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To: Jonathan Lennox <jonathan@vidyo.com>
References: <1f020dc6-1ac8-b71b-aee4-a711d15f1588@ericsson.com> <D2A63CD5-61AB-47DD-B946-6A17A94F76E7@vidyo.com>
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [rtcweb] [MMUSIC] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
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Hi Jonathan,

Thanks for the feedback. See inline for my attempt to answer them.

Den 2017-01-31 kl. 18:43, skrev Jonathan Lennox:
> In general, this looks good.
>
> I have one question, however.  What should happen if an RTP packet
> (without a MID) is received for a stream that has an existing mapping
> to an m= line, but whose PT is not valid for that m= line?
>

So, the text in JSEP-18 didn't specify this either. However, I do have 
an opinion on this.

> Should it 1. Cause the stream’s mapping to be moved to another m=
> line, if the received PT is in the payload type table for some other
> m= line? or

I would recommend against this due to that we will have to go into under 
which circumstances that one can accept this, and when not. Bo Burman 
and I had some discussion on this. We arrived at some relevant aspects 
of this. The first is that one have one to consider precedence rules 
between PTs, as well as a=ssrc and MID. In general we think mismatch 
between the information is to be considered an error. However, using PT 
and MID could cause strange error cases, where packets without MID ends 
up in on m= line with the PT and the ones with MID are dropped as 
inconsistent. I would claim that this stream is always in error.

I would also note that Bundle specification currently forbids PT based 
switching, but for a different reasons. Section 10.1 says:

    o  A given SSRC MUST NOT transmit RTP packets using payload types
       that originate from different bundled "m=" lines.

    NOTE: The last bullet above is to avoid sending multiple media types
    from the same SSRC.  If transmission of multiple media types are done
    with time overlap, RTP and RTCP fail to function.  Even if done in
    proper sequence this causes RTP Timestamp rate switching issues
    [RFC7160].  However, once an SSRC has left the RTP session (by
    sending an RTCP BYE packet), that SSRC can be reused by another
    source (possibly associated with a different bundled "m=" line) after
    a delay of 5 RTCP reporting intervals (the delay is to ensure the
    SSRC has timed out, in case the RTCP BYE packet was lost [RFC3550]).

But, we do note that the limitation is not strictly necessary in regards 
to the motivation. The minimal limitation based on the motivation would 
be to require that a SSRC only uses PTs of the same media type and clock 
rate within a single bundle group. Using shared PTs, which the current 
limitation requires automatically meets that goal, and is easier to 
check for.

We also noted that BUNDLE should clarify how MID relates to a=ssrc 
(RFC5576). My interpretation is that a=ssrc locks a RTP stream to a 
particular m= line, and can't be moved. So any MID value other than that 
matches the m= line that has the a=ssrc value would be an error case. 
For usages that intend to have a RTP stream to bounce between m= lines 
should not use a=ssrc, only MID. I think it would be good to clarify 
this in the BUNDLE specification.


 > 2. Be an error?

This would be my preferred choice.

>
> In other words, should it be possible for PT-mapping to cause streams
> to change their mappings?

To conclude: No, as it would cause several additional complexities and 
possible branches in the algorithm. The use case for moving SSRCs 
between m= lines are supported when using MID, and the m= lines are 
correctly configured with shared PTs.

Cheers

Magnus

>
> (Should this decision depend on what caused the stream to be mapped
> to its current m= line?)
>
>> On Jan 27, 2017, at 11:43 AM, Magnus Westerlund
>> <magnus.westerlund@ericsson.com> wrote:
>>
>> MMUSIC and RTCWEB,
>>
>> Here is now a more complete text proposal that also considers the
>> RTCP. It also goes further than what Appendix B defines in one
>> aspect, namely explicitly considering third party RTCP reporting
>> and what to do with it.
>>
>> Feedback is much appreciated. I plan to put this into a PR towards
>> JSEP APPENDIX B on monday. If only to get the JSEP authors
>> attention ;-). However, the text is really intended for the next
>> version of BUNDLE.
>>
>>
>> X.  Associating RTP/RTCP With Correct SDP Media Description
>>
>> As described in [RFC3550], RTP packets are associated with RTP
>> streams [RFC7656].  Each RTP stream is identified by an SSRC
>> value, and each RTP packet carries an SSRC value that is used to
>> associate the packet with the correct RTP stream.  RTCP packets
>> also uses SSRCs to identify on which RTP streams any report or
>> feedback relate to. Thus, an RTCP packet will commonly carry
>> multiple SSRC values, and might therefore be providing feedback or
>> report on multiple RTP streams.
>>
>> In order to be able to process received RTP/RTCP packets correctly
>> it must be possible to associate an RTP stream with the correct
>> "m=" line, as the "m=" line and SDP attributes associated with the
>> "m=" line contain information needed to process the packets.
>>
>> As all RTP streams associated with a BUNDLE group are part of the
>> same RTP session and using the same address:port combination for
>> sending and receiving RTP/RTCP packets, the local address:port
>> combination cannot be used to associate an RTP stream with the
>> correct "m=" line.  In addition, multiple RTP streams might be
>> associated with the same "m=" line.
>>
>> Also, as described in Section 10.1.1, the same payload type value
>> might be used by multiple RTP streams, in which case the payload
>> type value cannot be used to associate an RTP stream with the
>> correct "m=" line.  However, there are cases where each "m=" line
>> has unique payload type values, and then the payload type could
>> serve as indicator to the relevant "m=" line the RTP stream is
>> associated with.
>>
>> An offerer and answerer can inform each other which SSRC values
>> they will use for an RTP stream by using the SDP 'ssrc' attribute
>> [RFC5576].  However, an offerer will not know which SSRC values
>> the answerer will use until the offerer has received the answer
>> providing
>>
>>
>>
>> Name                      Expires July 31, 2017
>> [Page 2]
>>
>> Internet-Draft              Abbreviated-Title               January
>> 2017
>>
>>
>> that information.  Due to this, before the offerer has received
>> the answer, the offerer will not be able to associate an RTP stream
>> with the correct "m=" line using the SSRC value associated with the
>> RTP stream.  In addition, the offerer and answerer may start using
>> new SSRC values mid-session, without informing each other using the
>> SDP 'ssrc' attribute.
>>
>> In order for an offerer and answerer to always be able to
>> associate an RTP stream with the correct "m=" line, the offerer and
>> answerer using the BUNDLE extension MUST support the mechanism
>> defined in Section 14, where the offerer and answerer includes the
>> identification-tag (provided by the remote peer) associated with
>> an "m=" line in the RTP Streams and in RTCP SDES packets part of a
>> BUNDLE group.
>>
>> The mapping from an SSRC to an identification-tag is carried in
>> RTCP SDES packets or in RTP header extensions (Section 14).  Since
>> a compound RTCP packet can contain multiple RTCP SDES packets, and
>> each RTCP SDES packet can contain multiple chunks, an RTCP packet
>> can contain several SSRC to identification-tag mappings.  The
>> offerer and answerer maintain tables mapping RTP streams identified
>> by SSRC to "m=" lines identified by the identification-tag.  These
>> tables are updated each time new information that affects how
>> packets should be processed and routed are received.
>>
>> To prepare for demultiplexing RTP streams to the correct "m="
>> line, the following steps MUST be followed for each BUNDLE group
>> based on the SDP signalling information.
>>
>> Construct a table mapping MID to "m=" line for each "m=" line in
>> this BUNDLE group.  Note that an "m=" line may only have one MID.
>>
>> Construct a table mapping incoming RTP streams (SSRCs) to their
>> "m=" line for each "m=" line in this BUNDLE group and for each RTP
>> stream explicitly signalled for receiving in that "m=" line.
>>
>> Construct a table mapping payload types to "m=" line for each "m="
>> line in the BUNDLE group and for each payload type configured for
>> receiving in that "m=" line.  If any payload type is configured for
>> receiving in more than one "m=" line in the BUNDLE group, do not it
>> include it in the table.
>>
>> Note that for each of these tables, there can only be one mapping
>> for any given key (MID, SSRC, or PT).  In other words, the tables
>> are not multimaps.
>>
>> As "m=" lines are added or removed from the BUNDLE groups, or
>> their configurations are changed, the tables above MUST also be
>> updated.
>>
>>
>>
>> Name                      Expires July 31, 2017
>> [Page 3]
>>
>> Internet-Draft              Abbreviated-Title               January
>> 2017
>>
>>
>> Received RTP packets that are syntactically correct are processed
>> by the RTP/RTCP protocol implementation for statistics etc.  After
>> this processing they need to be routed to the higher layer context
>> associated with the "m=" line within the BUNDLE group.  Somewhere
>> in the process where an received RTP packet is processed to be
>> delivered to the higher layer by RTP the matching step below MUST
>> be performed:
>>
>> 1.  Reception of an RTP packet for an RTP stream that has an
>> existing mapping in the RTP stream to m= line table.  Before
>> proceeding in delivering the packet to the higher layer context
>> according to the RTP stream to "m=" line mapping table the
>> following checks MUST be performed:
>>
>> A.  If the packet carries an RTP header extension with a SDES MID
>> value that is not in the table mapping MID to "m=" line, then do
>> not deliver the RTP packet to higher layers.
>>
>> B.  If the packet carries an RTP header extension with a SDES MID
>> value that is in the table mapping MID to "m=" line, and the value
>> indicates a different "m=" line than the current RTP stream to "m="
>> line mapping table, then update the RTP stream to "m=" line
>> mapping.
>>
>> 2.  Reception of an RTP packet for an RTP stream that has no
>> existing mapping to an m= line.  In this case the following actions
>> MUST be performed:
>>
>> A.  If the packet carries an RTP header extension with a SDES MID
>> value that is in the table mapping MID to "m=" line, then create an
>> entry in the RTP stream to "m=" line mapping table for this RTP
>> stream (SSRC).  Then deliver the RTP packet to the "m=" line
>> context of the created mapping and stop.
>>
>> B.  If the packet carries a Payload Type that is in the payload
>> type table, then create an entry in the RTP stream to "m=" line
>> mapping table for this RTP stream (SSRC).  Then deliver the RTP
>> packet to the "m=" line context of the created mapping and stop.
>>
>> C.  Otherwise do not deliver the RTP packet to higher layers. Note,
>> this includes unknown MID values.
>>
>> For each RTCP packet received (including each RTCP packet that is
>> part of a compound RTCP packet), the RTCP packet needs to be
>> processed by the RTP/RTCP implementation and relevant information
>> and data from the RTCP packets needs to be routed to the
>> appropriate handler for the related RTP streams.  The appropriate
>> handler is determined by using the RTP stream to "m=" line mapping
>> table.
>>
>>
>>
>> Name                      Expires July 31, 2017
>> [Page 4]
>>
>> Internet-Draft              Abbreviated-Title               January
>> 2017
>>
>>
>> On reception of any compound RTCP packet prior to dispatching the
>> received information and data, if there is an RTCP SDES packet
>> included that SHOULD be processed first.  If that SDES packet
>> contains SDES MID entries, this can results in updates and
>> additions to the RTP stream to "m=" line mapping table.  Thus each
>> of the SDES MID items are processed and the current table entries
>> are checked if the corresponding MID value matches the current RTP
>> stream to "m=" line mapping, else the entry is updated.  If there
>> is no RTP stream to "m=" line table mapping entry for the received
>> SDES item's SSRC, such an entry is created.  Note, that in the
>> process of updating the table entries, update flap suppression as
>> discussed in Section 4.2.6 of [RFC7941] should be considered.
>>
>> The various different RTCP packets as well as their various sub
>> parts, such as the various RTCP Feedback message types, relates to
>> the RTP streams in a couple of different ways.  The currently
>> known patterns are the following:
>>
>> Reports on outgoing RTP streams:  For all RTP streams that this
>> endpoint is the source of, it can expect to receive report blocks
>> of several types identified as relating to an outgoing stream. The
>> basic pattern for these blocks are that the RTCP packet header
>> identifies the source of the reports, as identified by an SSRC, and
>> containing one or more report blocks, where each report block
>> identifies the RTP stream, using the SSRC, the report relates to.
>> For this pattern the relevant report information is provided to the
>> higher layer associated with the "m=" line the RTP Stream to "m="
>> line table identifies.  The source SSRC as identifier of the
>> endpoint that the report originates are relevant for interpreting
>> the information, but not necessarily for routing.  Example of this
>> is pattern are:
>>
>> Sender Report (SR) and Receiver Report (RR)  The basic receiver
>> report blocks from RFC3550 start with the SSRC of the RTP stream
>> they report on.
>>
>> Extended Reports (XR):  RFC3611 is a framework that enables a large
>> number of different reports.  However, a large number of these
>> report formats are reporting on specific RTP streams and thus each
>> individual report of these types contains a SSRC field to identify
>> the RTP stream.
>>
>> RTCP Feedback Messages for outgoing RTP streams:  The RFC 4585
>> RTCP feedback messages allow for a number of different type of
>> feedback messages.  However, the RTCP feedback message header
>> contains the SSRC identifying the source of the feedback messages
>> as well as the actual type of the feedback.  Some of the feedback
>> messages also uses the target SSRC field in the header to identify
>> which
>>
>>
>>
>> Name                      Expires July 31, 2017
>> [Page 5]
>>
>> Internet-Draft              Abbreviated-Title               January
>> 2017
>>
>>
>> RTP stream the feedback is related to.  For these types all the FCI
>> entries, if multiple ones are forwarded to the identified handler.
>> Examples of this pattern are:
>>
>> Picture Loss Indication (PLI):  RFC 4585 (PT=PSFB, FMT=1).
>>
>> Slice Loss Indication (SLI):  RFC 4585 (PT=PSFB, FMT=2).
>>
>> Reference Picture Selection Indication (RPSI):  RFC 4585 (PT=PSFB,
>> FMT=3).
>>
>> Generic NACK:  [RFC4585] (PT=RTPFB and FMT=1).
>>
>> Other feedback messages includes the target SSRC in the Feedback
>> Control Information (FCI).  Here each FCI needs to be processed and
>> the SSRC field identified.  And the individual FCI combined with
>> the RTCP packet header context needs to be forwarded to the
>> identified handler.  Example of this pattern are:
>>
>> Full Intra Request (FIR):  [RFC5104] (PT=PSFB, FMT=4).
>>
>> Temporal-Spatial Trade-off Request (TSTR):  [RFC5104] (PT=PSFB,
>> FMT=5).
>>
>> Temporal-Spatial Trade-off Notification (TSTN):  This [RFC5104]
>> defined message (PT=PSFB, FMT=5) is actually a notifciation in
>> response to a TSTR this endpoint sent using the SSRC the FCI
>> identifies as source.
>>
>> H.271 Video Back Channel Message (VBCM):  [RFC5104] (PT=PSFB,
>> FMT=7).
>>
>> Layer Refresh Request (LRR):  [I-D.ietf-avtext-lrr] (PT=PSFB,
>> FMT=TBD).
>>
>> Descriptive or Notifications for an incoming RTP stream:  There
>> exist some RTCP packet types that provides additional information
>> or notifies about events related to the RTP stream identified.  In
>> these cases the RTP stream is identified using the SSRC field
>> value, and the information is provided to the higher layer
>> associated with the "m=" line for the incoming RTP stream as
>> identified by the current RTP stream to "m=" line table.  For this
>> type of pattern it is common that the RTCP packets and information
>> is repeated, either periodically (e.g.  SDES items), or for a
>> duration (e.g.  BYE), thus suppression of repetitions can be
>> considered.  Examples of these are:
>>
>>
>>
>>
>>
>> Name                      Expires July 31, 2017
>> [Page 6]
>>
>> Internet-Draft              Abbreviated-Title               January
>> 2017
>>
>>
>> Source Description (SDES) RTCP Packet:  The base RTP/RTCP protocol
>> [RFC3550] defines SDES RTCP packets as a way of providing per
>> source (SSRC) specific information about the source.  An SDES
>> packet contains zero or more chunks, where a chunk contains the
>> SSRC for the source being described by the one or more items
>> included in the chunk.  Forward the SDES items in each chunk to the
>> RTP stream's handler.
>>
>> Goodbye (BYE) RTCP Packet:  This RTP/RTCP protocol [RFC3550]
>> mechanism indicates that a particular RTP stream is leaving the RTP
>> session.  Thus, a most relevant event to inform the handler for
>> this RTP steam of.
>>
>> Third Party Targeted Reports or Feedback:  There exist some
>> multiparty RTP Topologies that results in that an endpoint receives
>> third party reports (SR [RFC3550], RR [RFC3550] or XR [RFC3611]) as
>> well as Feedback Messages [RFC4585] that relates to or targets an
>> SSRC that originates from another endpoint, and where the source of
>> the RTCP packet is also another endpoint. This type of packets
>> should be forwarded to the higher layer function dealing with the
>> third party reporting.  And if none exist then they can be
>> suppressed.  As the third party handler can be focused on
>> determining the conditions for the source of the reports and
>> feedback, or focused on how the RTP stream source progresses, or
>> both recommendations can't be made.
>>
>> APP Packets:  The RTCP APP Packets [RFC3550] are a mechanism that
>> enables experimentation.  The APP packet only specifies the source
>> of the packet.  Thus this information can be related to any of the
>> "m=" lines, thus deliver a copy of the packet to each "m=" line or
>> an APP specific handler.
>>
>> -- Cheers
>>
>> Magnus Westerlund
>>
>> ----------------------------------------------------------------------
>>
>>
Services, Media and Network features, Ericsson Research EAB/TXM
>> ----------------------------------------------------------------------
>>
>>
Ericsson AB                 | Phone  +46 10 7148287
>> Färögatan 6                 | Mobile +46 73 0949079 SE-164 80
>> Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>> ----------------------------------------------------------------------
>>
>>
>>
_______________________________________________
>> mmusic mailing list mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>>
>
>


-- 

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
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: Christer Holmberg <christer.holmberg@ericsson.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>, Jonathan Lennox <jonathan@vidyo.com>
Thread-Topic: [MMUSIC] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
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Date: Wed, 1 Feb 2017 19:43:33 +0000
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Subject: Re: [rtcweb] [MMUSIC] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
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Hi,

>> I have one question, however.  What should happen if an RTP packet=20
>> (without a MID) is received for a stream that has an existing mapping=20
>> to an m=3D line, but whose PT is not valid for that m=3D line?

I don't think that case is specific to BUNDLE, is it? It could happen even =
without multiplexing.

I agree with Magnus that it should be treated as an error.

(There could even be cases where the PT isn't valid for ANY m- line).

Regards,

Christer


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From: Jonathan Lennox <jonathan@vidyo.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: [MMUSIC] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
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Date: Wed, 1 Feb 2017 20:02:25 +0000
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References: <1f020dc6-1ac8-b71b-aee4-a711d15f1588@ericsson.com> <D2A63CD5-61AB-47DD-B946-6A17A94F76E7@vidyo.com> <e679714c-1f09-1d01-2440-b61c7e850370@ericsson.com>
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Subject: Re: [rtcweb] [MMUSIC] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
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> On Feb 1, 2017, at 10:21 AM, Magnus Westerlund <magnus.westerlund@ericsso=
n.com> wrote:
>=20
> Hi Jonathan,
>=20
> Thanks for the feedback. See inline for my attempt to answer them.
>=20
> Den 2017-01-31 kl. 18:43, skrev Jonathan Lennox:
>> In general, this looks good.
>>=20
>> I have one question, however.  What should happen if an RTP packet
>> (without a MID) is received for a stream that has an existing mapping
>> to an m=3D line, but whose PT is not valid for that m=3D line?
>>=20
>=20
> So, the text in JSEP-18 didn't specify this either. However, I do have an=
 opinion on this.
>=20
>> Should it 1. Cause the stream=92s mapping to be moved to another m=3D
>> line, if the received PT is in the payload type table for some other
>> m=3D line? or
>=20
> I would recommend against this due to that we will have to go into under =
which circumstances that one can accept this, and when not. Bo Burman and I=
 had some discussion on this. We arrived at some relevant aspects of this. =
The first is that one have one to consider precedence rules between PTs, as=
 well as a=3Dssrc and MID. In general we think mismatch between the informa=
tion is to be considered an error. However, using PT and MID could cause st=
range error cases, where packets without MID ends up in on m=3D line with t=
he PT and the ones with MID are dropped as inconsistent. I would claim that=
 this stream is always in error.
>=20
> I would also note that Bundle specification currently forbids PT based sw=
itching, but for a different reasons. Section 10.1 says:
>=20
>   o  A given SSRC MUST NOT transmit RTP packets using payload types
>      that originate from different bundled "m=3D" lines.
>=20
>   NOTE: The last bullet above is to avoid sending multiple media types
>   from the same SSRC.  If transmission of multiple media types are done
>   with time overlap, RTP and RTCP fail to function.  Even if done in
>   proper sequence this causes RTP Timestamp rate switching issues
>   [RFC7160].  However, once an SSRC has left the RTP session (by
>   sending an RTCP BYE packet), that SSRC can be reused by another
>   source (possibly associated with a different bundled "m=3D" line) after
>   a delay of 5 RTCP reporting intervals (the delay is to ensure the
>   SSRC has timed out, in case the RTCP BYE packet was lost [RFC3550]).
>=20
> But, we do note that the limitation is not strictly necessary in regards =
to the motivation. The minimal limitation based on the motivation would be =
to require that a SSRC only uses PTs of the same media type and clock rate =
within a single bundle group. Using shared PTs, which the current limitatio=
n requires automatically meets that goal, and is easier to check for.

Okay, that=92s reasonable, I think.

> We also noted that BUNDLE should clarify how MID relates to a=3Dssrc (RFC=
5576). My interpretation is that a=3Dssrc locks a RTP stream to a particula=
r m=3D line, and can't be moved. So any MID value other than that matches t=
he m=3D line that has the a=3Dssrc value would be an error case. For usages=
 that intend to have a RTP stream to bounce between m=3D lines should not u=
se a=3Dssrc, only MID. I think it would be good to clarify this in the BUND=
LE specification.

That seems reasonable, but it=92s not what your proposed algorithm does, as=
 far as I can tell.  Receiving a MID for a different m=3D line moves the so=
urce, regardless of how the source initially ended up associated.=20


> > 2. Be an error?
>=20
> This would be my preferred choice.
>=20
>>=20
>> In other words, should it be possible for PT-mapping to cause streams
>> to change their mappings?
>=20
> To conclude: No, as it would cause several additional complexities and po=
ssible branches in the algorithm. The use case for moving SSRCs between m=
=3D lines are supported when using MID, and the m=3D lines are correctly co=
nfigured with shared PTs.
>=20
> Cheers
>=20
> Magnus
>=20
>>=20
>> (Should this decision depend on what caused the stream to be mapped
>> to its current m=3D line?)
>>=20
>>> On Jan 27, 2017, at 11:43 AM, Magnus Westerlund
>>> <magnus.westerlund@ericsson.com> wrote:
>>>=20
>>> MMUSIC and RTCWEB,
>>>=20
>>> Here is now a more complete text proposal that also considers the
>>> RTCP. It also goes further than what Appendix B defines in one
>>> aspect, namely explicitly considering third party RTCP reporting
>>> and what to do with it.
>>>=20
>>> Feedback is much appreciated. I plan to put this into a PR towards
>>> JSEP APPENDIX B on monday. If only to get the JSEP authors
>>> attention ;-). However, the text is really intended for the next
>>> version of BUNDLE.
>>>=20
>>>=20
>>> X.  Associating RTP/RTCP With Correct SDP Media Description
>>>=20
>>> As described in [RFC3550], RTP packets are associated with RTP
>>> streams [RFC7656].  Each RTP stream is identified by an SSRC
>>> value, and each RTP packet carries an SSRC value that is used to
>>> associate the packet with the correct RTP stream.  RTCP packets
>>> also uses SSRCs to identify on which RTP streams any report or
>>> feedback relate to. Thus, an RTCP packet will commonly carry
>>> multiple SSRC values, and might therefore be providing feedback or
>>> report on multiple RTP streams.
>>>=20
>>> In order to be able to process received RTP/RTCP packets correctly
>>> it must be possible to associate an RTP stream with the correct
>>> "m=3D" line, as the "m=3D" line and SDP attributes associated with the
>>> "m=3D" line contain information needed to process the packets.
>>>=20
>>> As all RTP streams associated with a BUNDLE group are part of the
>>> same RTP session and using the same address:port combination for
>>> sending and receiving RTP/RTCP packets, the local address:port
>>> combination cannot be used to associate an RTP stream with the
>>> correct "m=3D" line.  In addition, multiple RTP streams might be
>>> associated with the same "m=3D" line.
>>>=20
>>> Also, as described in Section 10.1.1, the same payload type value
>>> might be used by multiple RTP streams, in which case the payload
>>> type value cannot be used to associate an RTP stream with the
>>> correct "m=3D" line.  However, there are cases where each "m=3D" line
>>> has unique payload type values, and then the payload type could
>>> serve as indicator to the relevant "m=3D" line the RTP stream is
>>> associated with.
>>>=20
>>> An offerer and answerer can inform each other which SSRC values
>>> they will use for an RTP stream by using the SDP 'ssrc' attribute
>>> [RFC5576].  However, an offerer will not know which SSRC values
>>> the answerer will use until the offerer has received the answer
>>> providing
>>>=20
>>>=20
>>>=20
>>> Name                      Expires July 31, 2017
>>> [Page 2]
>>>=20
>>> Internet-Draft              Abbreviated-Title               January
>>> 2017
>>>=20
>>>=20
>>> that information.  Due to this, before the offerer has received
>>> the answer, the offerer will not be able to associate an RTP stream
>>> with the correct "m=3D" line using the SSRC value associated with the
>>> RTP stream.  In addition, the offerer and answerer may start using
>>> new SSRC values mid-session, without informing each other using the
>>> SDP 'ssrc' attribute.
>>>=20
>>> In order for an offerer and answerer to always be able to
>>> associate an RTP stream with the correct "m=3D" line, the offerer and
>>> answerer using the BUNDLE extension MUST support the mechanism
>>> defined in Section 14, where the offerer and answerer includes the
>>> identification-tag (provided by the remote peer) associated with
>>> an "m=3D" line in the RTP Streams and in RTCP SDES packets part of a
>>> BUNDLE group.
>>>=20
>>> The mapping from an SSRC to an identification-tag is carried in
>>> RTCP SDES packets or in RTP header extensions (Section 14).  Since
>>> a compound RTCP packet can contain multiple RTCP SDES packets, and
>>> each RTCP SDES packet can contain multiple chunks, an RTCP packet
>>> can contain several SSRC to identification-tag mappings.  The
>>> offerer and answerer maintain tables mapping RTP streams identified
>>> by SSRC to "m=3D" lines identified by the identification-tag.  These
>>> tables are updated each time new information that affects how
>>> packets should be processed and routed are received.
>>>=20
>>> To prepare for demultiplexing RTP streams to the correct "m=3D"
>>> line, the following steps MUST be followed for each BUNDLE group
>>> based on the SDP signalling information.
>>>=20
>>> Construct a table mapping MID to "m=3D" line for each "m=3D" line in
>>> this BUNDLE group.  Note that an "m=3D" line may only have one MID.
>>>=20
>>> Construct a table mapping incoming RTP streams (SSRCs) to their
>>> "m=3D" line for each "m=3D" line in this BUNDLE group and for each RTP
>>> stream explicitly signalled for receiving in that "m=3D" line.
>>>=20
>>> Construct a table mapping payload types to "m=3D" line for each "m=3D"
>>> line in the BUNDLE group and for each payload type configured for
>>> receiving in that "m=3D" line.  If any payload type is configured for
>>> receiving in more than one "m=3D" line in the BUNDLE group, do not it
>>> include it in the table.
>>>=20
>>> Note that for each of these tables, there can only be one mapping
>>> for any given key (MID, SSRC, or PT).  In other words, the tables
>>> are not multimaps.
>>>=20
>>> As "m=3D" lines are added or removed from the BUNDLE groups, or
>>> their configurations are changed, the tables above MUST also be
>>> updated.
>>>=20
>>>=20
>>>=20
>>> Name                      Expires July 31, 2017
>>> [Page 3]
>>>=20
>>> Internet-Draft              Abbreviated-Title               January
>>> 2017
>>>=20
>>>=20
>>> Received RTP packets that are syntactically correct are processed
>>> by the RTP/RTCP protocol implementation for statistics etc.  After
>>> this processing they need to be routed to the higher layer context
>>> associated with the "m=3D" line within the BUNDLE group.  Somewhere
>>> in the process where an received RTP packet is processed to be
>>> delivered to the higher layer by RTP the matching step below MUST
>>> be performed:
>>>=20
>>> 1.  Reception of an RTP packet for an RTP stream that has an
>>> existing mapping in the RTP stream to m=3D line table.  Before
>>> proceeding in delivering the packet to the higher layer context
>>> according to the RTP stream to "m=3D" line mapping table the
>>> following checks MUST be performed:
>>>=20
>>> A.  If the packet carries an RTP header extension with a SDES MID
>>> value that is not in the table mapping MID to "m=3D" line, then do
>>> not deliver the RTP packet to higher layers.
>>>=20
>>> B.  If the packet carries an RTP header extension with a SDES MID
>>> value that is in the table mapping MID to "m=3D" line, and the value
>>> indicates a different "m=3D" line than the current RTP stream to "m=3D"
>>> line mapping table, then update the RTP stream to "m=3D" line
>>> mapping.
>>>=20
>>> 2.  Reception of an RTP packet for an RTP stream that has no
>>> existing mapping to an m=3D line.  In this case the following actions
>>> MUST be performed:
>>>=20
>>> A.  If the packet carries an RTP header extension with a SDES MID
>>> value that is in the table mapping MID to "m=3D" line, then create an
>>> entry in the RTP stream to "m=3D" line mapping table for this RTP
>>> stream (SSRC).  Then deliver the RTP packet to the "m=3D" line
>>> context of the created mapping and stop.
>>>=20
>>> B.  If the packet carries a Payload Type that is in the payload
>>> type table, then create an entry in the RTP stream to "m=3D" line
>>> mapping table for this RTP stream (SSRC).  Then deliver the RTP
>>> packet to the "m=3D" line context of the created mapping and stop.
>>>=20
>>> C.  Otherwise do not deliver the RTP packet to higher layers. Note,
>>> this includes unknown MID values.
>>>=20
>>> For each RTCP packet received (including each RTCP packet that is
>>> part of a compound RTCP packet), the RTCP packet needs to be
>>> processed by the RTP/RTCP implementation and relevant information
>>> and data from the RTCP packets needs to be routed to the
>>> appropriate handler for the related RTP streams.  The appropriate
>>> handler is determined by using the RTP stream to "m=3D" line mapping
>>> table.
>>>=20
>>>=20
>>>=20
>>> Name                      Expires July 31, 2017
>>> [Page 4]
>>>=20
>>> Internet-Draft              Abbreviated-Title               January
>>> 2017
>>>=20
>>>=20
>>> On reception of any compound RTCP packet prior to dispatching the
>>> received information and data, if there is an RTCP SDES packet
>>> included that SHOULD be processed first.  If that SDES packet
>>> contains SDES MID entries, this can results in updates and
>>> additions to the RTP stream to "m=3D" line mapping table.  Thus each
>>> of the SDES MID items are processed and the current table entries
>>> are checked if the corresponding MID value matches the current RTP
>>> stream to "m=3D" line mapping, else the entry is updated.  If there
>>> is no RTP stream to "m=3D" line table mapping entry for the received
>>> SDES item's SSRC, such an entry is created.  Note, that in the
>>> process of updating the table entries, update flap suppression as
>>> discussed in Section 4.2.6 of [RFC7941] should be considered.
>>>=20
>>> The various different RTCP packets as well as their various sub
>>> parts, such as the various RTCP Feedback message types, relates to
>>> the RTP streams in a couple of different ways.  The currently
>>> known patterns are the following:
>>>=20
>>> Reports on outgoing RTP streams:  For all RTP streams that this
>>> endpoint is the source of, it can expect to receive report blocks
>>> of several types identified as relating to an outgoing stream. The
>>> basic pattern for these blocks are that the RTCP packet header
>>> identifies the source of the reports, as identified by an SSRC, and
>>> containing one or more report blocks, where each report block
>>> identifies the RTP stream, using the SSRC, the report relates to.
>>> For this pattern the relevant report information is provided to the
>>> higher layer associated with the "m=3D" line the RTP Stream to "m=3D"
>>> line table identifies.  The source SSRC as identifier of the
>>> endpoint that the report originates are relevant for interpreting
>>> the information, but not necessarily for routing.  Example of this
>>> is pattern are:
>>>=20
>>> Sender Report (SR) and Receiver Report (RR)  The basic receiver
>>> report blocks from RFC3550 start with the SSRC of the RTP stream
>>> they report on.
>>>=20
>>> Extended Reports (XR):  RFC3611 is a framework that enables a large
>>> number of different reports.  However, a large number of these
>>> report formats are reporting on specific RTP streams and thus each
>>> individual report of these types contains a SSRC field to identify
>>> the RTP stream.
>>>=20
>>> RTCP Feedback Messages for outgoing RTP streams:  The RFC 4585
>>> RTCP feedback messages allow for a number of different type of
>>> feedback messages.  However, the RTCP feedback message header
>>> contains the SSRC identifying the source of the feedback messages
>>> as well as the actual type of the feedback.  Some of the feedback
>>> messages also uses the target SSRC field in the header to identify
>>> which
>>>=20
>>>=20
>>>=20
>>> Name                      Expires July 31, 2017
>>> [Page 5]
>>>=20
>>> Internet-Draft              Abbreviated-Title               January
>>> 2017
>>>=20
>>>=20
>>> RTP stream the feedback is related to.  For these types all the FCI
>>> entries, if multiple ones are forwarded to the identified handler.
>>> Examples of this pattern are:
>>>=20
>>> Picture Loss Indication (PLI):  RFC 4585 (PT=3DPSFB, FMT=3D1).
>>>=20
>>> Slice Loss Indication (SLI):  RFC 4585 (PT=3DPSFB, FMT=3D2).
>>>=20
>>> Reference Picture Selection Indication (RPSI):  RFC 4585 (PT=3DPSFB,
>>> FMT=3D3).
>>>=20
>>> Generic NACK:  [RFC4585] (PT=3DRTPFB and FMT=3D1).
>>>=20
>>> Other feedback messages includes the target SSRC in the Feedback
>>> Control Information (FCI).  Here each FCI needs to be processed and
>>> the SSRC field identified.  And the individual FCI combined with
>>> the RTCP packet header context needs to be forwarded to the
>>> identified handler.  Example of this pattern are:
>>>=20
>>> Full Intra Request (FIR):  [RFC5104] (PT=3DPSFB, FMT=3D4).
>>>=20
>>> Temporal-Spatial Trade-off Request (TSTR):  [RFC5104] (PT=3DPSFB,
>>> FMT=3D5).
>>>=20
>>> Temporal-Spatial Trade-off Notification (TSTN):  This [RFC5104]
>>> defined message (PT=3DPSFB, FMT=3D5) is actually a notifciation in
>>> response to a TSTR this endpoint sent using the SSRC the FCI
>>> identifies as source.
>>>=20
>>> H.271 Video Back Channel Message (VBCM):  [RFC5104] (PT=3DPSFB,
>>> FMT=3D7).
>>>=20
>>> Layer Refresh Request (LRR):  [I-D.ietf-avtext-lrr] (PT=3DPSFB,
>>> FMT=3DTBD).
>>>=20
>>> Descriptive or Notifications for an incoming RTP stream:  There
>>> exist some RTCP packet types that provides additional information
>>> or notifies about events related to the RTP stream identified.  In
>>> these cases the RTP stream is identified using the SSRC field
>>> value, and the information is provided to the higher layer
>>> associated with the "m=3D" line for the incoming RTP stream as
>>> identified by the current RTP stream to "m=3D" line table.  For this
>>> type of pattern it is common that the RTCP packets and information
>>> is repeated, either periodically (e.g.  SDES items), or for a
>>> duration (e.g.  BYE), thus suppression of repetitions can be
>>> considered.  Examples of these are:
>>>=20
>>>=20
>>>=20
>>>=20
>>>=20
>>> Name                      Expires July 31, 2017
>>> [Page 6]
>>>=20
>>> Internet-Draft              Abbreviated-Title               January
>>> 2017
>>>=20
>>>=20
>>> Source Description (SDES) RTCP Packet:  The base RTP/RTCP protocol
>>> [RFC3550] defines SDES RTCP packets as a way of providing per
>>> source (SSRC) specific information about the source.  An SDES
>>> packet contains zero or more chunks, where a chunk contains the
>>> SSRC for the source being described by the one or more items
>>> included in the chunk.  Forward the SDES items in each chunk to the
>>> RTP stream's handler.
>>>=20
>>> Goodbye (BYE) RTCP Packet:  This RTP/RTCP protocol [RFC3550]
>>> mechanism indicates that a particular RTP stream is leaving the RTP
>>> session.  Thus, a most relevant event to inform the handler for
>>> this RTP steam of.
>>>=20
>>> Third Party Targeted Reports or Feedback:  There exist some
>>> multiparty RTP Topologies that results in that an endpoint receives
>>> third party reports (SR [RFC3550], RR [RFC3550] or XR [RFC3611]) as
>>> well as Feedback Messages [RFC4585] that relates to or targets an
>>> SSRC that originates from another endpoint, and where the source of
>>> the RTCP packet is also another endpoint. This type of packets
>>> should be forwarded to the higher layer function dealing with the
>>> third party reporting.  And if none exist then they can be
>>> suppressed.  As the third party handler can be focused on
>>> determining the conditions for the source of the reports and
>>> feedback, or focused on how the RTP stream source progresses, or
>>> both recommendations can't be made.
>>>=20
>>> APP Packets:  The RTCP APP Packets [RFC3550] are a mechanism that
>>> enables experimentation.  The APP packet only specifies the source
>>> of the packet.  Thus this information can be related to any of the
>>> "m=3D" lines, thus deliver a copy of the packet to each "m=3D" line or
>>> an APP specific handler.
>>>=20
>>> -- Cheers
>>>=20
>>> Magnus Westerlund
>>>=20
>>> ----------------------------------------------------------------------
>>>=20
>>>=20
> Services, Media and Network features, Ericsson Research EAB/TXM
>>> ----------------------------------------------------------------------
>>>=20
>>>=20
> 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
>>>=20
>>>=20
> _______________________________________________
>>> mmusic mailing list mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>=20
>>=20
>>=20
>=20
>=20
> --=20
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> 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


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From: Jonathan Lennox <jonathan@vidyo.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
Thread-Topic: [MMUSIC] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
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Subject: Re: [rtcweb] [MMUSIC] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Jonathan Lennox <jonathan@vidyo.com>
Thread-Topic: [MMUSIC] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
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Subject: Re: [rtcweb] [MMUSIC] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: mmusic <mmusic@ietf.org>, Jonathan Lennox <jonathan@vidyo.com>, "draft-ietf-rtcweb-jsep@tools.ietf.org" <draft-ietf-rtcweb-jsep@tools.ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "draft-ietf-mmusic-sdp-bundle-negotiation@ietf.org" <draft-ietf-mmusic-sdp-bundle-negotiation@ietf.org>
Subject: Re: [rtcweb] [MMUSIC] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
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*Any* RTP packet routed to an m-line needs to have a PT field that is valid f=
or that m-line. This is true whether the routing was done via MID match, SSR=
C match or PT match.=20

Also, Magnus mentioned a case where a BYE is received for an SSRC then subse=
quently the SSRC is reused, potentially with a new MID and/or PT.

Does the proposed text handle this? I do not think so. SSRC entries that are=
 "latched" need to be removed when a BYE is received.

> On Feb 1, 2017, at 11:43 AM, Christer Holmberg <christer.holmberg@ericsson=
.com> wrote:
>=20
> Hi,
>=20
>>> I have one question, however.  What should happen if an RTP packet=20
>>> (without a MID) is received for a stream that has an existing mapping=20=

>>> to an m=3D line, but whose PT is not valid for that m=3D line?
>=20
> I don't think that case is specific to BUNDLE, is it? It could happen even=
 without multiplexing.
>=20
> I agree with Magnus that it should be treated as an error.
>=20
> (There could even be cases where the PT isn't valid for ANY m- line).
>=20
> Regards,
>=20
> Christer
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [rtcweb] [MMUSIC] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
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Den 2017-02-01 kl. 21:02, skrev Jonathan Lennox:
>
>> On Feb 1, 2017, at 10:21 AM, Magnus Westerlund
>> <magnus.westerlund@ericsson.com> wrote:

>> We also noted that BUNDLE should clarify how MID relates to a=ssrc
>> (RFC5576). My interpretation is that a=ssrc locks a RTP stream to a
>> particular m= line, and can't be moved. So any MID value other than
>> that matches the m= line that has the a=ssrc value would be an
>> error case. For usages that intend to have a RTP stream to bounce
>> between m= lines should not use a=ssrc, only MID. I think it would
>> be good to clarify this in the BUNDLE specification.
>
> That seems reasonable, but it’s not what your proposed algorithm
> does, as far as I can tell.  Receiving a MID for a different m= line
> moves the source, regardless of how the source initially ended up
> associated.
>

No, this was something that was realized in the discussion that I had 
with Bo after your question. And, I think a possible way to realize this 
is to add to the entries in the table RTP stream to "m=" line a property 
"set by a=ssrc" that makes the entry immutable, with the exception for 
SDP signalling updates. That way one can also handle if RTCP BYE should 
after a timeout remove the entry from this table, which is what Bernard 
asked about.

Cheers

Magnus Westerlund

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Subject: Re: [rtcweb] [MMUSIC] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
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Den 2017-02-01 kl. 21:41, skrev Bernard Aboba:
> *Any* RTP packet routed to an m-line needs to have a PT field that is
> valid for that m-line. This is true whether the routing was done via
> MID match, SSRC match or PT match.

Agreed.

>
> Also, Magnus mentioned a case where a BYE is received for an SSRC
> then subsequently the SSRC is reused, potentially with a new MID
> and/or PT.
>
> Does the proposed text handle this? I do not think so. SSRC entries
> that are "latched" need to be removed when a BYE is received.
>

Yes, that is not covered here. I think this could be clarified. There 
needs to exist a timeout before the entry is actually removed, to avoid 
rebinding in case of packet re-order. Thus the regular SSRC timeout 
should be sufficient for this, i.e. 5*Td (using timeout mins). However, 
I think also this case needs to consider a=ssrc. If a=ssrc is set, then 
the RTP stream to m= line mapping is maintained, until signalling change.

Cheers

Magnus Westerlund

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From: Roni Even <roni.even@huawei.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>, "mmusic (E-mail)" <mmusic@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "draft-ietf-mmusic-sdp-bundle-negotiation@ietf.org" <draft-ietf-mmusic-sdp-bundle-negotiation@ietf.org>, "draft-ietf-rtcweb-jsep@tools.ietf.org" <draft-ietf-rtcweb-jsep@tools.ietf.org>
Thread-Topic: [rtcweb] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
Thread-Index: AQHSeLyNdlGsubKC20y06tuA5xxgmqFVs8jg
Date: Thu, 2 Feb 2017 12:49:12 +0000
Message-ID: <6E58094ECC8D8344914996DAD28F1CCD7716D7@DGGEMM506-MBX.china.huawei.com>
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Archived-At: <https://mailarchive.ietf.org/arch/msg/rtcweb/36_1DI16mmmlvMR4J3DKQj6aeGQ>
Subject: Re: [rtcweb] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
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From nobody Thu Feb  2 14:20:06 2017
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From: Cullen Jennings <fluffy@iii.ca>
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To: Jonathan Lennox <jonathan@vidyo.com>
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Subject: Re: [rtcweb] [MMUSIC] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
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> On Jan 31, 2017, at 10:43 AM, Jonathan Lennox <jonathan@vidyo.com> =
wrote:
>=20
> In general, this looks good.
>=20
> I have one question, however.  What should happen if an RTP packet =
(without a MID) is received for a stream that has an existing mapping to =
an m=3D line, but whose PT is not valid for that m=3D line?
>=20
> Should it
> 1. Cause the stream=E2=80=99s mapping to be moved to another m=3D =
line, if the received PT is in the payload type table for some other m=3D =
line?
> or=20
> 2. Be an error?


Imagine a case where we have two sends call A and B and they each use =
one PT that is unique and go to different m lines on receiver C via a =
SFU.  Initially C get a packet from B with a given SSRC and the PT and =
creates a mapping for it. But imagine that A and B had an SSRC collision =
which they sort that out and A ends up having the SSRC that B initially =
used. If you don't have the PT override the SSRC, the packets are going =
to go to the wrong m-line because the PT points at the m-line A is using =
but the old SSRC incorrectly points at the m-line B is using.=20

To put this a different way, if the packet has a mid, I think it is very =
clear the mid has to override the PT. Same for the PT. The mid is =
effectively just an extended PT.=20

I'm not thinking to much about the case where SSRC is signalled in SDP =
but ignoring that case for a second, I think the really easy thing to =
specify, implement, and understand works is simply a priority order =
where roughly=20

1) if you have mid, use that and latch the SSRC
2) if you have a unique pt, use that and latch the SSRC=20
3) if you have a latched SSRC use that=20

If you had signaled SSRC, guess I would insert that as step 1.5 of use =
that=20


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: [rtcweb] [MMUSIC] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
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Date: Thu, 2 Feb 2017 22:19:52 +0000
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Subject: Re: [rtcweb] [MMUSIC] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 2 Feb 2017 23:25:14 +0100
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Subject: Re: [rtcweb] [MMUSIC] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
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2017-02-02 23:19 GMT+01:00 Cullen Jennings <fluffy@iii.ca>:
> But imagine that A and B had an SSRC collision

Very likely :)


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>


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Subject: Re: [rtcweb] [MMUSIC] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
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Assuming the collision is detected and B sends a BYE then once the SSRC latc=
h is removed then A's packets will match on PT, a new SSRC latch is put in p=
lace and everything works.

> On Feb 2, 2017, at 2:19 PM, Cullen Jennings <fluffy@iii.ca> wrote:
>=20
>=20
>> On Jan 31, 2017, at 10:43 AM, Jonathan Lennox <jonathan@vidyo.com> wrote:=

>>=20
>> In general, this looks good.
>>=20
>> I have one question, however.  What should happen if an RTP packet (witho=
ut a MID) is received for a stream that has an existing mapping to an m=3D l=
ine, but whose PT is not valid for that m=3D line?
>>=20
>> Should it
>> 1. Cause the stream=E2=80=99s mapping to be moved to another m=3D line, i=
f the received PT is in the payload type table for some other m=3D line?
>> or=20
>> 2. Be an error?
>=20
>=20
> Imagine a case where we have two sends call A and B and they each use one P=
T that is unique and go to different m lines on receiver C via a SFU.  Initi=
ally C get a packet from B with a given SSRC and the PT and creates a mappin=
g for it. But imagine that A and B had an SSRC collision which they sort tha=
t out and A ends up having the SSRC that B initially used. If you don't have=
 the PT override the SSRC, the packets are going to go to the wrong m-line b=
ecause the PT points at the m-line A is using but the old SSRC incorrectly p=
oints at the m-line B is using.=20
>=20
> To put this a different way, if the packet has a mid, I think it is very c=
lear the mid has to override the PT. Same for the PT. The mid is effectively=
 just an extended PT.=20
>=20
> I'm not thinking to much about the case where SSRC is signalled in SDP but=
 ignoring that case for a second, I think the really easy thing to specify, i=
mplement, and understand works is simply a priority order where roughly=20
>=20
> 1) if you have mid, use that and latch the SSRC
> 2) if you have a unique pt, use that and latch the SSRC=20
> 3) if you have a latched SSRC use that=20
>=20
> If you had signaled SSRC, guess I would insert that as step 1.5 of use tha=
t=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
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Subject: Re: [rtcweb] [MMUSIC] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
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On 02/02/2017 23:19, Cullen Jennings wrote:
>
> Imagine a case where we have two sends call A and B and they each use one PT that is unique and go to different m lines on receiver C via a SFU.  Initially C get a packet from B with a given SSRC and the PT and creates a mapping for it. But imagine that A and B had an SSRC collision which they sort that out and A ends up having the SSRC that B initially used. If you don't have the PT override the SSRC, the packets are going to go to the wrong m-line because the PT points at the m-line A is using but the old SSRC incorrectly points at the m-line B is using.
>

Sorry if I am late to the party and I say something stupid, but 
shouldn't the SFU terminate RTP/RTCP streams and rewrite PT, packet 
numbers *AND* SSRC? so the SFU SHALL ensure in this case that there is 
no SSRC collision on C.

Best regards
Sergio


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Subject: Re: [rtcweb] [MMUSIC] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
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--94eb2c07664caaa47e05479deffc
Content-Type: text/plain; charset=UTF-8

I always thought of the MID as a port extension, since without bundle the
port would identify the m= line. It that case would the priority be:
1) MID
2) latched SSRC
3) signalled SSRC
4) unique PT

On 3 February 2017 at 08:46, Sergio Garcia Murillo <
sergio.garcia.murillo@gmail.com> wrote:

> On 02/02/2017 23:19, Cullen Jennings wrote:
>
>>
>> Imagine a case where we have two sends call A and B and they each use one
>> PT that is unique and go to different m lines on receiver C via a SFU.
>> Initially C get a packet from B with a given SSRC and the PT and creates a
>> mapping for it. But imagine that A and B had an SSRC collision which they
>> sort that out and A ends up having the SSRC that B initially used. If you
>> don't have the PT override the SSRC, the packets are going to go to the
>> wrong m-line because the PT points at the m-line A is using but the old
>> SSRC incorrectly points at the m-line B is using.
>>
>>
> Sorry if I am late to the party and I say something stupid, but shouldn't
> the SFU terminate RTP/RTCP streams and rewrite PT, packet numbers *AND*
> SSRC? so the SFU SHALL ensure in this case that there is no SSRC collision
> on C.
>
> Best regards
> Sergio
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--94eb2c07664caaa47e05479deffc
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<div dir=3D"ltr">I always thought of the MID as a port extension, since wit=
hout bundle the port would identify the m=3D line. It that case would the p=
riority be:<div>1) MID</div><div>2) latched SSRC</div><div>3) signalled SSR=
C</div><div>4) unique PT</div></div><div class=3D"gmail_extra"><br><div cla=
ss=3D"gmail_quote">On 3 February 2017 at 08:46, Sergio Garcia Murillo <span=
 dir=3D"ltr">&lt;<a href=3D"mailto:sergio.garcia.murillo@gmail.com" target=
=3D"_blank">sergio.garcia.murillo@gmail.com</a>&gt;</span> wrote:<br><block=
quote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc=
 solid;padding-left:1ex"><span class=3D"">On 02/02/2017 23:19, Cullen Jenni=
ngs wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
Imagine a case where we have two sends call A and B and they each use one P=
T that is unique and go to different m lines on receiver C via a SFU.=C2=A0=
 Initially C get a packet from B with a given SSRC and the PT and creates a=
 mapping for it. But imagine that A and B had an SSRC collision which they =
sort that out and A ends up having the SSRC that B initially used. If you d=
on&#39;t have the PT override the SSRC, the packets are going to go to the =
wrong m-line because the PT points at the m-line A is using but the old SSR=
C incorrectly points at the m-line B is using.<br>
<br>
</blockquote>
<br></span>
Sorry if I am late to the party and I say something stupid, but shouldn&#39=
;t the SFU terminate RTP/RTCP streams and rewrite PT, packet numbers *AND* =
SSRC? so the SFU SHALL ensure in this case that there is no SSRC collision =
on C.<br>
<br>
Best regards<span class=3D"HOEnZb"><font color=3D"#888888"><br>
Sergio</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
______________________________<wbr>_________________<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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/rtcweb</a><br=
>
</div></div></blockquote></div><br></div>

--94eb2c07664caaa47e05479deffc--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>, Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: [rtcweb] [MMUSIC] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
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Cc: "draft-ietf-rtcweb-jsep@tools.ietf.org" <draft-ietf-rtcweb-jsep@tools.ietf.org>, RTCWeb IETF <rtcweb@ietf.org>, "mmusic \(E-mail\)" <mmusic@ietf.org>, "draft-ietf-mmusic-sdp-bundle-negotiation@ietf.org" <draft-ietf-mmusic-sdp-bundle-negotiation@ietf.org>
Subject: Re: [rtcweb] [MMUSIC] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
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Hi,

>2) both the proposed and current text seem lacking in dealing with
>multiple bundle groups

No matter what text we move forward with, I think it shall be explicitly
said that the procedures are per IP address:port. So, if you have multiple
bundle groups, you will apply the procedures to each group, independent of
each other.

Regards,

Christer

>
>> On Jan 30, 2017, at 6:18 AM, Magnus Westerlund
>><magnus.westerlund@ericsson.com> wrote:
>>=20
>> Hi,
>>=20
>> I have now generated a pull request for this text, with some very minor
>>changes.
>>=20
>> https://github.com/rtcweb-wg/jsep/pull/538
>>=20
>> Cheers
>>=20
>> Magnus
>>=20
>> Den 2017-01-27 kl. 17:43, skrev Magnus Westerlund:
>>> MMUSIC and RTCWEB,
>>>=20
>>> Here is now a more complete text proposal that also considers the RTCP.
>>> It also goes further than what Appendix B defines in one aspect, namely
>>> explicitly considering third party RTCP reporting and what to do with
>>>it.
>>>=20
>>> Feedback is much appreciated. I plan to put this into a PR towards JSEP
>>> APPENDIX B on monday. If only to get the JSEP authors attention ;-).
>>> However, the text is really intended for the next version of BUNDLE.
>>>=20
>>>=20
>>> X.  Associating RTP/RTCP With Correct SDP Media Description
>>>=20
>>>   As described in [RFC3550], RTP packets are associated with RTP
>>>   streams [RFC7656].  Each RTP stream is identified by an SSRC value,
>>>   and each RTP packet carries an SSRC value that is used to associate
>>>   the packet with the correct RTP stream.  RTCP packets also uses SSRCs
>>>   to identify on which RTP streams any report or feedback relate to.
>>>   Thus, an RTCP packet will commonly carry multiple SSRC values, and
>>>   might therefore be providing feedback or report on multiple RTP
>>>   streams.
>>>=20
>>>   In order to be able to process received RTP/RTCP packets correctly it
>>>   must be possible to associate an RTP stream with the correct "m=3D"
>>>   line, as the "m=3D" line and SDP attributes associated with the "m=3D=
"
>>>   line contain information needed to process the packets.
>>>=20
>>>   As all RTP streams associated with a BUNDLE group are part of the
>>>   same RTP session and using the same address:port combination for
>>>   sending and receiving RTP/RTCP packets, the local address:port
>>>   combination cannot be used to associate an RTP stream with the
>>>   correct "m=3D" line.  In addition, multiple RTP streams might be
>>>   associated with the same "m=3D" line.
>>>=20
>>>   Also, as described in Section 10.1.1, the same payload type value
>>>   might be used by multiple RTP streams, in which case the payload type
>>>   value cannot be used to associate an RTP stream with the correct "m=
=3D"
>>>   line.  However, there are cases where each "m=3D" line has unique
>>>   payload type values, and then the payload type could serve as
>>>   indicator to the relevant "m=3D" line the RTP stream is associated
>>>   with.
>>>=20
>>>   An offerer and answerer can inform each other which SSRC values they
>>>   will use for an RTP stream by using the SDP 'ssrc' attribute
>>>   [RFC5576].  However, an offerer will not know which SSRC values the
>>>   answerer will use until the offerer has received the answer providing
>>>=20
>>>=20
>>>=20
>>> Name                      Expires July 31, 2017                 [Page
>>>2]
>>>=20
>>> Internet-Draft              Abbreviated-Title               January
>>>2017
>>>=20
>>>=20
>>>   that information.  Due to this, before the offerer has received the
>>>   answer, the offerer will not be able to associate an RTP stream with
>>>   the correct "m=3D" line using the SSRC value associated with the RTP
>>>   stream.  In addition, the offerer and answerer may start using new
>>>   SSRC values mid-session, without informing each other using the SDP
>>>   'ssrc' attribute.
>>>=20
>>>   In order for an offerer and answerer to always be able to associate
>>>   an RTP stream with the correct "m=3D" line, the offerer and answerer
>>>   using the BUNDLE extension MUST support the mechanism defined in
>>>   Section 14, where the offerer and answerer includes the
>>>   identification-tag (provided by the remote peer) associated with an
>>>   "m=3D" line in the RTP Streams and in RTCP SDES packets part of a
>>>   BUNDLE group.
>>>=20
>>>   The mapping from an SSRC to an identification-tag is carried in RTCP
>>>   SDES packets or in RTP header extensions (Section 14).  Since a
>>>   compound RTCP packet can contain multiple RTCP SDES packets, and each
>>>   RTCP SDES packet can contain multiple chunks, an RTCP packet can
>>>   contain several SSRC to identification-tag mappings.  The offerer and
>>>   answerer maintain tables mapping RTP streams identified by SSRC to
>>>   "m=3D" lines identified by the identification-tag.  These tables are
>>>   updated each time new information that affects how packets should be
>>>   processed and routed are received.
>>>=20
>>>   To prepare for demultiplexing RTP streams to the correct "m=3D" line,
>>>   the following steps MUST be followed for each BUNDLE group based on
>>>   the SDP signalling information.
>>>=20
>>>      Construct a table mapping MID to "m=3D" line for each "m=3D" line =
in
>>>      this BUNDLE group.  Note that an "m=3D" line may only have one MID=
.
>>>=20
>>>      Construct a table mapping incoming RTP streams (SSRCs) to their
>>>      "m=3D" line for each "m=3D" line in this BUNDLE group and for each=
 RTP
>>>      stream explicitly signalled for receiving in that "m=3D" line.
>>>=20
>>>      Construct a table mapping payload types to "m=3D" line for each "m=
=3D"
>>>      line in the BUNDLE group and for each payload type configured for
>>>      receiving in that "m=3D" line.  If any payload type is configured
>>>      for receiving in more than one "m=3D" line in the BUNDLE group, do
>>>      not it include it in the table.
>>>=20
>>>   Note that for each of these tables, there can only be one mapping for
>>>   any given key (MID, SSRC, or PT).  In other words, the tables are not
>>>   multimaps.
>>>=20
>>>   As "m=3D" lines are added or removed from the BUNDLE groups, or their
>>>   configurations are changed, the tables above MUST also be updated.
>>>=20
>>>=20
>>>=20
>>> Name                      Expires July 31, 2017                 [Page
>>>3]
>>>=20
>>> Internet-Draft              Abbreviated-Title               January
>>>2017
>>>=20
>>>=20
>>>   Received RTP packets that are syntactically correct are processed by
>>>   the RTP/RTCP protocol implementation for statistics etc.  After this
>>>   processing they need to be routed to the higher layer context
>>>   associated with the "m=3D" line within the BUNDLE group.  Somewhere i=
n
>>>   the process where an received RTP packet is processed to be delivered
>>>   to the higher layer by RTP the matching step below MUST be performed:
>>>=20
>>>   1.  Reception of an RTP packet for an RTP stream that has an existing
>>>       mapping in the RTP stream to m=3D line table.  Before proceeding =
in
>>>       delivering the packet to the higher layer context according to
>>>       the RTP stream to "m=3D" line mapping table the following checks
>>>       MUST be performed:
>>>=20
>>>       A.  If the packet carries an RTP header extension with a SDES MID
>>>           value that is not in the table mapping MID to "m=3D" line, th=
en
>>>           do not deliver the RTP packet to higher layers.
>>>=20
>>>       B.  If the packet carries an RTP header extension with a SDES MID
>>>           value that is in the table mapping MID to "m=3D" line, and th=
e
>>>           value indicates a different "m=3D" line than the current RTP
>>>           stream to "m=3D" line mapping table, then update the RTP stre=
am
>>>           to "m=3D" line mapping.
>>>=20
>>>   2.  Reception of an RTP packet for an RTP stream that has no existing
>>>       mapping to an m=3D line.  In this case the following actions MUST
>>>       be performed:
>>>=20
>>>       A.  If the packet carries an RTP header extension with a SDES MID
>>>           value that is in the table mapping MID to "m=3D" line, then
>>>           create an entry in the RTP stream to "m=3D" line mapping tabl=
e
>>>           for this RTP stream (SSRC).  Then deliver the RTP packet to
>>>           the "m=3D" line context of the created mapping and stop.
>>>=20
>>>       B.  If the packet carries a Payload Type that is in the payload
>>>           type table, then create an entry in the RTP stream to "m=3D"
>>>           line mapping table for this RTP stream (SSRC).  Then deliver
>>>           the RTP packet to the "m=3D" line context of the created
>>>           mapping and stop.
>>>=20
>>>       C.  Otherwise do not deliver the RTP packet to higher layers.
>>>           Note, this includes unknown MID values.
>>>=20
>>>   For each RTCP packet received (including each RTCP packet that is
>>>   part of a compound RTCP packet), the RTCP packet needs to be
>>>   processed by the RTP/RTCP implementation and relevant information and
>>>   data from the RTCP packets needs to be routed to the appropriate
>>>   handler for the related RTP streams.  The appropriate handler is
>>>   determined by using the RTP stream to "m=3D" line mapping table.
>>>=20
>>>=20
>>>=20
>>> Name                      Expires July 31, 2017                 [Page
>>>4]
>>>=20
>>> Internet-Draft              Abbreviated-Title               January
>>>2017
>>>=20
>>>=20
>>>   On reception of any compound RTCP packet prior to dispatching the
>>>   received information and data, if there is an RTCP SDES packet
>>>   included that SHOULD be processed first.  If that SDES packet
>>>   contains SDES MID entries, this can results in updates and additions
>>>   to the RTP stream to "m=3D" line mapping table.  Thus each of the SDE=
S
>>>   MID items are processed and the current table entries are checked if
>>>   the corresponding MID value matches the current RTP stream to "m=3D"
>>>   line mapping, else the entry is updated.  If there is no RTP stream
>>>   to "m=3D" line table mapping entry for the received SDES item's SSRC,
>>>   such an entry is created.  Note, that in the process of updating the
>>>   table entries, update flap suppression as discussed in Section 4.2.6
>>>   of [RFC7941] should be considered.
>>>=20
>>>   The various different RTCP packets as well as their various sub
>>>   parts, such as the various RTCP Feedback message types, relates to
>>>   the RTP streams in a couple of different ways.  The currently known
>>>   patterns are the following:
>>>=20
>>>   Reports on outgoing RTP streams:  For all RTP streams that this
>>>      endpoint is the source of, it can expect to receive report blocks
>>>      of several types identified as relating to an outgoing stream.
>>>      The basic pattern for these blocks are that the RTCP packet header
>>>      identifies the source of the reports, as identified by an SSRC,
>>>      and containing one or more report blocks, where each report block
>>>      identifies the RTP stream, using the SSRC, the report relates to.
>>>      For this pattern the relevant report information is provided to
>>>      the higher layer associated with the "m=3D" line the RTP Stream to
>>>      "m=3D" line table identifies.  The source SSRC as identifier of th=
e
>>>      endpoint that the report originates are relevant for interpreting
>>>      the information, but not necessarily for routing.  Example of this
>>>      is pattern are:
>>>=20
>>>      Sender Report (SR) and Receiver Report (RR)  The basic receiver
>>>         report blocks from RFC3550 start with the SSRC of the RTP
>>>         stream they report on.
>>>=20
>>>      Extended Reports (XR):  RFC3611 is a framework that enables a
>>>         large number of different reports.  However, a large number of
>>>         these report formats are reporting on specific RTP streams and
>>>         thus each individual report of these types contains a SSRC
>>>         field to identify the RTP stream.
>>>=20
>>>   RTCP Feedback Messages for outgoing RTP streams:  The RFC 4585 RTCP
>>>      feedback messages allow for a number of different type of feedback
>>>      messages.  However, the RTCP feedback message header contains the
>>>      SSRC identifying the source of the feedback messages as well as
>>>      the actual type of the feedback.  Some of the feedback messages
>>>      also uses the target SSRC field in the header to identify which
>>>=20
>>>=20
>>>=20
>>> Name                      Expires July 31, 2017                 [Page
>>>5]
>>>=20
>>> Internet-Draft              Abbreviated-Title               January
>>>2017
>>>=20
>>>=20
>>>      RTP stream the feedback is related to.  For these types all the
>>>      FCI entries, if multiple ones are forwarded to the identified
>>>      handler.  Examples of this pattern are:
>>>=20
>>>      Picture Loss Indication (PLI):  RFC 4585 (PT=3DPSFB, FMT=3D1).
>>>=20
>>>      Slice Loss Indication (SLI):  RFC 4585 (PT=3DPSFB, FMT=3D2).
>>>=20
>>>      Reference Picture Selection Indication (RPSI):  RFC 4585 (PT=3DPSF=
B,
>>>         FMT=3D3).
>>>=20
>>>      Generic NACK:  [RFC4585] (PT=3DRTPFB and FMT=3D1).
>>>=20
>>>      Other feedback messages includes the target SSRC in the Feedback
>>>      Control Information (FCI).  Here each FCI needs to be processed
>>>      and the SSRC field identified.  And the individual FCI combined
>>>      with the RTCP packet header context needs to be forwarded to the
>>>      identified handler.  Example of this pattern are:
>>>=20
>>>      Full Intra Request (FIR):  [RFC5104] (PT=3DPSFB, FMT=3D4).
>>>=20
>>>      Temporal-Spatial Trade-off Request (TSTR):  [RFC5104] (PT=3DPSFB,
>>>         FMT=3D5).
>>>=20
>>>      Temporal-Spatial Trade-off Notification (TSTN):  This [RFC5104]
>>>         defined message (PT=3DPSFB, FMT=3D5) is actually a notifciation=
 in
>>>         response to a TSTR this endpoint sent using the SSRC the FCI
>>>         identifies as source.
>>>=20
>>>      H.271 Video Back Channel Message (VBCM):  [RFC5104] (PT=3DPSFB,
>>>         FMT=3D7).
>>>=20
>>>      Layer Refresh Request (LRR):  [I-D.ietf-avtext-lrr] (PT=3DPSFB,
>>>         FMT=3DTBD).
>>>=20
>>>   Descriptive or Notifications for an incoming RTP stream:  There exist
>>>      some RTCP packet types that provides additional information or
>>>      notifies about events related to the RTP stream identified.  In
>>>      these cases the RTP stream is identified using the SSRC field
>>>      value, and the information is provided to the higher layer
>>>      associated with the "m=3D" line for the incoming RTP stream as
>>>      identified by the current RTP stream to "m=3D" line table.  For th=
is
>>>      type of pattern it is common that the RTCP packets and information
>>>      is repeated, either periodically (e.g.  SDES items), or for a
>>>      duration (e.g.  BYE), thus suppression of repetitions can be
>>>      considered.  Examples of these are:
>>>=20
>>>=20
>>>=20
>>>=20
>>>=20
>>> Name                      Expires July 31, 2017                 [Page
>>>6]
>>>=20
>>> Internet-Draft              Abbreviated-Title               January
>>>2017
>>>=20
>>>=20
>>>      Source Description (SDES) RTCP Packet:  The base RTP/RTCP protocol
>>>         [RFC3550] defines SDES RTCP packets as a way of providing per
>>>         source (SSRC) specific information about the source.  An SDES
>>>         packet contains zero or more chunks, where a chunk contains the
>>>         SSRC for the source being described by the one or more items
>>>         included in the chunk.  Forward the SDES items in each chunk to
>>>         the RTP stream's handler.
>>>=20
>>>      Goodbye (BYE) RTCP Packet:  This RTP/RTCP protocol [RFC3550]
>>>         mechanism indicates that a particular RTP stream is leaving the
>>>         RTP session.  Thus, a most relevant event to inform the handler
>>>         for this RTP steam of.
>>>=20
>>>   Third Party Targeted Reports or Feedback:  There exist some
>>>      multiparty RTP Topologies that results in that an endpoint
>>>      receives third party reports (SR [RFC3550], RR [RFC3550] or XR
>>>      [RFC3611]) as well as Feedback Messages [RFC4585] that relates to
>>>      or targets an SSRC that originates from another endpoint, and
>>>      where the source of the RTCP packet is also another endpoint.
>>>      This type of packets should be forwarded to the higher layer
>>>      function dealing with the third party reporting.  And if none
>>>      exist then they can be suppressed.  As the third party handler can
>>>      be focused on determining the conditions for the source of the
>>>      reports and feedback, or focused on how the RTP stream source
>>>      progresses, or both recommendations can't be made.
>>>=20
>>>   APP Packets:  The RTCP APP Packets [RFC3550] are a mechanism that
>>>      enables experimentation.  The APP packet only specifies the source
>>>      of the packet.  Thus this information can be related to any of the
>>>      "m=3D" lines, thus deliver a copy of the packet to each "m=3D" lin=
e or
>>>      an APP specific handler.
>>>=20
>>=20
>>=20
>> --=20
>>=20
>> Magnus Westerlund
>>=20
>> ----------------------------------------------------------------------
>> Services, Media and Network features, Ericsson Research EAB/TXM
>> ----------------------------------------------------------------------
>> 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
>


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] [MMUSIC] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
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Date: Fri, 3 Feb 2017 11:25:00 +0000
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Subject: Re: [rtcweb] [MMUSIC] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
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Hi,

Wherever the streams origin, all streams associated with a bundle group
(received by C) by definition belong to the same RTP session (no matter
whatever we want to use that wording or not), so SSRC collision cannot
occur (if it does, it=B9s an error).

As Sergio said, the SFU needs to handle it.

=8Auntil we publish draft-bundle-with-multiple-rtp-streams, at least.

Regards,

Christer





On 03/02/17 10:46, "rtcweb on behalf of Sergio Garcia Murillo"
<rtcweb-bounces@ietf.org on behalf of sergio.garcia.murillo@gmail.com>
wrote:

>On 02/02/2017 23:19, Cullen Jennings wrote:
>>
>> Imagine a case where we have two sends call A and B and they each use
>>one PT that is unique and go to different m lines on receiver C via a
>>SFU.  Initially C get a packet from B with a given SSRC and the PT and
>>creates a mapping for it. But imagine that A and B had an SSRC collision
>>which they sort that out and A ends up having the SSRC that B initially
>>used. If you don't have the PT override the SSRC, the packets are going
>>to go to the wrong m-line because the PT points at the m-line A is using
>>but the old SSRC incorrectly points at the m-line B is using.
>>
>
>Sorry if I am late to the party and I say something stupid, but
>shouldn't the SFU terminate RTP/RTCP streams and rewrite PT, packet
>numbers *AND* SSRC? so the SFU SHALL ensure in this case that there is
>no SSRC collision on C.
>
>Best regards
>Sergio
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb


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To: Victor Pascual Avila <victor.pascual.avila@gmail.com>, Bernard Aboba <bernard.aboba@gmail.com>
References: <24b5eee4-29b1-33b0-ecea-58c766636311@alvestrand.no> <50b49dbf-7de1-a7c4-923f-f702ac14215e@alvestrand.no> <0df3cbb0-a3e7-011c-c4d5-63aad9350283@alvestrand.no> <e4d73f68-4e70-05f9-bfc8-4429210c5313@cisco.com> <CAOW+2dsatEFie4WuaGhprxVw9z-DfMZhPOYetDr+itP5TSnoAw@mail.gmail.com> <CAGTXFp8uG1Rtyj78j+6rxQ-dSSBUuj1xXeNhO19M9dXOa4ZrMw@mail.gmail.com>
From: Flemming Andreasen <fandreas@cisco.com>
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Date: Fri, 3 Feb 2017 08:55:19 -0500
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Subject: Re: [rtcweb] [MMUSIC] Change Alert [Re: Modifying an approved document: MSID]
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Based on the feedback received, it seems fine to make this change.

Thanks

-- Flemming (as MMUSIC co-chair)

On 1/30/17 8:28 AM, Victor Pascual Avila wrote:
> fine with me
>
> On Fri, Jan 27, 2017 at 7:20 PM, Bernard Aboba <bernard.aboba@gmail.com> wrote:
>> I am in favor of the change.
>>
>> On Thu, Jan 26, 2017 at 5:54 AM, Flemming Andreasen <fandreas@cisco.com>
>> wrote:
>>> Hi Harald
>>>
>>> Since the document is not yet in Auth48 and the change is relatively
>>> minor, it is possible to update it during Auth48.
>>>
>>> However, from an MMUSIC chair point of view, I would like to get positive
>>> confirmation from some of the stakeholders that this change is indeed
>>> desired. Similarly, if anybody has any concerns with the suggested change,
>>> please send an e-mail to that effect.
>>>
>>> Thanks
>>>
>>> -- Flemming (as MMUSIC co-chair)
>>>
>>>
>>>
>>> On 1/23/17 10:18 AM, Harald Alvestrand wrote:
>>>> Den 19. jan. 2017 13:31, skrev Harald Alvestrand:
>>>>> Ted pointed out to me that I sent this to the wrong group.
>>>>>
>>>>> Chairs and members, please advise.
>>>> My proposed change is here:
>>>>
>>>> https://github.com/alvestrand/rtcweb-msid/pull/16
>>>>
>>>> The filed issue is here:
>>>>
>>>> https://github.com/alvestrand/rtcweb-msid/issues/15
>>>>
>>>> (CCing RTCWEB - please keep discussion, if any, on mmusic)
>>>>
>>>>>
>>>>> -------- Forwarded Message --------
>>>>> Subject:        [rtcweb] Modifying an approved document: MSID
>>>>> Date:   Wed, 18 Jan 2017 23:22:14 +0100
>>>>> From:   Harald Alvestrand <harald@alvestrand.no>
>>>>> To:     rtcweb@ietf.org <rtcweb@ietf.org>
>>>>>
>>>>>
>>>>>
>>>>> When reviewing the implications of the PeerConnection API change to
>>>>> support AddTrack rather than AddStream, I found an issue.
>>>>>
>>>>> This issue concerns draft-ietf-rtcweb-msid, which is currently in
>>>>> REF-WAIT state (I believe).
>>>>>
>>>>> The issue is that it is possible to add a track without specifying a
>>>>> stream. Since the track's ID needs to be carried, we have to send an
>>>>> "a=msid" line, but the track's ID is the *second* field on that line,
>>>>> with the first being the stream's ID.
>>>>>
>>>>> This creates a problem.
>>>>>
>>>>> Suggested fix: Insert two lines in the document:
>>>>>
>>>>> 1) On SDP generation:
>>>>>
>>>>> "If there is no stream associated with the track, use the reserved ID
>>>>> value '-'"
>>>>>
>>>>> 2) On SDP parsing
>>>>>
>>>>> "If the stream ID is the reserved value '-', the track is not associated
>>>>> with a stream, and no stream is signalled or created."
>>>>>
>>>>> If this is OK with the community, I'll issue an updated draft with this
>>>>> change.
>>>>>
>>>>>
>>>>> --
>>>>> Surveillance is pervasive. Go Dark.
>>>>>
>>>>> _______________________________________________
>>>>> rtcweb mailing list
>>>>> rtcweb@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> mmusic mailing list
>>>>> mmusic@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>>>
>>>> _______________________________________________
>>>> mmusic mailing list
>>>> mmusic@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>> .
>>>>
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>>
>
>


From nobody Tue Feb  7 05:19:46 2017
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To: Flemming Andreasen <fandreas@cisco.com>, Victor Pascual Avila <victor.pascual.avila@gmail.com>, Bernard Aboba <bernard.aboba@gmail.com>
References: <24b5eee4-29b1-33b0-ecea-58c766636311@alvestrand.no> <50b49dbf-7de1-a7c4-923f-f702ac14215e@alvestrand.no> <0df3cbb0-a3e7-011c-c4d5-63aad9350283@alvestrand.no> <e4d73f68-4e70-05f9-bfc8-4429210c5313@cisco.com> <CAOW+2dsatEFie4WuaGhprxVw9z-DfMZhPOYetDr+itP5TSnoAw@mail.gmail.com> <CAGTXFp8uG1Rtyj78j+6rxQ-dSSBUuj1xXeNhO19M9dXOa4ZrMw@mail.gmail.com> <f444ec35-a61d-9210-62df-c65dfdac7589@cisco.com>
From: Harald Alvestrand <harald@alvestrand.no>
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Date: Tue, 7 Feb 2017 14:19:38 +0100
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Subject: Re: [rtcweb] [MMUSIC] Change Alert [Re: Modifying an approved document: MSID]
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Den 03. feb. 2017 14:55, skrev Flemming Andreasen:
> Based on the feedback received, it seems fine to make this change.

Thank you; I've submitted -16 with this change.


> 
> Thanks
> 
> -- Flemming (as MMUSIC co-chair)
> 
> On 1/30/17 8:28 AM, Victor Pascual Avila wrote:
>> fine with me
>>
>> On Fri, Jan 27, 2017 at 7:20 PM, Bernard Aboba
>> <bernard.aboba@gmail.com> wrote:
>>> I am in favor of the change.
>>>
>>> On Thu, Jan 26, 2017 at 5:54 AM, Flemming Andreasen <fandreas@cisco.com>
>>> wrote:
>>>> Hi Harald
>>>>
>>>> Since the document is not yet in Auth48 and the change is relatively
>>>> minor, it is possible to update it during Auth48.
>>>>
>>>> However, from an MMUSIC chair point of view, I would like to get
>>>> positive
>>>> confirmation from some of the stakeholders that this change is indeed
>>>> desired. Similarly, if anybody has any concerns with the suggested
>>>> change,
>>>> please send an e-mail to that effect.
>>>>
>>>> Thanks
>>>>
>>>> -- Flemming (as MMUSIC co-chair)
>>>>
>>>>
>>>>
>>>> On 1/23/17 10:18 AM, Harald Alvestrand wrote:
>>>>> Den 19. jan. 2017 13:31, skrev Harald Alvestrand:
>>>>>> Ted pointed out to me that I sent this to the wrong group.
>>>>>>
>>>>>> Chairs and members, please advise.
>>>>> My proposed change is here:
>>>>>
>>>>> https://github.com/alvestrand/rtcweb-msid/pull/16
>>>>>
>>>>> The filed issue is here:
>>>>>
>>>>> https://github.com/alvestrand/rtcweb-msid/issues/15
>>>>>
>>>>> (CCing RTCWEB - please keep discussion, if any, on mmusic)
>>>>>
>>>>>>
>>>>>> -------- Forwarded Message --------
>>>>>> Subject:        [rtcweb] Modifying an approved document: MSID
>>>>>> Date:   Wed, 18 Jan 2017 23:22:14 +0100
>>>>>> From:   Harald Alvestrand <harald@alvestrand.no>
>>>>>> To:     rtcweb@ietf.org <rtcweb@ietf.org>
>>>>>>
>>>>>>
>>>>>>
>>>>>> When reviewing the implications of the PeerConnection API change to
>>>>>> support AddTrack rather than AddStream, I found an issue.
>>>>>>
>>>>>> This issue concerns draft-ietf-rtcweb-msid, which is currently in
>>>>>> REF-WAIT state (I believe).
>>>>>>
>>>>>> The issue is that it is possible to add a track without specifying a
>>>>>> stream. Since the track's ID needs to be carried, we have to send an
>>>>>> "a=msid" line, but the track's ID is the *second* field on that line,
>>>>>> with the first being the stream's ID.
>>>>>>
>>>>>> This creates a problem.
>>>>>>
>>>>>> Suggested fix: Insert two lines in the document:
>>>>>>
>>>>>> 1) On SDP generation:
>>>>>>
>>>>>> "If there is no stream associated with the track, use the reserved ID
>>>>>> value '-'"
>>>>>>
>>>>>> 2) On SDP parsing
>>>>>>
>>>>>> "If the stream ID is the reserved value '-', the track is not
>>>>>> associated
>>>>>> with a stream, and no stream is signalled or created."
>>>>>>
>>>>>> If this is OK with the community, I'll issue an updated draft with
>>>>>> this
>>>>>> change.
>>>>>>
>>>>>>
>>>>>> -- 
>>>>>> Surveillance is pervasive. Go Dark.
>>>>>>
>>>>>> _______________________________________________
>>>>>> rtcweb mailing list
>>>>>> rtcweb@ietf.org
>>>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> mmusic mailing list
>>>>>> mmusic@ietf.org
>>>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>>>>
>>>>> _______________________________________________
>>>>> mmusic mailing list
>>>>> mmusic@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>>> .
>>>>>
>>>> _______________________________________________
>>>> mmusic mailing list
>>>> mmusic@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>
>>>
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>
>>
>>
> 


From nobody Wed Feb  8 07:20:02 2017
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Subject: [rtcweb] rtcweb - Update to a Meeting Session Request for IETF 98
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An update to a meeting session request has just been submitted by Alissa Cooper, a ART Area Director.


---------------------------------------------------------
Working Group Name: Real-Time Communication in WEB-browsers
Area Name: Applications and Real-Time Area
Session Requester: Alissa Cooper

Number of Sessions: 2
Length of Session(s):  2 Hours, 1 Hour
Number of Attendees: 70
Conflicts to Avoid: 
 First Priority: tls rmcat aqm avtcore avtext clue dispatch httpbis stir acme mmusic payload sipcore perc quic
 Second Priority: dprive tcpinc straw tsvwg tsvarea ace uta netvc capport
 Third Priority: insipid irtfopen dane opsawg


People who must be present:
  Eric Rescorla
  Sean Turner
  Cullen Jennings
  Martin Thomson
  Alissa Cooper
  Ted Hardie
  Justin Uberti
  Harald Alvestrand

Resources Requested:
  

Special Requests:
  
---------------------------------------------------------


From nobody Wed Feb  8 10:05:12 2017
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Archived-At: <https://mailarchive.ietf.org/arch/msg/rtcweb/UYv6_8dVdw9mHZ3JOPYkaT08ORA>
Subject: Re: [rtcweb] rtcweb - Update to a Meeting Session Request for IETF 98
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Hi,

I wonder whether people would be interested/available to have an off-line d=
iscussion about the RTP mapping issue (unless we managed to agree on text b=
efore Chicago, that is). E.g., on Sunday?

I think it would be a waste of time trying to solve the issue behind the mi=
crophone during the meeting...

Regards,

Christer


-----Original Message-----
From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of "IETF Meeting Se=
ssion Request Tool"
Sent: 08 February 2017 17:20
To: session-request@ietf.org
Cc: rtcweb@ietf.org; rtcweb-chairs@ietf.org
Subject: [rtcweb] rtcweb - Update to a Meeting Session Request for IETF 98



An update to a meeting session request has just been submitted by Alissa Co=
oper, a ART Area Director.


---------------------------------------------------------
Working Group Name: Real-Time Communication in WEB-browsers Area Name: Appl=
ications and Real-Time Area Session Requester: Alissa Cooper

Number of Sessions: 2
Length of Session(s):  2 Hours, 1 Hour
Number of Attendees: 70
Conflicts to Avoid:=20
 First Priority: tls rmcat aqm avtcore avtext clue dispatch httpbis stir ac=
me mmusic payload sipcore perc quic  Second Priority: dprive tcpinc straw t=
svwg tsvarea ace uta netvc capport  Third Priority: insipid irtfopen dane o=
psawg


People who must be present:
  Eric Rescorla
  Sean Turner
  Cullen Jennings
  Martin Thomson
  Alissa Cooper
  Ted Hardie
  Justin Uberti
  Harald Alvestrand

Resources Requested:
 =20

Special Requests:
 =20
---------------------------------------------------------

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


From nobody Wed Feb  8 10:07:16 2017
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "rtcweb-chairs@ietf.org" <rtcweb-chairs@ietf.org>
Thread-Topic: [rtcweb] rtcweb - Update to a Meeting Session Request for IETF 98
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Archived-At: <https://mailarchive.ietf.org/arch/msg/rtcweb/mDkDeoRvX3ced0f7QttVre3DyxE>
Subject: Re: [rtcweb] rtcweb - Update to a Meeting Session Request for IETF 98
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Also, having a phone meeting to discuss the issue before Chicago would also=
 be good, I think.

Regards,

Christer

-----Original Message-----
From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Christer Holmber=
g
Sent: 08 February 2017 20:05
To: rtcweb@ietf.org; rtcweb-chairs@ietf.org
Subject: Re: [rtcweb] rtcweb - Update to a Meeting Session Request for IETF=
 98

Hi,

I wonder whether people would be interested/available to have an off-line d=
iscussion about the RTP mapping issue (unless we managed to agree on text b=
efore Chicago, that is). E.g., on Sunday?

I think it would be a waste of time trying to solve the issue behind the mi=
crophone during the meeting...

Regards,

Christer


-----Original Message-----
From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of "IETF Meeting Se=
ssion Request Tool"
Sent: 08 February 2017 17:20
To: session-request@ietf.org
Cc: rtcweb@ietf.org; rtcweb-chairs@ietf.org
Subject: [rtcweb] rtcweb - Update to a Meeting Session Request for IETF 98



An update to a meeting session request has just been submitted by Alissa Co=
oper, a ART Area Director.


---------------------------------------------------------
Working Group Name: Real-Time Communication in WEB-browsers Area Name: Appl=
ications and Real-Time Area Session Requester: Alissa Cooper

Number of Sessions: 2
Length of Session(s):  2 Hours, 1 Hour
Number of Attendees: 70
Conflicts to Avoid:=20
 First Priority: tls rmcat aqm avtcore avtext clue dispatch httpbis stir ac=
me mmusic payload sipcore perc quic  Second Priority: dprive tcpinc straw t=
svwg tsvarea ace uta netvc capport  Third Priority: insipid irtfopen dane o=
psawg


People who must be present:
  Eric Rescorla
  Sean Turner
  Cullen Jennings
  Martin Thomson
  Alissa Cooper
  Ted Hardie
  Justin Uberti
  Harald Alvestrand

Resources Requested:
 =20

Special Requests:
 =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


From nobody Wed Feb  8 13:13:42 2017
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From: Cullen Jennings <fluffy@iii.ca>
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Date: Wed, 8 Feb 2017 13:13:23 -0800
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Subject: [rtcweb] normative dependencies in draft-ietf-mmusic-sdp-bundle-negotiation
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Looking at draft-ietf-mmusic-sdp-bundle-negotiation, it seems that it =
would be fine to normatively depend on RFC 5245 instead of =
ietf-mmusic-ice-sip-sdp. It does not rely on any changes in made from =
4245 to ice-sip-sdp.=20

We don't want JSEP to normatively depend on ietf-mmusic-ice-sip-sdp =
since that is not needed. Could you change =
draft-ietf-mmusic-sdp-bundle-negotiation to not normatively depend on =
ietf-mmusic-ice-sip-sdp?

Thanks



From nobody Wed Feb  8 13:22:13 2017
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Cullen Jennings <fluffy@iii.ca>, mmusic <mmusic@ietf.org>
Thread-Topic: [MMUSIC] normative dependencies in draft-ietf-mmusic-sdp-bundle-negotiation
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Cc: RTCWeb IETF <rtcweb@ietf.org>
Subject: Re: [rtcweb] [MMUSIC] normative dependencies in draft-ietf-mmusic-sdp-bundle-negotiation
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Hi,

Don't you also reference trickle in JSEP?

Regards,

Christer

-----Original Message-----
From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of Cullen Jennings
Sent: 08 February 2017 23:13
To: mmusic <mmusic@ietf.org>
Cc: RTCWeb IETF <rtcweb@ietf.org>
Subject: [MMUSIC] normative dependencies in draft-ietf-mmusic-sdp-bundle-ne=
gotiation


Looking at draft-ietf-mmusic-sdp-bundle-negotiation, it seems that it would=
 be fine to normatively depend on RFC 5245 instead of ietf-mmusic-ice-sip-s=
dp. It does not rely on any changes in made from 4245 to ice-sip-sdp.=20

We don't want JSEP to normatively depend on ietf-mmusic-ice-sip-sdp since t=
hat is not needed. Could you change draft-ietf-mmusic-sdp-bundle-negotiatio=
n to not normatively depend on ietf-mmusic-ice-sip-sdp?

Thanks


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


From nobody Fri Feb 10 10:16:45 2017
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Subject: [rtcweb] =?iso-8859-1?q?14h00_au_Pavillon_Pouliot_de_l=27Universi?= =?iso-8859-1?q?t=E9_Laval_=28Salle_PLT-3903=29=2E?=
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14h00 au Pavillon Pouliot de l=92Universit=E9 Laval (Salle PLT-3903).


Pascal Cardin : pascal.cardin@cnesst.gouv.qc.ca
Mathieu Simard : mathieu.simard@cnesst.gouv.qc.ca
Jacques Martin : jacques.martin@cnesst.gouv.qc.ca
Louis Gosselin : louis.gosselin@hotmail.com
Ren=E9 L=E9vesque : rene.levesques@lorantech.com



-----Message d'origine-----
De=A0: rtcweb [mailto:rtcweb-bounces@ietf.org] De la part de
rtcweb-request@ietf.org
Envoy=E9=A0: 9 f=E9vrier 2017 15:00
=C0=A0: rtcweb@ietf.org
Objet=A0: rtcweb Digest, Vol 72, Issue 9

Send rtcweb mailing list submissions to
	rtcweb@ietf.org

To subscribe or unsubscribe via the World Wide Web, visit
	https://www.ietf.org/mailman/listinfo/rtcweb
or, via email, send a message with subject or body 'help' to
	rtcweb-request@ietf.org

You can reach the person managing the list at
	rtcweb-owner@ietf.org

When replying, please edit your Subject line so it is more specific than
"Re: Contents of rtcweb digest..."


From nobody Sat Feb 11 22:16:06 2017
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From: Roni Even <roni.even@huawei.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "rtcweb-chairs@ietf.org" <rtcweb-chairs@ietf.org>
Thread-Topic: [rtcweb] Associating RTP/RTCP With Correct SDP Media Description - bundle
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Subject: Re: [rtcweb] Associating RTP/RTCP With Correct SDP Media Description - bundle
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Changing the subject

I am OK with both options and agree that it will not work on the microphone
Roni=20

> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Christer
> Holmberg
> Sent: =E9=E5=ED=A0=E3 08 =F4=E1=F8=E5=E0=F8 2017 20:06
> To: Christer Holmberg; rtcweb@ietf.org; rtcweb-chairs@ietf.org
> Subject: Re: [rtcweb] rtcweb - Update to a Meeting Session Request for IE=
TF
> 98
>=20
> Also, having a phone meeting to discuss the issue before Chicago would al=
so
> be good, I think.
>=20
> Regards,
>=20
> Christer
>=20
> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Christer
> Holmberg
> Sent: 08 February 2017 20:05
> To: rtcweb@ietf.org; rtcweb-chairs@ietf.org
> Subject: Re: [rtcweb] rtcweb - Update to a Meeting Session Request for IE=
TF
> 98
>=20
> Hi,
>=20
> I wonder whether people would be interested/available to have an off-line
> discussion about the RTP mapping issue (unless we managed to agree on
> text before Chicago, that is). E.g., on Sunday?
>=20
> I think it would be a waste of time trying to solve the issue behind the
> microphone during the meeting...
>=20
> Regards,
>=20
> Christer
>=20
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> -----Original Message-----
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acme
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> People who must be present:
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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
References: <1f020dc6-1ac8-b71b-aee4-a711d15f1588@ericsson.com> <2e0ea537-d03e-f263-ad64-cdd65ecd3fb5@ericsson.com> <D701B5B7-C221-4D0E-B10A-D01D3FE5E4AD@cisco.com>
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Subject: Re: [rtcweb] [MMUSIC] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
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Hi,

Thanks for the feedback. Sorry for my delay in responding, a cold have
kept from work.

Den 2017-02-02 kl. 23:19, skrev Cullen Jennings (fluffy):
>
> So I much prefer the current text and think there are a bunch of
> problems with this text. If we actually had emails explaining what
> problems in the current text this was trying to fix, with individual
> PRs for those, this would be much easier to resolve each of them and
> get them fixed.
>
> 1) we have been trying to avoid the use of "RTP session" as it has
> been very unclear to implementors what it is. I think this would be
> better if we could rephrase to not use that

Okay, but the RTP session is very easy to make clear in the context in
of BUNDLE, where this text is intended to be. I can improve that.

>
> 2) both the proposed and current text seem lacking in dealing with
> multiple bundle groups

Okay, that can be fixed by clarifying that each bundle group results in
its own RTP session, thus the procedures in this is per bundle group.

>
> 3) Stats are typically maintained by things after the packet is
> routed - not before.

So this comes a question of ones view of RTP stack and the question of
layering. And this is exactly why I think the current text is
problematic. It takes one very particular view, why I attempted to be
much more neutral on which order things happens. There are a number of
functions that are in the RTP protocol layer, not in the higher layers.
There are however some things, like XR VoIP metrcis that are metrics in
the higher layers. So, yes this is not clear cut. I think ones view of
this depends on if one have a very integrated RTP implementation, then
what you say makes sense, but if one has a very layered and modularized
design, then my viewpoint makes more sense.

  From my perspective the most important thing here is that this text if
it contains any RFC 2119 words can't prevent some possible
implementation choices of the RTP stack.

>
> 4) Need to explain how the SDES in compound RTCP causes updates
>

I can attempt to clarify this. However, there is a potential issue here
in that some implementations may not be able to force the receiver to
process the content of the SDES RTCP packet prior to some or even all
the other RTCP packets in a compound packet.

What in the current text:

     On reception of any compound RTCP packet prior to dispatching the
     received information and data, if there is an RTCP SDES packet
     included that SHOULD be processed first.  If that SDES packet
     contains SDES MID entries, this can results in updates and additions
     to the RTP stream to "m=" line mapping table.  Thus each of the SDES
     MID items are processed and the current table entries are checked if
     the corresponding MID value matches the current RTP stream to "m="
     line mapping, else the entry is updated.  If there is no RTP stream
     to "m=" line table mapping entry for the received SDES item's SSRC,
     such an entry is created.  Note, that in the process of updating the
     table entries, update flap suppression as discussed in Section 4.2.6
     of [RFC7941] should be considered.

Is insufficient in that regards. Is it only the placement prior to the
individual RTCP packet types that is the issue? Should with the
exception of the first sentence be moved under the SDES text?

> 5) given this removes the outgoing SSRC table, not clear how it
> routes RTCP reports. I think this needs to be clarified.
>

Okay, I think I understand that. I have an implicit assumption that the
implementation knows how its local (outgoing) RTP Streams are related to
the media sources and thus the related RTPsender. I can update the text
to address this.

> 6) I don't think most implementers are going to have a clue what to
> do for the "Third Party Targeted Reports or Feedback" section
>

I can understand that, but I think it is important to call out that this
bucket do exist, and if you don't know what to do I think it is fine to
ignore these.

> I will try and take your PR and break it up into some bit size pieces
> so we can try and see if we can get the easy ones out of the way and
> focus on the parts that are key changes.
>

Ok, I have seen that you generated a lot of individual issues, I will 
attempt to look through them and comment if there are things that was 
unintentional or where I have additional aspects to add.

I intend to update my PR based on the feedback I received.

Cheers

Magnus Westerlund

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----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
FÃ¤rÃ¶gatan 6                 | Mobile +46 73 0949079
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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
To: Cullen Jennings <fluffy@iii.ca>, Jonathan Lennox <jonathan@vidyo.com>
References: <1f020dc6-1ac8-b71b-aee4-a711d15f1588@ericsson.com> <D2A63CD5-61AB-47DD-B946-6A17A94F76E7@vidyo.com> <7ED35CA1-D90C-444A-93EF-445C894B6859@iii.ca>
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Subject: Re: [rtcweb] [MMUSIC] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
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Hi,

Please see below for my feedback on this.

Den 2017-02-02 kl. 23:19, skrev Cullen Jennings:
>
>> On Jan 31, 2017, at 10:43 AM, Jonathan Lennox <jonathan@vidyo.com>
>> wrote:
>>
>> In general, this looks good.
>>
>> I have one question, however.  What should happen if an RTP packet
>> (without a MID) is received for a stream that has an existing
>> mapping to an m= line, but whose PT is not valid for that m= line?
>>
>> Should it 1. Cause the streamâ€™s mapping to be moved to another m=
>> line, if the received PT is in the payload type table for some
>> other m= line? or 2. Be an error?
>
>
> Imagine a case where we have two sends call A and B and they each use
> one PT that is unique and go to different m lines on receiver C via a
> SFU.  Initially C get a packet from B with a given SSRC and the PT
> and creates a mapping for it. But imagine that A and B had an SSRC
> collision which they sort that out and A ends up having the SSRC that
> B initially used. If you don't have the PT override the SSRC, the
> packets are going to go to the wrong m-line because the PT points at
> the m-line A is using but the old SSRC incorrectly points at the
> m-line B is using.
>
> To put this a different way, if the packet has a mid, I think it is
> very clear the mid has to override the PT. Same for the PT. The mid
> is effectively just an extended PT.
>
> I'm not thinking to much about the case where SSRC is signalled in
> SDP but ignoring that case for a second, I think the really easy
> thing to specify, implement, and understand works is simply a
> priority order where roughly
>
> 1) if you have mid, use that and latch the SSRC
> 2) if you have a unique pt, use that and latch the SSRC
> 3) if you have a latched SSRC use that
>
> If you had signaled SSRC, guess I would insert that as step 1.5 of
> use  that
>

The issue as I see it is when and how you update the RTP stream to m= 
line mapping table. I think we have to be clear and separate the actions 
for how the RTP stream to m= line table is built and how it is updated.

Signalling:

Build mid to m= line table
If a=ssrc is used, enter SSRC in RTP stream to m= line table. Add flag 
(signalled) to entry.


On Reception of RTP stream not in RTP stream to m= line table:

1) if you have mid, use that and add to RTP stream table.
2) if you have a unique pt, use that and add to RTP stream table

If the RTP stream is in the RTP stream to m= line table. On RTP packet 
reception check if update is needed:

1) New MID, then update RTP stream entry, unless signalled
2) If PT indicates different m= line, update entry unless signalled or 
set by MID.

The point of looking on the problem from this direction is that we 
actually need to care about what set the entry originally and if it has 
precedence.

Yes, the question of how one removes or updates stale entries are 
important. But this brings out the question for the above. Is an entry 
set by MID to be overwritten by a PT change? Yes, it will be possible to 
construct edge cases where one or the other approach will be 
sub-optimal. However, I think biasing the usage for working correctly 
with MID and having sub-optimal behaviors in other cases that can be 
avoided by for example the SFU not making bad choices, like not 
suppressing SSRC collisions between the different SSRC spaces.

So entry updates are happening for the following reasons.

New SDP: Remove all signalled entries not explicitly included in new 
signalling message. Explicitly listed entries are updated to indicate 
the correct m= line.

RTCP BYE: Remove SSRC from table (using timeout procedure to avoid RTP 
packet reordering from reinstating entry), unless signalled?

SSRC timeout. If the SSRC times out on RTP/RTCP level, then the entry is 
also removed, unless signaled.

Cheers

Magnus Westerlund

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Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
FÃ¤rÃ¶gatan 6                 | Mobile +46 73 0949079
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: [rtcweb] [MMUSIC] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
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Date: Wed, 15 Feb 2017 13:09:56 +0000
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References: <1f020dc6-1ac8-b71b-aee4-a711d15f1588@ericsson.com> <2e0ea537-d03e-f263-ad64-cdd65ecd3fb5@ericsson.com> <D701B5B7-C221-4D0E-B10A-D01D3FE5E4AD@cisco.com> <0e84fe0e-8e50-7b9f-bede-76ebf293a0d8@ericsson.com>
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Subject: Re: [rtcweb] [MMUSIC] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
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From: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: Describe parsing of SDP identity,  and how multiple identities are handled, in security-architecture
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Subject: [rtcweb] Describe parsing of SDP identity, and how multiple identities are handled, in security-architecture
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Hi,=0A=
=0A=
after some discussing in in W3C webrtc-pc [1] and JSEP [2][3][4] github =0A=
repos it seems the conclusion is that how identity is handled carried in =
=0A=
the SDP, and how identity in the SDP is parsed, and how multiple =0A=
identity attributes (if we are to support that) should be describe in =0A=
draft-ietf-rtcweb-security-arch [5].=0A=
=0A=
Identity in the SDP is currently described in =0A=
draft-ietf-rtcweb-security-arch, but only for a single identity =0A=
attribute, and parsing is not described, so it needs to be updated.=0A=
=0A=
I decided to send this to the list as the activity level at the github =0A=
repo [6] for draft-ietf-rtcweb-security-arch seems to be low (indicating =
=0A=
the action happens elsewhere).=0A=
=0A=
Stefan=0A=
=0A=
=0A=
[1] https://github.com/w3c/webrtc-pc/issues/964=0A=
[2] https://github.com/rtcweb-wg/jsep/issues/546=0A=
[3] =0A=
https://github.com/rtcweb-wg/jsep/commit/a40fb5101ae530e97cea2073b507d7b417=
df9e49=0A=
[4] https://github.com/rtcweb-wg/jsep/issues/601=0A=
[5] https://tools.ietf.org/html/draft-ietf-rtcweb-security-arch=0A=
[6] https://github.com/rtcweb-wg/security-arch=0A=


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From: Eric Rescorla <ekr@rtfm.com>
Date: Thu, 16 Feb 2017 07:20:16 -0800
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Subject: Re: [rtcweb] Describe parsing of SDP identity, and how multiple identities are handled, in security-architecture
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--001a114f0996fedb7a0548a757d0
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On Thu, Feb 16, 2017 at 2:09 AM, Stefan H=C3=A5kansson LK <
stefan.lk.hakansson@ericsson.com> wrote:

> Hi,
>
> after some discussing in in W3C webrtc-pc [1] and JSEP [2][3][4] github
> repos it seems the conclusion is that how identity is handled carried in
> the SDP, and how identity in the SDP is parsed, and how multiple
> identity attributes (if we are to support that) should be describe in
> draft-ietf-rtcweb-security-arch [5].
>
> Identity in the SDP is currently described in
> draft-ietf-rtcweb-security-arch, but only for a single identity
> attribute, and parsing is not described, so it needs to be updated.
>
> I decided to send this to the list as the activity level at the github
> repo [6] for draft-ietf-rtcweb-security-arch seems to be low (indicating
> the action happens elsewhere).
>

No, it's just that I'm too busy doing JSEP and TLS to do anything here.

-Ekr


>
> Stefan
>
>
> [1] https://github.com/w3c/webrtc-pc/issues/964
> [2] https://github.com/rtcweb-wg/jsep/issues/546
> [3]
> https://github.com/rtcweb-wg/jsep/commit/a40fb5101ae530e97cea2073b507d7
> b417df9e49
> [4] https://github.com/rtcweb-wg/jsep/issues/601
> [5] https://tools.ietf.org/html/draft-ietf-rtcweb-security-arch
> [6] https://github.com/rtcweb-wg/security-arch
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--001a114f0996fedb7a0548a757d0
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Thu, Feb 16, 2017 at 2:09 AM, Stefan H=C3=A5kansson LK <span dir=3D"=
ltr">&lt;<a href=3D"mailto:stefan.lk.hakansson@ericsson.com" target=3D"_bla=
nk">stefan.lk.hakansson@ericsson.com</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">Hi,<br>
<br>
after some discussing in in W3C webrtc-pc [1] and JSEP [2][3][4] github<br>
repos it seems the conclusion is that how identity is handled carried in<br=
>
the SDP, and how identity in the SDP is parsed, and how multiple<br>
identity attributes (if we are to support that) should be describe in<br>
draft-ietf-rtcweb-security-<wbr>arch [5].<br>
<br>
Identity in the SDP is currently described in<br>
draft-ietf-rtcweb-security-<wbr>arch, but only for a single identity<br>
attribute, and parsing is not described, so it needs to be updated.<br>
<br>
I decided to send this to the list as the activity level at the github<br>
repo [6] for draft-ietf-rtcweb-security-<wbr>arch seems to be low (indicati=
ng<br>
the action happens elsewhere).<br></blockquote><div><br></div><div>No, it&#=
39;s just that I&#39;m too busy doing JSEP and TLS to do anything here.</di=
v><div><br></div><div>-Ekr</div><div>=C2=A0</div><blockquote class=3D"gmail=
_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:=
1ex">
<br>
Stefan<br>
<br>
<br>
[1] <a href=3D"https://github.com/w3c/webrtc-pc/issues/964" rel=3D"noreferr=
er" target=3D"_blank">https://github.com/w3c/webrtc-<wbr>pc/issues/964</a><=
br>
[2] <a href=3D"https://github.com/rtcweb-wg/jsep/issues/546" rel=3D"norefer=
rer" target=3D"_blank">https://github.com/rtcweb-wg/<wbr>jsep/issues/546</a=
><br>
[3]<br>
<a href=3D"https://github.com/rtcweb-wg/jsep/commit/a40fb5101ae530e97cea207=
3b507d7b417df9e49" rel=3D"noreferrer" target=3D"_blank">https://github.com/=
rtcweb-wg/<wbr>jsep/commit/<wbr>a40fb5101ae530e97cea2073b507d7<wbr>b417df9e=
49</a><br>
[4] <a href=3D"https://github.com/rtcweb-wg/jsep/issues/601" rel=3D"norefer=
rer" target=3D"_blank">https://github.com/rtcweb-wg/<wbr>jsep/issues/601</a=
><br>
[5] <a href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-security-arch"=
 rel=3D"noreferrer" target=3D"_blank">https://tools.ietf.org/html/<wbr>draf=
t-ietf-rtcweb-security-<wbr>arch</a><br>
[6] <a href=3D"https://github.com/rtcweb-wg/security-arch" rel=3D"noreferre=
r" target=3D"_blank">https://github.com/rtcweb-wg/<wbr>security-arch</a><br=
>
<br>
______________________________<wbr>_________________<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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/rtcweb</a><br=
>
</blockquote></div><br></div></div>

--001a114f0996fedb7a0548a757d0--


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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Describe parsing of SDP identity, and how multiple identities are handled, in security-architecture
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Thanks Ekr, I'll create a github Issue then.=0A=
=0A=
On 16/02/17 16:22, Eric Rescorla wrote:=0A=
>=0A=
>=0A=
> On Thu, Feb 16, 2017 at 2:09 AM, Stefan H=E5kansson LK=0A=
> <stefan.lk.hakansson@ericsson.com=0A=
> <mailto:stefan.lk.hakansson@ericsson.com>> wrote:=0A=
>=0A=
>     Hi,=0A=
>=0A=
>     after some discussing in in W3C webrtc-pc [1] and JSEP [2][3][4] gith=
ub=0A=
>     repos it seems the conclusion is that how identity is handled carried=
 in=0A=
>     the SDP, and how identity in the SDP is parsed, and how multiple=0A=
>     identity attributes (if we are to support that) should be describe in=
=0A=
>     draft-ietf-rtcweb-security-arch [5].=0A=
>=0A=
>     Identity in the SDP is currently described in=0A=
>     draft-ietf-rtcweb-security-arch, but only for a single identity=0A=
>     attribute, and parsing is not described, so it needs to be updated.=
=0A=
>=0A=
>     I decided to send this to the list as the activity level at the githu=
b=0A=
>     repo [6] for draft-ietf-rtcweb-security-arch seems to be low (indicat=
ing=0A=
>     the action happens elsewhere).=0A=
>=0A=
>=0A=
> No, it's just that I'm too busy doing JSEP and TLS to do anything here.=
=0A=
>=0A=
> -Ekr=0A=
>=0A=
>=0A=
>=0A=
>     Stefan=0A=
>=0A=
>=0A=
>     [1] https://github.com/w3c/webrtc-pc/issues/964=0A=
>     <https://github.com/w3c/webrtc-pc/issues/964>=0A=
>     [2] https://github.com/rtcweb-wg/jsep/issues/546=0A=
>     <https://github.com/rtcweb-wg/jsep/issues/546>=0A=
>     [3]=0A=
>     https://github.com/rtcweb-wg/jsep/commit/a40fb5101ae530e97cea2073b507=
d7b417df9e49=0A=
>     <https://github.com/rtcweb-wg/jsep/commit/a40fb5101ae530e97cea2073b50=
7d7b417df9e49>=0A=
>     [4] https://github.com/rtcweb-wg/jsep/issues/601=0A=
>     <https://github.com/rtcweb-wg/jsep/issues/601>=0A=
>     [5] https://tools.ietf.org/html/draft-ietf-rtcweb-security-arch=0A=
>     <https://tools.ietf.org/html/draft-ietf-rtcweb-security-arch>=0A=
>     [6] https://github.com/rtcweb-wg/security-arch=0A=
>     <https://github.com/rtcweb-wg/security-arch>=0A=
>=0A=
>     _______________________________________________=0A=
>     rtcweb mailing list=0A=
>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>=0A=
>     https://www.ietf.org/mailman/listinfo/rtcweb=0A=
>     <https://www.ietf.org/mailman/listinfo/rtcweb>=0A=
>=0A=
>=0A=
=0A=


From nobody Fri Feb 17 03:04:42 2017
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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 of the IETF.

        Title           : Overview: Real Time Protocols for Browser-based Applications
        Author          : Harald T. Alvestrand
	Filename        : draft-ietf-rtcweb-overview-17.txt
	Pages           : 23
	Date            : 2017-02-17

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 an Applicability Statement - it does not itself
   specify any protocol, but specifies which other specifications WebRTC
   compliant implementations are supposed to follow.

   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:
https://tools.ietf.org/html/draft-ietf-rtcweb-overview-17

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-overview-17


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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From: Sean Turner <sean@sn3rd.com>
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Subject: [rtcweb] PRs for draft-ietf-rtcweb-overview-17 ID-nits
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I submitted PRs to address the ID-nits uncovered during my Shepherd =
review of -17:
- https://github.com/rtcweb-wg/rtcweb-overview/pull/11
- https://github.com/rtcweb-wg/rtcweb-overview/pull/10

spt=


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From: Sean Turner <sean@sn3rd.com>
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Note that after consultation with Alissa, I=E2=80=99m going to go ahead =
and push this draft to her plate.

spt

> On Feb 21, 2017, at 09:38, Sean Turner <sean@sn3rd.com> wrote:
>=20
> I submitted PRs to address the ID-nits uncovered during my Shepherd =
review of -17:
> - https://github.com/rtcweb-wg/rtcweb-overview/pull/11
> - https://github.com/rtcweb-wg/rtcweb-overview/pull/10
>=20
> spt


From nobody Wed Feb 22 13:15:02 2017
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I have reviewed this document in preparation for IETF last call. While =
the document content appears mostly ready, I don't think it makes sense =
to send this document out for IETF LC while the security documents are =
expired. For community review and IESG evaluation I think those doing =
the reviewing need at a minimum to feel like the normative dependencies =
in this document are in roughly the shape they will be in when those =
dependency documents get published. So at a minimum I think revisions of =
the security documents need to be published before IETF LC for this =
document makes sense.

Otherwise I have one substantive comment below and some nits to be =
resolved together with Sean=E2=80=99s PRs.

Substantive comment:

=3D Section 9 =3D

"Certain parts of the system SHOULD conform to certain properties, for
   instance:

   o  Echo cancellation should be good enough to achieve the suppression
      of acoustical feedback loops below a perceptually noticeable
      level.

   o  Privacy concerns MUST be satisfied; for instance, if remote
      control of camera is offered, the APIs should be available to let
      the local participant figure out who's controlling the camera, and
      possibly decide to revoke the permission for camera usage.

   o  Automatic gain control, if present, should normalize a speaking
      voice into a reasonable dB range."

I think there are some problematic uses of normative language here. =
Saying systems SHOULD conform to "certain properties" but not listing =
what all of those properties are and not defining them precisely seems =
problematic since implementers aren't being given clear directives. =
Likewise, the second bullet's MUST is quite vague and again just gives =
examples of how to comply rather than specifying precisely how. It's =
also confusing to use normative SHOULD in the heading but non-normative =
"should" in the bullets.

My suggestion would be to not use normative language in this part at =
all, since these aren't really hard requirements anyway. E.g.,=20

"Ideally certain parts of the system would conform to certain =
properties, for
   instance:

   o  Echo cancellation needs to be good enough to achieve the =
suppression
      of acoustical feedback loops below a perceptually noticeable
      level.=20
  =20
   o  Privacy concerns need to be satisfied; for instance, if remote
      control of camera is offered, the APIs should be available to let
      the local participant figure out who's controlling the camera, and
      possibly decide to revoke the permission for camera usage.

   o  Automatic gain control, if present, needs to normalize a speaking
      voice into a reasonable dB range."

If you really want to use normative language here, I would suggest that =
the "shoulds" in the bullets become SHOULDs and the middle bullet =
provide precise guidance about what is required.


Nits to be resolved together with Sean's PRs:

=3D Section 2.2 =3D

Would be good to cite the protocol spec in the bulleted list at the top.

s/Javascript API defined above/Javascript API cited above/

=3D Section 2.4 =3D

A citation for Jingle should be included.

I don't really understand why this text is included or what it means:

"NOTE: Where common definitions exist for these terms, those
   definitions should be used to the greatest extent possible."

=3D Section 7 =3D

It would be good to include citations for Bundle, RTCP-mux and Trickle =
ICE.


