the IP =
address) has set up a<br>&gt; DNS PTR record for the TURN server in the =
in-addr.arpa domain.<br>&gt;<br>&gt; If the device got IP =
173.164.252.149, then make a query for the PTR record<br>&gt; =
for:<br>&gt; _turn._udp.149.252.164.173.in-addr.arpa.<br>&gt; Then the =
SRV record would return the actual address to the TURN server =
(and<br>&gt; you may find several for load balancing and failover I =
guess)<br>&gt;<br>&gt; If the device is on a LAN, the IP 173.164.252.149 =
to query would be the WAN<br>&gt; IP you get via STUN in the ICE =
process. But if the LAN administrator<br>&gt; provides a local TURN =
server, then he also should have blocked STUN in the<br>&gt; firewall, =
and the browser should query the device's local host address (as<br>&gt; =
in the ICE procedure) and the local LAN DNS server should answer the =
query<br>&gt; to give the local TURN server address on the =
LAN.<br>&gt;<br>&gt; Shouldn't this work to allow network provider to =
offer his TURN server<br>&gt; &quot;automatically and =
generally&quot;?<br>&gt;<br>&gt; And wouldn't this work nicely for =
mobile devices, whenever the device is on<br>&gt; a &quot;WebRTC-ready =
access&quot; (actually good for everything using ICE), whether<br>&gt; =
fixed, WiFi or 3G/4G OTT, you get also a TURN server (and the =
network<br>&gt; provider hopefully offers a prioritized pipe there as =
one usage of this<br>&gt; mechanism).<br>&gt;<o:p></o:p></p><p>Does your =
mobile provider currently do reverse dns for your mobile's real ip =
address?<o:p></o:p></p><p>CB<o:p></o:p></p><p>&gt; Not to slow down =
every call setup by doing this in the ICE process, I guess<br>&gt; this =
could be done when starting the browser and when the device gets a =
new<br>&gt; IP address, to have the TURN server address ready for later =
use.<br>&gt;<br>&gt; /Karl<br>&gt;<br>&gt;<br>&gt; -----Ursprungligt =
meddelande-----<br>&gt; Fr=E5n: <a =
href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a> =
[mailto:<a =
href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a>] =
F=F6r<br>&gt; <a =
href=3D"mailto:Markus.Isomaki@nokia.com">Markus.Isomaki@nokia.com</a><br>=
&gt; Skickat: den 26 september 2013 14:06<br>&gt; Till: <a =
href=3D"mailto:fluffy@cisco.com">fluffy@cisco.com</a>; <a =
href=3D"mailto:cb.list6@gmail.com">cb.list6@gmail.com</a><br>&gt; Kopia: =
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>&gt; =C4mne: =
Re: [rtcweb] TURN server address via DHCP, WGLC of<br>&gt; =
draft-ietf-rtcweb-use-cases-and-requirements-11<br>&gt;<br>&gt; =
Hi,<br>&gt;<br>&gt; Cullen Jennings wrote:<br>&gt; &gt;<br>&gt; &gt; I =
will note that browsers have many ways to learn about HTTP =
proxies<br>&gt; &gt; from the network and it seems to me that using some =
of theses same<br>&gt; &gt; technique might also be a good way to learn =
about TURN servers.<br>&gt; &gt;<br>&gt;<br>&gt; I agree. I assume that =
in enterprises it would be the same people managing<br>&gt; both of =
these.<br>&gt;<br>&gt; Is there something the IETF can or should do =
about this, or do we just<br>&gt; assume it will happen? A new DHCP =
option is something the IETF could easily<br>&gt; do, but I also doubt =
how usable that would be.<br>&gt;<br>&gt; Markus<br>&gt; =
_______________________________________________<br>&gt; rtcweb mailing =
list<br>&gt; <a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>&gt; <a =
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.or=
g/mailman/listinfo/rtcweb</a><br>&gt;<o:p></o:p></p></div></body></html>
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Subject: Re: [rtcweb] JSEP and video stream properties selection (a=imageattr usage)
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Since we have a clear path forward for a signaling-plane mechanism for
these features, I think we should proceed on this path.

If it eventually turns out that this path does not provide good enough
performance in certain situations, we can look into a media-plane mechanism
as an alternative.


On Tue, Sep 24, 2013 at 8:58 AM, Matt Fredrickson <creslin@digium.com>wrote=
:

> I tend also to agree with this.  Maybe I'm completely off, but it would
> seem that pause and resume might be done using some semi-inband media
> related path, perhaps like RTCP or something like that.  I'm not sure whe=
re
> to fit resolution in though...
>
> Matthew Fredrickson
> Digium, Inc.
>
>
> On Fri, Sep 20, 2013 at 5:57 AM, Stefan H=C3=A5kansson LK <
> stefan.lk.hakansson@ericsson.com> wrote:
>
>> On 2013-09-20 09:25, Harald Alvestrand wrote:
>> > On 09/20/2013 08:02 AM, Magnus Westerlund wrote:
>> >> WG,
>> >>
>> >> In the JSEP call I commented regarding a=3Dimageattr open issue that =
we
>> >> had an original agreement from over a year ago (Vancouver) to go with
>> >> Harald's proposal in then
>> >> http://tools.ietf.org/id/draft-alvestrand-rtcweb-resolution-00.txt
>> >>
>> >> This is documented in the minutes:
>> >> http://www.ietf.org/proceedings/84/minutes/minutes-84-rtcweb
>> >>
>> >>
>> >> However, I did forget to note that there has been discussion on this
>> >> issue after that between Harald Alvestrand and Stefan H=C3=A5kansson
>> >> primarily on the W3C list
>> >>
>> >> http://lists.w3.org/Archives/Public/public-webrtc/2013Aug/0051.html
>> >>
>> >> Where they are seriously discussing only using WebRTC API for this.
>> >>
>> >> It is in the context of
>> >>
>> https://datatracker.ietf.org/doc/draft-alvestrand-constraints-resolution=
/
>> >>
>> >> Thus I would like to point out that there appear to be some indicatio=
n
>> >> of desire move away from the consensus last year.
>> >
>> > My thinking is that there are 3 points on the spectrum of solutions:
>> >
>> > - Constraints at source only (the approach most thoroughly described i=
n
>> > draft-alvestrand-constraints-resolution).
>> >
>> > - Constraints at destination can be sigalled in SDP, and acted on at
>> > source (described in draft-alvestrand-rtcweb-resolution). This spec
>> > could not be pursued as long as the "Plan X" discussions were still
>> > unsettled, but now it should really be "a piece of cake".
>>
>> My personal view is still that SDP is the wrong level for carrying this
>> kind of signaling (resolution, and perhaps further down the road
>> pause/resume related signaling). Each update requires an O/A, and locks
>> out other changes to the session. And since the SDPs must be handled by
>> the application (that is responsible for sending them over and applying
>> them) I wonder if there is any advantage compared to just having an API
>> on the sending side. If the receiving application wants another
>> resolution sent, it could just tell the sending application so, rather
>> than going through a createOffer/setLocal/ send-offer
>> /receive-answer/setRemote cycle.
>>
>> >
>> > - Defining a new way of signalling. This was the approach rejected in
>> > Vancouver.
>> >
>> > Formalistically, if the IETF decides to offer a well defined mechanism
>> > for negotiating resolution through SDP, the W3C will certainly take up
>> > the discussion on whether or not we should connect the API to that
>> > functionality.
>> >
>> > Happy to discuss more.
>> >
>> > _______________________________________________
>> > rtcweb mailing list
>> > rtcweb@ietf.org
>> > https://www.ietf.org/mailman/listinfo/rtcweb
>> >
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--089e013a027065a82404e7611f9e
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr">Since we have a clear path forward for a signaling-plane m=
echanism for these features, I think we should proceed on this path.=C2=A0<=
div><br></div><div>If it eventually turns out that this path does not provi=
de good enough performance in certain situations, we can look into a media-=
plane mechanism as an alternative.</div>

</div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Tue,=
 Sep 24, 2013 at 8:58 AM, Matt Fredrickson <span dir=3D"ltr">&lt;<a href=3D=
"mailto:creslin@digium.com" target=3D"_blank">creslin@digium.com</a>&gt;</s=
pan> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr">I tend also to agree with t=
his. =C2=A0Maybe I&#39;m completely off, but it would seem that pause and r=
esume might be done using some semi-inband media related path, perhaps like=
 RTCP or something like that. =C2=A0I&#39;m not sure where to fit resolutio=
n in though...<div>


<br></div><div>Matthew Fredrickson</div><div>Digium, Inc.</div></div><div c=
lass=3D"HOEnZb"><div class=3D"h5"><div class=3D"gmail_extra"><br><br><div c=
lass=3D"gmail_quote">On Fri, Sep 20, 2013 at 5:57 AM, Stefan H=C3=A5kansson=
 LK <span dir=3D"ltr">&lt;<a href=3D"mailto:stefan.lk.hakansson@ericsson.co=
m" target=3D"_blank">stefan.lk.hakansson@ericsson.com</a>&gt;</span> wrote:=
<br>


<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div><div>On 2013-09-20 09:25, Harald Alvest=
rand wrote:<br>
&gt; On 09/20/2013 08:02 AM, Magnus Westerlund wrote:<br>
&gt;&gt; WG,<br>
&gt;&gt;<br>
&gt;&gt; In the JSEP call I commented regarding a=3Dimageattr open issue th=
at we<br>
&gt;&gt; had an original agreement from over a year ago (Vancouver) to go w=
ith<br>
&gt;&gt; Harald&#39;s proposal in then<br>
&gt;&gt; <a href=3D"http://tools.ietf.org/id/draft-alvestrand-rtcweb-resolu=
tion-00.txt" target=3D"_blank">http://tools.ietf.org/id/draft-alvestrand-rt=
cweb-resolution-00.txt</a><br>
&gt;&gt;<br>
&gt;&gt; This is documented in the minutes:<br>
&gt;&gt; <a href=3D"http://www.ietf.org/proceedings/84/minutes/minutes-84-r=
tcweb" target=3D"_blank">http://www.ietf.org/proceedings/84/minutes/minutes=
-84-rtcweb</a><br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; However, I did forget to note that there has been discussion on th=
is<br>
&gt;&gt; issue after that between Harald Alvestrand and Stefan H=C3=A5kanss=
on<br>
&gt;&gt; primarily on the W3C list<br>
&gt;&gt;<br>
&gt;&gt; <a href=3D"http://lists.w3.org/Archives/Public/public-webrtc/2013A=
ug/0051.html" target=3D"_blank">http://lists.w3.org/Archives/Public/public-=
webrtc/2013Aug/0051.html</a><br>
&gt;&gt;<br>
&gt;&gt; Where they are seriously discussing only using WebRTC API for this=
.<br>
&gt;&gt;<br>
&gt;&gt; It is in the context of<br>
&gt;&gt; <a href=3D"https://datatracker.ietf.org/doc/draft-alvestrand-const=
raints-resolution/" target=3D"_blank">https://datatracker.ietf.org/doc/draf=
t-alvestrand-constraints-resolution/</a><br>
&gt;&gt;<br>
&gt;&gt; Thus I would like to point out that there appear to be some indica=
tion<br>
&gt;&gt; of desire move away from the consensus last year.<br>
&gt;<br>
&gt; My thinking is that there are 3 points on the spectrum of solutions:<b=
r>
&gt;<br>
&gt; - Constraints at source only (the approach most thoroughly described i=
n<br>
&gt; draft-alvestrand-constraints-resolution).<br>
&gt;<br>
&gt; - Constraints at destination can be sigalled in SDP, and acted on at<b=
r>
&gt; source (described in draft-alvestrand-rtcweb-resolution). This spec<br=
>
&gt; could not be pursued as long as the &quot;Plan X&quot; discussions wer=
e still<br>
&gt; unsettled, but now it should really be &quot;a piece of cake&quot;.<br=
>
<br>
</div></div>My personal view is still that SDP is the wrong level for carry=
ing this<br>
kind of signaling (resolution, and perhaps further down the road<br>
pause/resume related signaling). Each update requires an O/A, and locks<br>
out other changes to the session. And since the SDPs must be handled by<br>
the application (that is responsible for sending them over and applying<br>
them) I wonder if there is any advantage compared to just having an API<br>
on the sending side. If the receiving application wants another<br>
resolution sent, it could just tell the sending application so, rather<br>
than going through a createOffer/setLocal/ send-offer<br>
/receive-answer/setRemote cycle.<br>
<div><div><br>
&gt;<br>
&gt; - Defining a new way of signalling. This was the approach rejected in<=
br>
&gt; Vancouver.<br>
&gt;<br>
&gt; Formalistically, if the IETF decides to offer a well defined mechanism=
<br>
&gt; for negotiating resolution through SDP, the W3C will certainly take up=
<br>
&gt; the discussion on whether or not we should connect the API to that<br>
&gt; functionality.<br>
&gt;<br>
&gt; Happy to discuss more.<br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</=
a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt;<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>
</div></div><br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

--089e013a027065a82404e7611f9e--

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From: Justin Uberti <juberti@google.com>
Date: Fri, 27 Sep 2013 11:01:48 -0700
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Subject: Re: [rtcweb] JSEP-04: Some comments on Section 5.2.1. and 5.2.2 (19th september)
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On Fri, Sep 20, 2013 at 11:22 AM, Cullen Jennings (fluffy) <fluffy@cisco.co=
m
> wrote:

>
> Good points - proposed resolutions inline =E2=80=A6
>
> On Sep 19, 2013, at 3:08 AM, Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
>
> > Hi,
> >
> > Section 5.2.1:
> > -----------------
> >
> > Q_1:      o- line <sess-version> initial value - deviation
> >
> > What is the reason for recommending a <sess-version> zero value in the
> initial Offer, when the RFC says SHOULD use an NTP format timestamp?
> >
> > I don=E2=80=99t have any strong feelings, but in general: whenever we d=
eviate
> from the =E2=80=9Cbase=E2=80=9D SDP procedures I think we should justify =
why.
>
> Lets use NTP
>

It's not necessary to use NTP, since the session-id is already random.
Starting the session id at zero makes it easier for developers see the
changes to the session id. I can spell this out in the next version of the
document.


> >
> >
> > Q_2:      o- line <sess-version> initial value - forking
> >
> > When a new PeerConnection is created due to forking, the <sess-version>
> value must be based (incremented) on the value associated with the =E2=80=
=9Cmother=E2=80=9D
> PeerConnection, for which the initial Offer of the whole communication
> session was created.
>
>
> uh, that is old text sorry. I'm proposing the principal that any time the
> SDP changes (including new candidate lines) the version gets larger.
>

agree

>
> >
> >
> > Q_3:      RTP
> >
> > The text says:
> >
> > =E2=80=9CThe <proto> field MUST be set to "RTP/SAVPF". =E2=80=9C
> >
> > But, that of course only applies to RTP based streams (not the data
> channel). Also, in general, it needs to be clear what information needs t=
o
> be in every m- line, and what information is protocol specific. I would
> suggest to have a =E2=80=9CGeneral=E2=80=9D sub-section, a =E2=80=9CRTP=
=E2=80=9D sub-section, etc.
>
> Yep - I think this should be offers are created with RTP/SAVPF for audio
> and video and system can reeve offers with SAVPF or SAVP
>

Currently the document specifies what to do with media lines, with later
discussion of handling the SCTP line. Do you think the current information
is not sufficiently descriptive?

> >
> >
> > Q_4:      BUNDLE
> >
> > The text says:
> >
> > =E2=80=9CIf a m=3D section is not being bundled into another m=3D secti=
on, it MUST
> >                generate a unique set of ICE credentials and gather its
> own set of
> >                candidates.  Otherwise, it MUST use the same ICE
> credentials and
> >                candidates that were used in the m=3D section that it is
> being bundled
> >                into.=E2=80=9D
> >
> > As, when BUNDLE is used, the initial Offer will contain identical ICE
> candidates, does that mean that we will also include identical address
> information in the initial Offer?
>
> no the initial offer will not have identical stuff other than on lines
> that "bundle-only" and we still need to add more text on this but idea is
> to make just like what is in united-plan


> >
> > I don=E2=80=99t object to that =E2=80=93 I just want to clarify, as it =
has impacts on
> the text in the BUNDLE spec :)
>

I'm not sure what the exact concern is here - I just put this here to be
clear about the exact processing needed, since the BUNDLE draft is also a
moving target.

> >
> >
> > Section 5.2.2:
> > -----------------
> >
> >
> > Q_5:      Offer when in =E2=80=9Clocal-offer=E2=80=9D
> >
> > The text says:
> >
> > =E2=80=9CIf the initial offer was applied using setLocalDescription, bu=
t an
> >                answer from the remote side has not yet been applied,
> meaning the
> >                PeerConnection is still in the "local-offer" state, the
> steps for
> >                generating an initial offer should be followed,=E2=80=9D
> >
> > I don=E2=80=99t understand this. Why would you create a new Offer while=
 you are
> waiting for an Answer to the previously sent Offer?
>
> Justin has asked for this in the case when the offer has not been sent to
> anyone else. This is not going to be useful for typical systems trying to
> have SIP or Jingle compatible flow but who knows =E2=80=A6
>

It's possible that the initial offer is never sent, because you just want
to set it to warm up the stack. So you can update the offer without any
consequences.

>
> >
> > Also, when setRemote() is called, to which Offer does it apply?
>
> The most recent one. And the JS app would have to make sure it passed the
> write one or you run high odds of getting completely broken behavior.
>

exactly. This will be spelled out in setRemote processing.

>
> >
> >
> > Q_6:      BUNDLE
> >
> > We=E2=80=99ll probably also need some text about BUNDLE.
> >
>
> yep - but plan is to match up with unified plan
>

Can you be specific about what more you think needs to be said? The
createAnswer treatment of BUNDLE is one clear thing that is currently
missing.

>
> >
> > Regards,
> >
> > Christer
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--089e013a0270f1255b04e761481b
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Fri, Sep 20, 2013 at 11:22 AM, Cullen Jennings (fluffy) <span di=
r=3D"ltr">&lt;<a href=3D"mailto:fluffy@cisco.com" target=3D"_blank">fluffy@=
cisco.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><br>
Good points - proposed resolutions inline =E2=80=A6<br>
<div class=3D"im"><br>
On Sep 19, 2013, at 3:08 AM, Christer Holmberg &lt;<a href=3D"mailto:christ=
er.holmberg@ericsson.com">christer.holmberg@ericsson.com</a>&gt; wrote:<br>
<br>
&gt; Hi,<br>
&gt;<br>
&gt; Section 5.2.1:<br>
&gt; -----------------<br>
&gt;<br>
&gt; Q_1: =C2=A0 =C2=A0 =C2=A0o- line &lt;sess-version&gt; initial value - =
deviation<br>
&gt;<br>
&gt; What is the reason for recommending a &lt;sess-version&gt; zero value =
in the initial Offer, when the RFC says SHOULD use an NTP format timestamp?=
<br>
&gt;<br>
&gt; I don=E2=80=99t have any strong feelings, but in general: whenever we =
deviate from the =E2=80=9Cbase=E2=80=9D SDP procedures I think we should ju=
stify why.<br>
<br>
</div>Lets use NTP<br></blockquote><div><br></div><div>It&#39;s not necessa=
ry to use NTP, since the session-id is already random. Starting the session=
 id at zero makes it easier for developers see the changes to the session i=
d. I can spell this out in the next version of the document.</div>

<div><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0p=
x 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-lef=
t-style:solid;padding-left:1ex">
<div class=3D"im"><br>
&gt;<br>
&gt;<br>
&gt; Q_2: =C2=A0 =C2=A0 =C2=A0o- line &lt;sess-version&gt; initial value - =
forking<br>
&gt;<br>
&gt; When a new PeerConnection is created due to forking, the &lt;sess-vers=
ion&gt; value must be based (incremented) on the value associated with the =
=E2=80=9Cmother=E2=80=9D PeerConnection, for which the initial Offer of the=
 whole communication session was created.<br>


<br>
<br>
</div>uh, that is old text sorry. I&#39;m proposing the principal that any =
time the SDP changes (including new candidate lines) the version gets large=
r.<br></blockquote><div><br></div><div>agree=C2=A0</div><blockquote class=
=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;bo=
rder-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">


<div class=3D"im"><br>
&gt;<br>
&gt;<br>
&gt; Q_3: =C2=A0 =C2=A0 =C2=A0RTP<br>
&gt;<br>
&gt; The text says:<br>
&gt;<br>
&gt; =E2=80=9CThe &lt;proto&gt; field MUST be set to &quot;RTP/SAVPF&quot;.=
 =E2=80=9C<br>
&gt;<br>
&gt; But, that of course only applies to RTP based streams (not the data ch=
annel). Also, in general, it needs to be clear what information needs to be=
 in every m- line, and what information is protocol specific. I would sugge=
st to have a =E2=80=9CGeneral=E2=80=9D sub-section, a =E2=80=9CRTP=E2=80=9D=
 sub-section, etc.<br>


<br>
</div>Yep - I think this should be offers are created with RTP/SAVPF for au=
dio and video and system can reeve offers with SAVPF or SAVP<br></blockquot=
e><div><br></div><div>Currently the document specifies what to do with medi=
a lines, with later discussion of handling the SCTP line. Do you think the =
current information is not sufficiently descriptive?=C2=A0</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<div class=3D"im">&gt;<br>
&gt;<br>
&gt; Q_4: =C2=A0 =C2=A0 =C2=A0BUNDLE<br>
&gt;<br>
&gt; The text says:<br>
&gt;<br>
&gt; =E2=80=9CIf a m=3D section is not being bundled into another m=3D sect=
ion, it MUST<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0generate a uniq=
ue set of ICE credentials and gather its own set of<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0candidates. =C2=
=A0Otherwise, it MUST use the same ICE credentials and<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0candidates that=
 were used in the m=3D section that it is being bundled<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0into.=E2=80=9D<=
br>
&gt;<br>
&gt; As, when BUNDLE is used, the initial Offer will contain identical ICE =
candidates, does that mean that we will also include identical address info=
rmation in the initial Offer?<br>
<br>
</div>no the initial offer will not have identical stuff other than on line=
s that &quot;bundle-only&quot; and we still need to add more text on this b=
ut idea is to make just like what is in united-plan=C2=A0</blockquote><bloc=
kquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-=
width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;paddin=
g-left:1ex">


<div class=3D"im"><br>
&gt;<br>
&gt; I don=E2=80=99t object to that =E2=80=93 I just want to clarify, as it=
 has impacts on the text in the BUNDLE spec :)<br></div></blockquote><div><=
br></div><div>I&#39;m not sure what the exact concern is here - I just put =
this here to be clear about the exact processing needed, since the BUNDLE d=
raft is also a moving target.=C2=A0</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div class=3D"im">
&gt;<br>
&gt;<br>
&gt; Section 5.2.2:<br>
&gt; -----------------<br>
&gt;<br>
&gt;<br>
&gt; Q_5: =C2=A0 =C2=A0 =C2=A0Offer when in =E2=80=9Clocal-offer=E2=80=9D<b=
r>
&gt;<br>
&gt; The text says:<br>
&gt;<br>
&gt; =E2=80=9CIf the initial offer was applied using setLocalDescription, b=
ut an<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0answer from the=
 remote side has not yet been applied, meaning the<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0PeerConnection =
is still in the &quot;local-offer&quot; state, the steps for<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0generating an i=
nitial offer should be followed,=E2=80=9D<br>
&gt;<br>
&gt; I don=E2=80=99t understand this. Why would you create a new Offer whil=
e you are waiting for an Answer to the previously sent Offer?<br>
<br>
</div>Justin has asked for this in the case when the offer has not been sen=
t to anyone else. This is not going to be useful for typical systems trying=
 to have SIP or Jingle compatible flow but who knows =E2=80=A6<br></blockqu=
ote>

<div><br></div><div>It&#39;s possible that the initial offer is never sent,=
 because you just want to set it to warm up the stack. So you can update th=
e offer without any consequences.=C2=A0</div><blockquote class=3D"gmail_quo=
te" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-col=
or:rgb(204,204,204);border-left-style:solid;padding-left:1ex">


<div class=3D"im"><br>
&gt;<br>
&gt; Also, when setRemote() is called, to which Offer does it apply?<br>
<br>
</div>The most recent one. And the JS app would have to make sure it passed=
 the write one or you run high odds of getting completely broken behavior.<=
br></blockquote><div><br></div><div>exactly. This will be spelled out in se=
tRemote processing.=C2=A0</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<div class=3D"im"><br>
&gt;<br>
&gt;<br>
&gt; Q_6: =C2=A0 =C2=A0 =C2=A0BUNDLE<br>
&gt;<br>
&gt; We=E2=80=99ll probably also need some text about BUNDLE.<br>
&gt;<br>
<br>
</div>yep - but plan is to match up with unified plan<br></blockquote><div>=
<br></div><div>Can you be specific about what more you think needs to be sa=
id? The createAnswer treatment of BUNDLE is one clear thing that is current=
ly missing.</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<div class=3D""><div class=3D"h5"><br>
&gt;<br>
&gt; Regards,<br>
&gt;<br>
&gt; Christer<br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div></div>

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Subject: Re: [rtcweb] JSEP-04: Some comments on Section 5.2.1. and 5.2.2 (19th september)
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--90e6ba25dbdd77528804e7614da0
Content-Type: text/plain; charset=UTF-8

On Fri, Sep 20, 2013 at 3:17 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi Cullen,
>
> >> Q_2:      o- line <sess-version> initial value - forking
> >>
> >> When a new PeerConnection is created due to forking, the <sess-version>
> value must be based (incremented) on the value associated with the "mother"
> PeerConnection, for which the initial Offer of the whole communication
> session was created.
> >
> > uh, that is old text sorry. I'm proposing the principal that any time
> the SDP changes (including new candidate lines) the version gets larger.
>
> Yes. But keep in mind that this is an Offer created for a new
> PeerConnection.
>
>
> ----------------
>
>
> >> Q_4:      BUNDLE
> >>
> >> The text says:
> >>
> >> "If a m= section is not being bundled into another m= section, it MUST
> >>                generate a unique set of ICE credentials and gather its
> own set of
> >>                candidates.  Otherwise, it MUST use the same ICE
> credentials and
> >>                candidates that were used in the m= section that it is
> being bundled
> >>                into."
> >>
> >> As, when BUNDLE is used, the initial Offer will contain identical ICE
> candidates, does that mean that we will also include identical address
> information in the initial Offer?
> >
> > no the initial offer will not have identical stuff other than on lines
> that "bundle-only" and we still need to add more text on this but idea is
> to make just like what is in united-plan
>
> Regarding the details (usage of port zero etc) of "bundle-only", that is
> still something we have to agree upon. But, that discussion belongs to
> MMUSIC (and, I believe there is a thread about that already :)
>
> But again, in the case of forking, when a new PeerConnection is created, I
> think we should allow identical stuff in the initial Offer of that
> PeerConnection. Because, the remote party has already indicated support of
> BUNDLE, in the Answer that was sent for the "mother" PeerConnection, so the
> initial Offer for the new PeerConnection will be the second Offer for the
> remote party.
>
> At some point, we are going to have to add much more details about
> forking, in the case multiple PeerConnections are used. It is not enough to
> only say "create a new PeerConnection" :)
>
>
> ----------------
>
>
> >> Section 5.2.2:
> >> -----------------
> >>
> >>
> >> Q_5:      Offer when in "local-offer"
> >>
> >> The text says:
> >>
> >> "If the initial offer was applied using setLocalDescription, but an
> >>                answer from the remote side has not yet been applied,
> meaning the
> >>                PeerConnection is still in the "local-offer" state, the
> steps for
> >>                generating an initial offer should be followed,"
> >>
> >> I don't understand this. Why would you create a new Offer while you are
> waiting for an Answer to the previously sent Offer?
> >
> > Justin has asked for this in the case when the offer has not been sent
> to anyone else. This is not going to be useful for typical systems trying
> to have SIP or Jingle compatible flow but who knows ...
>
> Again, we have consensus to base JSEP on 3264. And, if we are going to
> deviate from 3264, I'd like to have some discussion and justification for
> that.
>

We have been through this many times before. The state machine that has
been in the JSEP document for a while clearly shows that
local-offer->local-offer state transitions are permitted.

--90e6ba25dbdd77528804e7614da0
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Fri, Sep 20, 2013 at 3:17 PM, Christer Holmberg <span dir=3D"ltr=
">&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">c=
hrister.holmberg@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi Cullen,<br>
<div class=3D"im"><br>
&gt;&gt; Q_2: =C2=A0 =C2=A0 =C2=A0o- line &lt;sess-version&gt; initial valu=
e - forking<br>
&gt;&gt;<br>
&gt;&gt; When a new PeerConnection is created due to forking, the &lt;sess-=
version&gt; value must be based (incremented) on the value associated with =
the &quot;mother&quot; PeerConnection, for which the initial Offer of the w=
hole communication session was created.<br>


&gt;<br>
&gt; uh, that is old text sorry. I&#39;m proposing the principal that any t=
ime the SDP changes (including new candidate lines) the version gets larger=
.<br>
<br>
</div>Yes. But keep in mind that this is an Offer created for a new PeerCon=
nection.<br>
<br>
<br>
----------------<br>
<div class=3D"im"><br>
<br>
&gt;&gt; Q_4: =C2=A0 =C2=A0 =C2=A0BUNDLE<br>
&gt;&gt;<br>
&gt;&gt; The text says:<br>
&gt;&gt;<br>
&gt;&gt; &quot;If a m=3D section is not being bundled into another m=3D sec=
tion, it MUST<br>
&gt;&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0generate a =
unique set of ICE credentials and gather its own set of<br>
&gt;&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0candidates.=
 =C2=A0Otherwise, it MUST use the same ICE credentials and<br>
&gt;&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0candidates =
that were used in the m=3D section that it is being bundled<br>
&gt;&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0into.&quot;=
<br>
&gt;&gt;<br>
&gt;&gt; As, when BUNDLE is used, the initial Offer will contain identical =
ICE candidates, does that mean that we will also include identical address =
information in the initial Offer?<br>
&gt;<br>
&gt; no the initial offer will not have identical stuff other than on lines=
 that &quot;bundle-only&quot; and we still need to add more text on this bu=
t idea is to make just like what is in united-plan<br>
<br>
</div>Regarding the details (usage of port zero etc) of &quot;bundle-only&q=
uot;, that is still something we have to agree upon. But, that discussion b=
elongs to MMUSIC (and, I believe there is a thread about that already :)<br=
>


<br>
But again, in the case of forking, when a new PeerConnection is created, I =
think we should allow identical stuff in the initial Offer of that PeerConn=
ection. Because, the remote party has already indicated support of BUNDLE, =
in the Answer that was sent for the &quot;mother&quot; PeerConnection, so t=
he initial Offer for the new PeerConnection will be the second Offer for th=
e remote party.<br>


<br>
At some point, we are going to have to add much more details about forking,=
 in the case multiple PeerConnections are used. It is not enough to only sa=
y &quot;create a new PeerConnection&quot; :)<br>
<br>
<br>
----------------<br>
<div class=3D"im"><br>
<br>
&gt;&gt; Section 5.2.2:<br>
&gt;&gt; -----------------<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; Q_5: =C2=A0 =C2=A0 =C2=A0Offer when in &quot;local-offer&quot;<br>
&gt;&gt;<br>
&gt;&gt; The text says:<br>
&gt;&gt;<br>
&gt;&gt; &quot;If the initial offer was applied using setLocalDescription, =
but an<br>
&gt;&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0answer from=
 the remote side has not yet been applied, meaning the<br>
&gt;&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0PeerConnect=
ion is still in the &quot;local-offer&quot; state, the steps for<br>
&gt;&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0generating =
an initial offer should be followed,&quot;<br>
&gt;&gt;<br>
&gt;&gt; I don&#39;t understand this. Why would you create a new Offer whil=
e you are waiting for an Answer to the previously sent Offer?<br>
&gt;<br>
</div>&gt; Justin has asked for this in the case when the offer has not bee=
n sent to anyone else. This is not going to be useful for typical systems t=
rying to have SIP or Jingle compatible flow but who knows ...<br>
<br>
Again, we have consensus to base JSEP on 3264. And, if we are going to devi=
ate from 3264, I&#39;d like to have some discussion and justification for t=
hat.<br></blockquote><div><br></div><div>We have been through this many tim=
es before. The state machine that has been in the JSEP document for a while=
 clearly shows that local-offer-&gt;local-offer state transitions are permi=
tted.=C2=A0</div>

</div></div></div>

--90e6ba25dbdd77528804e7614da0--

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From: Justin Uberti <juberti@google.com>
Date: Fri, 27 Sep 2013 11:04:24 -0700
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Subject: Re: [rtcweb] Unified Plan: 'bundle-only' documentation
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I think that the BUNDLE draft is the right place for this.


On Fri, Sep 27, 2013 at 6:59 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>  Hi,
>
>
>
> On the MMUSIC list I have suggested updated SDP Offer/Answer text for
> BUNDLE.
>
>
>
> Part of that text is the definition, and usage, of the SDP 'bundle-only'
> attribute.
>
>
>
> However, eventhough I personally think it fits well there (and I believe
> Adam had the same opinion, when I talked with him off-line in Berlin),
> afaik there has never been any formal WG agreement that the 'bundle-only'
> attribute will be defined in the BUNDLE draft.
>
>
>
> So, without going into the details (there are some issues, and if you are
> interested there are mail threads on the MMUSIC list dealing with those) do
> people have any issues with defining the SDP 'bundle-only' attribute, and
> the usage of it, in BUNDLE?
>
>
>
> If not, can we make a WG decision on the matter?
>
>
>
> Regards,
>
>
>
> Christer
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr">I think that the BUNDLE draft is the right place for this.=
</div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Fri,=
 Sep 27, 2013 at 6:59 AM, Christer Holmberg <span dir=3D"ltr">&lt;<a href=
=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.holmb=
erg@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">




<div>
<div style=3D"direction:ltr;font-size:10pt;font-family:Tahoma">
<p>Hi,</p>
<p>=C2=A0</p>
<p>On the MMUSIC list I have suggested updated SDP Offer/Answer text for BU=
NDLE.</p>
<p>=C2=A0</p>
<p>Part of that text is the definition, and usage, of the SDP &#39;bundle-o=
nly&#39; attribute.</p>
<p>=C2=A0</p>
<p>However, eventhough I personally think it fits well there (and I believe=
 Adam had the same opinion, when I talked with him off-line in Berlin), afa=
ik there has never been any formal WG agreement that the &#39;bundle-only&#=
39; attribute will be defined in the BUNDLE
 draft.</p>
<p>=C2=A0</p>
<p>So, without going into the details (there are some=C2=A0issues, and=C2=
=A0if you are interested there are mail threads on the MMUSIC list dealing =
with those)=C2=A0do people have any issues with defining the SDP &#39;bundl=
e-only&#39; attribute, and the usage of it, in BUNDLE?</p>


<p>=C2=A0</p>
<p>If not, can we make a WG decision on the matter?</p>
<p>=C2=A0</p>
<p>Regards,</p><span class=3D"HOEnZb"><font color=3D"#888888">
<p>=C2=A0</p>
<p>Christer</p>
</font></span></div>
</div>

<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

--047d7b2e0abb57735f04e7615282--

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From: Suhas Nandakumar <suhasietf@gmail.com>
To: Justin Uberti <juberti@google.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Unified Plan: 'bundle-only' documentation
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+1


On Fri, Sep 27, 2013 at 11:04 AM, Justin Uberti <juberti@google.com> wrote:

> I think that the BUNDLE draft is the right place for this.
>
>
> On Fri, Sep 27, 2013 at 6:59 AM, Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
>
>>  Hi,
>>
>>
>>
>> On the MMUSIC list I have suggested updated SDP Offer/Answer text for
>> BUNDLE.
>>
>>
>>
>> Part of that text is the definition, and usage, of the SDP 'bundle-only'
>> attribute.
>>
>>
>>
>> However, eventhough I personally think it fits well there (and I believe
>> Adam had the same opinion, when I talked with him off-line in Berlin),
>> afaik there has never been any formal WG agreement that the 'bundle-only'
>> attribute will be defined in the BUNDLE draft.
>>
>>
>>
>> So, without going into the details (there are some issues, and if you are
>> interested there are mail threads on the MMUSIC list dealing with those) do
>> people have any issues with defining the SDP 'bundle-only' attribute, and
>> the usage of it, in BUNDLE?
>>
>>
>>
>> If not, can we make a WG decision on the matter?
>>
>>
>>
>> Regards,
>>
>>
>>
>> Christer
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--047d7bb04f686c1ef504e76163eb
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<div dir=3D"ltr">+1</div><div class=3D"gmail_extra"><br><br><div class=3D"g=
mail_quote">On Fri, Sep 27, 2013 at 11:04 AM, Justin Uberti <span dir=3D"lt=
r">&lt;<a href=3D"mailto:juberti@google.com" target=3D"_blank">juberti@goog=
le.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr">I think that the BUNDLE dra=
ft is the right place for this.</div><div class=3D"gmail_extra"><br><br><di=
v class=3D"gmail_quote">
<div><div class=3D"h5">On Fri, Sep 27, 2013 at 6:59 AM, Christer Holmberg <=
span dir=3D"ltr">&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" targ=
et=3D"_blank">christer.holmberg@ericsson.com</a>&gt;</span> wrote:<br>

</div></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bo=
rder-left:1px #ccc solid;padding-left:1ex"><div><div class=3D"h5">




<div>
<div style=3D"direction:ltr;font-size:10pt;font-family:Tahoma">
<p>Hi,</p>
<p>=A0</p>
<p>On the MMUSIC list I have suggested updated SDP Offer/Answer text for BU=
NDLE.</p>
<p>=A0</p>
<p>Part of that text is the definition, and usage, of the SDP &#39;bundle-o=
nly&#39; attribute.</p>
<p>=A0</p>
<p>However, eventhough I personally think it fits well there (and I believe=
 Adam had the same opinion, when I talked with him off-line in Berlin), afa=
ik there has never been any formal WG agreement that the &#39;bundle-only&#=
39; attribute will be defined in the BUNDLE
 draft.</p>
<p>=A0</p>
<p>So, without going into the details (there are some=A0issues, and=A0if yo=
u are interested there are mail threads on the MMUSIC list dealing with tho=
se)=A0do people have any issues with defining the SDP &#39;bundle-only&#39;=
 attribute, and the usage of it, in BUNDLE?</p>



<p>=A0</p>
<p>If not, can we make a WG decision on the matter?</p>
<p>=A0</p>
<p>Regards,</p><span><font color=3D"#888888">
<p>=A0</p>
<p>Christer</p>
</font></span></div>
</div>

<br></div></div>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

--047d7bb04f686c1ef504e76163eb--

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] JSEP-04: Some comments on Section 5.2.1. and 5.2.2 (19th september)
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Subject: Re: [rtcweb] JSEP-04: Some comments on Section 5.2.1. and 5.2.2 (19th september)
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Hi,

>>>> Section 5.2.2:
>>>> -----------------
>>>>
>>>>
>>>> Q_5:      Offer when in "local-offer"
>>>>
>>>> The text says:
>>>>
>>>> "If the initial offer was applied using setLocalDescription, but an
>>>>                answer from the remote side has not yet been applied, m=
eaning the
>>>>                PeerConnection is still in the "local-offer" state, the=
 steps for
>>>>                generating an initial offer should be followed,"
>>>>
>>>> I don't understand this. Why would you create a new Offer while you ar=
e waiting for an Answer to the previously sent Offer?
>>
>>

>>> Justin has asked for this in the case when the offer has not been sent =
to anyone else. This is not going to be useful for typical systems trying t=
o have SIP or Jingle compatible flow but who knows ...
>>
>> Again, we have consensus to base JSEP on 3264. And, if we are going to d=
eviate from 3264, I'd like to have some discussion and justification for th=
at.
>
>

> We have been through this many times before.
>
> The state machine that has been in the JSEP document for a while clearly =
shows that local-offer->local-offer state transitions are permitted.

If we've been through this many times, I must have missed, or forgotten, ea=
ch time :) Well, that can happen, so feel free to send me a link to a discu=
ssion/decission on this.

But, still, whatever deviations we agree to make to 3264, I would like to h=
ave it clearly documented - with justification.

In this case, I would also want to have it documented that the previous Off=
er is "cancelled", and that any associated pranswer/answer etc only applies=
 to the latest offer.

Regards,

Christer



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<p>Hi,</p>
<p><br>
&gt;&gt;&gt;&gt; Section 5.2.2:<br>
&gt;&gt;&gt;&gt; -----------------<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; Q_5: &nbsp; &nbsp; &nbsp;Offer when in &quot;local-offer&q=
uot;<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; The text says:<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; &quot;If the initial offer was applied using setLocalDescr=
iption, but an<br>
&gt;&gt;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;ans=
wer from the remote side has not yet been applied, meaning the<br>
&gt;&gt;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Pee=
rConnection is still in the &quot;local-offer&quot; state, the steps for<br=
>
&gt;&gt;&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;gen=
erating an initial offer should be followed,&quot;<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; I don't understand this. Why would you create a new Offer =
while you are waiting for an Answer to the previously sent Offer?<br>
&gt;&gt;<br>
&gt;&gt;</p>
<p>&gt;&gt;&gt; Justin has asked for this in the case when the offer has no=
t been sent to anyone else. This is not going to be useful for typical syst=
ems trying to have SIP or Jingle compatible flow but who knows ...<br>
&gt;&gt;<br>
&gt;&gt; Again, we have consensus to base JSEP on 3264. And, if we are goin=
g to deviate from 3264, I'd like to have some discussion and justification =
for that.<br>
&gt;<br>
&gt;</p>
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>&gt; We have been through this many times before.</div>
<div>&gt;</div>
<div>&gt;&nbsp;The state machine that has been in the JSEP document for a w=
hile clearly shows that local-offer-&gt;local-offer state transitions are p=
ermitted.&nbsp;</div>
<div>&nbsp;</div>
<div>If we've been through this many times, I must have missed, or forgotte=
n, each time :) Well, that can happen, so feel free to send me a link to a =
discussion/decission on this.</div>
<div>&nbsp;</div>
<div>But, still, whatever deviations we&nbsp;agree to&nbsp;make to 3264, I =
would like to have it clearly documented - with justification.</div>
<div>&nbsp;</div>
<div>In this case, I would&nbsp;also want to have it documented that the pr=
evious Offer is &quot;cancelled&quot;, and that&nbsp;any associated pranswe=
r/answer etc only applies to the latest offer.</div>
<div>&nbsp;</div>
<div>Regards,</div>
<div>&nbsp;</div>
<div>Christer</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
</div>
</div>
</div>
</div>
</div>
</body>
</html>

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From christer.holmberg@ericsson.com  Fri Sep 27 12:15:32 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Justin Uberti <juberti@google.com>, "Cullen Jennings (fluffy)" <fluffy@cisco.com>
Thread-Topic: [rtcweb] JSEP-04: Some comments on Section 5.2.1. and 5.2.2 (19th september)
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] JSEP-04: Some comments on Section 5.2.1. and 5.2.2 (19th september)
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Hi,



>>> Section 5.2.1:
>>> -----------------
>>>
>>> Q_1:      o- line <sess-version> initial value - deviation
>>>
>>> What is the reason for recommending a <sess-version> zero value in the =
initial Offer, when the RFC says SHOULD use an NTP format timestamp?
>>>
>>> I don=92t have any strong feelings, but in general: whenever we deviate=
 from the =93base=94 SDP procedures I think we should justify why.
>>

>> Lets use NTP
>

> It's not necessary to use NTP, since the session-id is already random. St=
arting the session id at zero makes it easier for developers see the change=
s to the session id. I can spell this out in the next version of the docume=
nt.

I assume you mean the session version, because that is what is currently ze=
ro.

But, I am sure there is a reason why SDP recommends NTP. So, again, if we w=
ant to change that, we shall discuss it in MMUSIC.

And, if we agree to change it, clearly document and justify it.



>>> Q_3:      RTP
>>>
>>> The text says:
>>>
>>> =93The <proto> field MUST be set to "RTP/SAVPF". =93
>>>
>>> But, that of course only applies to RTP based streams (not the data cha=
nnel). Also, in general, it needs to be clear what information needs to be =
in every m- line, and what information is protocol specific. I would sugges=
t to have a =93General=94 sub-section, a =93RTP=94 sub-section, etc.
>>>
>>>
>> Yep - I think this should be offers are created with RTP/SAVPF for audio=
 and video and system can reeve offers with SAVPF or SAVP
>
> Currently the document specifies what to do with media lines, with later =
discussion of handling the SCTP line. Do you think the current information =
is not sufficiently descriptive?

My point was that we need to be more clear on what is generic, what is RTP,=
 and what is SCTP. But, as you say SCTP text is still to be added, that cla=
rification can be done when the SCTP text is added.



>> Q_4:      BUNDLE
>>
>> The text says:
>>
>> =93If a m=3D section is not being bundled into another m=3D section, it =
MUST
>>                generate a unique set of ICE credentials and gather its o=
wn set of
>>                candidates.  Otherwise, it MUST use the same ICE credenti=
als and
>>                candidates that were used in the m=3D section that it is =
being bundled
>>                into.=94
>>
>> As, when BUNDLE is used, the initial Offer will contain identical ICE ca=
ndidates, does that mean that we will also include identical address inform=
ation in the initial Offer?
>
>
> no the initial offer will not have identical stuff other than on lines th=
at "bundle-only"

Assuming you by identical mean port zero, we'll still have to agree on that=
. But, that discussion belongs to MMUSIC.







>>> Q_6:      BUNDLE
>>>
>>> We=92ll probably also need some text about BUNDLE.
>>>
>>

>> yep - but plan is to match up with unified plan
>>
>Can you be specific about what more you think needs to be said? The create=
Answer treatment of BUNDLE is one clear thing that is currently missing.



I want to have text covering both the 1st (unique address) and 2nd (shared =
address) Offer.



And, when a PeerConnection is created due to forking (ie you use a separate=
 PeerConnection for each forked leg of a session), I assume already the 1st=
 Offer for that PeerConnection can have a shared address (as the remote ent=
ity has indicated support for BUNDLE). That also needs to be covered.





Regards,



Christer




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<div style=3D"color: rgb(0, 0, 0); font-family: Tahoma; font-size: 10pt; di=
rection: ltr;">
<p>Hi,</p>
<p>&nbsp;</p>
<p>&gt;&gt;&gt; Section 5.2.1:<br>
&gt;&gt;&gt; -----------------<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Q_1: &nbsp; &nbsp; &nbsp;o- line &lt;sess-version&gt; initial =
value - deviation<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; What is the reason for recommending a &lt;sess-version&gt; zer=
o value in the initial Offer, when the RFC says SHOULD use an NTP format ti=
mestamp?<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; I don=92t have any strong feelings, but in general: whenever w=
e deviate from the =93base=94 SDP procedures I think we should justify why.=
<br>
&gt;&gt;</p>
<p>&gt;&gt; Lets use NTP<br>
&gt;</p>
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>&gt; It's not necessary to use NTP, since the session-id is already ra=
ndom. Starting the session id at zero makes it easier for developers see th=
e changes to the session id. I can spell this out in the next version of th=
e document.</div>
<div>&nbsp;</div>
<div>I assume you mean the session version, because that is what is current=
ly zero.</div>
<div>&nbsp;</div>
<div>But, I am sure there is a reason why SDP recommends NTP. So, again, if=
 we want to change that, we shall discuss it in MMUSIC.</div>
<div>&nbsp;</div>
<div>And, if we agree to change it, clearly document and justify it.</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div>&gt;&gt;&gt; Q_3: &nbsp; &nbsp; &nbsp;RTP<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; The text says:<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; =93The &lt;proto&gt; field MUST be set to &quot;RTP/SAVPF&quot=
;. =93<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; But, that of course only applies to RTP based streams (not the=
 data channel). Also, in general, it needs to be clear what information nee=
ds to be in every m- line, and what information is protocol specific. I wou=
ld suggest to have a =93General=94 sub-section,
 a =93RTP=94 sub-section, etc.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;</div>
<div>&gt;&gt; Yep - I think this should be offers are created with RTP/SAVP=
F for audio and video and system can reeve offers with SAVPF or SAVP<br>
&gt;</div>
<div>&gt; Currently the document specifies what to do with media lines, wit=
h later discussion of handling the SCTP line. Do you think the current info=
rmation is not sufficiently descriptive?&nbsp;</div>
<div>&nbsp;</div>
<div>My point was that we need to be more clear on what is generic, what is=
 RTP, and what is SCTP. But, as you say SCTP text is still to be added, tha=
t clarification can be done when the SCTP text is added.</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div><br>
&gt;&gt; Q_4: &nbsp; &nbsp; &nbsp;BUNDLE<br>
&gt;&gt;<br>
&gt;&gt; The text says:<br>
&gt;&gt;<br>
&gt;&gt; =93If a m=3D section is not being bundled into another m=3D sectio=
n, it MUST<br>
&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;generate a =
unique set of ICE credentials and gather its own set of<br>
&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;candidates.=
 &nbsp;Otherwise, it MUST use the same ICE credentials and<br>
&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;candidates =
that were used in the m=3D section that it is being bundled<br>
&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;into.=94<br=
>
&gt;&gt;<br>
&gt;&gt; As, when BUNDLE is used, the initial Offer will contain identical =
ICE candidates, does that mean that we will also include identical address =
information in the initial Offer?<br>
&gt;<br>
&gt;</div>
<div>&gt; no the initial offer will not have identical stuff other than on =
lines that &quot;bundle-only&quot;</div>
<div>&nbsp;</div>
<div>Assuming you by identical mean port zero, we'll still have to agree on=
 that. But, that discussion belongs to MMUSIC.</div>
<p>&nbsp;</p>
<p>&nbsp;</p>
<p>&nbsp;</p>
<p>&gt;&gt;&gt; Q_6: &nbsp; &nbsp; &nbsp;BUNDLE<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; We=92ll probably also need some text about BUNDLE.<br>
&gt;&gt;&gt;<br>
&gt;&gt;</p>
<p>&gt;&gt; yep - but plan is to match up with unified plan<br>
&gt;&gt;<br>
&gt;Can you be specific about what more you think needs to be said? The cre=
ateAnswer treatment of BUNDLE is one clear thing that is currently missing.=
</p>
<p>&nbsp;</p>
<p>I want to have text covering both&nbsp;the 1st (unique address) and 2nd =
(shared address) Offer.</p>
<p>&nbsp;</p>
<p>And, when a PeerConnection is created due to forking (ie you use a separ=
ate PeerConnection for each forked leg of a session), I assume already the =
1st Offer for that PeerConnection can have a shared address (as the remote =
entity has indicated support for
 BUNDLE). That also needs to be covered.</p>
<p>&nbsp;</p>
<p>&nbsp;</p>
<p>Regards,</p>
<p>&nbsp;</p>
<p>Christer</p>
<p>&nbsp;</p>
</div>
<br>
</div>
</div>
</div>
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From jlaurens@cisco.com  Fri Sep 27 12:44:12 2013
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From: "Jeremy Laurenson (jlaurens)" <jlaurens@cisco.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] TURN server address via DHCP,	WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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Subject: Re: [rtcweb] TURN server address via DHCP, WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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What is the usual "business model" for this? Assume in most cases the =
servers are hosted by folks who actually have a revenue stream from the =
app that is leveraging the servers.

Do we have some idea as to how the browser folks want to pay for the =
"WebRTC turn tax"?




On Sep 27, 2013, at 11:58 AM, Justin Uberti <juberti@google.com> wrote:

> Agree. I still think that extending PAC files to include TURN =
information is the right way to go.
>=20
> PAC files can already be discovered via DHCP or DNS, there is no need =
to reinvent this wheel.
>=20
>=20
> On Thu, Sep 26, 2013 at 4:55 PM, Cullen Jennings (fluffy) =
<fluffy@cisco.com> wrote:
>=20
> I think we need to give some advise to the browsers vendors on what =
they should implement to find turn servers
>=20
> On Sep 26, 2013, at 5:13 AM, "Markus.Isomaki@nokia.com" =
<Markus.Isomaki@nokia.com> wrote:
>=20
> > Hi,
> >
> > Cullen Jennings wrote:
> >>
> >> I will note that browsers have many ways to learn about HTTP =
proxies from
> >> the network and it seems to me that using some of theses same =
technique
> >> might also be a good way to learn about TURN servers.
> >
> > I agree. I assume that in enterprises it would be the same people =
managing both of these.
> >
> > Is there something the IETF can or should do about this, or do we =
just assume it will happen? A new DHCP option is something the IETF =
could easily do, but I also doubt how usable that would be.
> >
> > Markus
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dus-ascii"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;">What =
is the usual "business model" for this? Assume in most cases the servers =
are hosted by folks who actually have a revenue stream from the app that =
is leveraging the servers.<div><br></div><div>Do we have some idea as to =
how the browser folks want to pay for the "WebRTC turn =
tax"?</div><div><br></div><div><div apple-content-edited=3D"true"><div =
style=3D"color: rgb(0, 0, 0); letter-spacing: normal; orphans: auto; =
text-align: start; text-indent: 0px; text-transform: none; white-space: =
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style=3D"color: rgb(0, 0, 0); font-family: 'Century Gothic'; font-style: =
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-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;"><div =
style=3D"color: rgb(0, 0, 0); font-family: 'Century Gothic'; font-style: =
normal; font-variant: normal; font-weight: normal; letter-spacing: =
normal; line-height: normal; orphans: 2; text-align: -webkit-auto; =
text-indent: 0px; text-transform: none; white-space: normal; widows: 2; =
word-spacing: 0px; -webkit-text-stroke-width: 0px; word-wrap: =
break-word; -webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space;"><div style=3D"color: rgb(0, 0, 0); font-family: =
'Century Gothic'; font-style: normal; font-variant: normal; font-weight: =
normal; letter-spacing: normal; line-height: normal; orphans: 2; =
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white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-text-stroke-width: 0px; word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space;"><div><br></div><table bgcolor=3D"WHITE" border=3D"0" =
style=3D"font-family: 'Century Gothic'; letter-spacing: normal; orphans: =
2; text-indent: 0px; text-transform: none; widows: 2; word-spacing: 0px; =
-webkit-text-stroke-width: 0px;"><tbody><tr><td align=3D"CENTER" =
valign=3D"TOP"><br></td></tr></tbody></table></div></div></div></div></div=
></div></div></div>
</div>
<br><div><div>On Sep 27, 2013, at 11:58 AM, Justin Uberti &lt;<a =
href=3D"mailto:juberti@google.com">juberti@google.com</a>&gt; =
wrote:</div><br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><meta http-equiv=3D"Content-Type" content=3D"text/html; =
charset=3Dutf-8"><div dir=3D"ltr">Agree. I still think that extending =
PAC files to include TURN information is the right way to =
go.<div><br></div><div>PAC files can already be discovered via DHCP or =
DNS, there is no need to reinvent this wheel.</div>

</div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On =
Thu, Sep 26, 2013 at 4:55 PM, Cullen Jennings (fluffy) <span =
dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@cisco.com" =
target=3D"_blank">fluffy@cisco.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 =
.8ex;border-left:1px #ccc solid;padding-left:1ex"><br>
I think we need to give some advise to the browsers vendors on what they =
should implement to find turn servers<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
On Sep 26, 2013, at 5:13 AM, "<a =
href=3D"mailto:Markus.Isomaki@nokia.com">Markus.Isomaki@nokia.com</a>" =
&lt;<a =
href=3D"mailto:Markus.Isomaki@nokia.com">Markus.Isomaki@nokia.com</a>&gt; =
wrote:<br>
<br>
&gt; Hi,<br>
&gt;<br>
&gt; Cullen Jennings wrote:<br>
&gt;&gt;<br>
&gt;&gt; I will note that browsers have many ways to learn about HTTP =
proxies from<br>
&gt;&gt; the network and it seems to me that using some of theses same =
technique<br>
&gt;&gt; might also be a good way to learn about TURN servers.<br>
&gt;<br>
&gt; I agree. I assume that in enterprises it would be the same people =
managing both of these.<br>
&gt;<br>
&gt; Is there something the IETF can or should do about this, or do we =
just assume it will happen? A new DHCP option is something the IETF =
could easily do, but I also doubt how usable that would be.<br>
&gt;<br>
&gt; Markus<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" =
target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>
_______________________________________________<br>rtcweb mailing =
list<br><a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>https://www.ietf.or=
g/mailman/listinfo/rtcweb<br></blockquote></div><br></div></body></html>=

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From: Justin Uberti <juberti@google.com>
Date: Fri, 27 Sep 2013 14:42:21 -0700
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Subject: Re: [rtcweb] TURN server address via DHCP, WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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--089e013c69c6afb54a04e7645daf
Content-Type: text/plain; charset=UTF-8

The business model is the same as for HTTP proxies. ISPs or enterprises
that want to better control their traffic will pay.


On Fri, Sep 27, 2013 at 12:44 PM, Jeremy Laurenson (jlaurens) <
jlaurens@cisco.com> wrote:

> What is the usual "business model" for this? Assume in most cases the
> servers are hosted by folks who actually have a revenue stream from the app
> that is leveraging the servers.
>
> Do we have some idea as to how the browser folks want to pay for the
> "WebRTC turn tax"?
>
>
>
>
> On Sep 27, 2013, at 11:58 AM, Justin Uberti <juberti@google.com> wrote:
>
> Agree. I still think that extending PAC files to include TURN information
> is the right way to go.
>
> PAC files can already be discovered via DHCP or DNS, there is no need to
> reinvent this wheel.
>
>
> On Thu, Sep 26, 2013 at 4:55 PM, Cullen Jennings (fluffy) <
> fluffy@cisco.com> wrote:
>
>>
>> I think we need to give some advise to the browsers vendors on what they
>> should implement to find turn servers
>>
>> On Sep 26, 2013, at 5:13 AM, "Markus.Isomaki@nokia.com" <
>> Markus.Isomaki@nokia.com> wrote:
>>
>> > Hi,
>> >
>> > Cullen Jennings wrote:
>> >>
>> >> I will note that browsers have many ways to learn about HTTP proxies
>> from
>> >> the network and it seems to me that using some of theses same technique
>> >> might also be a good way to learn about TURN servers.
>> >
>> > I agree. I assume that in enterprises it would be the same people
>> managing both of these.
>> >
>> > Is there something the IETF can or should do about this, or do we just
>> assume it will happen? A new DHCP option is something the IETF could easily
>> do, but I also doubt how usable that would be.
>> >
>> > Markus
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--089e013c69c6afb54a04e7645daf
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">The business model is the same as for HTTP proxies. ISPs o=
r enterprises that want to better control their traffic will pay.</div><div=
 class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Fri, Sep 27, 2=
013 at 12:44 PM, Jeremy Laurenson (jlaurens) <span dir=3D"ltr">&lt;<a href=
=3D"mailto:jlaurens@cisco.com" target=3D"_blank">jlaurens@cisco.com</a>&gt;=
</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div style=3D"word-wrap:break-word">What is =
the usual &quot;business model&quot; for this? Assume in most cases the ser=
vers are hosted by folks who actually have a revenue stream from the app th=
at is leveraging the servers.<div>

<br></div><div>Do we have some idea as to how the browser folks want to pay=
 for the &quot;WebRTC turn tax&quot;?</div><div><div class=3D"h5"><div><br>=
</div><div><div><div style=3D"text-indent:0px;letter-spacing:normal;text-al=
ign:start;text-transform:none;white-space:normal;word-wrap:break-word;word-=
spacing:0px">

<div style=3D"text-indent:0px;letter-spacing:normal;text-align:start;text-t=
ransform:none;white-space:normal;word-wrap:break-word;word-spacing:0px"><di=
v style=3D"text-indent:0px;letter-spacing:normal;text-align:start;text-tran=
sform:none;white-space:normal;word-wrap:break-word;word-spacing:0px">

<div style=3D"text-indent:0px;letter-spacing:normal;font-variant:normal;tex=
t-align:-webkit-auto;font-style:normal;font-weight:normal;line-height:norma=
l;text-transform:none;white-space:normal;font-family:&#39;Century Gothic&#3=
9;;word-wrap:break-word;word-spacing:0px">

<div style=3D"text-indent:0px;letter-spacing:normal;font-variant:normal;tex=
t-align:-webkit-auto;font-style:normal;font-weight:normal;line-height:norma=
l;text-transform:none;white-space:normal;font-family:&#39;Century Gothic&#3=
9;;word-wrap:break-word;word-spacing:0px">

<div style=3D"text-indent:0px;letter-spacing:normal;font-variant:normal;tex=
t-align:-webkit-auto;font-style:normal;font-weight:normal;line-height:norma=
l;text-transform:none;white-space:normal;font-family:&#39;Century Gothic&#3=
9;;word-wrap:break-word;word-spacing:0px">

<div style=3D"text-indent:0px;letter-spacing:normal;font-variant:normal;tex=
t-align:-webkit-auto;font-style:normal;font-weight:normal;line-height:norma=
l;text-transform:none;white-space:normal;font-family:&#39;Century Gothic&#3=
9;;word-wrap:break-word;word-spacing:0px">

<div style=3D"text-indent:0px;letter-spacing:normal;font-variant:normal;tex=
t-align:-webkit-auto;font-style:normal;font-weight:normal;line-height:norma=
l;text-transform:none;white-space:normal;font-family:&#39;Century Gothic&#3=
9;;word-wrap:break-word;word-spacing:0px">

<div><br></div><table bgcolor=3D"WHITE" border=3D"0" style=3D"font-family:&=
#39;Century Gothic&#39;;letter-spacing:normal;text-indent:0px;text-transfor=
m:none;word-spacing:0px"><tbody><tr><td align=3D"CENTER" valign=3D"TOP"><br=
></td>

</tr></tbody></table></div></div></div></div></div></div></div></div>
</div>
<br><div><div>On Sep 27, 2013, at 11:58 AM, Justin Uberti &lt;<a href=3D"ma=
ilto:juberti@google.com" target=3D"_blank">juberti@google.com</a>&gt; wrote=
:</div><br><blockquote type=3D"cite"><div dir=3D"ltr">Agree. I still think =
that extending PAC files to include TURN information is the right way to go=
.<div>

<br></div><div>PAC files can already be discovered via DHCP or DNS, there i=
s no need to reinvent this wheel.</div>

</div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Thu,=
 Sep 26, 2013 at 4:55 PM, Cullen Jennings (fluffy) <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:fluffy@cisco.com" target=3D"_blank">fluffy@cisco.com</a>&gt=
;</span> wrote:<br>



<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><br>
I think we need to give some advise to the browsers vendors on what they sh=
ould implement to find turn servers<br>
<div><div><br>
On Sep 26, 2013, at 5:13 AM, &quot;<a href=3D"mailto:Markus.Isomaki@nokia.c=
om" target=3D"_blank">Markus.Isomaki@nokia.com</a>&quot; &lt;<a href=3D"mai=
lto:Markus.Isomaki@nokia.com" target=3D"_blank">Markus.Isomaki@nokia.com</a=
>&gt; wrote:<br>


<br>
&gt; Hi,<br>
&gt;<br>
&gt; Cullen Jennings wrote:<br>
&gt;&gt;<br>
&gt;&gt; I will note that browsers have many ways to learn about HTTP proxi=
es from<br>
&gt;&gt; the network and it seems to me that using some of theses same tech=
nique<br>
&gt;&gt; might also be a good way to learn about TURN servers.<br>
&gt;<br>
&gt; I agree. I assume that in enterprises it would be the same people mana=
ging both of these.<br>
&gt;<br>
&gt; Is there something the IETF can or should do about this, or do we just=
 assume it will happen? A new DHCP option is something the IETF could easil=
y do, but I also doubt how usable that would be.<br>
&gt;<br>
&gt; Markus<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>
_______________________________________________<br>rtcweb mailing list<br><=
a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br>=
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>

</blockquote></div><br></div></div></div></div><br>________________________=
_______________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

--089e013c69c6afb54a04e7645daf--

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Subject: Re: [rtcweb] JSEP-04: Some comments on Section 5.2.1. and 5.2.2 (19th september)
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On Fri, Sep 27, 2013 at 12:15 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>  Hi,
>
>
>
> >>> Section 5.2.1:
> >>> -----------------
> >>>
> >>> Q_1:      o- line <sess-version> initial value - deviation
> >>>
> >>> What is the reason for recommending a <sess-version> zero value in th=
e
> initial Offer, when the RFC says SHOULD use an NTP format timestamp?
> >>>
> >>> I don=E2=80=99t have any strong feelings, but in general: whenever we=
 deviate
> from the =E2=80=9Cbase=E2=80=9D SDP procedures I think we should justify =
why.
> >>
>
> >> Lets use NTP
> >
>   > It's not necessary to use NTP, since the session-id is already
> random. Starting the session id at zero makes it easier for developers se=
e
> the changes to the session id. I can spell this out in the next version o=
f
> the document.
>
> I assume you mean the session version, because that is what is currently
> zero.
>

whoops :)


>
> But, I am sure there is a reason why SDP recommends NTP. So, again, if we
> want to change that, we shall discuss it in MMUSIC.
>
> And, if we agree to change it, clearly document and justify it.
>

As long as what we are proposing is consistent with 3261 (the suggestion of
NTP is only SHOULD strength), and we explain why it should be done this way
in the rtcweb document (which the document already does, to some degree), I
don't think we need to argue about it in mmusic.

There are bigger fish to fry...

>
>
>
> >>> Q_3:      RTP
> >>>
> >>> The text says:
> >>>
> >>> =E2=80=9CThe <proto> field MUST be set to "RTP/SAVPF". =E2=80=9C
> >>>
> >>> But, that of course only applies to RTP based streams (not the data
> channel). Also, in general, it needs to be clear what information needs t=
o
> be in every m- line, and what information is protocol specific. I would
> suggest to have a =E2=80=9CGeneral=E2=80=9D sub-section, a =E2=80=9CRTP=
=E2=80=9D sub-section, etc.
> >>>
> >>>
> >> Yep - I think this should be offers are created with RTP/SAVPF for
> audio and video and system can reeve offers with SAVPF or SAVP
> >
> > Currently the document specifies what to do with media lines, with late=
r
> discussion of handling the SCTP line. Do you think the current informatio=
n
> is not sufficiently descriptive?
>
> My point was that we need to be more clear on what is generic, what is
> RTP, and what is SCTP. But, as you say SCTP text is still to be added, th=
at
> clarification can be done when the SCTP text is added.
>

Sorry I was unclear - there is SCTP-specific text in the current document.

>
>
>
> >> Q_4:      BUNDLE
> >>
> >> The text says:
> >>
> >> =E2=80=9CIf a m=3D section is not being bundled into another m=3D sect=
ion, it MUST
> >>                generate a unique set of ICE credentials and gather its
> own set of
> >>                candidates.  Otherwise, it MUST use the same ICE
> credentials and
> >>                candidates that were used in the m=3D section that it i=
s
> being bundled
> >>                into.=E2=80=9D
> >>
> >> As, when BUNDLE is used, the initial Offer will contain identical ICE
> candidates, does that mean that we will also include identical address
> information in the initial Offer?
> >
> >
> > no the initial offer will not have identical stuff other than on lines
> that "bundle-only"
>
> Assuming you by identical mean port zero, we'll still have to agree on
> that. But, that discussion belongs to MMUSIC.
>
>
>
>
>
>
>
> >>> Q_6:      BUNDLE
> >>>
> >>> We=E2=80=99ll probably also need some text about BUNDLE.
> >>>
> >>
>
> >> yep - but plan is to match up with unified plan
> >>
> >Can you be specific about what more you think needs to be said? The
> createAnswer treatment of BUNDLE is one clear thing that is currently
> missing.
>
>
>
> I want to have text covering both the 1st (unique address) and 2nd (share=
d
> address) Offer.
>
Acknowledged.

>
>
> And, when a PeerConnection is created due to forking (ie you use a
> separate PeerConnection for each forked leg of a session), I assume alrea=
dy
> the 1st Offer for that PeerConnection can have a shared address (as the
> remote entity has indicated support for BUNDLE). That also needs to be
> covered.
>
>
>
Yes. Which may indicate the need for a setting on a PeerConnection to use a
shared address on the initial offer, since it might have come from such a
fork.

>
>
> Regards,
>
>
>
> Christer
>
>
>
>

--001a11330a1681405f04e76472d2
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Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Fri, Sep 27, 2013 at 12:15 PM, Christer Holmberg <span dir=3D"lt=
r">&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">=
christer.holmberg@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">




<div>
<div style=3D"direction:ltr;font-size:10pt;font-family:Tahoma"><div class=
=3D"im">
<p>Hi,</p>
<p>=C2=A0</p>
<p>&gt;&gt;&gt; Section 5.2.1:<br>
&gt;&gt;&gt; -----------------<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Q_1: =C2=A0 =C2=A0 =C2=A0o- line &lt;sess-version&gt; initial =
value - deviation<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; What is the reason for recommending a &lt;sess-version&gt; zer=
o value in the initial Offer, when the RFC says SHOULD use an NTP format ti=
mestamp?<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; I don=E2=80=99t have any strong feelings, but in general: when=
ever we deviate from the =E2=80=9Cbase=E2=80=9D SDP procedures I think we s=
hould justify why.<br>
&gt;&gt;</p>
<p>&gt;&gt; Lets use NTP<br>
&gt;</p>
</div><div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote"><div class=3D"im">
<div>&gt; It&#39;s not necessary to use NTP, since the session-id is alread=
y random. Starting the session id at zero makes it easier for developers se=
e the changes to the session id. I can spell this out in the next version o=
f the document.</div>


<div>=C2=A0</div>
</div><div>I assume you mean the session version, because that is what is c=
urrently zero.</div>
<div></div></div></div></div></div></div></div></blockquote><div><br></div>=
<div>whoops :)</div><div>=C2=A0</div><blockquote class=3D"gmail_quote" styl=
e=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><d=
iv style=3D"direction:ltr;font-size:10pt;font-family:Tahoma">

<div><div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"=
><div>=C2=A0</div>
<div>But, I am sure there is a reason why SDP recommends NTP. So, again, if=
 we want to change that, we shall discuss it in MMUSIC.</div>
<div>=C2=A0</div>
<div>And, if we agree to change it, clearly document and justify it.</div><=
/div></div></div></div></div></div></blockquote><div><br></div><div>As long=
 as what we are proposing is consistent with 3261 (the suggestion of NTP is=
 only SHOULD strength), and we explain why it should be done this way in th=
e rtcweb document (which the document already does, to some degree), I don&=
#39;t think we need to argue about it in mmusic.=C2=A0</div>

<div><br></div><div>There are bigger fish to fry...</div><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex"><div><div style=3D"direction:ltr;font-size:10pt;font-family:T=
ahoma">

<div><div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"=
><div class=3D"im">
<div>=C2=A0</div>
<div>=C2=A0</div>
<div>=C2=A0</div>
<div>&gt;&gt;&gt; Q_3: =C2=A0 =C2=A0 =C2=A0RTP<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; The text says:<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; =E2=80=9CThe &lt;proto&gt; field MUST be set to &quot;RTP/SAVP=
F&quot;. =E2=80=9C<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; But, that of course only applies to RTP based streams (not the=
 data channel). Also, in general, it needs to be clear what information nee=
ds to be in every m- line, and what information is protocol specific. I wou=
ld suggest to have a =E2=80=9CGeneral=E2=80=9D sub-section,
 a =E2=80=9CRTP=E2=80=9D sub-section, etc.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;</div>
<div>&gt;&gt; Yep - I think this should be offers are created with RTP/SAVP=
F for audio and video and system can reeve offers with SAVPF or SAVP<br>
&gt;</div>
<div>&gt; Currently the document specifies what to do with media lines, wit=
h later discussion of handling the SCTP line. Do you think the current info=
rmation is not sufficiently descriptive?=C2=A0</div>
<div>=C2=A0</div>
</div><div>My point was that we need to be more clear on what is generic, w=
hat is RTP, and what is SCTP. But, as you say SCTP text is still to be adde=
d, that clarification can be done when the SCTP text is added.</div></div>

</div></div></div></div></div></blockquote><div><br></div><div>Sorry I was =
unclear - there is SCTP-specific text in the current document.=C2=A0</div><=
blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px=
 #ccc solid;padding-left:1ex">

<div><div style=3D"direction:ltr;font-size:10pt;font-family:Tahoma"><div><d=
iv dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><div c=
lass=3D"im">
<div>=C2=A0</div>
<div>=C2=A0</div>
<div><br>
&gt;&gt; Q_4: =C2=A0 =C2=A0 =C2=A0BUNDLE<br>
&gt;&gt;<br>
&gt;&gt; The text says:<br>
&gt;&gt;<br>
&gt;&gt; =E2=80=9CIf a m=3D section is not being bundled into another m=3D =
section, it MUST<br>
&gt;&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0generate a =
unique set of ICE credentials and gather its own set of<br>
&gt;&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0candidates.=
 =C2=A0Otherwise, it MUST use the same ICE credentials and<br>
&gt;&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0candidates =
that were used in the m=3D section that it is being bundled<br>
&gt;&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0into.=E2=80=
=9D<br>
&gt;&gt;<br>
&gt;&gt; As, when BUNDLE is used, the initial Offer will contain identical =
ICE candidates, does that mean that we will also include identical address =
information in the initial Offer?<br>
&gt;<br>
&gt;</div>
<div>&gt; no the initial offer will not have identical stuff other than on =
lines that &quot;bundle-only&quot;</div>
<div>=C2=A0</div>
</div><div>Assuming you by identical mean port zero, we&#39;ll still have t=
o agree on that. But, that discussion belongs to MMUSIC.</div><div class=3D=
"im">
<p>=C2=A0</p>
<p>=C2=A0</p>
<p>=C2=A0</p>
<p>&gt;&gt;&gt; Q_6: =C2=A0 =C2=A0 =C2=A0BUNDLE<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; We=E2=80=99ll probably also need some text about BUNDLE.<br>
&gt;&gt;&gt;<br>
&gt;&gt;</p>
<p>&gt;&gt; yep - but plan is to match up with unified plan<br>
&gt;&gt;<br>
&gt;Can you be specific about what more you think needs to be said? The cre=
ateAnswer treatment of BUNDLE is one clear thing that is currently missing.=
</p>
<p>=C2=A0</p>
</div><p>I want to have text covering both=C2=A0the 1st (unique address) an=
d 2nd (shared address) Offer.</p></div></div></div></div></div></div></bloc=
kquote><div>Acknowledged.=C2=A0<br></div><blockquote class=3D"gmail_quote" =
style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

<div><div style=3D"direction:ltr;font-size:10pt;font-family:Tahoma"><div><d=
iv dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">
<p>=C2=A0</p>
<p>And, when a PeerConnection is created due to forking (ie you use a separ=
ate PeerConnection for each forked leg of a session), I assume already the =
1st Offer for that PeerConnection can have a shared address (as the remote =
entity has indicated support for
 BUNDLE). That also needs to be covered.</p>
<p>=C2=A0</p></div></div></div></div></div></div></blockquote><div>Yes. Whi=
ch may indicate the need for a setting on a PeerConnection to use a shared =
address on the initial offer, since it might have come from such a fork.=C2=
=A0</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div><div style=3D"direction:ltr;font-size:1=
0pt;font-family:Tahoma"><div><div dir=3D"ltr"><div class=3D"gmail_extra"><d=
iv class=3D"gmail_quote">


<p>=C2=A0</p>
<p>Regards,</p><span class=3D"HOEnZb"><font color=3D"#888888">
<p>=C2=A0</p>
<p>Christer</p>
<p>=C2=A0</p>
</font></span></div>
<br>
</div>
</div>
</div>
</div>
</div>

</blockquote></div><br></div></div>

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] JSEP-04: Some comments on Section 5.2.1. and 5.2.2 (19th september)
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Cc: "Cullen Jennings \(fluffy\)" <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] JSEP-04: Some comments on Section 5.2.1. and 5.2.2 (19th september)
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Hi,



>>>>> Q_3:      RTP
>>>>>
>>>>> The text says:
>>>>>
>>>>> =93The <proto> field MUST be set to "RTP/SAVPF". =93
>>>>>
>>>>> But, that of course only applies to RTP based streams (not the data c=
hannel). Also, in general, it needs to be clear what information
>>>>> needs to be in every m- line, and what information is protocol specif=
ic. I would suggest to have a =93General=94 sub-section, a =93RTP=94 sub-se=
ction, etc.
>>>>>
>>>>>
>>>> Yep - I think this should be offers are created with RTP/SAVPF for aud=
io and video and system can reeve offers with SAVPF or SAVP
>>>
>>> Currently the document specifies what to do with media lines, with late=
r discussion of handling the SCTP line. Do you think the current informatio=
n is not sufficiently descriptive?
>>
>> My point was that we need to be more clear on what is generic, what is R=
TP, and what is SCTP. But, as you say SCTP text is still to be added, that =
clarification can be done when the SCTP text is added.

>Sorry I was unclear - there is SCTP-specific text in the current document.

Ok, so while the content itself might be ok, at some point I'd like to have=
 separate sub-sections for the generic stuff, the RTP specicif stuff, and t=
he SCTP specific stuff.



>>>>> Q_6:      BUNDLE
>>>>>
>>>>> We=92ll probably also need some text about BUNDLE.
>>>>>
>>>>

>>>> yep - but plan is to match up with unified plan
>>>>
>>>Can you be specific about what more you think needs to be said? The crea=
teAnswer treatment of BUNDLE is one clear thing that is currently missing.

>>

>> I want to have text covering both the 1st (unique address) and 2nd (shar=
ed address) Offer.

> Acknowledged.

>

>> And, when a PeerConnection is created due to forking (ie you use a separ=
ate PeerConnection for each forked leg of a session), I assume

>> already the 1st Offer for that PeerConnection can have a shared address =
(as the remote entity has indicated support for BUNDLE). That also needs to=
 be covered.

>>

> Yes. Which may indicate the need for a setting on a PeerConnection to use=
 a shared address on the initial offer, since it might have come from such =
a fork.

Excellent.

Regards,

Christer


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<html dir=3D"ltr">
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3DWindows-1=
252">
<style id=3D"owaParaStyle">P {=0A=
	MARGIN-BOTTOM: 0px; MARGIN-TOP: 0px=0A=
}=0A=
</style>
</head>
<body ocsi=3D"0" fpstyle=3D"1">
<div style=3D"color: rgb(0, 0, 0); font-family: Tahoma; font-size: 10pt; di=
rection: ltr;">
<p>Hi,</p>
<p>&nbsp;</p>
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin: 0px 0px 0px 0.8ex; paddi=
ng-left: 1ex; border-left-color: rgb(204, 204, 204); border-left-width: 1px=
; border-left-style: solid;">
<div>
<div style=3D"font-family: Tahoma; font-size: 10pt; direction: ltr;">
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div class=3D"im">
<div>&gt;&gt;&gt;&gt;&gt; Q_3: &nbsp; &nbsp; &nbsp;RTP<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; The text says:<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; =93The &lt;proto&gt; field MUST be set to &quot;RTP/SA=
VPF&quot;. =93<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; But, that of course only applies to RTP based streams =
(not the data channel). Also, in general, it needs to be clear what informa=
tion</div>
<div>&gt;&gt;&gt;&gt;&gt;&nbsp;needs to be in every m- line, and what infor=
mation is protocol specific. I would suggest to have a =93General=94 sub-se=
ction, a =93RTP=94 sub-section, etc.<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;</div>
<div>&gt;&gt;&gt;&gt; Yep - I think this should be offers are created with =
RTP/SAVPF for audio and video and system can reeve offers with SAVPF or SAV=
P<br>
&gt;&gt;&gt;</div>
<div>&gt;&gt;&gt; Currently the document specifies what to do with media li=
nes, with later discussion of handling the SCTP line. Do you think the curr=
ent information is not sufficiently descriptive?&nbsp;</div>
<div>&gt;&gt;</div>
</div>
<div>&gt;&gt; My point was that we need to be more clear on what is generic=
, what is RTP, and what is SCTP. But, as you say SCTP text is still to be a=
dded, that clarification can be done when the SCTP text is added.</div>
</div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>&gt;Sorry I was unclear - there is SCTP-specific text in the current d=
ocument.&nbsp;</div>
<div>&nbsp;</div>
<div>Ok, so while the content itself might be ok, at some point I'd like to=
 have separate sub-sections for the generic stuff, the RTP specicif stuff, =
and the SCTP specific stuff.
</div>
<div>&nbsp;</div>
<div>&nbsp;&nbsp;</div>
<blockquote class=3D"gmail_quote" style=3D"margin: 0px 0px 0px 0.8ex; paddi=
ng-left: 1ex; border-left-color: rgb(204, 204, 204); border-left-width: 1px=
; border-left-style: solid;">
<div>
<div style=3D"font-family: Tahoma; font-size: 10pt; direction: ltr;">
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div class=3D"im">
<p>&gt;&gt;&gt;&gt;&gt; Q_6: &nbsp; &nbsp; &nbsp;BUNDLE<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; We=92ll probably also need some text about BUNDLE.<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;</p>
<p>&gt;&gt;&gt;&gt; yep - but plan is to match up with unified plan<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;Can you be specific about what more you think needs to be said?=
 The createAnswer treatment of BUNDLE is one clear thing that is currently =
missing.</p>
<p>&gt;&gt;&nbsp;</p>
</div>
<p>&gt;&gt; I want to have text covering both&nbsp;the 1st (unique address)=
 and 2nd (shared address) Offer.</p>
</div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
<div>&gt; Acknowledged.&nbsp;<br>
</div>
<blockquote class=3D"gmail_quote" style=3D"margin: 0px 0px 0px 0.8ex; paddi=
ng-left: 1ex; border-left-color: rgb(204, 204, 204); border-left-width: 1px=
; border-left-style: solid;">
<div>
<div style=3D"font-family: Tahoma; font-size: 10pt; direction: ltr;">
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<p>&gt;&nbsp;</p>
<p>&gt;&gt; And, when a PeerConnection is created due to forking (ie you us=
e a separate PeerConnection for each forked leg of a session), I assume
</p>
<p>&gt;&gt; already the 1st Offer for that PeerConnection can have a shared=
 address (as the remote entity has indicated support for BUNDLE). That also=
 needs to be covered.</p>
<p>&gt;&gt;</p>
</div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
<div>&gt; Yes. Which may indicate the need for a setting on a PeerConnectio=
n to use a shared address on the initial offer, since it might have come fr=
om such a fork.&nbsp;</div>
<div>&nbsp;</div>
<div>Excellent.</div>
<div>&nbsp;</div>
<div>Regards,</div>
<div>&nbsp;</div>
<div>Christer</div>
<div>&nbsp;</div>
</div>
</div>
</div>
</div>
</div>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1C4AE733ESESSMB209erics_--

From sergio.garcia.murillo@gmail.com  Sat Sep 28 15:08:44 2013
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Date: Sun, 29 Sep 2013 00:08:41 +0200
From: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
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To: "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: [rtcweb] Camera rotation on mobile phones
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This is a multi-part message in MIME format.
--------------010403020909070001000908
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Hi all,

Currently, in order to keep image property displayed on the receiver 
side if the sender rotates the phone (and the camera), it is needed that 
the sender rotates the image from WxH to HxW so it is sent in upward 
position over the wire. Also, if the receiver is a mobile phone and has 
it rotated, it will have to rotate it again to match the current phone 
orientation.

How about using Coordination of Video Orientation as in 3GPP TS 26.114 
(http://www.3gpp.org/ftp/Specs/html-info/26114.htm)?

Coordination of Video Orientation consists in signalling of the current 
orientation of the image captured on the sender side to the receiver for 
appropriate rendering and displaying. When CVO is succesfully negotiated 
it shall be signalled by the MTSI client. The signalling of the CVO uses 
RTP Header Extensions as specified in IETF RFC 5285 [95]. The one-byte 
form of the header shall be used. CVO information for a 2 bit 
granularity of Rotation (corresponding to urn:3gpp:video-orientation) is 
carried as a byte formatted as follows:

Bit#76543210(LSB)
Definition0000CFR1R0

With the following definitions:

C = Camera: indicates the direction of the camera used for this video 
stream. It can be used by the MTSI client in receiver to e.g. display 
the received video differently depending on the source camera.

     0: Front-facing camera, facing the user. If camera direction is 
unknown by the sending MTSI client in the terminal then this is the 
default value used.

1: Back-facing camera, facing away from the user.

F = Flip: indicates a horizontal (left-right flip) mirror operation on 
the video as sent on the link.

0: No flip operation. If the sending MTSI client in terminal does not 
know if a horizontal mirror operation is necessary, then this is the 
default value used.

1: Horizontal flip operation

R1, R0 = Rotation: indicates the rotation of the video as transmitted on 
the link. The receiver should rotate the video to compensate that 
rotation. E.g. a 90° Counter Clockwise rotation should be compensated by 
the receiver with a 90° Clockwise rotation prior to displaying.

Table 7.2: Rotation signalling for 2 bit granularity

R1

	

R0

	

Rotation of the video as sent on the link

	

Rotation on the receiver before display

0

	

0

	

0° rotation

	

None

0

	

1

	

90° Counter Clockwise (CCW) rotation or 270° Clockwise (CW) rotation

	

90° CW rotation

1

	

0

	

180° CCW rotation or 180° CW rotation

	

180° CW rotation

1

	

1

	

270° CCW rotation or 90° CW rotation

	

90° CCW rotation

CVO information for a higher granularity of Rotation (corresponding to 
urn:3GPP:video-orientation:6) is carried as a byteformatted as follows:**

Bit#76543210(LSB)
DefinitionR5R4R3R2CFR1R0

where Cand Fare as defined above and the bits R5,R4,R3,R2,R1,R0 
represent the Rotation, which indicates the rotation of the video as 
transmitted on the link. Table 7.3 describes the rotation to be applied 
by the receiver based on the rotation bits.

Table 7.3: Rotation signalling for 6 bit granularity

R1

	

R0

	

R5

	

R4

	

R3

	

R2

	

Rotation of the video as sent on the link

	

Rotation on the receiver before display

0

	

0

	

0

	

0

	

0

	

0

	

0° rotation

	

None

0

	

0

	

0

	

0

	

0

	

1

	

(360/64)° Counter Clockwise (CCW) rotation

	

(360/64)° CW rotation

0

	

0

	

0

	

0

	

1

	

0

	

(2*360/64)° CCW rotation

	

(2*360/64)° CW rotation

*.*

	

*.*

	

*.*

	

*.*

	

*.*

	

*.*

	

*.*

	

*.*

*.*

	

*.*

	

*.*

	

*.*

	

*.*

	

*.*

	

*.*

	

*.*

*.*

	

*.*

	

*.*

	

*.*

	

*.*

	

*.*

	

*.*

	

*.*

1

	

1

	

1

	

1

	

1

	

0

	

(62*360/64)° CCW rotation

	

(2*360/64)° CCW rotation

1

	

1

	

1

	

1

	

1

	

1

	

(63*360/64)° CCW rotation

	

(360/64)° CCW rotation

Best regards
Sergio


--------------010403020909070001000908
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<html>
  <head>

    <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
  </head>
  <body text="#000000" bgcolor="#FFFFFF">
    Hi all,<br>
    <br>
    Currently, in order to keep image property displayed on the&nbsp;
    receiver side if the sender rotates the phone (and the camera), it
    is needed that the sender rotates the image from WxH to HxW so it is
    sent in upward position over the wire. Also, if the receiver is a
    mobile phone and has it rotated, it will have to rotate it again to
    match the current phone orientation.<br>
    <br>
    How about using <span style="font-size:10.0pt;font-family:
      &quot;Times New
      Roman&quot;,&quot;serif&quot;;mso-fareast-font-family:&quot;Times
      New Roman&quot;;mso-ansi-language:
      EN-GB;mso-fareast-language:EN-US;mso-bidi-language:AR-SA"
      lang="EN-GB"> Coordination of Video Orientation</span> as in 3GPP
    TS 26.114 (<a class="moz-txt-link-freetext"
      href="http://www.3gpp.org/ftp/Specs/html-info/26114.htm">http://www.3gpp.org/ftp/Specs/html-info/26114.htm</a>)?<span
      lang="EN-GB"> <br>
      <br>
      Coordination of Video Orientation consists in signalling of the
      current orientation of the image captured on the sender side to
      the receiver for appropriate rendering and displaying. When CVO is
      succesfully negotiated it shall be signalled by the MTSI client.
      The signalling of the CVO uses RTP Header Extensions as specified
      in IETF RFC 5285 [95]. The one-byte form of the header shall be
      used. CVO information for a 2 bit granularity of Rotation
      (corresponding to urn:3gpp:video-orientation) is carried as a byte
      formatted as follows:</span>
    <p class="MsoNormal" style="margin-left:14.2pt"><span lang="EN-GB">Bit#<span
          style="mso-tab-count:1"> </span></span><span
        style="font-family: Courier" lang="EN-GB"><span
          style="mso-tab-count:3">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        lang="EN-GB">7<span style="mso-tab-count:1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        style="font-family:Courier" lang="EN-GB"><span
          style="mso-tab-count:1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        lang="EN-GB">6<span style="mso-tab-count:1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        style="font-family:Courier" lang="EN-GB"><span
          style="mso-tab-count:1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        lang="EN-GB">5<span style="mso-tab-count:1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        style="font-family:Courier" lang="EN-GB"><span
          style="mso-tab-count:1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        lang="EN-GB">4<span style="mso-tab-count:1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        style="font-family:Courier" lang="EN-GB"><span
          style="mso-tab-count:1">&nbsp;&nbsp;&nbsp; </span></span><span lang="EN-GB">3<span
          style="mso-tab-count:1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        style="font-family:Courier" lang="EN-GB"><span
          style="mso-tab-count:1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        lang="EN-GB">2<span style="mso-tab-count:1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        style="font-family:Courier" lang="EN-GB"><span
          style="mso-tab-count:1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        lang="EN-GB">1<span style="mso-tab-count:1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        style="font-family:Courier" lang="EN-GB"><span
          style="mso-tab-count:1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        lang="EN-GB">0</span><span style="font-family:Courier"
        lang="EN-GB">(LSB)</span><span lang="EN-GB"><br>
      </span><span style="font-family:Courier" lang="EN-GB">Definition<span
          style="mso-tab-count:1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        style="font-family: Courier;mso-ansi-language:EN-US"
        lang="EN-US">0<span style="mso-tab-count:1">&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        style="font-family:Courier" lang="EN-GB"><span
          style="mso-tab-count:1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        style="font-family:Courier;mso-ansi-language:EN-US" lang="EN-US">0</span><span
        style="font-family:Courier" lang="EN-GB"><span
          style="mso-tab-count:2">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        style="font-family:Courier;mso-ansi-language:EN-US" lang="EN-US">0</span><span
        style="font-family:Courier" lang="EN-GB"><span
          style="mso-tab-count:2">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        style="font-family:Courier;mso-ansi-language:EN-US" lang="EN-US">0</span><span
        style="font-family:Courier" lang="EN-GB"><span
          style="mso-tab-count:2">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>C<span
          style="mso-tab-count:2">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>F<span
          style="mso-tab-count:2">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>R1<span
          style="mso-tab-count:2">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>R0</span></p>
    <p class="MsoNormal"><span style="mso-ansi-language:EN-US"
        lang="EN-US">With the following definitions:<o:p></o:p></span></p>
    <p class="MsoNormal"><span style="mso-ansi-language:EN-US"
        lang="EN-US">C = Camera: <st1:state w:st="on"><st1:place
            w:st="on">ind</st1:place></st1:state>icates the direction of
        the camera used for this video stream. It can be used by the
        MTSI client in receiver to e.g. display the received video
        differently depending on the source camera.<o:p></o:p></span></p>
    <p class="MsoNormal"><span style="mso-ansi-language:EN-US"
        lang="EN-US">&nbsp;&nbsp;&nbsp; 0: Front-facing camera, facing the user. If
        camera direction is unknown by the sending MTSI client in the
        terminal then this is the default value used.<o:p></o:p></span></p>
    <p class="MsoNormal" style="text-indent:14.2pt"><span
        style="mso-ansi-language:EN-US" lang="EN-US">1: Back-facing
        camera, facing away from the user.</span></p>
    <p class="MsoNormal"><span style="mso-ansi-language:EN-US"
        lang="EN-US">F = Flip: indicates a horizontal (left-right flip)
        mirror operation on the video as sent on the link.<o:p></o:p></span></p>
    <p class="MsoNormal"><span style="mso-ansi-language:EN-US"
        lang="EN-US"><span style="mso-tab-count:1">&nbsp;&nbsp;&nbsp; </span>0: No
        flip operation. If the sending MTSI client in terminal does not
        know if a horizontal mirror operation is necessary, then this is
        the default value used.<o:p></o:p></span></p>
    <p class="MsoNormal"><span style="mso-ansi-language:EN-US"
        lang="EN-US"><span style="mso-tab-count:1">&nbsp;&nbsp;&nbsp; </span>1:
        Horizontal flip operation<o:p></o:p></span></p>
    <p class="MsoNormal"><span style="mso-ansi-language:EN-US"
        lang="EN-US">R1, R0 = Rotation: <st1:state w:st="on"><st1:place
            w:st="on">ind</st1:place></st1:state>icates
        the rotation of the video as transmitted on the link. The
        receiver should rotate the video to compensate that rotation.
        E.g. a 90&deg; Counter Clockwise rotation should be compensated by
        the receiver with a 90&deg; Clockwise rotation prior to displaying.<o:p></o:p></span></p>
    <p class="TH"><span lang="EN-GB">Table 7.2: <st1:place w:st="on">Rota</st1:place>tion

        signalling for 2 bit granularity</span></p>
    <div align="center">
      <table class="MsoNormalTable"
        style="border-collapse:collapse;border:none;mso-border-alt:solid
        windowtext .5pt; mso-yfti-tbllook:480;mso-padding-alt:0cm 5.4pt
        0cm 1.4pt;mso-border-insideh: .5pt solid
        windowtext;mso-border-insidev:.5pt solid windowtext" border="1"
        cellpadding="0" cellspacing="0">
        <tbody>
          <tr style="mso-yfti-irow:0;mso-yfti-firstrow:yes">
            <td style="width:24.5pt;border:solid windowtext 1.0pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="top" width="46">
              <p class="TAH"
                style="margin-top:3.0pt;mso-pagination:lines-together;
                tab-stops:70.9pt 5.0cm 212.65pt 10.0cm 354.4pt 15.0cm"><span
                  lang="EN-GB">R1</span></p>
            </td>
            <td style="width:24.55pt;border:solid windowtext 1.0pt;
              border-left:none;mso-border-left-alt:solid windowtext
              .5pt;mso-border-alt: solid windowtext .5pt;padding:0cm
              5.4pt 0cm 1.4pt" valign="top" width="46">
              <p class="TAH"
                style="margin-top:3.0pt;mso-pagination:lines-together;
                tab-stops:70.9pt 5.0cm 212.65pt 10.0cm 354.4pt 15.0cm"><span
                  lang="EN-GB">R0</span></p>
            </td>
            <td style="width:226.7pt;border:solid windowtext 1.0pt;
              border-left:none;mso-border-left-alt:solid windowtext
              .5pt;mso-border-alt: solid windowtext .5pt;padding:0cm
              5.4pt 0cm 1.4pt" valign="top" width="422">
              <p class="TAH"
                style="margin-top:3.0pt;mso-pagination:lines-together;
                tab-stops:70.9pt 5.0cm 212.65pt 10.0cm 354.4pt 15.0cm"><span
                  lang="EN-GB">Rotation of the video as sent on the link</span></p>
            </td>
            <td style="width:213.1pt;border:solid windowtext 1.0pt;
              border-left:none;mso-border-left-alt:solid windowtext
              .5pt;mso-border-alt: solid windowtext .5pt;padding:0cm
              5.4pt 0cm 1.4pt" valign="top" width="397">
              <p class="TAH"
                style="margin-top:3.0pt;mso-pagination:lines-together;
                tab-stops:70.9pt 5.0cm 212.65pt 10.0cm 354.4pt 15.0cm"><span
                  lang="EN-GB">Rotation on the receiver before display</span></p>
            </td>
          </tr>
          <tr style="mso-yfti-irow:1">
            <td style="width:24.5pt;border:solid windowtext 1.0pt;
              border-top:none;mso-border-top-alt:solid windowtext
              .5pt;mso-border-alt:solid windowtext .5pt; padding:0cm
              5.4pt 0cm 1.4pt" valign="top" width="46">
              <p class="TAL" style="margin-top:3.0pt;text-align:center;
                mso-pagination:lines-together;tab-stops:70.9pt 5.0cm
                212.65pt 10.0cm 354.4pt 15.0cm" align="center"><span
                  lang="EN-GB">0</span></p>
            </td>
            <td style="width:24.55pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="top" width="46">
              <p class="TAL" style="margin-top:3.0pt;text-align:center;
                mso-pagination:lines-together;tab-stops:70.9pt 5.0cm
                212.65pt 10.0cm 354.4pt 15.0cm" align="center"><span
                  style="mso-fareast-language:ZH-CN" lang="EN-GB">0<o:p></o:p></span></p>
            </td>
            <td style="width:226.7pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="top" width="422">
              <p class="TAL"
                style="margin-top:3.0pt;mso-pagination:lines-together;
                tab-stops:70.9pt 5.0cm 212.65pt 10.0cm 354.4pt 15.0cm"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">0&deg;
                  rotation</span></p>
            </td>
            <td style="width:213.1pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="top" width="397">
              <p class="TAL"
                style="margin-top:3.0pt;mso-pagination:lines-together;
                tab-stops:70.9pt 5.0cm 212.65pt 10.0cm 354.4pt 15.0cm"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">None<o:p></o:p></span></p>
            </td>
          </tr>
          <tr style="mso-yfti-irow:2">
            <td style="width:24.5pt;border:solid windowtext 1.0pt;
              border-top:none;mso-border-top-alt:solid windowtext
              .5pt;mso-border-alt:solid windowtext .5pt; padding:0cm
              5.4pt 0cm 1.4pt" valign="top" width="46">
              <p class="TAL" style="margin-top:3.0pt;text-align:center;
                mso-pagination:lines-together;tab-stops:70.9pt 5.0cm
                212.65pt 10.0cm 354.4pt 15.0cm" align="center"><span
                  lang="EN-GB">0</span></p>
            </td>
            <td style="width:24.55pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="top" width="46">
              <p class="TAL" style="margin-top:3.0pt;text-align:center;
                mso-pagination:lines-together;tab-stops:70.9pt 5.0cm
                212.65pt 10.0cm 354.4pt 15.0cm" align="center"><span
                  style="mso-fareast-language:ZH-CN" lang="EN-GB">1<o:p></o:p></span></p>
            </td>
            <td style="width:226.7pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="top" width="422">
              <p class="TAL"
                style="margin-top:3.0pt;mso-pagination:lines-together;
                tab-stops:70.9pt 5.0cm 212.65pt 10.0cm 354.4pt 15.0cm"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">90&deg;
                  Counter Clockwise (CCW) rotation or 270&deg; Clockwise
                  (CW) rotation</span></p>
            </td>
            <td style="width:213.1pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="top" width="397">
              <p class="TAL"
                style="margin-top:3.0pt;mso-pagination:lines-together;
                tab-stops:70.9pt 5.0cm 212.65pt 10.0cm 354.4pt 15.0cm"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">90&deg; CW
                  rotation<o:p></o:p></span></p>
            </td>
          </tr>
          <tr style="mso-yfti-irow:3">
            <td style="width:24.5pt;border:solid windowtext 1.0pt;
              border-top:none;mso-border-top-alt:solid windowtext
              .5pt;mso-border-alt:solid windowtext .5pt; padding:0cm
              5.4pt 0cm 1.4pt" valign="top" width="46">
              <p class="TAL" style="margin-top:3.0pt;text-align:center;
                mso-pagination:lines-together;tab-stops:70.9pt 5.0cm
                212.65pt 10.0cm 354.4pt 15.0cm" align="center"><span
                  lang="EN-GB">1</span></p>
            </td>
            <td style="width:24.55pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="top" width="46">
              <p class="TAL" style="margin-top:3.0pt;text-align:center;
                mso-pagination:lines-together;tab-stops:70.9pt 5.0cm
                212.65pt 10.0cm 354.4pt 15.0cm" align="center"><span
                  style="mso-fareast-language:ZH-CN" lang="EN-GB">0<o:p></o:p></span></p>
            </td>
            <td style="width:226.7pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="top" width="422">
              <p class="TAL"
                style="margin-top:3.0pt;mso-pagination:lines-together;
                tab-stops:70.9pt 5.0cm 212.65pt 10.0cm 354.4pt 15.0cm"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">180&deg; CCW
                  rotation or 180&deg; CW rotation</span></p>
            </td>
            <td style="width:213.1pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="top" width="397">
              <p class="TAL"
                style="margin-top:3.0pt;mso-pagination:lines-together;
                tab-stops:70.9pt 5.0cm 212.65pt 10.0cm 354.4pt 15.0cm"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">180&deg; CW
                  rotation<o:p></o:p></span></p>
            </td>
          </tr>
          <tr style="mso-yfti-irow:4;mso-yfti-lastrow:yes">
            <td style="width:24.5pt;border:solid windowtext 1.0pt;
              border-top:none;mso-border-top-alt:solid windowtext
              .5pt;mso-border-alt:solid windowtext .5pt; padding:0cm
              5.4pt 0cm 1.4pt" valign="top" width="46">
              <p class="TAL" style="margin-top:3.0pt;text-align:center;
                mso-pagination:lines-together;tab-stops:70.9pt 5.0cm
                212.65pt 10.0cm 354.4pt 15.0cm" align="center"><span
                  lang="EN-GB">1</span></p>
            </td>
            <td style="width:24.55pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="top" width="46">
              <p class="TAL" style="margin-top:3.0pt;text-align:center;
                mso-pagination:lines-together;tab-stops:70.9pt 5.0cm
                212.65pt 10.0cm 354.4pt 15.0cm" align="center"><span
                  style="mso-fareast-language:ZH-CN" lang="EN-GB">1<o:p></o:p></span></p>
            </td>
            <td style="width:226.7pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="top" width="422">
              <p class="TAL"
                style="margin-top:3.0pt;mso-pagination:lines-together;
                tab-stops:70.9pt 5.0cm 212.65pt 10.0cm 354.4pt 15.0cm"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">270&deg; CCW
                  rotation or 90&deg; CW rotation</span></p>
            </td>
            <td style="width:213.1pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="top" width="397">
              <p class="TAL"
                style="margin-top:3.0pt;mso-pagination:lines-together;
                tab-stops:70.9pt 5.0cm 212.65pt 10.0cm 354.4pt 15.0cm"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">90&deg; CCW
                  rotation<o:p></o:p></span></p>
            </td>
          </tr>
        </tbody>
      </table>
    </div>
    <p class="FP"><span lang="EN-GB"><o:p>&nbsp;</o:p></span></p>
    <p class="MsoNormal"><span lang="EN-GB">CVO information for a higher
        granularity of Rotation (corresponding to
        urn:3GPP:video-orientation:6) is carried as a byte<span
          style="mso-spacerun:yes">&nbsp; </span>formatted as follows:<b><span
            style="color:black"><o:p></o:p></span></b></span></p>
    <p class="MsoNormal" style="margin-left:14.2pt"><span
        style="mso-ansi-language:PT-BR" lang="PT-BR">Bit#<span
          style="mso-tab-count:1"> </span></span><span
        style="font-family:Courier;mso-ansi-language:PT-BR" lang="PT-BR"><span
          style="mso-tab-count:3">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        style="mso-ansi-language:PT-BR" lang="PT-BR">7<span
          style="mso-tab-count:1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        style="font-family:Courier;mso-ansi-language:PT-BR" lang="PT-BR"><span
          style="mso-tab-count:1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        style="mso-ansi-language: PT-BR" lang="PT-BR">6<span
          style="mso-tab-count:1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        style="font-family:Courier;mso-ansi-language:PT-BR" lang="PT-BR"><span
          style="mso-tab-count:1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        style="mso-ansi-language: PT-BR" lang="PT-BR">5<span
          style="mso-tab-count:1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        style="font-family:Courier;mso-ansi-language:PT-BR" lang="PT-BR"><span
          style="mso-tab-count:1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        style="mso-ansi-language: PT-BR" lang="PT-BR">4<span
          style="mso-tab-count:1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        style="font-family:Courier;mso-ansi-language:PT-BR" lang="PT-BR"><span
          style="mso-tab-count:1">&nbsp;&nbsp;&nbsp; </span></span><span
        style="mso-ansi-language: PT-BR" lang="PT-BR">3<span
          style="mso-tab-count:1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        style="font-family:Courier;mso-ansi-language:PT-BR" lang="PT-BR"><span
          style="mso-tab-count:1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        style="mso-ansi-language: PT-BR" lang="PT-BR">2<span
          style="mso-tab-count:1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        style="font-family:Courier;mso-ansi-language:PT-BR" lang="PT-BR"><span
          style="mso-tab-count:1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        style="mso-ansi-language: PT-BR" lang="PT-BR">1<span
          style="mso-tab-count:1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        style="font-family:Courier;mso-ansi-language:PT-BR" lang="PT-BR"><span
          style="mso-tab-count:1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        style="mso-ansi-language: PT-BR" lang="PT-BR">0</span><span
        style="font-family:Courier;mso-ansi-language: PT-BR"
        lang="PT-BR">(LSB)</span><span style="mso-ansi-language:PT-BR"
        lang="PT-BR"><br>
      </span><span style="font-family:Courier;mso-ansi-language:PT-BR"
        lang="PT-BR">Definition<span style="mso-tab-count:1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        style="font-family: Courier;mso-ansi-language:EN-US"
        lang="EN-US">R5<span style="mso-tab-count:1">&nbsp;&nbsp;&nbsp; </span></span><span
        style="font-family:Courier;mso-ansi-language:PT-BR" lang="PT-BR"><span
          style="mso-tab-count:1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        style="font-family: Courier;mso-ansi-language:EN-US"
        lang="EN-US">R4</span><span style="font-family:
        Courier;mso-ansi-language:PT-BR" lang="PT-BR"><span
          style="mso-tab-count:1">&nbsp;&nbsp;&nbsp; </span><span
          style="mso-tab-count:1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><span
        style="font-family: Courier;mso-ansi-language:EN-US"
        lang="EN-US">R3</span><span style="font-family:
        Courier;mso-ansi-language:PT-BR" lang="PT-BR"><span
          style="mso-tab-count:1">&nbsp;&nbsp;&nbsp; </span><span
          style="mso-tab-count:1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>R2<span
          style="mso-tab-count:2">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>C<span
          style="mso-tab-count:2">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>F<span
          style="mso-tab-count:2">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>R1<span
          style="mso-tab-count:2">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>R0</span><span
        style="mso-ansi-language:PT-BR" lang="PT-BR"><o:p></o:p></span></p>
    <p class="MsoNormal"><span lang="EN-GB">where </span><span
        style="font-family:&quot;Courier New&quot;" lang="EN-GB">C</span><span
        lang="EN-GB"> and </span><span style="font-family:&quot;Courier
        New&quot;" lang="EN-GB">F</span><span lang="EN-GB"> are as
        defined above and the bits R5,R4,R3,R2,R1,R0 represent the
        Rotation, which indicates the rotation of the video as
        transmitted on the link. Table 7.3 describes the rotation to be
        applied by the receiver based on the rotation bits.</span></p>
    <p class="TH"><span lang="EN-GB">Table 7.3: Rotation signalling for
        6 bit granularity</span></p>
    <div align="center">
      <table class="MsoNormalTable"
        style="width:337.7pt;border-collapse:collapse;border:none;mso-border-alt:solid

        windowtext .5pt; mso-yfti-tbllook:480;mso-padding-alt:0cm 5.4pt
        0cm 1.4pt;mso-border-insideh: .5pt solid
        windowtext;mso-border-insidev:.5pt solid windowtext" width="628"
        border="1" cellpadding="0" cellspacing="0">
        <tbody>
          <tr
            style="mso-yfti-irow:0;mso-yfti-firstrow:yes;height:34.15pt">
            <td style="width:21.05pt;border:solid windowtext 1.0pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt;height:34.15pt" valign="top" width="39">
              <p class="TAH"
                style="margin-top:3.0pt;mso-pagination:lines-together;
                tab-stops:70.9pt 5.0cm 212.65pt 10.0cm 354.4pt 15.0cm"><span
                  lang="EN-GB">R1</span></p>
            </td>
            <td style="width:21.05pt;border:solid windowtext 1.0pt;
              border-left:none;mso-border-left-alt:solid windowtext
              .5pt;mso-border-alt: solid windowtext .5pt;padding:0cm
              5.4pt 0cm 1.4pt;height:34.15pt" valign="top" width="39">
              <p class="TAH"
                style="margin-top:3.0pt;mso-pagination:lines-together;
                tab-stops:70.9pt 5.0cm 212.65pt 10.0cm 354.4pt 15.0cm"><span
                  lang="EN-GB">R0</span></p>
            </td>
            <td style="width:21.0pt;border:solid windowtext 1.0pt;
              border-left:none;mso-border-left-alt:solid windowtext
              .5pt;mso-border-alt: solid windowtext .5pt;padding:0cm
              5.4pt 0cm 1.4pt;height:34.15pt" valign="top" width="39">
              <p class="TAH"
                style="margin-top:3.0pt;mso-pagination:lines-together;
                tab-stops:70.9pt 5.0cm 212.65pt 10.0cm 354.4pt 15.0cm"><span
                  lang="EN-GB">R5</span></p>
            </td>
            <td style="width:21.05pt;border:solid windowtext 1.0pt;
              border-left:none;mso-border-left-alt:solid windowtext
              .5pt;mso-border-alt: solid windowtext .5pt;padding:0cm
              5.4pt 0cm 1.4pt;height:34.15pt" valign="top" width="39">
              <p class="TAH"
                style="margin-top:3.0pt;mso-pagination:lines-together;
                tab-stops:70.9pt 5.0cm 212.65pt 10.0cm 354.4pt 15.0cm"><span
                  lang="EN-GB">R4</span></p>
            </td>
            <td style="width:21.05pt;border:solid windowtext 1.0pt;
              border-left:none;mso-border-left-alt:solid windowtext
              .5pt;mso-border-alt: solid windowtext .5pt;padding:0cm
              5.4pt 0cm 1.4pt;height:34.15pt" valign="top" width="39">
              <p class="TAH"
                style="margin-top:3.0pt;mso-pagination:lines-together;
                tab-stops:70.9pt 5.0cm 212.65pt 10.0cm 354.4pt 15.0cm"><span
                  lang="EN-GB">R3</span></p>
            </td>
            <td style="width:21.0pt;border:solid windowtext 1.0pt;
              border-left:none;mso-border-left-alt:solid windowtext
              .5pt;mso-border-alt: solid windowtext .5pt;padding:0cm
              5.4pt 0cm 1.4pt;height:34.15pt" valign="top" width="39">
              <p class="TAH"
                style="margin-top:3.0pt;mso-pagination:lines-together;
                tab-stops:70.9pt 5.0cm 212.65pt 10.0cm 354.4pt 15.0cm"><span
                  lang="EN-GB">R2</span></p>
            </td>
            <td style="width:124.8pt;border:solid windowtext 1.0pt;
              border-left:none;mso-border-left-alt:solid windowtext
              .5pt;mso-border-alt: solid windowtext .5pt;padding:0cm
              5.4pt 0cm 1.4pt;height:34.15pt" valign="top" width="232">
              <p class="TAH"
                style="margin-top:3.0pt;mso-pagination:lines-together;
                tab-stops:70.9pt 5.0cm 212.65pt 10.0cm 354.4pt 15.0cm"><span
                  lang="EN-GB">Rotation of the video as sent on the link</span></p>
            </td>
            <td style="width:86.7pt;border:solid windowtext 1.0pt;
              border-left:none;mso-border-left-alt:solid windowtext
              .5pt;mso-border-alt: solid windowtext .5pt;padding:0cm
              5.4pt 0cm 1.4pt;height:34.15pt" valign="top" width="161">
              <p class="TAH"
                style="margin-top:3.0pt;mso-pagination:lines-together;
                tab-stops:70.9pt 5.0cm 212.65pt 10.0cm 354.4pt 15.0cm"><span
                  lang="EN-GB">Rotation on the receiver before display</span></p>
            </td>
          </tr>
          <tr style="mso-yfti-irow:1">
            <td style="width:21.05pt;border:solid windowtext 1.0pt;
              border-top:none;mso-border-top-alt:solid windowtext
              .5pt;mso-border-alt:solid windowtext .5pt; padding:0cm
              5.4pt 0cm 1.4pt" valign="bottom" width="39">
              <p class="TAL" style="margin-top:3.0pt;text-align:center;
                mso-pagination:lines-together;tab-stops:70.9pt 5.0cm
                212.65pt 10.0cm 354.4pt 15.0cm" align="center"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">0<o:p></o:p></span></p>
            </td>
            <td style="width:21.05pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="bottom" width="39">
              <p class="TAL" style="margin-top:3.0pt;text-align:center;
                mso-pagination:lines-together;tab-stops:70.9pt 5.0cm
                212.65pt 10.0cm 354.4pt 15.0cm" align="center"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">0<o:p></o:p></span></p>
            </td>
            <td style="width:21.0pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="bottom" width="39">
              <p class="TAL" style="margin-top:3.0pt;text-align:center;
                mso-pagination:lines-together;tab-stops:70.9pt 5.0cm
                212.65pt 10.0cm 354.4pt 15.0cm" align="center"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">0<o:p></o:p></span></p>
            </td>
            <td style="width:21.05pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="bottom" width="39">
              <p class="TAL" style="margin-top:3.0pt;text-align:center;
                mso-pagination:lines-together;tab-stops:70.9pt 5.0cm
                212.65pt 10.0cm 354.4pt 15.0cm" align="center"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">0<o:p></o:p></span></p>
            </td>
            <td style="width:21.05pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="bottom" width="39">
              <p class="TAL" style="margin-top:3.0pt;text-align:center;
                mso-pagination:lines-together;tab-stops:70.9pt 5.0cm
                212.65pt 10.0cm 354.4pt 15.0cm" align="center"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">0<o:p></o:p></span></p>
            </td>
            <td style="width:21.0pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="bottom" width="39">
              <p class="TAL" style="margin-top:3.0pt;text-align:center;
                mso-pagination:lines-together;tab-stops:70.9pt 5.0cm
                212.65pt 10.0cm 354.4pt 15.0cm" align="center"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">0<o:p></o:p></span></p>
            </td>
            <td style="width:124.8pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="top" width="232">
              <p class="TAL"
                style="margin-top:3.0pt;mso-pagination:lines-together;
                tab-stops:70.9pt 5.0cm 212.65pt 10.0cm 354.4pt 15.0cm"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">0&deg;
                  rotation</span></p>
            </td>
            <td style="width:86.7pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="top" width="161">
              <p class="TAL"
                style="margin-top:3.0pt;mso-pagination:lines-together;
                tab-stops:70.9pt 5.0cm 212.65pt 10.0cm 354.4pt 15.0cm"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">None<o:p></o:p></span></p>
            </td>
          </tr>
          <tr style="mso-yfti-irow:2;height:15.7pt">
            <td style="width:21.05pt;border:solid windowtext 1.0pt;
              border-top:none;mso-border-top-alt:solid windowtext
              .5pt;mso-border-alt:solid windowtext .5pt; padding:0cm
              5.4pt 0cm 1.4pt;height:15.7pt" valign="bottom" width="39">
              <p class="TAL" style="margin-top:3.0pt;text-align:center;
                mso-pagination:lines-together;tab-stops:70.9pt 5.0cm
                212.65pt 10.0cm 354.4pt 15.0cm" align="center"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">0<o:p></o:p></span></p>
            </td>
            <td style="width:21.05pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt;height:15.7pt" valign="bottom" width="39">
              <p class="TAL" style="margin-top:3.0pt;text-align:center;
                mso-pagination:lines-together;tab-stops:70.9pt 5.0cm
                212.65pt 10.0cm 354.4pt 15.0cm" align="center"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">0<o:p></o:p></span></p>
            </td>
            <td style="width:21.0pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt;height:15.7pt" valign="bottom" width="39">
              <p class="TAL" style="margin-top:3.0pt;text-align:center;
                mso-pagination:lines-together;tab-stops:70.9pt 5.0cm
                212.65pt 10.0cm 354.4pt 15.0cm" align="center"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">0<o:p></o:p></span></p>
            </td>
            <td style="width:21.05pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt;height:15.7pt" valign="bottom" width="39">
              <p class="TAL" style="margin-top:3.0pt;margin-right:0cm;
                margin-bottom:0cm;margin-left:-26.5pt;margin-bottom:.0001pt;text-align:center;


                text-indent:25.95pt;mso-pagination:lines-together;tab-stops:70.9pt

                5.0cm 212.65pt 10.0cm 354.4pt 15.0cm" align="center"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">0<o:p></o:p></span></p>
            </td>
            <td style="width:21.05pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt;height:15.7pt" valign="bottom" width="39">
              <p class="TAL" style="margin-top:3.0pt;text-align:center;
                mso-pagination:lines-together;tab-stops:70.9pt 5.0cm
                212.65pt 10.0cm 354.4pt 15.0cm" align="center"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">0<o:p></o:p></span></p>
            </td>
            <td style="width:21.0pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt;height:15.7pt" valign="bottom" width="39">
              <p class="TAL" style="margin-top:3.0pt;text-align:center;
                mso-pagination:lines-together;tab-stops:70.9pt 5.0cm
                212.65pt 10.0cm 354.4pt 15.0cm" align="center"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">1<o:p></o:p></span></p>
            </td>
            <td style="width:124.8pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt;height:15.7pt" valign="top" width="232">
              <p class="TAL"
                style="margin-top:3.0pt;mso-pagination:lines-together;
                tab-stops:70.9pt 5.0cm 212.65pt 10.0cm 354.4pt 15.0cm"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">(360/64)&deg;
                  Counter Clockwise (CCW) rotation</span></p>
            </td>
            <td style="width:86.7pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt;height:15.7pt" valign="top" width="161">
              <p class="TAL"
                style="margin-top:3.0pt;mso-pagination:lines-together;
                tab-stops:70.9pt 5.0cm 212.65pt 10.0cm 354.4pt 15.0cm"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">(360/64)&deg;
                  CW rotation<o:p></o:p></span></p>
            </td>
          </tr>
          <tr style="mso-yfti-irow:3">
            <td style="width:21.05pt;border:solid windowtext 1.0pt;
              border-top:none;mso-border-top-alt:solid windowtext
              .5pt;mso-border-alt:solid windowtext .5pt; padding:0cm
              5.4pt 0cm 1.4pt" valign="bottom" width="39">
              <p class="TAL" style="margin-top:3.0pt;text-align:center;
                mso-pagination:lines-together;tab-stops:70.9pt 5.0cm
                212.65pt 10.0cm 354.4pt 15.0cm" align="center"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">0<o:p></o:p></span></p>
            </td>
            <td style="width:21.05pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="bottom" width="39">
              <p class="TAL" style="margin-top:3.0pt;text-align:center;
                mso-pagination:lines-together;tab-stops:70.9pt 5.0cm
                212.65pt 10.0cm 354.4pt 15.0cm" align="center"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">0<o:p></o:p></span></p>
            </td>
            <td style="width:21.0pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="bottom" width="39">
              <p class="TAL" style="margin-top:3.0pt;text-align:center;
                mso-pagination:lines-together;tab-stops:70.9pt 5.0cm
                212.65pt 10.0cm 354.4pt 15.0cm" align="center"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">0<o:p></o:p></span></p>
            </td>
            <td style="width:21.05pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="bottom" width="39">
              <p class="TAL" style="margin-top:3.0pt;text-align:center;
                mso-pagination:lines-together;tab-stops:70.9pt 5.0cm
                212.65pt 10.0cm 354.4pt 15.0cm" align="center"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">0<o:p></o:p></span></p>
            </td>
            <td style="width:21.05pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="bottom" width="39">
              <p class="TAL" style="margin-top:3.0pt;text-align:center;
                mso-pagination:lines-together;tab-stops:70.9pt 5.0cm
                212.65pt 10.0cm 354.4pt 15.0cm" align="center"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">1<o:p></o:p></span></p>
            </td>
            <td style="width:21.0pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="bottom" width="39">
              <p class="TAL" style="margin-top:3.0pt;text-align:center;
                mso-pagination:lines-together;tab-stops:70.9pt 5.0cm
                212.65pt 10.0cm 354.4pt 15.0cm" align="center"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">0<o:p></o:p></span></p>
            </td>
            <td style="width:124.8pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="top" width="232">
              <p class="TAL"
                style="margin-top:3.0pt;mso-pagination:lines-together;
                tab-stops:70.9pt 5.0cm 212.65pt 10.0cm 354.4pt 15.0cm"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">(2*360/64)&deg;
                  CCW rotation</span></p>
            </td>
            <td style="width:86.7pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="top" width="161">
              <p class="TAL"
                style="margin-top:3.0pt;mso-pagination:lines-together;
                tab-stops:70.9pt 5.0cm 212.65pt 10.0cm 354.4pt 15.0cm"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">(2*360/64)&deg;
                  CW rotation<o:p></o:p></span></p>
            </td>
          </tr>
          <tr style="mso-yfti-irow:4">
            <td style="width:21.05pt;border:solid windowtext 1.0pt;
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              mso-border-top-alt:solid windowtext
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                  style="mso-ansi-language:EN-US" lang="EN-US">0<o:p></o:p></span></p>
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              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="top" width="232">
              <p class="TAL"
                style="margin-top:3.0pt;mso-pagination:lines-together;
                tab-stops:70.9pt 5.0cm 212.65pt 10.0cm 354.4pt 15.0cm"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">(62*360/64)&deg;
                  CCW rotation<o:p></o:p></span></p>
            </td>
            <td style="width:86.7pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="top" width="161">
              <p class="TAL"
                style="margin-top:3.0pt;mso-pagination:lines-together;
                tab-stops:70.9pt 5.0cm 212.65pt 10.0cm 354.4pt 15.0cm"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">(2*360/64)&deg;
                  CCW rotation<o:p></o:p></span></p>
            </td>
          </tr>
          <tr style="mso-yfti-irow:8;mso-yfti-lastrow:yes">
            <td style="width:21.05pt;border:solid windowtext 1.0pt;
              border-top:none;mso-border-top-alt:solid windowtext
              .5pt;mso-border-alt:solid windowtext .5pt; padding:0cm
              5.4pt 0cm 1.4pt" valign="bottom" width="39">
              <p class="TAL" style="margin-top:3.0pt;text-align:center;
                mso-pagination:lines-together;tab-stops:70.9pt 5.0cm
                212.65pt 10.0cm 354.4pt 15.0cm" align="center"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">1<o:p></o:p></span></p>
            </td>
            <td style="width:21.05pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="bottom" width="39">
              <p class="TAL" style="margin-top:3.0pt;text-align:center;
                mso-pagination:lines-together;tab-stops:70.9pt 5.0cm
                212.65pt 10.0cm 354.4pt 15.0cm" align="center"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">1<o:p></o:p></span></p>
            </td>
            <td style="width:21.0pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="bottom" width="39">
              <p class="TAL" style="margin-top:3.0pt;text-align:center;
                mso-pagination:lines-together;tab-stops:70.9pt 5.0cm
                212.65pt 10.0cm 354.4pt 15.0cm" align="center"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">1<o:p></o:p></span></p>
            </td>
            <td style="width:21.05pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="bottom" width="39">
              <p class="TAL" style="margin-top:3.0pt;text-align:center;
                mso-pagination:lines-together;tab-stops:70.9pt 5.0cm
                212.65pt 10.0cm 354.4pt 15.0cm" align="center"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">1<o:p></o:p></span></p>
            </td>
            <td style="width:21.05pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="bottom" width="39">
              <p class="TAL" style="margin-top:3.0pt;text-align:center;
                mso-pagination:lines-together;tab-stops:70.9pt 5.0cm
                212.65pt 10.0cm 354.4pt 15.0cm" align="center"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">1<o:p></o:p></span></p>
            </td>
            <td style="width:21.0pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="bottom" width="39">
              <p class="TAL" style="margin-top:3.0pt;text-align:center;
                mso-pagination:lines-together;tab-stops:70.9pt 5.0cm
                212.65pt 10.0cm 354.4pt 15.0cm" align="center"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">1<o:p></o:p></span></p>
            </td>
            <td style="width:124.8pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="top" width="232">
              <p class="TAL"
                style="margin-top:3.0pt;mso-pagination:lines-together;
                tab-stops:70.9pt 5.0cm 212.65pt 10.0cm 354.4pt 15.0cm"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">(63*360/64)&deg;
                  CCW rotation<o:p></o:p></span></p>
            </td>
            <td style="width:86.7pt;border-top:none;border-left:
              none;border-bottom:solid windowtext
              1.0pt;border-right:solid windowtext 1.0pt;
              mso-border-top-alt:solid windowtext
              .5pt;mso-border-left-alt:solid windowtext .5pt;
              mso-border-alt:solid windowtext .5pt;padding:0cm 5.4pt 0cm
              1.4pt" valign="top" width="161">
              <p class="TAL"
                style="margin-top:3.0pt;mso-pagination:lines-together;
                tab-stops:70.9pt 5.0cm 212.65pt 10.0cm 354.4pt 15.0cm"><span
                  style="mso-ansi-language:EN-US" lang="EN-US">(360/64)&deg;
                  CCW rotation<o:p></o:p></span></p>
            </td>
          </tr>
        </tbody>
      </table>
    </div>
    <p class="FP"><span lang="EN-GB"><o:p>&nbsp;</o:p></span></p>
    Best regards<br>
    Sergio<br>
    <br>
  </body>
</html>

--------------010403020909070001000908--

From martin.thomson@gmail.com  Sat Sep 28 15:53:27 2013
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Date: Sat, 28 Sep 2013 15:53:12 -0700
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From: Martin Thomson <martin.thomson@gmail.com>
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Subject: Re: [rtcweb] TURN server address via DHCP, WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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On Sep 26, 2013 8:46 PM, "Harald Alvestrand" <harald@alvestrand.no> wrote:
> So far, neither the POSIX standard nor any OS vendor has offered a
generic facility to access information made available in DHCP packets.

Actually, Windows does provide an API for this exact purpose, it's just not
particularly well-suited to what geopriv needed, i.e. getting changing
values. It is good enough for the purpose we are taking about. That isn't
much help for other operating systems, I'll concede.

Applications can send their own DHCP requests...maybe. You need to listen
on port 69 (from memory) for which you have to have the usual special
dispensation, and you have to compete with the OS DHCP agent. It's not
pretty either way.

--001a11c38c82bca43404e7797707
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<p dir=3D"ltr"><br>
On Sep 26, 2013 8:46 PM, &quot;Harald Alvestrand&quot; &lt;<a href=3D"mailt=
o:harald@alvestrand.no">harald@alvestrand.no</a>&gt; wrote:<br>
&gt; So far, neither the POSIX standard nor any OS vendor has offered a gen=
eric facility to access information made available in DHCP packets.</p>
<p dir=3D"ltr">Actually, Windows does provide an API for this exact purpose=
, it&#39;s just not particularly well-suited to what geopriv needed, i.e. g=
etting changing values. It is good enough for the purpose we are taking abo=
ut. That isn&#39;t much help for other operating systems, I&#39;ll concede.=
</p>

<p dir=3D"ltr">Applications can send their own DHCP requests...maybe. You n=
eed to listen on port 69 (from memory) for which you have to have the usual=
 special dispensation, and you have to compete with the OS DHCP agent. It&#=
39;s not pretty either way.<br>

</p>

--001a11c38c82bca43404e7797707--

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Subject: Re: [rtcweb] TURN server address via DHCP, WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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--_c98599e2-c4f1-43c2-85ed-6b1c12957b25_
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On Sep 26=2C 2013 8:46 PM=2C "Harald Alvestrand" <harald@alvestrand.no> wro=
te:
"So far=2C neither the POSIX standard nor any OS vendor has offered a gener=
ic facility to access information made available in DHCP packets."
[BA] The Windows DHCP client API does provide this: http://msdn.microsoft.c=
om/en-us/library/windows/desktop/aa363351(v=3Dvs.85).aspx
In particular=2C the SendParams argument to the DhcpRequestParams function =
can be used to request a particular parameter (e.g. TURN server address)=2C=
 which will then be returned in the RecdParams variable. =20
Nevertheless=2C I still think that using DHCP to configure the TURN server =
address in a browser isn't a good idea.  For one thing=2C since DHCP is eff=
ectively unsecured=2C this mechanism could be used by a rogue DHCP server t=
o force traffic to a rogue turnserver.   Great for surveillance!
 		 	   		  =

--_c98599e2-c4f1-43c2-85ed-6b1c12957b25_
Content-Type: text/html; charset="iso-8859-1"
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<html>
<head>
<style><!--
.hmmessage P
{
margin:0px=3B
padding:0px
}
body.hmmessage
{
font-size: 12pt=3B
font-family:Calibri
}
--></style></head>
<body class=3D'hmmessage'><div dir=3D'ltr'><p dir=3D"ltr">On Sep 26=2C 2013=
 8:46 PM=2C "Harald Alvestrand" &lt=3B<a href=3D"mailto:harald@alvestrand.n=
o">harald@alvestrand.no</a>&gt=3B wrote:<br>"So far=2C neither the POSIX st=
andard nor any OS vendor has offered a generic facility to access informati=
on made available in DHCP packets."</p><p dir=3D"ltr"><br></p><p dir=3D"ltr=
">[BA] The Windows DHCP client API does provide this:&nbsp=3B</p><p dir=3D"=
ltr"><a href=3D"http://msdn.microsoft.com/en-us/library/windows/desktop/aa3=
63351(v=3Dvs.85).aspx" target=3D"_blank">http://msdn.microsoft.com/en-us/li=
brary/windows/desktop/aa363351(v=3Dvs.85).aspx</a></p><p dir=3D"ltr"><br></=
p><p dir=3D"ltr">In particular=2C the SendParams argument to the DhcpReques=
tParams function&nbsp=3B<span style=3D"font-size: 12pt=3B">can be used to r=
equest a particular parameter (e.g. TURN server address)=2C which will then=
 be returned in the RecdParams variable. &nbsp=3B</span></p><p dir=3D"ltr">=
<span style=3D"font-size: 12pt=3B"><br></span></p><p dir=3D"ltr"><span styl=
e=3D"font-size: 12pt=3B">Nevertheless=2C I still think that using DHCP to c=
onfigure the TURN server address in a browser isn't a good idea. &nbsp=3B</=
span><span style=3D"font-size: 12pt=3B">For one thing=2C since DHCP is effe=
ctively unsecured=2C this mechanism could be used by a rogue DHCP server to=
 force traffic to a rogue turnserver. &nbsp=3B Great for surveillance!</spa=
n></p><p dir=3D"ltr"><br></p> 		 	   		  </div></body>
</html>=

--_c98599e2-c4f1-43c2-85ed-6b1c12957b25_--

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From: "Jeremy Laurenson (jlaurens)" <jlaurens@cisco.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] TURN server address via DHCP, WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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Subject: Re: [rtcweb] TURN server address via DHCP, WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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Why not have this at least optionally specified in JavaScript in the browse=
r? This way those who "own" the apps have control and aren't relying on cor=
rect implementation of others (necessarily)

On Sep 28, 2013, at 8:41 PM, "Bernard Aboba" <bernard_aboba@hotmail.com<mai=
lto:bernard_aboba@hotmail.com>> wrote:


On Sep 26, 2013 8:46 PM, "Harald Alvestrand" <harald@alvestrand.no<mailto:h=
arald@alvestrand.no>> wrote:
"So far, neither the POSIX standard nor any OS vendor has offered a generic=
 facility to access information made available in DHCP packets."


[BA] The Windows DHCP client API does provide this:

http://msdn.microsoft.com/en-us/library/windows/desktop/aa363351(v=3Dvs.85)=
.aspx


In particular, the SendParams argument to the DhcpRequestParams function ca=
n be used to request a particular parameter (e.g. TURN server address), whi=
ch will then be returned in the RecdParams variable.


Nevertheless, I still think that using DHCP to configure the TURN server ad=
dress in a browser isn't a good idea.  For one thing, since DHCP is effecti=
vely unsecured, this mechanism could be used by a rogue DHCP server to forc=
e traffic to a rogue turnserver.   Great for surveillance!


_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org>
https://www.ietf.org/mailman/listinfo/rtcweb

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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body dir=3D"auto">
<div>Why not have this at least optionally specified in JavaScript in the b=
rowser? This way those who &quot;own&quot; the apps have control and aren't=
 relying on correct implementation of others (necessarily)<br>
</div>
<div><br>
On Sep 28, 2013, at 8:41 PM, &quot;Bernard Aboba&quot; &lt;<a href=3D"mailt=
o:bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a>&gt; wrote:<br>
<br>
</div>
<blockquote type=3D"cite">
<div><style><!--
.hmmessage P
{
margin:0px;
padding:0px
}
body.hmmessage
{
font-size: 12pt;
font-family:Calibri
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--></style>
<div dir=3D"ltr">
<p dir=3D"ltr">On Sep 26, 2013 8:46 PM, &quot;Harald Alvestrand&quot; &lt;<=
a href=3D"mailto:harald@alvestrand.no">harald@alvestrand.no</a>&gt; wrote:<=
br>
&quot;So far, neither the POSIX standard nor any OS vendor has offered a ge=
neric facility to access information made available in DHCP packets.&quot;<=
/p>
<p dir=3D"ltr"><br>
</p>
<p dir=3D"ltr">[BA] The Windows DHCP client API does provide this:&nbsp;</p=
>
<p dir=3D"ltr"><a href=3D"http://msdn.microsoft.com/en-us/library/windows/d=
esktop/aa363351(v=3Dvs.85).aspx" target=3D"_blank">http://msdn.microsoft.co=
m/en-us/library/windows/desktop/aa363351(v=3Dvs.85).aspx</a></p>
<p dir=3D"ltr"><br>
</p>
<p dir=3D"ltr">In particular, the SendParams argument to the DhcpRequestPar=
ams function&nbsp;<span style=3D"font-size: 12pt;">can be used to request a=
 particular parameter (e.g. TURN server address), which will then be return=
ed in the RecdParams variable. &nbsp;</span></p>
<p dir=3D"ltr"><span style=3D"font-size: 12pt;"><br>
</span></p>
<p dir=3D"ltr"><span style=3D"font-size: 12pt;">Nevertheless, I still think=
 that using DHCP to configure the TURN server address in a browser isn't a =
good idea. &nbsp;</span><span style=3D"font-size: 12pt;">For one thing, sin=
ce DHCP is effectively unsecured, this mechanism
 could be used by a rogue DHCP server to force traffic to a rogue turnserve=
r. &nbsp; Great for surveillance!</span></p>
<p dir=3D"ltr"><br>
</p>
</div>
</div>
</blockquote>
<blockquote type=3D"cite">
<div><span>_______________________________________________</span><br>
<span>rtcweb mailing list</span><br>
<span><a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br>
<span><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.=
ietf.org/mailman/listinfo/rtcweb</a></span><br>
</div>
</blockquote>
</body>
</html>

--_000_3CD8C0AB3E694E959ED7198AA4568A25ciscocom_--

From christer.holmberg@ericsson.com  Mon Sep 30 01:08:32 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: JSEP: Order of m- lines in multiple PeerConnections
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Date: Mon, 30 Sep 2013 08:07:47 +0000
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Subject: [rtcweb] JSEP: Order of m- lines in multiple PeerConnections
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Hi,

JSEP talks about the usage of multiple PeerConnection to support forking, i=
.e. for each new forked leg (SIP: early dialog) a new PeerConnection is cre=
ated.

As has been indicated, as each new PeerConnection will have its own set of =
address properties, ICE properties etc, so a new Offer will have to be crea=
ted and sent to inform the remote about the new properties.

So far so I good.

I also assume that the same camera/mic/etc sources are connection to each P=
eerConnection, so the number of m- lines in the Offer of the new PeerConnec=
tion should be the same.

However, according the 3264, the ORDER of the m- lines also need to be kept=
 the same.

So, my question is: how can I ensure that the order of the m- lines in an O=
ffer for a new PeerConnection is the same as in an Offer for an old PeerCon=
nection?

Regards,

Christer


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<p class=3D"MsoNormal"><span lang=3D"FI">Hi,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"FI"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal">JSEP talks about the usage of multiple PeerConnectio=
n to support forking, i.e. for each new forked leg (SIP: early dialog) a ne=
w PeerConnection is created.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">As has been indicated, as each new PeerConnection wi=
ll have its own set of address properties, ICE properties etc, so a new Off=
er will have to be created and sent to inform the remote about the new prop=
erties.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">So far so I good.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">I also assume that the same camera/mic/etc sources a=
re connection to each PeerConnection, so the number of m- lines in the Offe=
r of the new PeerConnection should be the same.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">However, according the 3264, the <b>ORDER of the m- =
lines</b> also need to be kept the same.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">So, my question is: how can I ensure that the order =
of the m- lines in an Offer for a new PeerConnection is the same as in an O=
ffer for an old PeerConnection?<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Christer<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</body>
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References: <7594FB04B1934943A5C02806D1A2204B1C4A77DB@ESESSMB209.ericsson.se> <C5E08FE080ACFD4DAE31E4BDBF944EB1166BED56@xmb-aln-x02.cisco.com> <CAOJ7v-35pjc-w_vgNCxE8dwfp9jh_cyGHR6_Cun8WAX4iCFNMQ@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1C4AE0DB@ESESSMB209.ericsson.se> <CAOJ7v-1MSWBLVf4WNrxoapHpp6Fe2UaR3JWyJ=6+cFvvrQ2MHQ@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1C4AE733@ESESSMB209.ericsson.se>
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Cc: "Cullen Jennings \(fluffy\)" <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] JSEP-04: Some comments on Section 5.2.1. and 5.2.2 (19th september)
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Can someone explain why the proto field is 'RTP/SAVPF' rather than
'UDP/TLS/RTP/SAVPF'?
We are using DTLS-SRTP after all.



On 28 September 2013 07:52, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>  Hi,
>
>
>
>>     >>>>> Q_3:      RTP
>> >>>>>
>> >>>>> The text says:
>> >>>>>
>> >>>>> =93The <proto> field MUST be set to "RTP/SAVPF". =93
>> >>>>>
>> >>>>> But, that of course only applies to RTP based streams (not the dat=
a
>> channel). Also, in general, it needs to be clear what information
>> >>>>> needs to be in every m- line, and what information is protocol
>> specific. I would suggest to have a =93General=94 sub-section, a =93RTP=
=94
>> sub-section, etc.
>> >>>>>
>> >>>>>
>> >>>> Yep - I think this should be offers are created with RTP/SAVPF for
>> audio and video and system can reeve offers with SAVPF or SAVP
>> >>>
>> >>> Currently the document specifies what to do with media lines, with
>> later discussion of handling the SCTP line. Do you think the current
>> information is not sufficiently descriptive?
>> >>
>>  >> My point was that we need to be more clear on what is generic, what
>> is RTP, and what is SCTP. But, as you say SCTP text is still to be added=
,
>> that clarification can be done when the SCTP text is added.
>>
>
>  >Sorry I was unclear - there is SCTP-specific text in the current
> document.
>
> Ok, so while the content itself might be ok, at some point I'd like to
> have separate sub-sections for the generic stuff, the RTP specicif stuff,
> and the SCTP specific stuff.
>
>
>
>>     >>>>> Q_6:      BUNDLE
>> >>>>>
>> >>>>> We=92ll probably also need some text about BUNDLE.
>> >>>>>
>> >>>>
>>
>> >>>> yep - but plan is to match up with unified plan
>> >>>>
>> >>>Can you be specific about what more you think needs to be said? The
>> createAnswer treatment of BUNDLE is one clear thing that is currently
>> missing.
>>
>> >>
>>
>> >> I want to have text covering both the 1st (unique address) and 2nd
>> (shared address) Offer.
>>
> > Acknowledged.
>
>>    >
>>
>> >> And, when a PeerConnection is created due to forking (ie you use a
>> separate PeerConnection for each forked leg of a session), I assume
>>
>> >> already the 1st Offer for that PeerConnection can have a shared
>> address (as the remote entity has indicated support for BUNDLE). That al=
so
>> needs to be covered.
>>
>> >>
>>
> > Yes. Which may indicate the need for a setting on a PeerConnection to
> use a shared address on the initial offer, since it might have come from
> such a fork.
>
> Excellent.
>
> Regards,
>
> Christer
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--001a11c38ba25da19f04e795ef1a
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<div dir=3D"ltr">Can someone explain why the proto field is &#39;RTP/SAVPF&=
#39; rather than &#39;<span style=3D"font-family:arial,sans-serif;font-size=
:1em">UDP/TLS/RTP/SAVPF&#39;? We are using DTLS-SRTP after all.</span><div>=
<span style=3D"font-family:arial,sans-serif;font-size:1em"><br>
</span></div></div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_q=
uote">On 28 September 2013 07:52, Christer Holmberg <span dir=3D"ltr">&lt;<=
a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer=
.holmberg@ericsson.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">




<div>
<div style=3D"direction:ltr;font-size:10pt;font-family:Tahoma">
<p>Hi,</p>
<p>=A0</p>
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote"><div class=3D"im">
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;padding=
-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-l=
eft-style:solid">
<div>
<div style=3D"font-family:Tahoma;font-size:10pt;direction:ltr">
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>
<div>&gt;&gt;&gt;&gt;&gt; Q_3: =A0 =A0 =A0RTP<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; The text says:<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; =93The &lt;proto&gt; field MUST be set to &quot;RTP/SA=
VPF&quot;. =93<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; But, that of course only applies to RTP based streams =
(not the data channel). Also, in general, it needs to be clear what informa=
tion</div>
<div>&gt;&gt;&gt;&gt;&gt;=A0needs to be in every m- line, and what informat=
ion is protocol specific. I would suggest to have a =93General=94 sub-secti=
on, a =93RTP=94 sub-section, etc.<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;</div>
<div>&gt;&gt;&gt;&gt; Yep - I think this should be offers are created with =
RTP/SAVPF for audio and video and system can reeve offers with SAVPF or SAV=
P<br>
&gt;&gt;&gt;</div>
<div>&gt;&gt;&gt; Currently the document specifies what to do with media li=
nes, with later discussion of handling the SCTP line. Do you think the curr=
ent information is not sufficiently descriptive?=A0</div>
<div>&gt;&gt;</div>
</div>
<div>&gt;&gt; My point was that we need to be more clear on what is generic=
, what is RTP, and what is SCTP. But, as you say SCTP text is still to be a=
dded, that clarification can be done when the SCTP text is added.</div>

</div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>&gt;Sorry I was unclear - there is SCTP-specific text in the current d=
ocument.=A0</div>
<div>=A0</div>
</div><div>Ok, so while the content itself might be ok, at some point I&#39=
;d like to have separate sub-sections for the generic stuff, the RTP specic=
if stuff, and the SCTP specific stuff.
</div><div class=3D"im">
<div>=A0</div>
<div>=A0=A0</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;padding=
-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-l=
eft-style:solid">
<div>
<div style=3D"font-family:Tahoma;font-size:10pt;direction:ltr">
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>
<p>&gt;&gt;&gt;&gt;&gt; Q_6: =A0 =A0 =A0BUNDLE<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; We=92ll probably also need some text about BUNDLE.<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;</p>
<p>&gt;&gt;&gt;&gt; yep - but plan is to match up with unified plan<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;Can you be specific about what more you think needs to be said?=
 The createAnswer treatment of BUNDLE is one clear thing that is currently =
missing.</p>
<p>&gt;&gt;=A0</p>
</div>
<p>&gt;&gt; I want to have text covering both=A0the 1st (unique address) an=
d 2nd (shared address) Offer.</p>
</div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
<div>&gt; Acknowledged.=A0<br>
</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;padding=
-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-l=
eft-style:solid">
<div>
<div style=3D"font-family:Tahoma;font-size:10pt;direction:ltr">
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<p>&gt;=A0</p>
<p>&gt;&gt; And, when a PeerConnection is created due to forking (ie you us=
e a separate PeerConnection for each forked leg of a session), I assume
</p>
<p>&gt;&gt; already the 1st Offer for that PeerConnection can have a shared=
 address (as the remote entity has indicated support for BUNDLE). That also=
 needs to be covered.</p>
<p>&gt;&gt;</p>
</div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
<div>&gt; Yes. Which may indicate the need for a setting on a PeerConnectio=
n to use a shared address on the initial offer, since it might have come fr=
om such a fork.=A0</div>
<div>=A0</div>
</div><div>Excellent.</div>
<div>=A0</div>
<div>Regards,</div><span class=3D"HOEnZb"><font color=3D"#888888">
<div>=A0</div>
<div>Christer</div>
<div>=A0</div>
</font></span></div>
</div>
</div>
</div>
</div>
</div>

<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

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From christer.holmberg@ericsson.com  Mon Sep 30 02:49:51 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Kevin Dempsey <kevindempsey70@gmail.com>
Thread-Topic: [rtcweb] JSEP-04: Some comments on Section 5.2.1. and 5.2.2 (19th september)
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Hi Kevin,

I agree. According to RFC 5763 the correct value is UDP/TLS/RTP/SAVP.

Regards,

Christer

From: Kevin Dempsey [mailto:kevindempsey70@gmail.com]
Sent: 30. syyskuuta 2013 11:51
To: Christer Holmberg
Cc: Justin Uberti; Cullen Jennings (fluffy); rtcweb@ietf.org
Subject: Re: [rtcweb] JSEP-04: Some comments on Section 5.2.1. and 5.2.2 (1=
9th september)

Can someone explain why the proto field is 'RTP/SAVPF' rather than 'UDP/TLS=
/RTP/SAVPF'? We are using DTLS-SRTP after all.


On 28 September 2013 07:52, Christer Holmberg <christer.holmberg@ericsson.c=
om<mailto:christer.holmberg@ericsson.com>> wrote:

Hi,


>>>>> Q_3:      RTP
>>>>>
>>>>> The text says:
>>>>>
>>>>> "The <proto> field MUST be set to "RTP/SAVPF". "
>>>>>
>>>>> But, that of course only applies to RTP based streams (not the data c=
hannel). Also, in general, it needs to be clear what information
>>>>> needs to be in every m- line, and what information is protocol specif=
ic. I would suggest to have a "General" sub-section, a "RTP" sub-section, e=
tc.
>>>>>
>>>>>
>>>> Yep - I think this should be offers are created with RTP/SAVPF for aud=
io and video and system can reeve offers with SAVPF or SAVP
>>>
>>> Currently the document specifies what to do with media lines, with late=
r discussion of handling the SCTP line. Do you think the current informatio=
n is not sufficiently descriptive?
>>
>> My point was that we need to be more clear on what is generic, what is R=
TP, and what is SCTP. But, as you say SCTP text is still to be added, that =
clarification can be done when the SCTP text is added.

>Sorry I was unclear - there is SCTP-specific text in the current document.

Ok, so while the content itself might be ok, at some point I'd like to have=
 separate sub-sections for the generic stuff, the RTP specicif stuff, and t=
he SCTP specific stuff.



>>>>> Q_6:      BUNDLE
>>>>>
>>>>> We'll probably also need some text about BUNDLE.
>>>>>
>>>>

>>>> yep - but plan is to match up with unified plan
>>>>
>>>Can you be specific about what more you think needs to be said? The crea=
teAnswer treatment of BUNDLE is one clear thing that is currently missing.

>>

>> I want to have text covering both the 1st (unique address) and 2nd (shar=
ed address) Offer.
> Acknowledged.

>

>> And, when a PeerConnection is created due to forking (ie you use a separ=
ate PeerConnection for each forked leg of a session), I assume

>> already the 1st Offer for that PeerConnection can have a shared address =
(as the remote entity has indicated support for BUNDLE). That also needs to=
 be covered.

>>
> Yes. Which may indicate the need for a setting on a PeerConnection to use=
 a shared address on the initial offer, since it might have come from such =
a fork.

Excellent.

Regards,

Christer


_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org>
https://www.ietf.org/mailman/listinfo/rtcweb


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<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi Kevin,<o:p></o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">I agree. According to
<b>RFC 5763</b> the correct value is <b>UDP/TLS/RTP/SAVP</b>.<o:p></o:p></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Regards,<o:p></o:p></span=
></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Christer<o:p></o:p></span=
></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Kevin De=
mpsey [mailto:kevindempsey70@gmail.com]
<br>
<b>Sent:</b> 30. syyskuuta 2013 11:51<br>
<b>To:</b> Christer Holmberg<br>
<b>Cc:</b> Justin Uberti; Cullen Jennings (fluffy); rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] JSEP-04: Some comments on Section 5.2.1. and 5=
.2.2 (19th september)<o:p></o:p></span></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">Can someone explain why the proto field is 'RTP/SAVP=
F' rather than '<span style=3D"font-family:&quot;Arial&quot;,&quot;sans-ser=
if&quot;">UDP/TLS/RTP/SAVPF'? We are using DTLS-SRTP after all.</span><o:p>=
</o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On 28 September 2013 07:52, Christer Holmberg &lt;<a=
 href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.=
holmberg@ericsson.com</a>&gt; wrote:<o:p></o:p></p>
<div>
<div>
<p><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;san=
s-serif&quot;">Hi,<o:p></o:p></span></p>
<p><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;san=
s-serif&quot;">&nbsp;<o:p></o:p></span></p>
<div>
<div>
<div>
<div>
<div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<div>
<div>
<div>
<div>
<div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ta=
homa&quot;,&quot;sans-serif&quot;">&gt;&gt;&gt;&gt;&gt; Q_3: &nbsp; &nbsp; =
&nbsp;RTP<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; The text says:<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; &#8220;The &lt;proto&gt; field MUST be set to &quot;RT=
P/SAVPF&quot;. &#8220;<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; But, that of course only applies to RTP based streams =
(not the data channel). Also, in general, it needs to be clear what informa=
tion<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ta=
homa&quot;,&quot;sans-serif&quot;">&gt;&gt;&gt;&gt;&gt;&nbsp;needs to be in=
 every m- line, and what information is protocol specific. I would suggest =
to have a &#8220;General&#8221; sub-section, a &#8220;RTP&#8221; sub-sectio=
n, etc.<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ta=
homa&quot;,&quot;sans-serif&quot;">&gt;&gt;&gt;&gt; Yep - I think this shou=
ld be offers are created with RTP/SAVPF for audio and video and system can =
reeve offers with SAVPF or SAVP<br>
&gt;&gt;&gt;<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ta=
homa&quot;,&quot;sans-serif&quot;">&gt;&gt;&gt; Currently the document spec=
ifies what to do with media lines, with later discussion of handling the SC=
TP line. Do you think the current information is not sufficiently
 descriptive?&nbsp;<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ta=
homa&quot;,&quot;sans-serif&quot;">&gt;&gt;<o:p>&nbsp;</o:p></span></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ta=
homa&quot;,&quot;sans-serif&quot;">&gt;&gt; My point was that we need to be=
 more clear on what is generic, what is RTP, and what is SCTP. But, as you =
say SCTP text is still to be added, that clarification can be
 done when the SCTP text is added.<o:p></o:p></span></p>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ta=
homa&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ta=
homa&quot;,&quot;sans-serif&quot;">&gt;Sorry I was unclear - there is SCTP-=
specific text in the current document.&nbsp;<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ta=
homa&quot;,&quot;sans-serif&quot;">&nbsp;<o:p></o:p></span></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ta=
homa&quot;,&quot;sans-serif&quot;">Ok, so while the content itself might be=
 ok, at some point I'd like to have separate sub-sections for the generic s=
tuff, the RTP specicif stuff, and the SCTP specific stuff.
<o:p></o:p></span></p>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ta=
homa&quot;,&quot;sans-serif&quot;">&nbsp;<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ta=
homa&quot;,&quot;sans-serif&quot;">&nbsp;&nbsp;<o:p></o:p></span></p>
</div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<div>
<div>
<div>
<div>
<div>
<div>
<div>
<p><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;san=
s-serif&quot;">&gt;&gt;&gt;&gt;&gt; Q_6: &nbsp; &nbsp; &nbsp;BUNDLE<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; We&#8217;ll probably also need some text about BUNDLE.=
<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;<o:p></o:p></span></p>
<p><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;san=
s-serif&quot;">&gt;&gt;&gt;&gt; yep - but plan is to match up with unified =
plan<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;Can you be specific about what more you think needs to be said?=
 The createAnswer treatment of BUNDLE is one clear thing that is currently =
missing.<o:p></o:p></span></p>
<p><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;san=
s-serif&quot;">&gt;&gt;&nbsp;<o:p></o:p></span></p>
</div>
<p><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;san=
s-serif&quot;">&gt;&gt; I want to have text covering both&nbsp;the 1st (uni=
que address) and 2nd (shared address) Offer.<o:p></o:p></span></p>
</div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ta=
homa&quot;,&quot;sans-serif&quot;">&gt; Acknowledged.&nbsp;<o:p></o:p></spa=
n></p>
</div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<div>
<div>
<div>
<div>
<div>
<div>
<p><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;san=
s-serif&quot;">&gt;&nbsp;<o:p></o:p></span></p>
<p><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;san=
s-serif&quot;">&gt;&gt; And, when a PeerConnection is created due to forkin=
g (ie you use a separate PeerConnection for each forked leg of a session), =
I assume
<o:p></o:p></span></p>
<p><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;san=
s-serif&quot;">&gt;&gt; already the 1st Offer for that PeerConnection can h=
ave a shared address (as the remote entity has indicated support for BUNDLE=
). That also needs to be covered.<o:p></o:p></span></p>
<p><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;san=
s-serif&quot;">&gt;&gt;<o:p>&nbsp;</o:p></span></p>
</div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ta=
homa&quot;,&quot;sans-serif&quot;">&gt; Yes. Which may indicate the need fo=
r a setting on a PeerConnection to use a shared address on the initial offe=
r, since it might have come from such a fork.&nbsp;<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ta=
homa&quot;,&quot;sans-serif&quot;">&nbsp;<o:p></o:p></span></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ta=
homa&quot;,&quot;sans-serif&quot;">Excellent.<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ta=
homa&quot;,&quot;sans-serif&quot;">&nbsp;<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ta=
homa&quot;,&quot;sans-serif&quot;">Regards,<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ta=
homa&quot;,&quot;sans-serif&quot;;color:#888888">&nbsp;<o:p></o:p></span></=
p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ta=
homa&quot;,&quot;sans-serif&quot;;color:#888888">Christer<o:p></o:p></span>=
</p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ta=
homa&quot;,&quot;sans-serif&quot;;color:#888888">&nbsp;<o:p></o:p></span></=
p>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1C4AFDAFESESSMB209erics_--

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From: "Rauschenbach, Uwe (NSN - DE/Munich)" <uwe.rauschenbach@nsn.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: No a=ice-lite in JSEP-04
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Date: Mon, 30 Sep 2013 15:28:18 +0000
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Subject: [rtcweb] No a=ice-lite in JSEP-04
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Hi Justin, Cullen,

JSEP-04 prohibits the use of a=3Dice-lite. It is stated that it is incompat=
ible with I-D.ietf-rtcweb-rtp-usage.
Can you point out which specific aspect of this draft ICE Lite is incompati=
ble with?

In deployments where one endpoint is a browser and another one is a gateway=
 connected to the public Internet, it makes sense to allow ICE lite.

=A0
Kind regards,
Uwe=20


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2013/9/30 Rauschenbach, Uwe (NSN - DE/Munich) <uwe.rauschenbach@nsn.com>:
> JSEP-04 prohibits the use of a=3Dice-lite. It is stated that it is incomp=
atible with I-D.ietf-rtcweb-rtp-usage.
> Can you point out which specific aspect of this draft ICE Lite is incompa=
tible with?
>
> In deployments where one endpoint is a browser and another one is a gatew=
ay connected to the public Internet, it makes sense to allow ICE lite.

Without reading JSEP-04 I expect that it is not allowed for a web
browser to indicate "a=3Dice-lite", but a server can do that. Anyhow,
AFAIR the usage of ICE lite is fully transparent for the non ICE lite
endpoint.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Date: Mon, 30 Sep 2013 15:30:08 -0700
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] No a=ice-lite in JSEP-04
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Yes. In the offers and answers that WebRTC endpoints create, a=3Dice-lite i=
s
prohibited, since WebRTC endpoints must support "Full" ICE.

The remote endpoint may of course support only ICE Lite, and WebRTC
implementations must work properly with such endpoints.

This isn't normatively indicated anywhere that I could find, so I plan to
add specifics on this to the next version of JSEP.




On Mon, Sep 30, 2013 at 8:32 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net> wr=
ote:

> 2013/9/30 Rauschenbach, Uwe (NSN - DE/Munich) <uwe.rauschenbach@nsn.com>:
> > JSEP-04 prohibits the use of a=3Dice-lite. It is stated that it is
> incompatible with I-D.ietf-rtcweb-rtp-usage.
> > Can you point out which specific aspect of this draft ICE Lite is
> incompatible with?
> >
> > In deployments where one endpoint is a browser and another one is a
> gateway connected to the public Internet, it makes sense to allow ICE lit=
e.
>
> Without reading JSEP-04 I expect that it is not allowed for a web
> browser to indicate "a=3Dice-lite", but a server can do that. Anyhow,
> AFAIR the usage of ICE lite is fully transparent for the non ICE lite
> endpoint.
>
>
> --
> I=C3=B1aki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr">Yes. In the offers and answers that WebRTC endpoints creat=
e, a=3Dice-lite is prohibited, since WebRTC endpoints must support &quot;Fu=
ll&quot; ICE.=C2=A0<div><br></div><div>The remote endpoint may of course su=
pport only ICE Lite, and WebRTC implementations must work properly with suc=
h endpoints.</div>


<div><br></div><div>This isn&#39;t normatively indicated anywhere that I co=
uld find, so I plan to add specifics on this to the next version of JSEP.<b=
r><div><br></div><div><br></div></div><div class=3D"gmail_extra"><br>
<br><div class=3D"gmail_quote">On Mon, Sep 30, 2013 at 8:32 AM, I=C3=B1aki =
Baz Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.net" target=
=3D"_blank">ibc@aliax.net</a>&gt;</span> wrote:<br><blockquote class=3D"gma=
il_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-lef=
t:1ex">


2013/9/30 Rauschenbach, Uwe (NSN - DE/Munich) &lt;<a href=3D"mailto:uwe.rau=
schenbach@nsn.com" target=3D"_blank">uwe.rauschenbach@nsn.com</a>&gt;:<br>
<div>&gt; JSEP-04 prohibits the use of a=3Dice-lite. It is stated that it i=
s incompatible with I-D.ietf-rtcweb-rtp-usage.<br>
&gt; Can you point out which specific aspect of this draft ICE Lite is inco=
mpatible with?<br>
&gt;<br>
&gt; In deployments where one endpoint is a browser and another one is a ga=
teway connected to the public Internet, it makes sense to allow ICE lite.<b=
r>
<br>
</div>Without reading JSEP-04 I expect that it is not allowed for a web<br>
browser to indicate &quot;a=3Dice-lite&quot;, but a server can do that. Any=
how,<br>
AFAIR the usage of ICE lite is fully transparent for the non ICE lite<br>
endpoint.<br>
<span><font color=3D"#888888"><br>
<br>
--<br>
I=C3=B1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt=
;<br>
</font></span><div><div>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div></div>

--20cf300e529d20898404e7a16250--

From karl.stahl@intertex.se  Mon Sep 30 17:13:26 2013
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From: "Karl Stahl" <karl.stahl@intertex.se>
To: "'Bernard Aboba'" <bernard_aboba@hotmail.com>, "'Harald Alvestrand'" <harald@alvestrand.no>, "'Justin Uberti'" <juberti@google.com>, "'Cullen Jennings \(fluffy\)'" <fluffy@cisco.com>, <draft-ietf-rtcweb-use-cases-and-requirements@tools.ietf.org>
References: <C5E08FE080ACFD4DAE31E4BDBF944EB11667BBA0@xmb-aln-x02.cisco.com>, <523c6d3d.c9d1440a.3b96.7499SMTPIN_ADDED_BROKEN@mx.google.com>, <CAD6AjGRXr5kPRQdN+4jkgXHciN3NE7HiRmsb7kaYuzwHEPa7ZA@mail.gmail.com>, <C5E08FE080ACFD4DAE31E4BDBF944EB1166CC702@xmb-aln-x02.cisco.com>, <5244104D.4010401@alvestrand.no>, <CABkgnnWyYCdpSxXyiYb+4BzMpME85671x5JzxJX08RiyQd+SFQ@mail.gmail.com> <BLU169-W98EC710F291837B36C14F0932B0@phx.gbl>
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Date: Tue, 1 Oct 2013 02:13:15 +0200
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Cc: rtcweb@ietf.org
Subject: [rtcweb]  [mmusic] TURN server address via DHCP, WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
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I happily withdraw my suggestion to use DHCP to find a network provider
offered TURN server in favor of the DNS-Based Service Discovery method
(inserted last below). The major problem with the DHCP usage was that =
you
then also have to do something similar for: RA - Router Advertisement - =
in
IPv6, addition to the IPCP protocol for PPPoE and something for the =
mobile
OTT channel =96 wherever DHCP is NOT the method to give you an IP =
address.

=20

Further:

> On Thu, Sep 27, 2013 Justin Uberti wrote:

> Agree. I still think that extending PAC files to include TURN =
information
is the right way to go.

> PAC files can already be discovered via DHCP or DNS, there is no need =
to
reinvent this wheel.

=20

>>On Thu, Sep 26, 2013 at 4:55 PM, Cullen Jennings (fluffy) <
<mailto:fluffy@cisco.com> fluffy@cisco.com> wrote:

>>I think we need to give some advise to the browsers vendors on what =
they
should implement to find turn servers

=20

I Googled a bit on PAC and WPAD and saw that a HTTP proxy config JS file =
can
be picked up through a URL based on the device name. Similar could work =
for
finding a TURN server on a LAN, but what about the case with a network
provider offered TURN server? The network provider hands out an IP =
address
and wants to announce a related TURN server address: =96 Is there a =
=93PAC-way=94
to announce it to a device on a LAN (behind a NAT/firewall)? The device =
must
find such configuration file based on its public IP address (that he can
find using STUN as in the suggestion below).

=20

But generally, finding a TURN-server is a network-thing, ICE (or even =
TURN
not using ICE), not a Web-thing, so for that reason the DNS-Based =
Service
Discovery method is at a better level. Such method could also be used by =
SIP
clients (even a Skype client could implement it to benefit from a =
real-time
path with better quality, if the network provider offers such path =
through
his TURN server).

=20

The DNS-Based Service Discovery method should be easy to implement in a
WebRTC browser (doing complex ICE things anyway). The question is rather =
how
easy it is for the network provider to provision. Do they all do reverse =
DNS
like with found with mobile operators TeliaSonera and Tele2 in Sweden? =
Then
it should not be too difficult.

=20

Or is there yet another better method to announce a TURN server address =
with
the IP address offered?

=20

/Karl

=20

=20

Fr=E5n: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] F=F6r =
Bernard
Aboba
Skickat: den 29 september 2013 02:41
Till: Harald Alvestrand
Kopia: rtcweb@ietf.org
=C4mne: Re: [rtcweb] TURN server address via DHCP, WGLC of
draft-ietf-rtcweb-use-cases-and-requirements-11

=20

On Sep 26, 2013 8:46 PM, "Harald Alvestrand" <harald@alvestrand.no> =
wrote:
"So far, neither the POSIX standard nor any OS vendor has offered a =
generic
facility to access information made available in DHCP packets."

=20

[BA] The Windows DHCP client API does provide this:=20

http://msdn.microsoft.com/en-us/library/windows/desktop/aa363351(v=3Dvs.8=
5).as
px

=20

In particular, the SendParams argument to the DhcpRequestParams function =
can
be used to request a particular parameter (e.g. TURN server address), =
which
will then be returned in the RecdParams variable. =20

=20

Nevertheless, I still think that using DHCP to configure the TURN server
address in a browser isn't a good idea.  For one thing, since DHCP is
effectively unsecured, this mechanism could be used by a rogue DHCP =
server
to force traffic to a rogue turnserver.   Great for surveillance!

=20

=20

On Sep 27, 2013 1:27 AM, "Karl Stahl" < <mailto:karl.stahl@intertex.se>
karl.stahl@intertex.se> wrote:

Here comes the suggestion I got from my developer that would allow a =
network
provider to offer his TURN server for the WebRTC browser to use. This =
would
require NO new DHCP-options or similar, and NO OS changes (I do realize
those should be avoided if possible...).

The idea is still, that whoever is responsible for giving a device an IP
address (the network provider or a LAN administrator) can also announce =
a
TURN server for the WebRTC browser to use.

The suggestion is to use RFC6763 (DNS-Based Service Discovery, see =
chapter
11) where the network provider (the owner of the IP address) has set up =
a
DNS PTR record for the TURN server in the in-addr.arpa domain.

If the device got IP 173.164.252.149, then make a query for the PTR =
record
for:
_turn._udp.149.252.164.173.in-addr.arpa.
Then the SRV record would return the actual address to the TURN server =
(and
you may find several for load balancing and failover I guess)

If the device is on a LAN, the IP 173.164.252.149 to query would be the =
WAN
IP you get via STUN in the ICE process. But if the LAN administrator
provides a local TURN server, then he also should have blocked STUN in =
the
firewall, and the browser should query the device's local host address =
(as
in the ICE procedure) and the local LAN DNS server should answer the =
query
to give the local TURN server address on the LAN.

Shouldn't this work to allow network provider to offer his TURN server
"automatically and generally"?

And wouldn't this work nicely for mobile devices, whenever the device is =
on
a "WebRTC-ready access" (actually good for everything using ICE), =
whether
fixed, WiFi or 3G/4G OTT, you get also a TURN server (and the network
provider hopefully offers a prioritized pipe there as one usage of this
mechanism).

Not to slow down every call setup by doing this in the ICE process, I =
guess
this could be done when starting the browser and when the device gets a =
new
IP address, to have the TURN server address ready for later use.

/Karl


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vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-family:"Calibri","sans-serif"'>I happily =
withdraw my suggestion to use DHCP to find a network provider offered =
TURN server in favor of the DNS-Based Service Discovery method (inserted =
last below). The major problem with the DHCP usage was that you then =
also have to do something similar for: RA - Router Advertisement - in =
IPv6, addition to the IPCP protocol for PPPoE and something for the =
mobile OTT channel &#8211; wherever DHCP is NOT the method to give you =
an IP address.<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'><o:p>&nbsp;</o:p></span></p>=
<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>Further:<o:p></o:p></span></=
p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>&gt; On Thu, Sep 27, 2013 =
</span><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>Justin Uberti</span><span =
lang=3DEN-US style=3D'font-family:"Calibri","sans-serif"'> =
wrote:<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>&gt; Agree. I still think =
that extending PAC files to include TURN information is the right way to =
go.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>&gt; PAC files can already =
be discovered via DHCP or DNS, there is no need to reinvent this =
wheel.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'><o:p>&nbsp;</o:p></span></p>=
<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>&gt;&gt;On Thu, Sep 26, =
2013 at 4:55 PM, Cullen Jennings (fluffy) &lt;</span><span =
style=3D'font-family:"Calibri","sans-serif"'><a =
href=3D"mailto:fluffy@cisco.com" target=3D"_blank"><span =
lang=3DEN-US>fluffy@cisco.com</span></a></span><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>&gt; =
wrote:<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>&gt;&gt;I think we need to =
give some advise to the browsers vendors on what they should implement =
to find turn servers<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'><o:p>&nbsp;</o:p></span></p>=
<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>I Googled a bit on PAC and =
WPAD and saw that a HTTP proxy config JS file can be picked up through a =
URL based on the device name. Similar could work for finding a TURN =
server on a LAN, but what about the case with a network provider offered =
TURN server? The network provider hands out an IP address and wants to =
announce a related TURN server address: &#8211; Is there a =
&#8220;PAC-way&#8221; to announce it to a device on a LAN (behind a =
NAT/firewall)? The device must find such configuration file based on its =
public IP address (that he can find using STUN as in the suggestion =
below).<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'><o:p>&nbsp;</o:p></span></p>=
<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>But generally, finding a =
TURN-server is a network-thing, ICE (or even TURN not using ICE), not a =
Web-thing, so for that reason the DNS-Based Service Discovery method is =
at a better level. Such method could also be used by SIP clients (even a =
Skype client could implement it to benefit from a real-time path with =
better quality, if the network provider offers such path through his =
TURN server).<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'><o:p>&nbsp;</o:p></span></p>=
<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>The DNS-Based Service =
Discovery method should be easy to implement in a WebRTC browser (doing =
complex ICE things anyway). The question is rather how easy it is for =
the network provider to provision. Do they all do reverse DNS like with =
found with mobile operators TeliaSonera and Tele2 in Sweden? Then it =
should not be too difficult.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'><o:p>&nbsp;</o:p></span></p>=
<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>Or is there yet another =
better method to announce a TURN server address with the IP address =
offered?<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'><o:p>&nbsp;</o:p></span></p>=
<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>/Karl<o:p></o:p></span></p><=
p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><div><div =
style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm'><p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Fr=E5n:</spa=
n></b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> =
rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] <b>F=F6r =
</b>Bernard Aboba<br><b>Skickat:</b> den 29 september 2013 =
02:41<br><b>Till:</b> Harald Alvestrand<br><b>Kopia:</b> =
rtcweb@ietf.org<br><b>=C4mne:</b> Re: [rtcweb] TURN server address via =
DHCP, WGLC of =
draft-ietf-rtcweb-use-cases-and-requirements-11<o:p></o:p></span></p></di=
v></div><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><p =
class=3DMsoNormal><span style=3D'font-family:"Calibri","sans-serif"'>On =
Sep 26, 2013 8:46 PM, &quot;Harald Alvestrand&quot; &lt;<a =
href=3D"mailto:harald@alvestrand.no">harald@alvestrand.no</a>&gt; =
wrote:<br>&quot;So far, neither the POSIX standard nor any OS vendor has =
offered a generic facility to access information made available in DHCP =
packets.&quot;<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-family:"Calibri","sans-serif"'><o:p>&nbsp;</o:p></span></p>=
<p class=3DMsoNormal><span =
style=3D'font-family:"Calibri","sans-serif"'>[BA] The Windows DHCP =
client API does provide this:&nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal><span style=3D'font-family:"Calibri","sans-serif"'><a =
href=3D"http://msdn.microsoft.com/en-us/library/windows/desktop/aa363351(=
v=3Dvs.85).aspx" =
target=3D"_blank">http://msdn.microsoft.com/en-us/library/windows/desktop=
/aa363351(v=3Dvs.85).aspx</a><o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'font-family:"Calibri","sans-serif"'><o:p>&nbsp;</o:p></span></p>=
<p class=3DMsoNormal><span =
style=3D'font-family:"Calibri","sans-serif"'>In particular, the =
SendParams argument to the DhcpRequestParams function&nbsp;can be used =
to request a particular parameter (e.g. TURN server address), which will =
then be returned in the RecdParams variable. =
&nbsp;<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-family:"Calibri","sans-serif"'><o:p>&nbsp;</o:p></span></p>=
<p class=3DMsoNormal><span =
style=3D'font-family:"Calibri","sans-serif"'>Nevertheless, I still think =
that using DHCP to configure the TURN server address in a browser isn't =
a good idea. &nbsp;For one thing, since DHCP is effectively unsecured, =
this mechanism could be used by a rogue DHCP server to force traffic to =
a rogue turnserver. &nbsp; Great for =
surveillance!<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US>On Sep =
27, 2013 1:27 AM, &quot;Karl Stahl&quot; &lt;</span><a =
href=3D"mailto:karl.stahl@intertex.se"><span =
lang=3DEN-US>karl.stahl@intertex.se</span></a><span lang=3DEN-US>&gt; =
wrote:<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US>Here =
comes the suggestion I got from my developer that would allow a =
network<br>provider to offer his TURN server for the WebRTC browser to =
use. This would<br>require NO new DHCP-options or similar, and NO OS =
changes (I do realize<br>those should be avoided if =
possible...).<br><br>The idea is still, that whoever is responsible for =
giving a device an IP<br>address (the network provider or a LAN =
administrator) can also announce a<br>TURN server for the WebRTC browser =
to use.<br><br>The suggestion is to use RFC6763 (DNS-Based Service =
Discovery, see chapter<br>11) where the network provider (the owner of =
the IP address) has set up a<br>DNS PTR record for the TURN server in =
the in-addr.arpa domain.<br><br>If the device got IP 173.164.252.149, =
then make a query for the PTR =
record<br>for:<br>_turn._udp.149.252.164.173.in-addr.arpa.<br>Then the =
SRV record would return the actual address to the TURN server =
(and<br>you may find several for load balancing and failover I =
guess)<br><br>If the device is on a LAN, the IP 173.164.252.149 to query =
would be the WAN<br>IP you get via STUN in the ICE process. But if the =
LAN administrator<br>provides a local TURN server, then he also should =
have blocked STUN in the<br>firewall, and the browser should query the =
device's local host address (as<br>in the ICE procedure) and the local =
LAN DNS server should answer the query<br>to give the local TURN server =
address on the LAN.<br><br>Shouldn't this work to allow network provider =
to offer his TURN server<br>&quot;automatically and =
generally&quot;?<br><br>And wouldn't this work nicely for mobile =
devices, whenever the device is on<br>a &quot;WebRTC-ready access&quot; =
(actually good for everything using ICE), whether<br>fixed, WiFi or =
3G/4G OTT, you get also a TURN server (and the network<br>provider =
hopefully offers a prioritized pipe there as one usage of =
this<br>mechanism).<br><br>Not to slow down every call setup by doing =
this in the ICE process, I guess<br>this could be done when starting the =
browser and when the device gets a new<br>IP address, to have the TURN =
server address ready for later use.<br><br></span>/Karl<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p></o:p></span></p></div></div></body></html>
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