From owner-tcp-impl@grc.nasa.gov  Thu Nov  1 02:13:19 2001
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Date: Thu, 01 Nov 2001 12:26:37 +0530
To: tcp-impl@grc.nasa.gov
From: Srinivas <srinia@qpackets.com>
Subject: Req:  Doubt regarding the USER call (send call)
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Hi,

This is srinivas from INDIA. Here my doubt follows.

Doubt regarding the user calls  (send call without urgent data and next 
send call with urgent data).

Take a case that TCP "A" and TCP "B" both are in established state.

Application sitting over "A" has given send call (without urgent bit) of 
16k bytes.
But the tcp stack of "A" is sending data to  "B" of 536 bytes due to MTU of 
the path.
(that is the MTU between "A" and "B" is 536).So TCP "A" takes some time to 
send all
the 16k bytes to the distant end. take a instant, that TCP "A" has sent 5k 
bytes and ready
to send the next 536 bytes segment. At this instant Application sitting 
over the TCP"A" has given
send call with urgent data of 10 bytes. So Here comes my doubt. Should 
TCP"A" has to send this data
  at this instant or It should queue this data and sent after the remaining 
11k bytes of data???

I think my question is clear.

Thanks and regards
Srinivas



From owner-tcp-impl@grc.nasa.gov  Thu Nov  1 02:38:35 2001
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Date: Thu, 01 Nov 2001 12:59:57 +0530
To: tcp-impl@grc.nasa.gov
From: Srinivas Amaravadi <srinia@trinc.com>
Subject: Req:  Any books or sites for USER calls.
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Hi,

Any books for user calls (section 3.9 of RFC 793.  Event processing,  Page 
52) which will give a little bit of explanation.

Thanks.

Regards,
Srinivas  .



From owner-tcp-impl@grc.nasa.gov  Fri Nov  2 10:54:02 2001
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Date: Fri, 02 Nov 2001 14:24:29 +0530
To: tcp-impl@grc.nasa.gov
From: Srinivas Amaravadi <srinia@trinc.com>
Subject: Req:  How to join this group !!!
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Hi,

This is A.Srinivas from INIDA.

I want to join this group, that is how can i get the questions and answers 
posted to this group.

Thanks

Regards,
Srinivas



From owner-tcp-impl@grc.nasa.gov  Tue Nov  6 03:02:47 2001
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Date: Tue, 06 Nov 2001 13:17:35 +0530
To: tcp-impl@grc.nasa.gov
From: Srinivas <srinia@qpackets.com>
Subject: Re:  Req:  How to join this group !!!
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Hello Sir,

Thanks for ur reply.
Sorry to disturb you, i am unable to find out the way from RFC 2142.Please 
help me in joining this group.

Thanks

regards,
Srinivas

At 12:47 PM 11/2/01 -0500, you wrote:
 > To: tcp-impl@grc.nasa.gov
 > I want to join this group, that is how can i get the questions and
 > answers posted to this group.

See RFC2142 section 6.

/~\ The ASCII der Mouse
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From owner-tcp-impl@grc.nasa.gov  Tue Nov  6 15:01:22 2001
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Date: Tue, 06 Nov 2001 20:46:29 +0100
To: tcp-impl@grc.nasa.gov
From: Ragon Philippe <pragon@mangoosta.fr>
Subject: Question about Solaris' way to compute RTO
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I am involved into TCP applications testing over a wireless network.
I am stumbling over a 'systematic' problem related to Solaris 2.6 when it 
is used as a server :  I am regurlarly getting spurrious retransmissions 
(Ack arriving a few tens of ms after the server has decided to retransmit) 
with a huge performance loss as a consequence.

This problem does not occur with Solaris 8, Linux or Windows NT 4 servers.
I have checked with traces and the use of Solaris' command 'ndd /dev/tcp 
tcp_status' that :
- Solaris 2.6 is taking one RTT sample per packet
- Solaris 2.6 seems to be RFC 2988 compliant in its computation of RTO from 
its RTT measurements (alpha =1/8, beta = 1/4, K= 4 and clock granularity 
is100 ms)

I am eager to get any information on how :
- Solaris 2.6 is actually computing RTO and whether there are some 
parameters to tune this computation.
- how Solaris 8 is different from Solaris 2.6 from this point of view

Any help will be appreciated.
---------------------------------------------------------------------
Philippe Ragon / Tour Atlas / 10, villa d'Este/ 75013 Paris/ FRANCE
Telephone & Fax    : (33) 01 45 70 79 53
E-mail                   :   pragon@mangoosta.fr
---------------------------------------------------------------------



From owner-tcp-impl@grc.nasa.gov  Wed Nov  7 04:20:34 2001
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Date: Wed, 07 Nov 2001 14:36:03 +0530
To: tcp-impl@grc.nasa.gov
From: Srinivas <srinia@qpackets.com>
Subject: Doubt regarding the time out option in OPEN call (user call)
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>
>Hi,
>
>This is regarding the OPEN call (user call). In open call there is a 
>timeout option.
>
>in RFC 793 it is stated that (page 45, open)  :-
>
>         The timeout, if present, permits the caller to set up a timeout
>         for all data submitted to TCP.  If data is not successfully
>         delivered to the destination within the timeout period, the TCP
>         will abort the connection.  The present global default is five
>         minutes.
>
>here comes my doubt:  is it the timeout for the entire connection ( that 
>is with in that timeout whole data of that particular connection has to 
>send to remote tcp )?? or within in the time out SYN segment should be 
>sent to the destination???
>
>i think i am clear.
>
>Thanks
>
>regards,
>Srinivas



From owner-tcp-impl@grc.nasa.gov  Wed Nov  7 11:40:48 2001
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          7 Nov 2001 11:17 EST
Date: Wed, 7 Nov 2001 11:17:58 -0500 (EST)
From: Constantinos Dovrolis <dovrolis@mail.eecis.udel.edu>
To: tcp-impl@grc.nasa.gov
MMDF-Warning:  Parse error in original version of preceding line at mail.eecis.udel.edu
Cc: Hao Jiang <hjiang@mail.eecis.udel.edu>
Subject: 3-way handshake delays
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Hi all,

we have a question about the generation of the SYN-ACK
packet at the callee.

Is it always true that the callee sends the SYN-ACK packet
immediately after receiving the SYN packet? I am not
referring to delays that can be caused by other
higher-priority activity at the kernel, but about delays
that would be caused by the code that processes the
SYN packet and sends the SYN-ACK packet.

Similarly, is it always true that the caller sends the
first ACK packet immediately after receicing the SYN-ACK
packet?

Thanks much,

Constantinos

Computer and Information Sciences - University of Delaware

http://www.cis.udel.edu/~dovrolis/



From owner-tcp-impl@grc.nasa.gov  Wed Nov  7 12:02:45 2001
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Date: Wed, 07 Nov 2001 08:50:44 -0800
From: Joe Touch <touch@ISI.EDU>
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To: Constantinos Dovrolis <dovrolis@mail.eecis.udel.edu>
Cc: tcp-impl@grc.nasa.gov, Hao Jiang <hjiang@mail.eecis.udel.edu>
Subject: Re: 3-way handshake delays
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Constantinos Dovrolis wrote:

> Hi all,
> 
> we have a question about the generation of the SYN-ACK
> packet at the callee.
> 
> Is it always true that the callee sends the SYN-ACK packet
> immediately after receiving the SYN packet? I am not
> referring to delays that can be caused by other
> higher-priority activity at the kernel, but about delays
> that would be caused by the code that processes the
> SYN packet and sends the SYN-ACK packet.
> 
> Similarly, is it always true that the caller sends the
> first ACK packet immediately after receicing the SYN-ACK
> packet?


Are you asking about a timeliness requirement, or about typical 
implementations?

Once the packet is being processed by TCP, the process-send exchange is 
typically immediate. The process/update state/emit packet exchange is 
considered atomic in RFC793, although there could be intermediate 
internal states, they don't serve much use given the input and output IP 
  queues.

However there can be delays in those IP queues. If there is a noticible 
delay, these might be the place to check first.

Joe



From owner-tcp-impl@grc.nasa.gov  Wed Nov  7 14:02:08 2001
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To: Constantinos Dovrolis <dovrolis@mail.eecis.udel.edu>
Cc: tcp-impl@grc.nasa.gov, Hao Jiang <hjiang@mail.eecis.udel.edu>
Subject: Re: 3-way handshake delays 
In-Reply-To: Your message of "Wed, 07 Nov 2001 11:17:58 EST."
             <Pine.GSO.4.33.0111071109520.2161-100000@louie.udel.edu> 
Date: Wed, 07 Nov 2001 13:50:19 -0500
From: Craig Partridge <craig@aland.bbn.com>
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In message <Pine.GSO.4.33.0111071109520.2161-100000@louie.udel.edu>, Constantin
os Dovrolis writes:

>Similarly, is it always true that the caller sends the
>first ACK packet immediately after receicing the SYN-ACK
>packet?

It is a long time since I've looked at this issue, but I believe it is
perfectly acceptable to pause very briefly before sending back the ACK to
piggy-back data with the ACK.  I don't know of any implementation that
does so, however.

Craig


From owner-tcp-impl@grc.nasa.gov  Wed Nov  7 14:18:36 2001
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From: Erich Nahum <nahum@watson.ibm.com>
Message-Id: <200111071910.OAA30594@orinoco.watson.ibm.com>
Subject: Re: 3-way handshake delays
To: craig@aland.bbn.com (Craig Partridge)
Date: Wed, 7 Nov 2001 14:10:03 -0500 (EST)
Cc: dovrolis@mail.eecis.udel.edu (Constantinos Dovrolis),
        tcp-impl@grc.nasa.gov, hjiang@mail.eecis.udel.edu (Hao Jiang)
In-Reply-To: <200111071850.fA7IoJq74826@aland.bbn.com> from "Craig Partridge" at Nov 07, 2001 01:50:19 PM
Reply-To: nahum@watson.ibm.com (Erich M. Nahum)
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Craig Partridge writes:
> 
> >Similarly, is it always true that the caller sends the
> >first ACK packet immediately after receicing the SYN-ACK
> >packet?
> 
> It is a long time since I've looked at this issue, but I believe it is
> perfectly acceptable to pause very briefly before sending back the ACK to
> piggy-back data with the ACK.  I don't know of any implementation that
> does so, however.

There's at least one that does so: AIX.  It's run-time configurable.
I have a paper that includes this very topic that will appear in ToN
soon.  FreeBSD may have something similar as well.

More broadly, the SYN-ACK may be delayed an arbitrary amount of time
depending on the implementation.  For example, some implementations
run their networking code off the interrupt stack, which provides
a very quick response, while others may use a kernel thread of
some kind (e.g. BSD soft interrupts or linux bottom-half handlers).
If you do the latter and you're hit with a lot of hardware interrupts,
it may take some time to respond to the SYN.

-Erich

-- 
Erich M. Nahum                  IBM T.J. Watson Research Center
Networking Research             P.O. Box 704
nahum@watson.ibm.com            Yorktown Heights NY 10598


From owner-tcp-impl@grc.nasa.gov  Wed Nov  7 14:21:41 2001
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Subject: Re: 3-way handshake delays
To: craig@aland.bbn.com (Craig Partridge)
Date: Wed, 7 Nov 2001 22:06:23 +0300 (MSK)
Cc: dovrolis@mail.eecis.udel.edu, tcp-impl@grc.nasa.gov,
        hjiang@mail.eecis.udel.edu
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Hello!

> piggy-back data with the ACK.  I don't know of any implementation that
> does so, however.

Linux does this, but only if application set a special option
on socket telling that active connector is going to send data first.

Alexey


From owner-tcp-impl@grc.nasa.gov  Wed Nov  7 16:31:31 2001
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Subject: Re: 3-way handshake delays
Cc: tcp-impl@grc.nasa.gov, hjiang@mail.eecis.udel.edu
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Some time ago, I remember seeing an idea which would cause this kind
of 3-way handshake delays.  I don't remember who suggested it or where.
Basically, the idea is to only send the SYN/ACK after the first I/O is
posted to the socket (read, write, probably select).  It "should" be
backwards-compatible, since programs usually do an I/O immediately after
accept(), but programs aware of the different API could, e.g., close the
connection (causing a RST, perhaps) based on IP address access control
(or some such), causing an attacker to think that the port is closed.

I never heard of an implementation of this, but I've always thought it
was a potentially neat idea.

  Bill


From owner-tcp-impl@grc.nasa.gov  Wed Nov  7 18:54:56 2001
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From: "Mehmet Yavuz" <myavuz@nortelnetworks.com>
To: tcp-impl@grc.nasa.gov
Subject: TCP Segment size
Date: Wed, 7 Nov 2001 17:42:32 -0600
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This message is in MIME format. Since your mail reader does not understand
this format, some or all of this message may not be legible.

------_=_NextPart_001_01C167E5.DE96EFD0
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	charset="iso-8859-1"


	Greetings,

	While running ftp tests ("put" command) with a Win95 laptop, I
observe that the advertised MSS is 960Bytes both for the server and client
(based on tcpdump logs). 
	However, tcpdump logs show that during the data transfer most
segments (more than 90%) are of size 512Bytes. Running "tracert" shows that
between the client and server the minimum MTU size used by routers is
1500Bytes.

	Does anybody have an idea why a smaller segment size (512B) is used
during the transfer compared to the MSS (960B)? 

	thanks,
	mehmet

BTW: Laptop has RS232 connection and for download tests ("get" command)
mostly a segment size of 960B is observed. 
	  

------_=_NextPart_001_01C167E5.DE96EFD0
Content-Type: text/html;
	charset="iso-8859-1"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2654.89">
<TITLE>TCP Segment size</TITLE>
</HEAD>
<BODY>
<BR>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>Greetings,</FONT>
</P>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>While =
running ftp tests (&quot;put&quot; command) with a Win95 laptop, I =
observe that the advertised MSS is 960Bytes both for the server and =
client (based on tcpdump logs). </FONT></P>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>However, =
tcpdump logs show that during the data transfer most segments (more =
than 90%) are of size 512Bytes. Running &quot;tracert&quot; shows that =
between the client and server the minimum MTU size used by routers is =
1500Bytes.</FONT></P>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>Does =
anybody have an idea why a smaller segment size (512B) is used during =
the transfer compared to the MSS (960B)? </FONT></P>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>thanks,</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>mehmet</FONT>
</P>

<P><FONT SIZE=3D2>BTW: Laptop has RS232 connection and for download =
tests (&quot;get&quot; command) mostly a segment size of 960B is =
observed. </FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>&nbsp; =
</FONT>
</P>

</BODY>
</HTML>
------_=_NextPart_001_01C167E5.DE96EFD0--


From owner-tcp-impl@grc.nasa.gov  Wed Nov  7 20:01:33 2001
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Date: Wed, 07 Nov 2001 16:52:07 -0800
From: Joe Touch <touch@ISI.EDU>
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To: Bill Fenner <fenner@research.att.com>
Cc: dovrolis@mail.eecis.udel.edu, tcp-impl@grc.nasa.gov,
        hjiang@mail.eecis.udel.edu
Subject: Re: 3-way handshake delays
References: <200111072118.NAA19703@windsor.research.att.com>
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Bill Fenner wrote:

> Some time ago, I remember seeing an idea which would cause this kind
> of 3-way handshake delays.  I don't remember who suggested it or where.
> Basically, the idea is to only send the SYN/ACK after the first I/O is
> posted to the socket (read, write, probably select).  It "should" be
> backwards-compatible, since programs usually do an I/O immediately after
> accept(), but programs aware of the different API could, e.g., close the
> connection (causing a RST, perhaps) based on IP address access control
> (or some such), causing an attacker to think that the port is closed.
> 
> I never heard of an implementation of this, but I've always thought it
> was a potentially neat idea.


Seems like a good way to cause a timeout.

The data can't go to the application anyway; the connection doesn't 
exist until after the ACK comes back, so making a decision based on any 
connection information seems invalid, IMO.

Joe



From owner-tcp-impl@grc.nasa.gov  Wed Nov  7 20:06:11 2001
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Date: Wed, 07 Nov 2001 16:56:38 -0800
From: Joe Touch <touch@ISI.EDU>
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To: Mehmet Yavuz <myavuz@nortelnetworks.com>
Cc: tcp-impl@grc.nasa.gov
Subject: Re: TCP Segment size
References: <1B54FA3A2709D51195C800508BF9386A027A5F00@zrc2c000.us.nortel.com>
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Mehmet Yavuz wrote:

> 
>         Greetings,
> 
>         While running ftp tests ("put" command) with a Win95 laptop, I 
> observe that the advertised MSS is 960Bytes both for the server and 
> client (based on tcpdump logs).
> 
>         However, tcpdump logs show that during the data transfer most 
> segments (more than 90%) are of size 512Bytes. Running "tracert" shows 
> that between the client and server the minimum MTU size used by routers 
> is 1500Bytes.
> 
>         Does anybody have an idea why a smaller segment size (512B) is 
> used during the transfer compared to the MSS (960B)?


If the destination is NOT the local subnet, it is likely that the 
Internet MSS of 576 is used, rather than the link MTU. Presuming that 
path MTU is not enabled. At that point, most implementations send 512 
bytes of data, rather than 536, favoring the power-of-two.

 > BTW: Laptop has RS232 connection and for download tests
 > ("get" command) mostly a segment size of 960B is observed.

When the destination is the local subnet (subnet coincident with an 
interface), some implementations use the link MTU even if path MTU isn't 
enabled, assuming that the 'path' is direct.

(this is in most of the TCP books, FWIW)

Joe







From owner-tcp-impl@grc.nasa.gov  Thu Nov  8 00:40:38 2001
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From: Erich Nahum <nahum@watson.ibm.com>
Message-Id: <200111080528.AAA39168@orinoco.watson.ibm.com>
Subject: Re: 3-way handshake delays
To: touch@ISI.EDU (Joe Touch)
Date: Thu, 8 Nov 2001 00:28:48 -0500 (EST)
Cc: fenner@research.att.com (Bill Fenner), dovrolis@mail.eecis.udel.edu,
        tcp-impl@grc.nasa.gov, hjiang@mail.eecis.udel.edu
In-Reply-To: <3BE9D737.3090706@isi.edu> from "Joe Touch" at Nov 07, 2001 04:52:07 PM
Reply-To: nahum@watson.ibm.com (Erich M. Nahum)
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Joe Touch writes:
> 
> > Some time ago, I remember seeing an idea which would cause this kind
> > of 3-way handshake delays.  I don't remember who suggested it or where.
> > Basically, the idea is to only send the SYN/ACK after the first I/O is
> > posted to the socket (read, write, probably select).  It "should" be
> > backwards-compatible, since programs usually do an I/O immediately after
> > accept(), but programs aware of the different API could, e.g., close the
> > connection (causing a RST, perhaps) based on IP address access control
> > (or some such), causing an attacker to think that the port is closed.
> > 
> > I never heard of an implementation of this, but I've always thought it
> > was a potentially neat idea.
> 
> Seems like a good way to cause a timeout.
> 
> The data can't go to the application anyway; the connection doesn't 
> exist until after the ACK comes back, so making a decision based on any 
> connection information seems invalid, IMO.

Joe is correct.  The connection can't be delivered to the application
(or accept queue) until the client ACK returns for the server's SYN-ACK.

I mis-understood the original question.  What can happen on AIX is that
the ACK of the SYN-ACK on the *client* can be delayed, hoping to piggyback 
on an outgoing segment (such as the first GET request) (since connect() only
blocks until the SYN is ACKed).  The server shouldn't deliberately delay 
the ACK of the SYN from the client.

-Erich

-- 
Erich M. Nahum                  IBM T.J. Watson Research Center
Networking Research             P.O. Box 704
nahum@watson.ibm.com            Yorktown Heights NY 10598


From owner-tcp-impl@grc.nasa.gov  Thu Nov  8 05:41:14 2001
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Date: Thu, 08 Nov 2001 15:58:11 +0530
To: tcp-impl@grc.nasa.gov
From: Srinivas <srinia@qpackets.com>
Subject: doubt regarding the OPEN ( passive) call
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Hi,

In RFC 793 page 45 ,[section OPEN second para] it is mentioned as :-

  If the active/passive flag is set to passive, then this is a
         call to LISTEN for an incoming connection.  A PASSIVE OPEN MAY
         HAVE EITHER A FULLY SPECIFIED FOREIGN SOCKET TO WAIT FOR A
         PARTICULAR CONNECTION

here comes my doubt:

(pls read the above BOLD letter lines) is there any real application which 
gives this type of calls?

i think iam clear

thanks

Regards,
Srinivas



From owner-tcp-impl@grc.nasa.gov  Thu Nov  8 07:13:00 2001
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Date: Thu, 08 Nov 2001 17:36:02 +0530
From: Balaji Ramakrishnan <balajir@qpackets.com>
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To: tcp-impl@egroups.com
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Subject: Doubt reg RFC 793 page 72
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Hi all,
     A basic doubt reg windows in TCP.

1. The window advt by a TCP is its receive window space.

2,       From Page 72 in rfc 793 ,

          If SND.UNA < SEG.ACK =< SND.NXT, the send window should be
          updated.  If (SND.WL1 < SEG.SEQ or (SND.WL1 = SEG.SEQ and
          SND.WL2 =< SEG.ACK)), set SND.WND <- SEG.WND, set
          SND.WL1 <- SEG.SEQ, and set SND.WL2 <- SEG.ACK.

     If we start to update ur send window evrytime like this means we 
will be basically
     shrinking our send window every time we get small window 
advertisement.Is this
     correct or what is the case the above condition is valid.

                     Pl respond and thanks in advance.
Regards
Balaji Ramakrishnan.








From owner-tcp-impl@grc.nasa.gov  Mon Nov 12 05:30:57 2001
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Date: Mon, 12 Nov 2001 15:33:30 +0530
From: Balaji Ramakrishnan <balajir_viz@yahoo.com>
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Hi
1.I have one basic doubt in TCP retransmissions.
   What is an loss in a network ?
   Eg.If u are in Congestion avoidance phase and we have sent 10 segments
   If one segment 's RTT gets timed out.Is it a case of  loss and we go to
   slow start.

2.Also reg SACK - SACK is used by Tx during retransmissions.
   Retransmisions occur
   1.Due to Fast Retransmit
   2.Due to Retransmission timeout also.
               Also in the end of RFC2018 they have specified that SACK 
info should be discarded
     when a timeout happens.(what this timeout refer to actually).

Expecting some reply.
Thankx in advance
Balaji.R
           


    



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Date: Tue, 13 Nov 2001 10:19:15 +0530
To: tcp-impl@grc.nasa.gov
From: Srinivas Amaravadi <srinia@trinc.com>
Subject: Req:   Doubt regarding the MANAGING THE RTO TIMER
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Hi  All,

Here is a small doubt in RFC  2988 section 5.3  page4


  (5.3) When an ACK is received that acknowledges new data, restart the
          retransmission timer so that it will expire after RTO seconds
          (for the current value of RTO).

Why should the timer restart ??? section 5.1 and 5.2 are clear but 5.3 is 
not clear

Thanks

Srinivas




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Date: Tue, 13 Nov 2001 11:09:12 +0530 (IST)
From: Chetan Kumar <chetan@protocol.ece.iisc.ernet.in>
To: tcp-impl@grc.nasa.gov
Subject: test suites for TCP
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Hi,

	I would like to know is there any standard test suites to test TCP
implementations.

Thanks
Chetan S



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From: "Andrei Gurtov" <gurtov@cs.Helsinki.FI>
To: <tcp-impl@grc.nasa.gov>
Subject: RE: test suites for TCP
Date:   Tue, 13 Nov 2001 09:17:49 +0200
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> -----Original Message-----
> From: owner-tcp-impl@grc.nasa.gov [mailto:owner-tcp-impl@grc.nasa.gov]On
> Behalf Of Chetan Kumar
> Sent: 13. marraskuuta 2001 7:39
> To: tcp-impl@grc.nasa.gov
> Subject: test suites for TCP
>
>
> Hi,
>
> 	I would like to know is there any standard test suites to test TCP
> implementations.

There is a number of 'fingerprinting' methods to check which TCP
implementation is used, see e.g.
http://www.networkice.com/Advice/Countermeasures/Scanners/Fingerprinting/def
ault.htm

TBIT tool by ACIRI can tell whether a given TCP uses congestion control, the
size of inital window, etc.
http://www.aciri.org/tbit/

However, it would be nice to have a standard test suit for a network
emulator like Dummynet, Nistnet, ONE or Seawind to be able to test different
TCPs by inserting a pattern of drops, delay or reodering into a connection.

Andrei



From owner-tcp-impl@grc.nasa.gov  Tue Nov 13 02:33:51 2001
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From: "Andrei Gurtov" <gurtov@cs.Helsinki.FI>
To: <tcp-impl@grc.nasa.gov>
Subject: RE: Doubt regarding the MANAGING THE RTO TIMER
Date:   Tue, 13 Nov 2001 09:01:29 +0200
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> -----Original Message-----
> From: owner-tcp-impl@grc.nasa.gov [mailto:owner-tcp-impl@grc.nasa.gov]On
> Behalf Of Srinivas Amaravadi
> Sent: 13. marraskuuta 2001 6:49
> To: tcp-impl@grc.nasa.gov
> Subject: Req: Doubt regarding the MANAGING THE RTO TIMER
>
>
> Hi  All,
>
> Here is a small doubt in RFC  2988 section 5.3  page4
>
>
>   (5.3) When an ACK is received that acknowledges new data, restart the
>           retransmission timer so that it will expire after RTO seconds
>           (for the current value of RTO).
>
> Why should the timer restart ??? section 5.1 and 5.2 are clear but 5.3 is
> not clear

If the timer is not restarted a spurious RTO can easily occur. Imagine the
slow start phase and that the RTO value is updated once per RTT. The RTT can
double between two samples due to queuing delay. If RTO is not restarted, a
spurious timeout will occur before a new RTT sample is collected.

More discussion can be found in:
Reiner Ludwig, Keith Sklower, The Eifel Retransmission Timer. Appears in ACM
Computer Communications Review, Vol. 30, No. 3, July 2000.
http://iceberg.cs.berkeley.edu/publications.html

In my opinion to avoid spurious RTOs TCP should collect RTT samples more
frequently than once per window, and restart the timer also on fast
retransmit, and possibly on partial ACKs and DUPACKs.


Andrei



From owner-tcp-impl@grc.nasa.gov  Tue Nov 13 04:28:58 2001
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From: Andi Kleen <ak@muc.de>
To: Chetan Kumar <chetan@protocol.ece.iisc.ernet.in>
Cc: tcp-impl@grc.nasa.gov
Subject: Re: test suites for TCP
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On Tue, Nov 13, 2001 at 06:39:12AM +0100, Chetan Kumar wrote:
> Hi,
> 
> 	I would like to know is there any standard test suites to test TCP
> implementations.
There is unfortunately no standard test suite for TCP. That is one of the
reasons why it is so hard to write a good TCP.

-Andi


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Date: Tue, 13 Nov 2001 17:26:58 +0530 (IST)
From: Chetan Kumar <chetan@protocol.ece.iisc.ernet.in>
To: Andrei Gurtov <gurtov@cs.Helsinki.FI>
Cc: tcp-impl@grc.nasa.gov
Subject: RE: test suites for TCP
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On Tue, 13 Nov 2001, Andrei Gurtov wrote:

*>> Hi,
*>> 	I would like to know is there any standard test suites to test TCP
*>> implementations.
*>There is a number of 'fingerprinting' methods to check which TCP
*>implementation is used, see e.g.
<snip>

*>
*>However, it would be nice to have a standard test suit for a network

Yes, I feal the same.. Just now mail Andi on the same issue 

(copied the same again)

Just a question: 
Why doesn't some one write something like "standard test suite for XX
protocol" make this part of standard IETF documents. Like there are
various Internet protocol defined, and it is a real good idea to have a
standard test suites so that it much easier for implementors to write a
good inter-operable code. Especially if this is IETF document..

Thanks
Chetan S

*>emulator like Dummynet, Nistnet, ONE or Seawind to be able to test different
*>TCPs by inserting a pattern of drops, delay or reodering into a connection.
*>
*>Andrei
*>

-- 



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Message-Id: <200111131633.QAA04067@gra.isi.edu>
To: chetan@protocol.ece.iisc.ernet.in, ak@muc.de
Subject: Re: test suites for TCP
Cc: tcp-impl@grc.nasa.gov
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  *> From owner-tcp-impl@grc.nasa.gov  Tue Nov 13 01:30:31 2001
  *> Date: Tue, 13 Nov 2001 10:12:54 +0100
  *> From: Andi Kleen <ak@muc.de>
  *> To: Chetan Kumar <chetan@protocol.ece.iisc.ernet.in>
  *> Cc: tcp-impl@grc.nasa.gov
  *> Subject: Re: test suites for TCP
  *> Mime-Version: 1.0
  *> 
  *> On Tue, Nov 13, 2001 at 06:39:12AM +0100, Chetan Kumar wrote:
  *> > Hi,
  *> > 
  *> > 	I would like to know is there any standard test suites to test TCP
  *> > implementations.
  *> There is unfortunately no standard test suite for TCP. That is one of the
  *> reasons why it is so hard to write a good TCP.
  *> 
  *> -Andi
  *> 

Unnnhhh, the lack of a standard test is the least of an implementor's
problems!!  I wrote my TCP in 1979, and I came away from the experience
with a profound respect for the sublety of the protocol.

Bob Braden


From owner-tcp-impl@grc.nasa.gov  Tue Nov 13 13:46:25 2001
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Date: Tue, 13 Nov 2001 10:28:53 -0800
From: Murali Bashyam <mbashyam@cisco.com>
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To: Andi Kleen <ak@muc.de>
Cc: Chetan Kumar <chetan@protocol.ece.iisc.ernet.in>, tcp-impl@grc.nasa.gov
Subject: Re: test suites for TCP
References: <Pine.LNX.4.21.0111131107540.5687-100000@protocol.ece.iisc.ernet.in> <20011113101254.50970@colin.muc.de>
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Andi Kleen wrote:

> On Tue, Nov 13, 2001 at 06:39:12AM +0100, Chetan Kumar wrote:
> > Hi,
> >
> >       I would like to know is there any standard test suites to test TCP
> > implementations.
> There is unfortunately no standard test suite for TCP. That is one of the
> reasons why it is so hard to write a good TCP.

ANVL test suite from Midnight networks would be a good place to start.

http://www.empirix.com/Empirix/voice+network+test/products/_anvl+faq.html.

Murali

>
>
> -Andi



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Message-ID: <639BAB6ABE61D411B3870090276A8975033323@hermes.corp.iready.com>
From: Steve Potuzak <spotuzak@corp.iready.com>
To: "'tcp-impl@grc.nasa.gov'" <tcp-impl@grc.nasa.gov>
Cc: Steve Potuzak <spotuzak@corp.iready.com>,
        Robin Uyeshiro <ruyeshiro@iready.com>,
        "'mailto:gurtov@cs.Helsinki.FI'" <mailto:gurtov@cs.Helsinki.FI>
Subject: RE: test suites for TCP
Date: Tue, 13 Nov 2001 12:28:02 -0800
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Hello,

You might want to look into the ANVL data-network product line from
Empirix..
it consists of a family of automated test suites that attempt to test,
as completely as possible, against relevant RFC's (especially 793).
At least some of the flexibility you mentioned concerning delay or
reordering
may possibly be introduced via their associated TCP development toolkit,
although since I have not attempted it myself, I can't say with certainty.
In any case, we are using the ANVL core TCP/IP test suite here at iReady
Corporation.

The following are related links..

http://www.empirix.com/empirix/voice+network+test/products/data+network+test
.html

http://www.empirix.com/Empirix/default.html


Steven Potuzak
sr. diagnostic software engineer
iReady Corporation
spotuzak@iready.com
808.944.5257 x21
1357 Kapiolani Blvd. Suite 102
Honolulu, HI 96814


-----Original Message-----
From: Robin Uyeshiro 
Sent: Tuesday, November 13, 2001 8:41 AM
To: Steve Potuzak
Subject: FW: test suites for TCP




-----Original Message-----
From: Andrei Gurtov [mailto:gurtov@cs.Helsinki.FI]
Sent: Monday, November 12, 2001 9:18 PM
To: tcp-impl@grc.nasa.gov
Subject: RE: test suites for TCP




> -----Original Message-----
> From: owner-tcp-impl@grc.nasa.gov [mailto:owner-tcp-impl@grc.nasa.gov]On
> Behalf Of Chetan Kumar
> Sent: 13. marraskuuta 2001 7:39
> To: tcp-impl@grc.nasa.gov
> Subject: test suites for TCP
>
>
> Hi,
>
> 	I would like to know is there any standard test suites to test TCP
> implementations.

There is a number of 'fingerprinting' methods to check which TCP
implementation is used, see e.g.
http://www.networkice.com/Advice/Countermeasures/Scanners/Fingerprinting/def
ault.htm

TBIT tool by ACIRI can tell whether a given TCP uses congestion control, the
size of inital window, etc.
http://www.aciri.org/tbit/

However, it would be nice to have a standard test suit for a network
emulator like Dummynet, Nistnet, ONE or Seawind to be able to test different
TCPs by inserting a pattern of drops, delay or reodering into a connection.

Andrei

------_=_NextPart_001_01C16C81.B1565290
Content-Type: text/html;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2653.12">
<TITLE>RE: test suites for TCP</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>Hello,</FONT>
</P>

<P><FONT SIZE=3D2>You might want to look into the ANVL data-network =
product line from Empirix..</FONT>
<BR><FONT SIZE=3D2>it consists of a family of automated test suites =
that attempt to test,</FONT>
<BR><FONT SIZE=3D2>as completely as possible, against relevant RFC's =
(especially 793).</FONT>
<BR><FONT SIZE=3D2>At least some of the flexibility you mentioned =
concerning delay or reordering</FONT>
<BR><FONT SIZE=3D2>may possibly be introduced via their associated TCP =
development toolkit,</FONT>
<BR><FONT SIZE=3D2>although since I have not attempted it myself, I =
can't say with certainty.</FONT>
<BR><FONT SIZE=3D2>In any case, we are using the ANVL core TCP/IP test =
suite here at iReady Corporation.</FONT>
</P>

<P><FONT SIZE=3D2>The following are related links..</FONT>
</P>

<P><FONT SIZE=3D2><A =
HREF=3D"http://www.empirix.com/empirix/voice+network+test/products/data+=
network+test.html" =
TARGET=3D"_blank">http://www.empirix.com/empirix/voice+network+test/prod=
ucts/data+network+test.html</A></FONT>
</P>

<P><FONT SIZE=3D2><A =
HREF=3D"http://www.empirix.com/Empirix/default.html" =
TARGET=3D"_blank">http://www.empirix.com/Empirix/default.html</A></FONT>=

</P>
<BR>

<P><FONT SIZE=3D2>Steven Potuzak</FONT>
<BR><FONT SIZE=3D2>sr. diagnostic software engineer</FONT>
<BR><FONT SIZE=3D2>iReady Corporation</FONT>
<BR><FONT SIZE=3D2>spotuzak@iready.com</FONT>
<BR><FONT SIZE=3D2>808.944.5257 x21</FONT>
<BR><FONT SIZE=3D2>1357 Kapiolani Blvd. Suite 102</FONT>
<BR><FONT SIZE=3D2>Honolulu, HI 96814</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Robin Uyeshiro </FONT>
<BR><FONT SIZE=3D2>Sent: Tuesday, November 13, 2001 8:41 AM</FONT>
<BR><FONT SIZE=3D2>To: Steve Potuzak</FONT>
<BR><FONT SIZE=3D2>Subject: FW: test suites for TCP</FONT>
</P>
<BR>
<BR>
<BR>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Andrei Gurtov [<A =
HREF=3D"mailto:gurtov@cs.Helsinki.FI">mailto:gurtov@cs.Helsinki.FI</A>]<=
/FONT>
<BR><FONT SIZE=3D2>Sent: Monday, November 12, 2001 9:18 PM</FONT>
<BR><FONT SIZE=3D2>To: tcp-impl@grc.nasa.gov</FONT>
<BR><FONT SIZE=3D2>Subject: RE: test suites for TCP</FONT>
</P>
<BR>
<BR>
<BR>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: owner-tcp-impl@grc.nasa.gov [<A =
HREF=3D"mailto:owner-tcp-impl@grc.nasa.gov">mailto:owner-tcp-impl@grc.na=
sa.gov</A>]On</FONT>
<BR><FONT SIZE=3D2>&gt; Behalf Of Chetan Kumar</FONT>
<BR><FONT SIZE=3D2>&gt; Sent: 13. marraskuuta 2001 7:39</FONT>
<BR><FONT SIZE=3D2>&gt; To: tcp-impl@grc.nasa.gov</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: test suites for TCP</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; Hi,</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; I would like to =
know is there any standard test suites to test TCP</FONT>
<BR><FONT SIZE=3D2>&gt; implementations.</FONT>
</P>

<P><FONT SIZE=3D2>There is a number of 'fingerprinting' methods to =
check which TCP</FONT>
<BR><FONT SIZE=3D2>implementation is used, see e.g.</FONT>
<BR><FONT SIZE=3D2><A =
HREF=3D"http://www.networkice.com/Advice/Countermeasures/Scanners/Finger=
printing/def" =
TARGET=3D"_blank">http://www.networkice.com/Advice/Countermeasures/Scann=
ers/Fingerprinting/def</A></FONT>
<BR><FONT SIZE=3D2>ault.htm</FONT>
</P>

<P><FONT SIZE=3D2>TBIT tool by ACIRI can tell whether a given TCP uses =
congestion control, the</FONT>
<BR><FONT SIZE=3D2>size of inital window, etc.</FONT>
<BR><FONT SIZE=3D2><A HREF=3D"http://www.aciri.org/tbit/" =
TARGET=3D"_blank">http://www.aciri.org/tbit/</A></FONT>
</P>

<P><FONT SIZE=3D2>However, it would be nice to have a standard test =
suit for a network</FONT>
<BR><FONT SIZE=3D2>emulator like Dummynet, Nistnet, ONE or Seawind to =
be able to test different</FONT>
<BR><FONT SIZE=3D2>TCPs by inserting a pattern of drops, delay or =
reodering into a connection.</FONT>
</P>

<P><FONT SIZE=3D2>Andrei</FONT>
</P>

</BODY>
</HTML>
------_=_NextPart_001_01C16C81.B1565290--


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You might find RFC 1025 either a useful source or a source of amusement,
depending on your viewpoint.

Bob Braden


From owner-tcp-impl@grc.nasa.gov  Thu Nov 15 19:37:15 2001
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To: Bob Braden <braden@ISI.EDU>
Cc: chetan@protocol.ece.iisc.ernet.in, ak@muc.de, tcp-impl@grc.nasa.gov
Subject: Re: test suites for TCP
References: <200111131633.QAA04067@gra.isi.edu>
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I wrote mine in 1999 so can remember the experience reasonable well.
TCP/IP may be subtle, RFC793 is a mess. I would have enormous respect
for anyone who could rewrite it a tight manner.

Regards

Bob Braden wrote:
> 
>   *> From owner-tcp-impl@grc.nasa.gov  Tue Nov 13 01:30:31 2001
>   *> Date: Tue, 13 Nov 2001 10:12:54 +0100
>   *> From: Andi Kleen <ak@muc.de>
>   *> To: Chetan Kumar <chetan@protocol.ece.iisc.ernet.in>
>   *> Cc: tcp-impl@grc.nasa.gov
>   *> Subject: Re: test suites for TCP
>   *> Mime-Version: 1.0
>   *>
>   *> On Tue, Nov 13, 2001 at 06:39:12AM +0100, Chetan Kumar wrote:
>   *> > Hi,
>   *> >
>   *> >  I would like to know is there any standard test suites to test TCP
>   *> > implementations.
>   *> There is unfortunately no standard test suite for TCP. That is one of the
>   *> reasons why it is so hard to write a good TCP.
>   *>
>   *> -Andi
>   *>
> 
> Unnnhhh, the lack of a standard test is the least of an implementor's
> problems!!  I wrote my TCP in 1979, and I came away from the experience
> with a profound respect for the sublety of the protocol.
> 
> Bob Braden


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From: Bob Braden <braden@ISI.EDU>
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Date: Fri, 16 Nov 2001 00:33:26 GMT
Message-Id: <200111160033.AAA05641@gra.isi.edu>
To: braden@ISI.EDU, charlese@cvs.com.au
Subject: Re: test suites for TCP
Cc: chetan@protocol.ece.iisc.ernet.in, ak@muc.de, tcp-impl@grc.nasa.gov
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  *> From charlese@cvs.com.au  Thu Nov 15 16:14:34 2001
  *> Date: Fri, 16 Nov 2001 22:46:15 +1100
  *> From: Charles Esson <charlese@cvs.com.au>
  *> X-Accept-Language: en
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  *> To: Bob Braden <braden@ISI.EDU>
  *> CC: chetan@protocol.ece.iisc.ernet.in, ak@muc.de, tcp-impl@grc.nasa.gov
  *> Subject: Re: test suites for TCP
  *> Content-Transfer-Encoding: 7bit
  *> 
  *> I wrote mine in 1999 so can remember the experience reasonable well.
  *> TCP/IP may be subtle, RFC793 is a mess. I would have enormous respect
  *> for anyone who could rewrite it a tight manner.
  *> 
  *> Regards
  *> 

Actually I have profound admiration for RFC 793.  At the time, it was
a remarkable achievement, and I believe that today very, very few
protocol documents achieve the same clarity and elegance.  Of course
it omits many subtleties that were added later -- slow start,
congestion avoidance, large windows, PAWS, SACK, ...

Bob Braden


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Date: Fri, 16 Nov 2001 09:54:20 +0100
From: Henk Langeveld - NL <Henk.Langeveld@Sun.COM>
Subject: Re: test suites for TCP
To: tcp-impl@grc.nasa.gov
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Bob Braden wrote:

>   *> From: Charles Esson <charlese@cvs.com.au>
>   *> 
>   *> I wrote mine in 1999 so can remember the experience reasonable well.
>   *> TCP/IP may be subtle, RFC793 is a mess. I would have enormous respect
>   *> for anyone who could rewrite it a tight manner.
> 
> Actually I have profound admiration for RFC 793.  At the time, it was
> a remarkable achievement, and I believe that today very, very few
> protocol documents achieve the same clarity and elegance.  Of course
> it omits many subtleties that were added later -- slow start,
> congestion avoidance, large windows, PAWS, SACK, ...



A risk of test-suites in any industry is that people may start to
confuse the test-suite with the standard.  No test-suite can ever
be exhaustive.

Internet Standards require interoperability.
This is why interop (and the various bake-offs[?]) was started.  You need
to keep coming back, and keep matching up new implementations of old and
new standards and extensions to them.

Henk Langeveld




From owner-tcp-impl@grc.nasa.gov  Wed Nov 21 16:54:16 2001
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From: info@ltssg3.epfl.ch
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To: tcp-impl@lerc.nasa.gov
Date: Wed Nov 21 23:47:37 2001
Subject: CALL FOR PAPERS: IEEE ICME 2002, Int.Conf.MM & Expo
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Apologies for any duplication of this announcement
_____________________________________________

CALL FOR PAPERS: ICME 2002
IEEE International Conference on Multimedia and Expo

Swiss Federal Institute of Technology,
EPFL, Lausanne, Switzerland
August 26-29 2002

GOALS OF THE CONFERENCE

International Conference on Multimedia & Expo (ICME) is a major annual 
international conference organized with the objective of bringing together 
researchers, developers and practitioners from academia and industry 
working in all areas of  multimedia. ICME serves as a forum for the
dissemination of state-of-the-art research, development, and implementations 
of multimedia systems, technologies and applications. 

Co-sponsored by four IEEE societies (the Circuits and Systems Society, 
the Communications Society, the Computer Society and the Signal Processing 
Society) the third edition of ICME will be held in Lausanne, Switzerland.

INSTRUCTIONS AND TOPICS

Authors should submit a four-page manuscript in double-column format 
including authors' names, affiliations and a short abstract. Only electronic 
submission will be accepted. A sample of the papers presented at the 
conference will be selected for a possible publication in an upcoming 
special issue of IEEE Transactions on Multimedia. Topics covered include 
but are not limited to the following:

* Audio, image and/or video processing
* Components and technologies for multimedia systems
* Human-machine interface and interaction
* Multimedia applications
* Multimedia hardware architectures
* Multimedia communication and networking
* Multimedia computing systems
* Multimedia content access and distribution
* Multimedia databases
* Signal processing for media integration
* Standards (e.g., MPEG) and related issues
* System integration, integration of art and technologies
* Virtual reality and computer graphics
* Watermarking and security

CONFERENCE WEB SITE AND CONTACT

Web: http://www.icme2002.org
Contact: info@icme02.epfl.ch

SPECIAL SESSIONS, TUTORIALS, DEMO/EXHIBITS

Proposals for Special Sessions, Tutorials and Demo/Exhibits are also 
strongly encouraged. Authors are referred to the ICME website to additional 
information regarding submissions.

IMPORTANT DATES

Special Session Proposal Due            December 1, 2001
Regular Papers Submission Due           February 15, 2002
Tutorial Proposal Due                   March 1, 2002
Demo/Exhibit Proposal Due               May 1, 2002


Thierry Pun, Univ. of Geneva, Switzerland
Jean-Luc Dugelay, Eurecom, France
ICME Publicity co-Chairs


PS: feel free to distribute!

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From owner-tcp-impl@grc.nasa.gov  Mon Nov 26 04:25:24 2001
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Date: Mon, 26 Nov 2001 14:42:15 +0530
From: Balaji Ramakrishnan <balajir@qpackets.com>
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Subject: Duplicate Ack implementation
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--------------070703060306010304060204
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Hi all,
      In the implementation of fast retransmit algorithm we check receipt of
      duplicate acks.

 Q1.So whenever a receiver ou-of order in-sequence segment
       it should send a Duplicate ACK segment immediately.
       So can data accompany this duplicate Ack segment.
 
Q2. If data accompanies a ACK response for out -of -segment then
      how to differentiate between the normal transmission of segments
     where the RCV.nxt is copied as Ack
     i.e we are just transmitting but no segments we received.
 
Q3.Is there any RFC which says what an duplicate ack segment should contain
      for receipt of out-of -order segment or it is implementation specific.

Thankx in advance
Regards
Balaji.R
   

     


--------------070703060306010304060204
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<html>
<head>
</head>
<body>
Hi all,<br>
&nbsp; &nbsp; &nbsp; In the implementation of fast retransmit algorithm we check receipt
of<br>
&nbsp; &nbsp; &nbsp; duplicate acks.<br>
<br>
<b>&nbsp;Q1.</b>So whenever a receiver ou-of order in-sequence segment<br>
&nbsp; &nbsp; &nbsp;&nbsp; it should send a Duplicate ACK segment immediately.<br>
&nbsp; &nbsp; &nbsp;&nbsp; So can data accompany this duplicate Ack segment.<br>
&nbsp;<br>
<b>Q2</b>. If data accompanies a ACK response for out -of -segment then <br>
&nbsp; &nbsp; &nbsp; how to differentiate between the normal transmission of segments<br>
&nbsp; &nbsp; &nbsp;where the RCV.nxt is copied as Ack <br>
&nbsp; &nbsp; &nbsp;i.e we are just transmitting but no segments we received.<br>
&nbsp; <br>
<b>Q3</b>.Is there any RFC which says what an duplicate ack segment should
contain<br>
&nbsp; &nbsp;&nbsp; &nbsp;for receipt of out-of -order segment or it is implementation specific.<br>
<br>
Thankx in advance<br>
Regards<br>
Balaji.R<br>
&nbsp; &nbsp; <br>
<br>
&nbsp; &nbsp; &nbsp; <br>
<br>
</body>
</html>

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