From exim@www1.ietf.org  Tue Jul  1 04:12:32 2003
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Date: Tue, 1 Jul 2003 09:07:15 +0100
From: Colin Perkins <csp@csperkins.org>
To: avt@ietf.org
Cc: casner@acm.org, magnus.westerlund@ericsson.com
Subject: Re: [AVT] Meeting in Vienna
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A reminder that the deadline for agenda slot requests is today. If you need
an agenda slot in AVT, and have not already contacted the chairs, please do
so shortly.

Colin


--> Colin Perkins <csp@csperkins.org> writes:
> Folks,
> 
> The AVT working group is scheduled for one slot at the 57th IETF meeting
> in Vienna.  On the tentative agenda, this is from 9:00 to 11:30 on Monday
> 14th July.  This date is subject to change - DO NOT USE IT TO PLAN
> TRAVEL.
> 
> Those wanting agenda time at the meeting should contact the working group
> chairs by 1st July. Submitting a draft does not automatically provide
> agenda time. 
> 
> You are reminded that the cut-off date for -00 Internet Draft submissions
> is Monday, 23rd June at 9:00am EDT (GMT -04:00). Unless you have prior
> approval from the chairs, -00 submissions should have filenames with the
> format draft-yourname-avt-...-00.txt. Be advised that updates to -00
> drafts received after 23rd June will not be accepted. 
> 
> All other Internet-Draft submissions must be made by Monday, 30th June at
> 9:00am EDT.
> 
> Drafts which are submitted late and do not make the cutoff date will not
> be discussed at the meeting.
> 
> -- 
> Colin Perkins
> csp@csperkins.org
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt


-- 
Colin Perkins
csp@csperkins.org

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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		 Mon, 30 Jun 2003 14:30:48 +0200
Subject: [AVT] CfP: Frist European Workshop on Wireless Sensor Networks
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>
> Dear Colleague,
>
> I would like to draw your attention to the " Frist European Workshop
> on Wireless Sensor Networks" Berlin, January 2004 (see Attachement).
> The submission deadline for EWSN has been extended to July 14th, 18:00 
> middle European daylight savings time (corresponding to noon Eastern 
> Standard Time or 9 am Pacific time). Details of the Call, see   
> http://www.ewsn.org
>
> Best regards
> Adam Wolisz
>

--------------020902060703010007080804
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<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
  <title></title>
</head>
<body>
<blockquote type="CITE"> 
  <div class="moz-text-html" lang="x-western"> 
  <div class="moz-text-html" lang="x-western"> 
  <div class="moz-text-html" lang="x-western"> 
  <div class="moz-text-html" lang="x-western"> 
  <div class="moz-text-html" lang="x-western"> 
  <div class="moz-text-html" lang="x-western">Dear Colleague, 
  <p>I would like to draw your attention to the " Frist European Workshop 
  <br>
on Wireless Sensor Networks" Berlin, January 2004 (see Attachement). <br>
The submission deadline for EWSN has been <b>extended to July 14th, 18:00
middle European daylight savings time </b>(corresponding to noon Eastern
Standard Time or 9 am Pacific time). Details of the Call, see &nbsp; <a
 class="moz-txt-link-freetext" href="http://www.ewsn.org">http://www.ewsn.org</a></p>
  </div>
  </div>
  </div>
  </div>
  </div>
  </div>
</blockquote>
<blockquote type="CITE">
  <div class="moz-text-html" lang="x-western">
  <div class="moz-text-html" lang="x-western">
  <div class="moz-text-html" lang="x-western">
  <div class="moz-text-html" lang="x-western">
  <div class="moz-text-html" lang="x-western">
  <div class="moz-text-html" lang="x-western">
  <p> </p>
  <p>Best regards <br>
Adam Wolisz</p>
  </div>
 </div>
 </div>
 </div>
 </div>
 </div>
 </blockquote>
 
</body>
</html>

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--------------060701000109040508000805--



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Tue Jul  1 06:47:31 2003
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Date: Mon, 30 Jun 2003 11:00:58 -0600
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Thread-Topic: Updated RFC 2833 Draft?
Thread-Index: AcM/KSwD/Ca5sQsoSueJ4mCwDhkRng==
From: "Konopelski, Peter J" <peter.j.konopelski@intel.com>
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Subject: [AVT] Updated RFC 2833 Draft?
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A colleague of mine was informed that the updated version of RFC 2833 =
draft would be available last week.

Can you tell me the status of it?

Thank You,
Pete=20

Peter J. Konopelski
Customer Engineering
Intel Corporation
peter.j.konopelski@intel.com
303-665-2286 x152

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Tue Jul  1 07:22:29 2003
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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 Audio/Video Transport Working Group of the IETF.

	Title		: RTP Payload Format for JPEG 2000 Video Streams
	Author(s)	: E. Edwards et al.
	Filename	: draft-ietf-avt-rtp-jpeg2000-03.txt
	Pages		: 20
	Date		: 2003-6-30
	
This document describes a payload format for transporting JPEG
2000 video streams using RTP (Real-time Transport Protocol). JPEG
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still images. This payload format will allow for scalability and
robustness of JPEG 2000's potential to be maximized in streaming
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Subject: Re: [AVT] Updated RFC 2833 Draft?
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It was submitted just before the deadline on Sunday. It should appear 
shortly.

Konopelski, Peter J wrote:
> A colleague of mine was informed that the updated version of RFC 2833 draft would be available last week.
> 
> Can you tell me the status of it?
> 
> Thank You,
> Pete 
> 
> Peter J. Konopelski
> Customer Engineering
> Intel Corporation
> peter.j.konopelski@intel.com
> 303-665-2286 x152
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt


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From exim@www1.ietf.org  Tue Jul  1 11:55:33 2003
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From: Muni Chandra <mcr_321@yahoo.com>
Subject: RE: [AVT] RFC 2190 : Clarification
To: Gary Sullivan <garysull@windows.microsoft.com>, avt@ietf.org
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Hi
 
 Thanks a lot for the reponse.
 
 One more clarification.
 
1) At  MODE-A packet without GOB header at start of it, it not possible to decode, becasue no way of finding out where this GOB belongs in the encoded frame in the case of packet loss case. This is since i feel MODE-A H.263 payload header will not give this information.
 
2) so based on above point, in MODE-A packet first GOB has header then solves the problem. Rest GOBs headers in MODE-A  packet are redundant. It is waste of network bandwidth.
 
Please clarify this.
 
Best Regards
Muni Chandra Reddy

Gary Sullivan <garysull@windows.microsoft.com> wrote:
A few comments in response:
 
- RFC 2190 should only be used for H.263 that pre-date 1998 in syntax design (i.e., no use of PLUSPTYPE header format).
 
- RFC 2429 is a better design than 2190, and 2429 can be used for both old and newer versions of H.263 syntax.
 

Regarding the questions:
 
1) I don't think RFC 2190 any prohibition against having additional GOB headers within Mode A packets.  The RFC says it "allows fragmentation at GOB boundaries", but I don't see anywhere where it says it *requires* such fragmentation or prohibits GOB headers within packets.
 
2) As far as the H.263 standard (and within RFC 2190, I believe) is concerned, it is up to the discretion of the encoder when to send GOB headers and when not to.  Even within a single coded picture, some GOBs can have headers while others do not.  In some particular system environments (e.g. in some H.32x environments), there are ways to indicate that GOB headers should be sent or do not need to be sent for all GOBs - but this is outside the scope of H.263 itself.  There is nothing in the H.263 video spec itself that requires any particular behavior for whether GOB headers are present or not.
 
3) Given the above statements about item 1, I don't think item 3 is especially relevant anymore.
 
Best Regards,
 
Gary Sullivan
 
 -----Original Message-----
From: avt-admin@ietf.org [mailto:avt-admin@ietf.org] On Behalf Of Muni Chandra
Sent: Saturday, June 28, 2003 5:07 AM
To: avt@ietf.org
Subject: [AVT] RFC 2190 : Clarification


Hello 
 
I am working on the implementation part of RFC 2190
for H.263 video packtization.
 
I have a clarification.
 
1) As per RFC 2190, In MODE A,   beginning of Packet should contain GOB with its Header. So, this means only first GOB (or part of it) in  each packet should contain GOB header, rest GOBs will not have GOB header in the same MODE-A packet.
 
2) But H.263 standard says ( this with respective to my discussion with H.263 developers) whether all GOB's should have headers and all should not have GOB headers in a paerticular frame.
 
3) So to acheive above point 1 with respective above point 2, we need to  calculate header for all GOB's and transmit GOB header only for first GOB of each MODE - A packet.
 
Please clarify above point 3 is correct or we can calculate GOB headers for only first GOB of each MODE -A packet. So that we can save time of GOB header computation at  rest of GOBs.
 
Thanking you in Advance
 
Muni Chanra Reddy




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<DIV>Hi</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;Thanks a lot for the reponse.</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;One more clarification.</DIV>
<DIV>&nbsp;</DIV>
<DIV>1)&nbsp;At&nbsp;&nbsp;MODE-A packet without GOB header at start of it, it not possible to decode, becasue no way of finding out where this GOB belongs in the encoded frame in the case of packet loss case. This is since i feel&nbsp;MODE-A H.263 payload header will not give&nbsp;this information.</DIV>
<DIV>&nbsp;</DIV>
<DIV>2) so based on above point, in MODE-A packet first GOB has header then solves the problem.&nbsp;Rest GOBs headers in MODE-A&nbsp; packet are&nbsp;redundant. It is waste of network bandwidth.</DIV>
<DIV>&nbsp;</DIV>
<DIV>Please clarify this.</DIV>
<DIV>&nbsp;</DIV>
<DIV>Best Regards</DIV>
<DIV>Muni Chandra Reddy<BR><BR><B><I>Gary Sullivan &lt;garysull@windows.microsoft.com&gt;</I></B> wrote:</DIV>
<DIV>
<BLOCKQUOTE style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid; WIDTH: 100%">
<META content="MSHTML 6.00.2800.1170" name=GENERATOR>
<DIV><SPAN class=745265317-30062003><FONT face="Courier New" color=#0000ff>A few comments in response:</FONT></SPAN></DIV>
<DIV><SPAN class=745265317-30062003><FONT face="Courier New" color=#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=745265317-30062003><FONT face="Courier New" color=#0000ff>- RFC 2190 should only be used for H.263 that pre-date 1998 in syntax design (i.e., no use of PLUSPTYPE header format).</FONT></SPAN></DIV>
<DIV><SPAN class=745265317-30062003><FONT face="Courier New" color=#0000ff>
<DIV><SPAN class=745265317-30062003><FONT face="Courier New" color=#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=745265317-30062003><FONT face="Courier New" color=#0000ff>- RFC 2429 is a better design than 2190, and 2429 can be used for both old and newer versions of H.263 syntax.</FONT></SPAN></DIV>
<DIV><SPAN class=745265317-30062003></SPAN>&nbsp;</DIV></FONT></SPAN></DIV>
<DIV><SPAN class=745265317-30062003><FONT face="Courier New" color=#0000ff>Regarding the questions:</FONT></SPAN></DIV>
<DIV><SPAN class=745265317-30062003><FONT face="Courier New" color=#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=745265317-30062003><FONT face="Courier New" color=#0000ff>1) I don't&nbsp;think&nbsp;RFC 2190&nbsp;any prohibition&nbsp;against having additional GOB headers within Mode A packets.&nbsp; The RFC says it "allows fragmentation at GOB boundaries", but I don't see anywhere where it says it *requires* such fragmentation or prohibits GOB headers within packets.</FONT></SPAN></DIV>
<DIV><SPAN class=745265317-30062003><FONT face="Courier New" color=#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=745265317-30062003><FONT face="Courier New" color=#0000ff>2) As far as the H.263 standard (and within RFC 2190, I believe) is concerned, it is up to the discretion of the encoder when to send GOB headers and when not to.&nbsp;&nbsp;Even within&nbsp;a single coded picture, some GOBs can have headers while others do not.&nbsp;&nbsp;In some particular system environments (e.g. in some H.32x environments), there are ways to indicate that GOB headers should be sent or do not need&nbsp;to be sent for all GOBs - but this is outside the scope of H.263 itself.&nbsp; There is nothing in the H.263 video spec itself that requires any particular behavior for whether GOB headers are present or not.</FONT></SPAN></DIV>
<DIV><SPAN class=745265317-30062003><FONT face="Courier New" color=#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=745265317-30062003><FONT face="Courier New" color=#0000ff>3) Given the above statements about item 1, I don't think item 3 is especially relevant anymore.</FONT></SPAN></DIV>
<DIV><SPAN class=745265317-30062003><FONT face="Courier New" color=#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=745265317-30062003><FONT face="Courier New" color=#0000ff>Best Regards,</FONT></SPAN></DIV>
<DIV><SPAN class=745265317-30062003><FONT face="Courier New" color=#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=745265317-30062003><FONT face="Courier New" color=#0000ff>Gary Sullivan</FONT></SPAN></DIV>
<DIV><SPAN class=745265317-30062003></SPAN><FONT face=Tahoma size=2><SPAN class=745265317-30062003><FONT face="Courier New" color=#0000ff size=3></FONT></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=Tahoma size=2><SPAN class=745265317-30062003>&nbsp;</SPAN>-----Original Message-----<BR><B>From:</B> avt-admin@ietf.org [mailto:avt-admin@ietf.org] <B>On Behalf Of </B>Muni Chandra<BR><B>Sent:</B> Saturday, June 28, 2003 5:07 AM<BR><B>To:</B> avt@ietf.org<BR><B>Subject:</B> [AVT] RFC 2190 : Clarification<BR><BR></DIV></FONT>
<BLOCKQUOTE style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px">
<DIV>
<DIV><FONT face=Arial size=2>Hello </FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=Arial size=2>I am working on the implementation part of RFC 2190<BR>for H.263 video packtization.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=Arial size=2>I have a clarification.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=Arial size=2>1) As per RFC 2190, In MODE A,&nbsp;&nbsp; beginning of Packet should contain GOB with its Header. So, this means only first GOB (or part of it) in&nbsp; each packet should contain GOB header, rest GOBs will not have GOB header in the same MODE-A packet.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=Arial size=2>2) But H.263 standard says ( this with respective to my discussion with H.263 developers) whether all GOB's should have headers and all should not have GOB headers in a paerticular frame.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=Arial size=2>3) So to acheive above point 1 with respective above point 2, we need to&nbsp; calculate header for all GOB's and transmit GOB header only for first GOB of each MODE - A packet.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=Arial size=2>Please clarify above point 3 is correct or we can calculate GOB headers for only first GOB of each MODE -A packet. So that we can save time of GOB header computation&nbsp;at &nbsp;rest of GOBs.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=Arial size=2>Thanking you in Advance</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=Arial size=2>Muni Chanra Reddy<BR></FONT></DIV></DIV>
<P>
<HR SIZE=1>
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From: Stephen Casner <casner@acm.org>
To: Oren Peleg <OrenP@audiocodes.com>
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Subject: [AVT] Re: Q about draft-ietf-avt-rtp-new
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On Thu, 26 Jun 2003, Oren Peleg wrote:

> Is this draft a potential RFC? or its like the avt-profile &
> avt-profile-new which one is the update of the other that is subject
> to future updates as well & therefore will never become an RFC.

Both draft-ietf-avt-rtp-new and draft-ietf-avt-profile-new are in the
final stage of editing with the RFC Editor (actually the "authors's 48
hours", i.e., the ball is in my court) and will be published as RFC's
soon.

I'm not sure where you got the idea that profile-new would never
become an RFC, other than the fact that the process has taken quite a
long time.

                                                        -- Steve


_______________________________________________
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From: =?ISO-8859-1?Q?Mats_N=E4slund?= <mats.naslund@era.ericsson.se>
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To: David Mcgrew <mcgrew@cisco.com>
CC: Colin Perkins <csp@csperkins.org>, Allison Mankin <mankin@psg.com>,
        mbaugher@cisco.com, "Rolf Blom (EAB)" <rolf.blom@era.ericsson.se>,
        "Elisabetta
 Carrara (EAB)" <Elisabetta.Carrara@era.ericsson.se>,
        "Karl Norrman (EAB)"
 <Karl.Norrman@era.ericsson.se>, oran@cisco.com,
        avt@ietf.org
Subject: Re: Fwd: Re: [AVT] IESG Review of draft-ietf-srtp-08.txt - another
 set of comments
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Hi all,

David Mcgrew wrote:
> Mats, Colin, and others,
> 
> I belatedly realized that there is another advantage to using the RTP 
> padding over other schemes: compactness.  If someone implements CBC without 
> using the RTP padding, then the plaintext needs to be turned into a 
> prefix-free code so that the receiver can unambiguously remove any padding 
> that may be present.  Because of this, AES CBC ciphertext will expand by an 
> extra 16-octet block on messages for which no padding is needed, if the 
> conventional method is used (see Appendix A of 
> http://csrc.nist.gov/publications/nistpubs/800-38a/sp800-38a.pdf for more 
> detail).  OTOH, if the RTP padding is used, then the padding bit indicates 
> the presence of padding, and there is never need to further expand the 
> ciphertext.  Also, the NIST document cited above makes it clear that any 
> suitable padding method is considered acceptable.   So this makes me think 
> that we should recommend RTP padding for use with modes that require padding.

Will someone concerned with compactness use a mode that requires
padding in the first place? Won't they rely on the pre-defined modes?

I honestly don't see how we can recommend a padding that we *know*
is "flawed", in particular when the flaw occurs in the absence
of integrity---something we had to fight allow at all!

I am fine with keeping the RTP padding as a sort of "default",
be honest about its pros and cons, but I would not go as far as
to *recommend* it.

> 
> As a side note, I would still prefer a CBC variant that doesn't use padding 

Agreed, but that is for a future extension of SRTP.

> as I'd mentioned before, but at any rate the compactness benefit of using 
> RTP padding deserves to be mentioned.

Absolutely, but so does the security problems.

/M


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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To: Mats =?iso-8859-1?Q?N=E4slund?= <mats.naslund@era.ericsson.se>,
        Colin Perkins <csp@csperkins.org>
From: David Mcgrew <mcgrew@cisco.com>
Subject: Fwd: Re: [AVT] IESG Review of draft-ietf-srtp-08.txt - another
  set of comments
Cc: Allison Mankin <mankin@psg.com>, mbaugher@cisco.com,
        "Rolf Blom (EAB)" <rolf.blom@era.ericsson.se>,
        "Elisabetta Carrara (EAB)" <Elisabetta.Carrara@era.ericsson.se>,
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Mats, Colin, and others,

I belatedly realized that there is another advantage to using the RTP=20
padding over other schemes: compactness.  If someone implements CBC without=
=20
using the RTP padding, then the plaintext needs to be turned into a=20
prefix-free code so that the receiver can unambiguously remove any padding=
=20
that may be present.  Because of this, AES CBC ciphertext will expand by an=
=20
extra 16-octet block on messages for which no padding is needed, if the=20
conventional method is used (see Appendix A of=20
http://csrc.nist.gov/publications/nistpubs/800-38a/sp800-38a.pdf for more=20
detail).  OTOH, if the RTP padding is used, then the padding bit indicates=
=20
the presence of padding, and there is never need to further expand the=20
ciphertext.  Also, the NIST document cited above makes it clear that any=20
suitable padding method is considered acceptable.   So this makes me think=
=20
that we should recommend RTP padding for use with modes that require=
 padding.

As a side note, I would still prefer a CBC variant that doesn't use padding=
=20
as I'd mentioned before, but at any rate the compactness benefit of using=20
RTP padding deserves to be mentioned.

David


>Date: Tue, 24 Jun 2003 09:05:00 -0700
>To: Mats N=E4slund <mats.naslund@era.ericsson.se>, Colin Perkins=20
><csp@csperkins.org>
>From: David Mcgrew <mcgrew@cisco.com>
>Subject: Re: [AVT] IESG Review of draft-ietf-srtp-08.txt - another set of=
=20
>comments
>Cc: Allison Mankin <mankin@psg.com>, mbaugher@cisco.com, "Rolf Blom (EAB)"=
=20
><rolf.blom@era.ericsson.se>, "Elisabetta Carrara (EAB)"=20
><Elisabetta.Carrara@era.ericsson.se>, "Karl Norrman (EAB)"=20
><Karl.Norrman@era.ericsson.se>, oran@cisco.com, avt@ietf.org
>
>Mats, Colin,
>
>The RTP padding is simple, and I can definitely see the value of using a=20
>standard mechanism.  However, it is not clear to me that we actually need=
=20
>a padding method.  AFAICT CBC is the only interesting mode that needs=20
>padding.  The other 'NIST standard' encryption modes (OFB, CFB,  ECB, and=
=20
>counter) and the new and interesting OCB mode don't.  Also, there is a CBC=
=20
>encryption variant that avoids padding by switching to CFB mode to encrypt=
=20
>the last block; it's been proven secure, though unfortunately NIST hasn't=
=20
>included it in a standard, at least not explicitly.   If we used this CBC=
=20
>variant - it would be the variant that would make the most sense for voice=
=20
>traffic - then AFAICT we don't need a padding method at all.
>
>I think that a brief discussion of the security details would be=20
>useful.  If we allow chosen-plaintext attacks (e.g., the attacker can=20
>manipulate the plaintexts directly or indirectly, causing the encryption=20
>of data of her choice), then the padding method can affect=20
>confidentiality.  Otherwise, I believe that it only affects message=20
>authentication.  If message authentication is used (before decryption, as=
=20
>required in SRTP), it defeats Vaudenay's attacks against CBC padding [1],=
=20
>and likely defeats all similar attacks.  The threat is that when message=20
>authentication isn't used, there are attacks that can violate=20
>confidentiality.   Using the RTP padding method with CBC would provide the=
=20
>same level of security as does the ESP padding: it's perfectly good unless=
=20
>there's no authentication, in which case there are some=20
>confidentiality-violating attacks.  At present, the two ciphers defined=20
>for SRTP don't require padding, and aren't vulnerable to any=20
>confidentiality-violating attacks even when message authentication is not=
=20
>used, as you (Mats) point out.
>
>One of the complicating factors is algorithms that provide message=20
>authentication.  Block cipher modes of operation that provide that=20
>service  (CBC MAC, XCBC MAC, OCB [2]) use their own padding schemes.   In=
=20
>each of these modes, the padding method that is used is vital to security,=
=20
>or at least is used in the security proof.   XCBC is widely regarded as=20
>the 'right' way to do CBC MAC, and is expected to be adopted by the=20
>industry, and it is closely tied to its particular padding scheme.  So if=
=20
>we mandate the RTP padding method for SRTP, we need to make clear that it=
=20
>is just for encryption modes (that need padding), and not for=20
>authentication modes.  Perhaps it would make more sense to state that=20
>encryption modes that need a padding scheme SHOULD use the RTP padding=20
>method, unless they need to use another method for security=20
>reasons.  OTOH, I personally would rather see the pad-less variant of CBC,=
=20
>though crypto hardware vendors with existing products might choose=
 otherwise.
>
>On a side note, the RTP padding could be used in conjunction with=20
>encryption as a length-hiding mechanism.  The sender can hide the length=20
>of a packet by adding padding, which the receiver will discard.  The=20
>encryption will hide the last padding byte, so that an adversary cannot=20
>tell by looking at a ciphertext packet what the length of the payload of=20
>the corresponding plaintext packet is.  I don't think that we stated this=
=20
>anywhere in the draft, though it seems like a legal use of RTP and SRTP to=
 me.
>
>David
>
>[1] Security Flaws Induced by CBC Padding - Applications to SSL, IPsec,=20
>and WTLS, Serge Vaudenay, online at=20
>http://lasecwww.epfl.ch/php_code/publications/search.php?ref=3DVau02a
>
>[2] NIST Table of Submitted Modes of Operation, online at=20
>http://csrc.nist.gov/CryptoToolkit/modes/proposedmodes/
>
>At 11:00 AM 6/24/2003 +0200, Mats N=E4slund wrote:
>
>>Hi,
>>
>>Colin Perkins wrote:
>>>[inline]
>>>--> Allison Mankin <mankin@psg.com> writes:
>>>
>>>>SRTP is on the agenda of the IESG again this week.
>>>>Some more comments on SRTP may still come in, but I thought
>>>>it would be worth sending on those of Russ Housley, the second Security
>>>>AD.  Steve Bellovin has supported the draft.  Russ has comments that=20
>>>>seem straightforward to address.  Wait for more comments before sending=
=20
>>>>in a revision, but so far things are going well.
>>>>
>>>>Allison
>>>>
>>>>
>>>>>>                    Yes    No-Objection  Discuss *  Abstain
>>>>>>
>>>>>>Russ Housley        [   ]     [   ]       [ X ]      [   ]
>>>>>
>>>>>I have six comments.
>>>>>
>>>>>1.  In section 1, spell out the first use of RTCP.
>>>>>
>>>>>2.  I find the structure of section 2 confusing.  I had to read it
>>>>>twice to understand it.  I think that a second level of indenting would
>>>>>be one way to fix it.  I am sure there are others.
>>>>>
>>>>>3.  In section 3.1, in the paragraph after figure 1, please delete:
>>>>>
>>>>>    "It is exact for the pre-defined transforms."
>>>>>
>>>>>This point is made more clearly later in the paragraph.  Then, at the
>>>>>end of the same paragraph, the document says:
>>>>>
>>>>>    "While it could seem more attractive to specify a fixed padding
>>>>>    scheme for all transforms, security and flexibility of transform
>>>>>    specifications REQUIRE that each transform specify a secure
>>>>>    padding method."
>>>>>
>>>>>I disagree.  IPsec and S/MIME both specify padding schemes that are=20
>>>>>employed by all of the ciphers.  Please reword.  Do not use "REQUIRE"
>>>>>in the replacement.
>>>If IPsec and S/MIME both define a standard padding mechanism, why cannot
>>>SRTP do the same?  And, perhaps, use the standard RTP padding mechanism?
>>
>>
>>To comply with Russ' comment, I have no objection to deleting
>>"REQUIRE". I would hesitate to go beyond that.
>>
>>We will open up new issues in doing so. Do we use the IPsec pad?
>>The S/MIME pad? Perhaps none of them is general enough? What about
>>new modes? (If I remember correctly, e.g. XCBC had some padding issues.)
>>David referenced a paper showing that some common pad schemes
>>were "weak". (Even if these weaknesses are "theoretical", we have
>>made great effort to make SRTP "maximally" secure, and I would hate
>>to sub-optimize at this point.)
>>
>>It seems to me that the only future proof approach is to let each
>>transform specify its own padding, or at least point to were a
>>padding spec. can be found.
>>
>>/Mats
>>
>>
>>_______________________________________________
>>Audio/Video Transport Working Group
>>avt@ietf.org
>>https://www1.ietf.org/mailman/listinfo/avt

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: RE: [AVT] RFC 2190 : Clarification
Date: Tue, 1 Jul 2003 16:43:17 -0700
Message-ID: <91D7F2CEE3425A4A9D11311D09FCE24602A34701@WIN-MSG-10.wingroup.windeploy.ntdev.microsoft.com>
Thread-Topic: [AVT] RFC 2190 : Clarification
Thread-Index: AcM/6PDaFXVeIiYgRWq4b9aTFozWFQAPfnsQ
From: "Gary Sullivan" <garysull@windows.microsoft.com>
To: "Muni Chandra" <mcr_321@yahoo.com>, <avt@ietf.org>
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Muni,
=20
I believe I basically agree with you on both points.
=20
Regarding item 1, yes - I believe you would not be able to decode the
data at the start of such a GOB-headerless packet unless you had
received the packet or packets before it that provided the necessary
header information.
=20
Regarding item 2, yes - Placing more than one GOB header within a single
RFC 2190 Mode A RTP packet appears to be basically a useless waste of
bits, unless you can argue that some form of data corruption is possible
somewhere in the transmission channel (possibly through a gateway to
some other non-IP environment).
=20
As I previously indicated, I am not too fond of RFC 2190.  RFC 2429 is a
superior design.  However, RFC 2190 was done earlier and is therefore
still needed for interoperability with devices made before RFc 2429 was
developed.
=20
Best Regards,
=20
Gary Sullivan

-----Original Message-----
From: Muni Chandra [mailto:mcr_321@yahoo.com]=20
Sent: Tuesday, July 01, 2003 8:53 AM
To: Gary Sullivan; avt@ietf.org
Subject: RE: [AVT] RFC 2190 : Clarification


Hi
=20
 Thanks a lot for the reponse.
=20
 One more clarification.
=20
1) At  MODE-A packet without GOB header at start of it, it not possible
to decode, becasue no way of finding out where this GOB belongs in the
encoded frame in the case of packet loss case. This is since i feel
MODE-A H.263 payload header will not give this information.
=20
2) so based on above point, in MODE-A packet first GOB has header then
solves the problem. Rest GOBs headers in MODE-A  packet are redundant.
It is waste of network bandwidth.
=20
Please clarify this.
=20
Best Regards
Muni Chandra Reddy

Gary Sullivan <garysull@windows.microsoft.com> wrote:

A few comments in response:
=20
- RFC 2190 should only be used for H.263 that pre-date 1998 in syntax
design (i.e., no use of PLUSPTYPE header format).

=20
- RFC 2429 is a better design than 2190, and 2429 can be used for both
old and newer versions of H.263 syntax.
=20
Regarding the questions:
=20
1) I don't think RFC 2190 any prohibition against having additional GOB
headers within Mode A packets.  The RFC says it "allows fragmentation at
GOB boundaries", but I don't see anywhere where it says it *requires*
such fragmentation or prohibits GOB headers within packets.
=20
2) As far as the H.263 standard (and within RFC 2190, I believe) is
concerned, it is up to the discretion of the encoder when to send GOB
headers and when not to.  Even within a single coded picture, some GOBs
can have headers while others do not.  In some particular system
environments (e.g. in some H.32x environments), there are ways to
indicate that GOB headers should be sent or do not need to be sent for
all GOBs - but this is outside the scope of H.263 itself.  There is
nothing in the H.263 video spec itself that requires any particular
behavior for whether GOB headers are present or not.
=20
3) Given the above statements about item 1, I don't think item 3 is
especially relevant anymore.
=20
Best Regards,
=20
Gary Sullivan
=20
 -----Original Message-----
From: avt-admin@ietf.org [mailto:avt-admin@ietf.org] On Behalf Of Muni
Chandra
Sent: Saturday, June 28, 2003 5:07 AM
To: avt@ietf.org
Subject: [AVT] RFC 2190 : Clarification



Hello=20
=20
I am working on the implementation part of RFC 2190
for H.263 video packtization.
=20
I have a clarification.
=20
1) As per RFC 2190, In MODE A,   beginning of Packet should contain GOB
with its Header. So, this means only first GOB (or part of it) in  each
packet should contain GOB header, rest GOBs will not have GOB header in
the same MODE-A packet.
=20
2) But H.263 standard says ( this with respective to my discussion with
H.263 developers) whether all GOB's should have headers and all should
not have GOB headers in a paerticular frame.
=20
3) So to acheive above point 1 with respective above point 2, we need to
calculate header for all GOB's and transmit GOB header only for first
GOB of each MODE - A packet.
=20
Please clarify above point 3 is correct or we can calculate GOB headers
for only first GOB of each MODE -A packet. So that we can save time of
GOB header computation at  rest of GOBs.
=20
Thanking you in Advance
=20
Muni Chanra Reddy




  _____ =20

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<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dus-ascii">
<TITLE>Message</TITLE>

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<BODY>
<DIV><SPAN class=3D879181723-01072003><FONT face=3D"Courier New"=20
color=3D#0000ff>Muni,</FONT></SPAN></DIV>
<DIV><SPAN class=3D879181723-01072003><FONT face=3D"Courier New"=20
color=3D#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D879181723-01072003><FONT face=3D"Courier New" =
color=3D#0000ff>I=20
believe I basically agree with you on both points.</FONT></SPAN></DIV>
<DIV><SPAN class=3D879181723-01072003><FONT face=3D"Courier New"=20
color=3D#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D879181723-01072003><FONT face=3D"Courier New"=20
color=3D#0000ff>Regarding item 1, yes - I believe you would not be able =
to decode=20
the data at the start of such a GOB-headerless packet unless you had =
received=20
the packet or packets before it that provided the necessary&nbsp;header=20
information.</FONT></SPAN></DIV>
<DIV><SPAN class=3D879181723-01072003><FONT face=3D"Courier New"=20
color=3D#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D879181723-01072003><FONT face=3D"Courier New"=20
color=3D#0000ff>Regarding item 2, yes -&nbsp;Placing more than =
one&nbsp;GOB header=20
within a single&nbsp;RFC 2190&nbsp;Mode A RTP packet&nbsp;appears to=20
be&nbsp;basically a useless waste of bits, unless you can argue that =
some form=20
of data corruption is possible somewhere in the transmission channel =
(possibly=20
through a gateway to some other non-IP environment).</FONT></SPAN></DIV>
<DIV><SPAN class=3D879181723-01072003><FONT face=3D"Courier New"=20
color=3D#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D879181723-01072003><FONT face=3D"Courier New" =
color=3D#0000ff>As I=20
previously indicated, I am not too fond of RFC 2190.&nbsp; RFC 2429 is a =

superior design.&nbsp; However, RFC 2190 was done earlier and is =
therefore still=20
needed for interoperability with devices made before RFc 2429 was=20
developed.</FONT></SPAN></DIV>
<DIV><SPAN class=3D879181723-01072003><FONT face=3D"Courier New"=20
color=3D#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D879181723-01072003><FONT face=3D"Courier New" =
color=3D#0000ff>Best=20
Regards,</FONT></SPAN></DIV>
<DIV><SPAN class=3D879181723-01072003><FONT face=3D"Courier New"=20
color=3D#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D879181723-01072003><FONT face=3D"Courier New" =
color=3D#0000ff>Gary=20
Sullivan</FONT></SPAN></DIV>
<BLOCKQUOTE=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV></DIV>
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft><FONT=20
  face=3DTahoma size=3D2>-----Original Message-----<BR><B>From:</B> Muni =
Chandra=20
  [mailto:mcr_321@yahoo.com] <BR><B>Sent:</B> Tuesday, July 01, 2003 =
8:53=20
  AM<BR><B>To:</B> Gary Sullivan; avt@ietf.org<BR><B>Subject:</B> RE: =
[AVT] RFC=20
  2190 : Clarification<BR><BR></FONT></DIV>
  <DIV>Hi</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>&nbsp;Thanks a lot for the reponse.</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>&nbsp;One more clarification.</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>1)&nbsp;At&nbsp;&nbsp;MODE-A packet without GOB header at start =
of it, it=20
  not possible to decode, becasue no way of finding out where this GOB =
belongs=20
  in the encoded frame in the case of packet loss case. This is since i=20
  feel&nbsp;MODE-A H.263 payload header will not give&nbsp;this=20
  information.</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>2) so based on above point, in MODE-A packet first GOB has header =
then=20
  solves the problem.&nbsp;Rest GOBs headers in MODE-A&nbsp; packet=20
  are&nbsp;redundant. It is waste of network bandwidth.</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>Please clarify this.</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>Best Regards</DIV>
  <DIV>Muni Chandra Reddy<BR><BR><B><I>Gary Sullivan=20
  &lt;garysull@windows.microsoft.com&gt;</I></B> wrote:</DIV>
  <DIV>
  <BLOCKQUOTE=20
  style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px =
solid; WIDTH: 100%">
    <META content=3D"MSHTML 6.00.2800.1170" name=3DGENERATOR>
    <DIV><SPAN class=3D745265317-30062003><FONT face=3D"Courier New" =
color=3D#0000ff>A=20
    few comments in response:</FONT></SPAN></DIV>
    <DIV><SPAN class=3D745265317-30062003><FONT face=3D"Courier New"=20
    color=3D#0000ff></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D745265317-30062003><FONT face=3D"Courier New" =
color=3D#0000ff>-=20
    RFC 2190 should only be used for H.263 that pre-date 1998 in syntax =
design=20
    (i.e., no use of PLUSPTYPE header format).</FONT></SPAN></DIV>
    <DIV><SPAN class=3D745265317-30062003><FONT face=3D"Courier New" =
color=3D#0000ff>
    <DIV><SPAN class=3D745265317-30062003><FONT face=3D"Courier New"=20
    color=3D#0000ff></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D745265317-30062003><FONT face=3D"Courier New" =
color=3D#0000ff>-=20
    RFC 2429 is a better design than 2190, and 2429 can be used for both =
old and=20
    newer versions of H.263 syntax.</FONT></SPAN></DIV>
    <DIV><SPAN =
class=3D745265317-30062003></SPAN>&nbsp;</DIV></FONT></SPAN></DIV>
    <DIV><SPAN class=3D745265317-30062003><FONT face=3D"Courier New"=20
    color=3D#0000ff>Regarding the questions:</FONT></SPAN></DIV>
    <DIV><SPAN class=3D745265317-30062003><FONT face=3D"Courier New"=20
    color=3D#0000ff></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D745265317-30062003><FONT face=3D"Courier New"=20
    color=3D#0000ff>1) I don't&nbsp;think&nbsp;RFC 2190&nbsp;any=20
    prohibition&nbsp;against having additional GOB headers within Mode A =

    packets.&nbsp; The RFC says it "allows fragmentation at GOB =
boundaries", but=20
    I don't see anywhere where it says it *requires* such fragmentation =
or=20
    prohibits GOB headers within packets.</FONT></SPAN></DIV>
    <DIV><SPAN class=3D745265317-30062003><FONT face=3D"Courier New"=20
    color=3D#0000ff></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D745265317-30062003><FONT face=3D"Courier New"=20
    color=3D#0000ff>2) As far as the H.263 standard (and within RFC =
2190, I=20
    believe) is concerned, it is up to the discretion of the encoder =
when to=20
    send GOB headers and when not to.&nbsp;&nbsp;Even within&nbsp;a =
single coded=20
    picture, some GOBs can have headers while others do =
not.&nbsp;&nbsp;In some=20
    particular system environments (e.g. in some H.32x environments), =
there are=20
    ways to indicate that GOB headers should be sent or do not =
need&nbsp;to be=20
    sent for all GOBs - but this is outside the scope of H.263 =
itself.&nbsp;=20
    There is nothing in the H.263 video spec itself that requires any =
particular=20
    behavior for whether GOB headers are present or =
not.</FONT></SPAN></DIV>
    <DIV><SPAN class=3D745265317-30062003><FONT face=3D"Courier New"=20
    color=3D#0000ff></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D745265317-30062003><FONT face=3D"Courier New"=20
    color=3D#0000ff>3) Given the above statements about item 1, I don't =
think item=20
    3 is especially relevant anymore.</FONT></SPAN></DIV>
    <DIV><SPAN class=3D745265317-30062003><FONT face=3D"Courier New"=20
    color=3D#0000ff></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D745265317-30062003><FONT face=3D"Courier New"=20
    color=3D#0000ff>Best Regards,</FONT></SPAN></DIV>
    <DIV><SPAN class=3D745265317-30062003><FONT face=3D"Courier New"=20
    color=3D#0000ff></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D745265317-30062003><FONT face=3D"Courier New"=20
    color=3D#0000ff>Gary Sullivan</FONT></SPAN></DIV>
    <DIV><SPAN class=3D745265317-30062003></SPAN><FONT face=3DTahoma =
size=3D2><SPAN=20
    class=3D745265317-30062003><FONT face=3D"Courier New" =
color=3D#0000ff=20
    size=3D3></FONT></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT face=3DTahoma size=3D2><SPAN=20
    class=3D745265317-30062003>&nbsp;</SPAN>-----Original=20
    Message-----<BR><B>From:</B> avt-admin@ietf.org =
[mailto:avt-admin@ietf.org]=20
    <B>On Behalf Of </B>Muni Chandra<BR><B>Sent:</B> Saturday, June 28, =
2003=20
    5:07 AM<BR><B>To:</B> avt@ietf.org<BR><B>Subject:</B> [AVT] RFC 2190 =
:=20
    Clarification<BR><BR></DIV></FONT>
    <BLOCKQUOTE=20
    style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff =
2px solid; MARGIN-RIGHT: 0px">
      <DIV>
      <DIV><FONT face=3DArial size=3D2>Hello </FONT></DIV>
      <DIV>&nbsp;</DIV>
      <DIV><FONT face=3DArial size=3D2>I am working on the =
implementation part of=20
      RFC 2190<BR>for H.263 video packtization.</FONT></DIV>
      <DIV>&nbsp;</DIV>
      <DIV><FONT face=3DArial size=3D2>I have a =
clarification.</FONT></DIV>
      <DIV>&nbsp;</DIV>
      <DIV><FONT face=3DArial size=3D2>1) As per RFC 2190, In MODE =
A,&nbsp;&nbsp;=20
      beginning of Packet should contain GOB with its Header. So, this =
means=20
      only first GOB (or part of it) in&nbsp; each packet should contain =
GOB=20
      header, rest GOBs will not have GOB header in the same MODE-A=20
      packet.</FONT></DIV>
      <DIV>&nbsp;</DIV>
      <DIV><FONT face=3DArial size=3D2>2) But H.263 standard says ( this =
with=20
      respective to my discussion with H.263 developers) whether all =
GOB's=20
      should have headers and all should not have GOB headers in a =
paerticular=20
      frame.</FONT></DIV>
      <DIV>&nbsp;</DIV>
      <DIV><FONT face=3DArial size=3D2>3) So to acheive above point 1 =
with=20
      respective above point 2, we need to&nbsp; calculate header for =
all GOB's=20
      and transmit GOB header only for first GOB of each MODE - A=20
      packet.</FONT></DIV>
      <DIV>&nbsp;</DIV>
      <DIV><FONT face=3DArial size=3D2>Please clarify above point 3 is =
correct or we=20
      can calculate GOB headers for only first GOB of each MODE -A =
packet. So=20
      that we can save time of GOB header computation&nbsp;at &nbsp;rest =
of=20
      GOBs.</FONT></DIV>
      <DIV>&nbsp;</DIV>
      <DIV><FONT face=3DArial size=3D2>Thanking you in =
Advance</FONT></DIV>
      <DIV>&nbsp;</DIV>
      <DIV><FONT face=3DArial size=3D2>Muni Chanra =
Reddy<BR></FONT></DIV></DIV>
      <P>
      <HR SIZE=3D1>
      Do you Yahoo!?<BR><A=20
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      Yahoo! DSL</A> - Now only $29.95 per =
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  <P>
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Subject: [AVT] Revised internet draft of RTP Payload Format for ATRAC-X
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Hi all,
 
We have submitted our revised internet draft as "draft-hatanaka-avt-rtp-atracx-02.txt".
You will be able to get it from the on-line Internet-Drafts directories.

We do not have a plan to discuss in AVT group at Vienna meeting,
 because this is slight changing version.
 
The revised items are as follows.
(1) Changing a terminology of "StreamID" to "ElementID" in section 2.3.
(2) Changing a unit of LENGTH "byte" to "bit" in section 2.3.
(3) Changing a unit of time stamp "millisecond of a presentation time"
 to "sample number of a presentation time" in section 7.
(4) Addition of the description about multicasting consideration in
 section 8.
 
We will evaluate the defined functions of this payload with the practical
 test application until next meeting, and we would like to reflect the
 results in next document.
 
Best Regards,
Mitsuyuki Hatanaka
 

---------------------------------------
 Mitsuyuki Hatanaka
 Sony Corporation
 E-mail: hatanaka@av.crl.sony.co.jp
---------------------------------------


_______________________________________________
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This draft is a work item of the Audio/Video Transport Working Group of the IETF.

	Title		: RTP Payload Format for Uncompressed Video
	Author(s)	: L. Gharai, C. Perkins
	Filename	: draft-ietf-avt-uncomp-video-03.txt
	Pages		: 16
	Date		: 2003-7-1
	
This memo specifies a packetization scheme for encapsulating
uncompressed video into a payload format for the Real-time
Transport Protocol, RTP. It supports a range of standard- and
high-definition video formats, including common television
formats such as ITU BT.601, and standards from the Society of
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274M and SMPTE 296M. The format is designed to be applicable
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	Title		: RTP Payload Format for MIDI
	Author(s)	: J. Lazzaro, J. Wawrzynek
	Filename	: draft-ietf-avt-mwpp-midi-rtp-08.txt
	Pages		: 101
	Date		: 2003-7-1
	
This memo describes an RTP payload format for the MIDI command
language.  The format encodes all commands that may legally appear
on a MIDI 1.0 DIN cable.  The format is suitable for interactive
applications (such as the remote operation of musical instruments)
and content-delivery applications (such as file streaming).  The
format may be used over unicast and multicast UDP as well as TCP,
and defines tools for graceful recovery from packet loss.  Stream
behavior, including the MIDI rendering method, may be customized
during session setup.  The format also serves as a mode for the
mpeg4-generic format, to support the MPEG 4 Audio Object Types for
General MIDI, Downloadable Sounds Level 2, and Structured Audio.

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Dear all,

An -01 update to draft-herlein-speex-rtp-profile is now available:

http://www.ietf.org/internet-drafts/draft-herlein-speex-rtp-profile-01.txt

Comments and feedback welcomed.

Regards

Phil



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This draft is a work item of the Audio/Video Transport Working Group of the IETF.

	Title		: RTP payload Format for JVT Video
	Author(s)	: S. Wenger et al.
	Filename	: draft-ietf-avt-rtp-h264-02.txt
	Pages		: 55
	Date		: 2003-7-2
	
This memo describes an RTP Payload format for the ITU-T 
Recommendation H.264 video codec.  This codec was designed as a 
joint project of the Video Coding Experts Group (VCEG) of ITU-T and
the Moving Picture Experts Group (MPEG) of ISO/IEC.  Recommendation 
H.264 was approved by ITU-T on May 2003, and the approved draft 
specification is available for public review.  ISO/IEC 
International Standard 14496-10 will be technically identical to 
ITU-T Recommendation H.264.

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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 Audio/Video Transport Working Group of the IETF.

	Title		: RTP Payload for DTMF Digits, Telephony Tones and 
                          Telephony Signals
	Author(s)	: H. Schulzrinne, S. Petrack
	Filename	: draft-ietf-avt-rfc2833bis-03.txt
	Pages		: 38
	Date		: 2003-7-2
	
This memo describes how to carry dual-tone multifrequency (DTMF)
signaling, other tone signals and telephony events in RTP packets.
This document updates RFC 2833.

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From exim@www1.ietf.org  Thu Jul  3 05:22:32 2003
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	(envelope-from csp@csperkins.org)
Date: Thu, 3 Jul 2003 11:21:26 +0200
From: Colin Perkins <csp@csperkins.org>
To: philkerr@elec.gla.ac.uk
Cc: avt@ietf.org
Subject: Re: [AVT] draft-herlein-speex-rtp-profile-01
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--> philkerr@elec.gla.ac.uk writes:
> Dear all,
> 
> An -01 update to draft-herlein-speex-rtp-profile is now available:
> 
> http://www.ietf.org/internet-drafts/draft-herlein-speex-rtp-profile-01.txt
> 
> Comments and feedback welcomed.

Can you summarize the changes from the previous version, and areas where
input is needed?

-- 
Colin Perkins
csp@csperkins.org

_______________________________________________
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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 Audio/Video Transport Working Group of the IETF.

	Title		: Internet Low Bit Rate Codec
	Author(s)	: S. Andersen et al.
	Filename	: draft-ietf-avt-ilbc-codec-02.txt
	Pages		: 177
	Date		: 2003-7-3
	
This document specifies a speech codec suitable for robust voice 
communication over IP. The codec is developed by Global IP Sound 
(GIPS). It is designed for narrow band speech and results in a payload 
bit rate of 13.33 kbit/s for 30 ms frames and 15.20 kbit/s for 20 ms 
frames. The codec enables graceful speech quality degradation in the 
case of lost frames, which occurs in connection with lost or delayed IP 
packets.

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_______________________________________________
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https://www1.ietf.org/mailman/listinfo/avt



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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 Audio/Video Transport Working Group of the IETF.

	Title		: RTCP Extensions for Single-Source Multicast Sessions 
                          with Unicase Feedback
	Author(s)	: J. Chesterfield et al.
	Filename	: draft-ietf-avt-rtcpssm-04.txt
	Pages		: 37
	Date		: 2003-7-3
	
This document specifies a modification to the Real-time Transport 
Control Protocol (RTCP) to use unicast feedback. The proposed 
extension is useful for single source multicast sessions such as 
Source Specific Multicast (SSM) communication where the traditional 
model of many-to-many group communication is either not possible or 
not preferred. In addition, it can be applied to any group that 
might benefit from a sender controlled summarised reporting 
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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 Audio/Video Transport Working Group of the IETF.

	Title		: RTP Payload Format for iLBC Speech
	Author(s)	: A. Duric, S. Andersen
	Filename	: draft-ietf-avt-rtp-ilbc-02.txt
	Pages		: 10
	Date		: 2003-7-3
	
This document describes the RTP payload format for the internet Low 
Bit Rate Coder (iLBC) Speech [1] developed by Global IP Sound 
(GIPS). Also, within the document there are included necessary 
details for the use of iLBC with MIME and SDP.

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_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Thu Jul  3 12:41:35 2003
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What is the expected behavior if only a b=RR line is specified and it's
value is greater than 5% of the stream bandwidth. The draft says that if
only one parameter is specified then the value of the other is determined
by subracting the value of the specified parameter from 5% of the stream
bitrate. In the case I have mentioned this would cause a negative
bandwidth to be calculated. That is obviously not valid. Could someone
please clarify what should happen in this case.

Thanks,
Aaron

_______________________________________________
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avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Thu, 03 Jul 2003 11:46:44 -0700
To: Allison Mankin <mankin@psg.com>
From: Mark Baugher <mbaugher@cisco.com>
Cc: mats.naslund@era.ericsson.se,
        "Rolf Blom (EAB)" <rolf.blom@era.ericsson.se>,
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Subject: [AVT] Re: IESG Review of draft-ietf-srtp-08.txt - another set of
 comments
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Allison,
    We have addressed five out of the six comments from Russ at 
http://www.rdrop.com/users/mbaugher/I-D/draft-ietf-avt-srtp-09.txt.  The 
document is available with highlighted changes at 
http://www.rdrop.com/users/mbaugher/I-D/draft-ietf-avt-srtp-09.pdf.  There 
have been some small editorial changes made as well and we paired down the 
authors list from seven authors to five to be in accordance with IETF 
policy.  Rolf Blom and Dave Oran asked that their names be removed to 
comply with the policy.  They played an important role in starting the SRTP 
effort and in guiding it and we thank them for that.

   I said that we addressed five out of six comments.  We could not agree 
on what if anything needed to be changed in Section 2 (Russ's item #2) and 
decided to leave it as it is.

   We'll submit this version to internet-drafts as well.

Mark
At 02:13 PM 6/23/2003 -0700, Allison Mankin wrote:
>SRTP is on the agenda of the IESG again this week.
>Some more comments on SRTP may still come in, but I thought
>it would be worth sending on those of Russ Housley, the second Security
>AD.  Steve Bellovin has supported the draft.  Russ has comments that
>seem straightforward to address.  Wait for more comments before
>sending in a revision, but so far things are going well.
>
>Allison
>
> >
> > >                     Yes    No-Objection  Discuss *  Abstain
> > >
> > >Russ Housley        [   ]     [   ]       [ X ]      [   ]
> >
> > I have six comments.
> >
> > 1.  In section 1, spell out the first use of RTCP.
> >
> > 2.  I find the structure of section 2 confusing.  I had to read it 
> twice to
> > understand it.  I think that a second level of indenting would be one way
> > to fix it.  I am sure there are others.
> >
> > 3.  In section 3.1, in the paragraph after figure 1, please delete:
> >
> >     "It is exact for the pre-defined transforms."
> >
> > This point is made more clearly later in the paragraph.  Then, at the end
> > of the same paragraph, the document says:
> >
> >     "While it could seem more attractive to specify a fixed padding
> >     scheme for all transforms, security and flexibility of transform
> >     specifications REQUIRE that each transform specify a secure
> >     padding method."
> >
> > I disagree.  IPsec and S/MIME both specify padding schemes that are
> > employed by all of the ciphers.  Please reword.  Do not use "REQUIRE" in
> > the replacement.
> >
> > 4.  In section 3.2.1, the document says: "the master key(s), which MUST be
> > random and kept secret."  This is quite a high bar.  I suggest that
> > pseudo-random is sufficient.  Please see the RFC 2828 definitions of
> > "random" and "pseudo-random."  The document says that it is following 
> these
> > terms.  Also, see RFC 1750.  Similarly, the requirement on the master salt
> > should also be reduced to pseudo-random.
> >
> > 5.  The last paragraph is section 3.2.3 says:
> >
> >    "If no valid context can be found for a packet corresponding to a
> >     certain context identifier, that packet MUST be discarded from
> >     further SRTP processing."
> >
> > Elsewhere, "discarded from further processing" is used.  This seems better
> > to me.  It is discarded, which seems different that no further SRTP
> > processing.  Also, SRTCP should be covered by this statement.
> >
> > 6.  In section 4.1.1, I am confused by the term "fixed key" in the 
> following:
> >
> >     "For a fixed Counter Mode key, each IV value used as an input MUST be
> >     distinct, in order to avoid the security exposure of a two-time pad
> >     situation (Section 9.1).  To satisfy this constraint, an
> >     implementation MUST ensure that the values of the SRTP packet index
> >     of ROC || SEQ, and the SSRC used in the construction of the IV are
> >     distinct for any fixed key.  The failure to ensure this uniqueness
> >     could  be catastrophic for Secure RTP.  This is in contrast to the
> >     situation for RTP itself, which may be able to tolerate such
> >     failures.  It is RECOMMENDED that, if a dedicated security module is
> >     present, the RTP sequence numbers and SSRC either be generated or
> >     checked by that module (i.e., sequence-number and SSRC processing in
> >     an SRTP system needs to be protected as well as the key). "
> >
> > I think that "fixed" means "any particular key" but I am not sure.
> >
> >
> >
> >



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: Stephen Casner <casner@acm.org>
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Subject: Re: [AVT] draft-ietf-avt-rtcp-bw-05.txt question
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Aaron,

> What is the expected behavior if only a b=RR line is specified and it's
> value is greater than 5% of the stream bandwidth. The draft says that if
> only one parameter is specified then the value of the other is determined
> by subracting the value of the specified parameter from 5% of the stream
> bitrate. In the case I have mentioned this would cause a negative
> bandwidth to be calculated. That is obviously not valid. Could someone
> please clarify what should happen in this case.

I propose to simply say that the unspecified parameter is zero in that
case.
                                                        -- Steve


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: Aaron Colwell <acolwell@real.com>
To: Stephen Casner <casner@acm.org>
cc: avt@ietf.org
Subject: Re: [AVT] draft-ietf-avt-rtcp-bw-05.txt question
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Steve,

This proposal would work fine if only b=RS is present. This would just
mean that the client would not send receiver reports.

In the case where b=RR is specified that would mean that no sender reports
or BYE packets could be sent. Is that allowed? In the A/V case the client
would not be able to sync the streams because it would never get their
CNAME. I suppose that is the server's fault for not specifying a value
for the sender reports. I guess this brings up another question. Does it
make sense for the RS value to be 0?

Perhaps we could modify your proposal to say, "If only b=RS is specified
and it's value is greater than 5% of the stream bitrate, then RR value is
0. If you need a b=RR line that is greater than 5% of the stream bitrate,
then you MUST supply a b=RS line." We could say that or we could just
make the rule that if you are allocating more than 5% of the stream
bitrate to RTCP, you MUST specify both the b=RR and b=RS lines.

Aaron



On Thu, 3 Jul 2003, Stephen Casner wrote:

> Aaron,
>
> > What is the expected behavior if only a b=RR line is specified and it's
> > value is greater than 5% of the stream bandwidth. The draft says that if
> > only one parameter is specified then the value of the other is determined
> > by subracting the value of the specified parameter from 5% of the stream
> > bitrate. In the case I have mentioned this would cause a negative
> > bandwidth to be calculated. That is obviously not valid. Could someone
> > please clarify what should happen in this case.
>
> I propose to simply say that the unspecified parameter is zero in that
> case.
>                                                         -- Steve
>
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
>

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: Colin Perkins <csp@csperkins.org>
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Cc: casner@acm.org, magnus.westerlund@ericsson.com
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Folks,

Enclosed is the agenda for the AVT meeting in Vienna. Please send any
comments or corrections to the chairs. 

Those who have agenda time scheduled are requested to prepare a summary
of the issues they wish to discuss, and to send this to the mailing list
ahead of the meeting. This gives those who won't be attending the
opportunity to post their comments and concerns, and potentially even
reach some consensus, before the meeting time is consumed. It also gives
other participants the opportunity to consider the issue in more depth.
Summary messages SHOULD be sent to the mailing list by Monday 7th July.

We also request that all presenters send us their slides ahead of time,
preferably in PDF format. This has multiple purposes:

  - It streamlines the presentation process by having the presentations 
    on a single laptop.

  - It avoids the delay that sometimes occurs in getting the presentations
    sent to us after the meeting, for inclusion in the minutes.

  - It allows us to preview the presentations to see if they are following
    the guidelines for what should be covered during the meeting.

Presentations SHOULD be submitted to the chairs by Friday 11th July.

Agenda time is conditional on sending both a summary of open issues and
a copy of your slides ahead of the meeting, and 5 minutes ahead doesn't
count!

Those presenting new drafts: briefly describe the basic problem and your
proposed solution, discuss known issues, limitations and problems,
outline where you want input from the working group, and how you wish us
to proceed.  When presenting work which has been previously discussed:
describe only the changes from the previous draft, highlight unresolved
problems, and propose future directions. Presenters are reminded of the
requirements for disclosure of intellectual property relating to
contributions, in section 10 of RFC 2026.

We strongly encourage all participants to read all relevent drafts before
the meeting, and to comment on the open issues raised on the mailing list.

Thanks for your assistance to try to conduct an effective meeting.

Colin, Steve and Magnus


===============================================================================

Audio/Video Transport Working Group

CHAIRS: Steve Casner <casner@acm.org>
        Colin Perkins <csp@csperkins.org>
	Magnus Westerlund <magnus.westerlund@ericsson.com>


Monday, 14th July 2003 at 09:00-11:30
=====================================

Introduction, Status and Charter Update			(Chairs)	20
	draft-ietf-avt-rtp-retransmission-08.txt	[Last call]
	draft-ietf-avt-rtcp-feedback-07.txt		[Last call]
	draft-burmeister-avt-rtcp-feedback-sim-02.txt	[Last call]
	draft-ietf-avt-crtp-enhance-07.txt		[RFC Editor]
	draft-ietf-avt-rtp-new-12.txt			[RFC Editor]
	draft-ietf-avt-profile-new-13.txt		[RFC Editor]
	draft-ietf-avt-rtp-mime-06.txt			[RFC Editor]
	draft-ietf-avt-rtcp-bw-05.txt			[RFC Editor]
	draft-ietf-avt-dsr-05.txt			[RFC Editor]
	draft-ietf-avt-evrc-smv-03.txt			[RFC Editor]
	draft-ietf-srtp-08.txt 				[IESG review]
	draft-ietf-avt-mpeg4-simple-07.txt		[IESG review]
	draft-ietf-avt-rtp-clearmode-00.txt
	draft-ietf-avt-mwpp-midi-rtp-07.txt
	draft-lazzaro-avt-mwpp-coding-guidelines-03.txt
	draft-ietf-avt-rtp-jpeg2000-03.txt
	draft-hatanaka-avt-rtp-atracx-02.txt 
	draft-flaks-avt-rtp-ac3-03.txt
	draft-ramalho-rgl-desc-01.txt
	draft-ramalho-rgl-rtpformat-01.txt
	draft-chen-rtp-bv-00.txt
	draft-lazzaro-avt-rtp-framing-contrans-01.txt
	FAX over RTP in ITU-T SG16 Q.14
	New charter

RTCP Extended Reports					(Friedman)	 5
	draft-ietf-avt-rtcp-report-extns-06.txt

RTP Payload Format for DTMF Digits, Tones, etc		(Schulzrinne)	 5
	draft-ietf-avt-rfc2833bis-03.txt

RTP Payload Format for Speex				(Kerr)		15
	draft-herlein-speex-rtp-profile-01.txt

RTP Payload Format for iLBC Speech/iLBC codec		(Duric)		15
	draft-ietf-avt-rtp-ilbc-02.txt
	draft-ietf-avt-ilbc-codec-02.txt

RTP Payload Format for Uncompressed Video		(Gharai)	15
	draft-ietf-avt-uncomp-video-03.txt

RTP Payload Format for H.264 video			(Wenger)	15
	draft-ietf-avt-rtp-h264-02.txt 

RTP Payload Format for 3GPP Timed Text			(Rey)		15
	draft-rey-avt-3gpp-timed-text-00.txt 

RTCP Extensions for SSM with Unicast Feedback		(Ott)		15
	draft-ietf-avt-rtcpssm-04.txt

CRTP over MPLS 						(Ash)		20
	draft-ash-e2e-crtp-hdr-compress-01.txt
	draft-ash-e2e-vompls-hdr-compress-01.txt
	draft-ash-e2e-voip-hdr-comp-rqmts-00.txt

				 - * -


_______________________________________________
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avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Fri Jul  4 04:54:29 2003
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Subject: Re: [AVT] draft-herlein-speex-rtp-profile-01
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Colin / all, 

Sorry for missing this information out.

Changes in the -01 draft.

Expanded RTP header section detailing bitfield usage particularly the M bit at
the suggestion of Henning Schulzrinne.

Expanded payload section detailing frame packing and fragmentation handling at
the suggestion of Federico Montesino Pouzols.

Expanded example packet section detailing single Speex frame with padding and
multiple Speex frame packing at the suggestion of Magnus Westerlund and Ross
Finlayson.

MIME type section expanded to clarify usage scenario of audio/speex.

General text changes to improve clarity.


Open issues:

Maximum path-MTU discovery to prevent fragmentation.


I've booked a slot in Vienna for the Speex I-D, as well as the Vorbis RTP draft.

Regards

Phil



Quoting Colin Perkins <csp@csperkins.org>:

> --> philkerr@elec.gla.ac.uk writes:
> > Dear all,
> > 
> > An -01 update to draft-herlein-speex-rtp-profile is now available:
> > 
> > http://www.ietf.org/internet-drafts/draft-herlein-speex-rtp-profile-01.txt
> > 
> > Comments and feedback welcomed.
> 
> Can you summarize the changes from the previous version, and areas where
> input is needed?
> 
> -- 
> Colin Perkins
> csp@csperkins.org
> 




-------------------------------------------------
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_______________________________________________
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avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Fri Jul  4 10:38:37 2003
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Subject: [AVT] Implementation of RFC 3389 (Comfort noise)?
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I'm looking for a library (C, C++ or Java) that generates comfort noise 
according to RFC 3389. Pointers are appreciated.

Thank you.

Henning Schulzrinne


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avt@ietf.org
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Subject: [AVT] RFC 2833bis: open issues
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As requested by Colin, the Vienna discussion items for 2833bis (named 
events and tones):

There have been no major changes, except a cleanup of MF description 
based on ITU and other documents. I believe this is now essentially correct.

The example was extended to include a sequence of packets, not just one. 
Based on feedback from a number of readers, editorial changes were made 
to make the document easier to read.

I believe the draft would be ready for last call if we weren't blocked 
on interop statements to progress the document to Draft Standard. There 
has been essentially no progress on this, unfortunately.

There are two choices:
(1) Cycle at Proposed and hope that somebody else has the energy to drum 
up interop reports so that the document can progress to Draft at some 
later point. I wouldn't hold my breath, so this means that 2833bis joins 
the list of Proposed Standards that run the Internet.

(2) Try to come up with a reasonably small list of items to test, such as
- support for end-of-event indication
- support for end-of-event replication
- support for long events
- RFC 2198
- frequency generation
- list of events supported

and hope that people will step up and report. Finding pairs of 
implementations that, say, support a particular code point is going to 
be rather challenging, unless we define support as "printf the correct 
number". I would suggest that a report of usage (transmit & receive) of 
an event type should be considered sufficient here since there's no 
dependency on the sender.

Henning



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From: John Lazzaro <lazzaro@CS.Berkeley.EDU>
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Subject: [AVT] Vienna status reports ...
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Hi everyone,

	I won't be presenting at Vienna, however, here are quick
status reports for the "RTP MIDI" and "RTP framing for TCP/TLS" I-Ds:

draft-ietf-avt-mwpp-midi-rtp-08
-------------------------------

	This is the normative RTP MIDI I-D. Two revisions since San
Francisco, to address MMA concerns and to achieve compatibility with
all of the MMA standards.

	Presently, I'm recoding sfront networking to support RTP MIDI.
In doing so, I expect to catch some RTP MIDI protocol bugs.  I may
also do a tiny RTSP RTP MIDI server, and add RTSP sfront support, to
catch content-streaming bugs, if time permits.

	Once this coding work is complete, I plan to update the I-D
with bugfixes, add in changes in response to any additional comments I
receive on the document, and issue a Last Call document.  Most
probably well before the Minneapolis submission deadline.


draft-lazzaro-avt-mwpp-coding-guidelines-03
-------------------------------------------

	This is the "RTP MIDI Implementation Guidelines" informative
I-D.  One revision since San Francisco, to trim away most RTP-generic
content, and leave a shorter I-D that focuses on RTP MIDI specific
implementation issues.

	Several months ago, I requested WG item status for this I-D,
because I believe that RTP MIDI implementors should read it before
starting their design.  If the WG agrees, I'll submit a relabeled I-D,
and then put this I-D and the normative I-D up for Last Call
concurrently.


draft-lazzaro-avt-rtp-framing-contrans-01
-----------------------------------------

	This is the RTP framing over connection-oriented transport
I-D.  In response to comments in San Francisco, I rewrote the I-D,
removing the RTSP support, and adding RTP/AVP fmt token definitions
for use with TCP and TLS.  The new document is only 8 pages long.

	Further action on this document awaits the revised AVT WG
charter, and also awaits the mmusic WG reaching consensus on
draft-ietf-mmusic-sdp-comedia.

----

	I had hoped to also have input on ideas for adding audio
synchronization support to the IETF multimedia toolkit, as a potential
AVT WG charter addition.  Unfortunately, the I-Ds above took
precedence ... I think its important to focus on moving RTP MIDI out
to Last Call, so further work on synchronization will probably start
after the RTP MIDI I-D writing work completes.

-------------------------------------------------------------------------
John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley
lazzaro [at] cs [dot] berkeley [dot] edu     www.cs.berkeley.edu/~lazzaro
-------------------------------------------------------------------------

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: [AVT] Revised agenda for Vienna
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Folks,

Here's an updated agenda for the Vienna AVT meeting. There are two
additions: the RTP payload format for Vorbis, and a mention of IAB 
concerns re congestion control for VoIP.

-- 
Colin Perkins
csp@csperkins.org

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Audio/Video Transport Working Group

CHAIRS: Steve Casner <casner@acm.org>
        Colin Perkins <csp@csperkins.org>
	Magnus Westerlund <magnus.westerlund@ericsson.com>


Monday, 14th July 2003 at 09:00-11:30
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

Introduction, Status and Charter Update			(Chairs)	20
	draft-ietf-avt-rtp-retransmission-08.txt	[Last call]
	draft-ietf-avt-rtcp-feedback-07.txt		[Last call]
	draft-burmeister-avt-rtcp-feedback-sim-02.txt	[Last call]
	draft-ietf-avt-crtp-enhance-07.txt		[RFC Editor]
	draft-ietf-avt-rtp-new-12.txt			[RFC Editor]
	draft-ietf-avt-profile-new-13.txt		[RFC Editor]
	draft-ietf-avt-rtp-mime-06.txt			[RFC Editor]
	draft-ietf-avt-rtcp-bw-05.txt			[RFC Editor]
	draft-ietf-avt-dsr-05.txt			[RFC Editor]
	draft-ietf-avt-evrc-smv-03.txt			[RFC Editor]
	draft-ietf-srtp-08.txt 				[IESG review]
	draft-ietf-avt-mpeg4-simple-07.txt		[IESG review]
	draft-ietf-avt-rtp-clearmode-00.txt
	draft-ietf-avt-mwpp-midi-rtp-07.txt
	draft-lazzaro-avt-mwpp-coding-guidelines-03.txt
	draft-ietf-avt-rtp-jpeg2000-03.txt
	draft-hatanaka-avt-rtp-atracx-02.txt=20
	draft-flaks-avt-rtp-ac3-03.txt
	draft-ramalho-rgl-desc-01.txt
	draft-ramalho-rgl-rtpformat-01.txt
	draft-chen-rtp-bv-00.txt
	draft-lazzaro-avt-rtp-framing-contrans-01.txt
	FAX over RTP in ITU-T SG16 Q.14
	New charter

IAB Concerns Regarding Congestion Control for VoIP	(Floyd)		 5
	draft-iab-congestion-00.txt=20

RTCP Extended Reports					(Friedman)	 5
	draft-ietf-avt-rtcp-report-extns-06.txt

RTP Payload Format for DTMF Digits, Tones, etc		(Schulzrinne)	 5
	draft-ietf-avt-rfc2833bis-03.txt

RTP Payload Format for Speex				(Kerr)		10
	draft-herlein-speex-rtp-profile-01.txt

RTP Payload Format for Vorbis				(Kerr)		10
	draft-kerr-avt-vorbis-rtp-02.txt

RTP Payload Format for iLBC Speech/iLBC codec		(Duric)		15
	draft-ietf-avt-rtp-ilbc-02.txt
	draft-ietf-avt-ilbc-codec-02.txt

RTP Payload Format for Uncompressed Video		(Gharai)	10
	draft-ietf-avt-uncomp-video-03.txt

RTP Payload Format for H.264 video			(Wenger)	15
	draft-ietf-avt-rtp-h264-02.txt=20

RTP Payload Format for 3GPP Timed Text			(Rey)		15
	draft-rey-avt-3gpp-timed-text-00.txt=20

RTCP Extensions for SSM with Unicast Feedback		(Ott)		15
	draft-ietf-avt-rtcpssm-04.txt

CRTP over MPLS 						(Ash)		20
	draft-ash-e2e-crtp-hdr-compress-01.txt
	draft-ash-e2e-vompls-hdr-compress-01.txt
	draft-ash-e2e-voip-hdr-comp-rqmts-00.txt

				 - * -


--Multipart_Sun__6_Jul_2003_17:08:05_+0200_08180800--

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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        magnus.westerlund@ericsson.com, mankin@psg.com, ned.freed@mrochek.com,
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Content-Type: multipart/alternative; boundary="=_alternative 005C46FCC1256D59_="
Subject: [AVT] Re: IESG Review of draft-ietf-avt-mpeg4-simple-07.txt - Discuss Comments
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Dear Allison, all,

Thanks for the IESG comments. In response I would like to suggest the=20
following:

1) Comment:  It is strange to have more than one section labeled
  "Introduction."  Please pick a new label for section 2.1.

My proposal is to label 2.1 as "Signaling by MIME format parameters"

2) Where is the security model defined for ECMAScript in this context?=20
   (Problems with the model have been part of the Javascript security=20
   problem for Web browsers.)

MPEG-4 defines some important constraints on the use of ECMA scripts in=20
MPEG-4. An annex to the MPEG-4 system spec describes the differences=20
between "regular" ECMA scipts and "MPEG-4 scripts". As far as I understand =

these constraints do not allow dangerous ECMA script constructs, which=20
means there is no need for an ECMA script security model in this context.=20
Below I attached the MPEG-4 annex that describes the differences.=20
I suggest to resolve this as follows:
a) replace in section 5 "security considerations" all current references=20
to ECMAScript by MPEG-4 script, and=20
b) add to the fifth paragraph (starting with "In ISO/IEC 14496-1 a=20
security model is defined for ...") the following trailing sentence "Note: =

MPEG-4 scripts are based on ECMA scripts, but there is no need for an ECMA =

script security model, as the use of insecure ECMA  script constructs is=20
impossible in MPEG-4 scripts."

3) 2.4:  Why is this form of application-level fragmentation better than
   IP fragmentation?

This is because of error resilience. If IP fragmentation occurs without=20
application-level fragmentation, then all data of the entire Access Unit=20
gets lost when one such fragmented IP packet is lost. When=20
application-level fragmentation is used, and one RTP packet with an AU=20
fragment gets lost, then the received RTP packet(s)  with the other AU=20
fragments can still be decoded. I suggest to remove "so as to avoid IP=20
layer fragmentation" from the first sentence of section 2.4, and to add=20
the following after the first sentence: "Hence when an IP packet is lost=20
after IP-level fragmentation, only an AU fragment may get lost instead of=20
the entire AU".

Any comments very welcome.


Best regards,

Jan van der Meer


*******begin of MPEG-4 system annex*******

        MPEG-4 Scripts Have a Rigid Representation=20
MPEG-4 scripts differ slightly from ECMA scripts. The most important=20
difference is that MPEG-4 scripts are not represented textually, but are=20
transmitted as a parse tree representation.  This means that only=20
constructs that can be represented by the MPEG-4 parse grammar can be=20
encoded and transmitted. Not all ECMA script constructs can be represented =

in MPEG-4 scripts.=20
The differences between ECMA scripts and scripts that can be represented=20
in MPEG-4 are given below.

        Keywords
MPEG-4 scripts cannot utilize the following keywords: catch delete do final=
ly in instanceof throw try typeof void with .
This means that  do ? while loops and  for ? in loops are not possible.
        Relational operators
The relational operators  "=3D=3D=3D" and "!=3D=3D" cannot be included in M=
PEG-4 Scripts.=20

        Labeled statements
In MPEG-4 scripts it is impossible to label statements and to break or cont=
inue to labeled statements.

        Switch statement restriction
MPEG-4 scripts with switch statements can only take numerical case expressi=
ons and always must have at least one case statement.
In particular this means that=20
switch {
        case (x+1):  ?.
}

is not possible, while
Switch {
        case 1:  ?.
}
is okay.
        Functions, not programs
The MPEG-4 event driven script model allows only functions to be called in =

response to events.=20

        Expressions=20
Statements that include statement blocks, such as for are represented in=20
the parse tree as having an empty statement block, where as in ECMA script =

they can omit this block. Functionally, the statements behave identically. =

For example, the expression:=20
        for ( <expr>; <expr>; <expr>)=20

must be represented as

        for ( <expr>; <expr>; <expr>) {}

        Array and Object Literals
Array and object literals of the form [value1, value2, .., valueN] and {pro=
perty1:value1, property2: value2, .. propertyN:valueN} cannot be used in MP=
EG-4 scripts.=20
*******end of MPEG-4 system annex*******












Allison Mankin <mankin@psg.com>
2003-06-27 06:19 AM
Please respond to mankin

=20
        To:     Jan vanderMeer/EHV/CE/PHILIPS@EMEA3
dmackie@apple.com
viswanathan.swaminathan@sun.com
singer@apple.com
Philippe Gentric/MP4-SUR/CE/PHILIPS@EMEA1
        cc:     casner@acm.org
magnus.westerlund@ericsson.com
csp@csperkins.org
smb@research.att.com
ned.freed@mrochek.com
mankin@psg.com
avt@ietf.org
        Subject:        IESG Review of draft-ietf-avt-mpeg4-simple-07.txt -=
 Discuss Comments
        Classification:=20



The IESG reviewed draft-ietf-avt-mpeg4-simple-07.txt and had a few=20
concerns
that should be addressed before the draft can advance.  Editorially:

   Comment:  It is strange to have more than one section labeled
  "Introduction."  Please pick a new label for section 2.1.

Steve Bellovin and Ned Freed both request that a reference be given in
the Security Considerations for security model for ECMAscript.  Here
is Steve's Discuss comment:

   Where is the security model defined for ECMAScript in this context?=20
   (Problems with the model have been part of the Javascript security=20
   problem for Web browsers.)

Steve also asked:

   2.4:  Why is this form of application-level fragmentation better than
   IP fragmentation?

Please discuss the issues in email and we'll see if the fixes to the draft
can be done as notes rather than a revised i-d, to be quicker.

Allison










--=_alternative 005C46FCC1256D59_=
Content-Type: text/html; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable


<br><font size=3D2 face=3D"sans-serif">Dear Allison, all,</font>
<br>
<br><font size=3D2 face=3D"sans-serif">Thanks for the IESG comments. In res=
ponse I would like to suggest the following:</font>
<br>
<br><font size=3D2 face=3D"Courier New">1) Comment: &nbsp;It is strange to =
have more than one section labeled<br>
 &nbsp;&quot;Introduction.&quot; &nbsp;Please pick a new label for section =
2.1.</font>
<br>
<br><font size=3D2 face=3D"sans-serif">My proposal is to label 2.1 as &quot=
;Signaling by MIME format parameters&quot;<br>
</font>
<br><font size=3D2 face=3D"Courier New">2) Where is the security model defi=
ned for ECMAScript in this context? &nbsp;<br>
 &nbsp; (Problems with the model have been part of the Javascript security =
<br>
 &nbsp; problem for Web browsers.)<br>
</font>
<br><font size=3D2 face=3D"sans-serif">MPEG-4 defines some important constr=
aints on the use of ECMA scripts in MPEG-4. An annex to the MPEG-4 system s=
pec describes the differences between &quot;regular&quot; ECMA scipts and &=
quot;MPEG-4 scripts&quot;. As far as I understand these constraints do not =
allow dangerous ECMA script constructs, which means there is no need for an=
 ECMA script security model in this context. Below I attached the MPEG-4 an=
nex that describes the differences. </font>
<br><font size=3D2 face=3D"sans-serif">I suggest to resolve this as follows=
:</font>
<br><font size=3D2 face=3D"sans-serif">a) replace in section 5 &quot;securi=
ty considerations&quot; all current references to ECMAScript by MPEG-4 scri=
pt, and </font>
<br><font size=3D2 face=3D"sans-serif">b) add to the fifth paragraph (start=
ing with &quot;In ISO/IEC 14496-1 a security model is defined for ...&quot;=
) the following trailing sentence &quot;Note: MPEG-4 scripts are based on E=
CMA scripts, but there is no need for an ECMA script security model, as the=
 use of insecure ECMA &nbsp;script constructs is impossible in MPEG-4 scrip=
ts.&quot;</font>
<br>
<br><font size=3D2 face=3D"Courier New">3) 2.4: &nbsp;Why is this form of a=
pplication-level fragmentation better than<br>
 &nbsp; IP fragmentation?<br>
</font>
<br><font size=3D2 face=3D"sans-serif">This is because of error resilience.=
 If IP fragmentation occurs without application-level fragmentation, then a=
ll data of the entire Access Unit gets lost when one such fragmented IP pac=
ket is lost. When application-level fragmentation is used, and one RTP pack=
et with an AU fragment gets lost, then the received RTP packet(s) &nbsp;wit=
h the other AU fragments can still be decoded. I suggest to remove &quot;so=
 as to avoid IP layer fragmentation&quot; from the first sentence of sectio=
n 2.4, and to add the following after the first sentence: &quot;Hence when =
an IP packet is lost after IP-level fragmentation, only an AU fragment may =
get lost instead of the entire AU&quot;.</font>
<br>
<br><font size=3D2 face=3D"sans-serif">Any comments very welcome.</font>
<br>
<br><font size=3D2 face=3D"sans-serif"><br>
Best regards,<br>
<br>
Jan van der Meer<br>
</font>
<br>
<br><font size=3D2 face=3D"sans-serif">*******begin of MPEG-4 system annex*=
******</font>
<br>
<br><font size=3D2 face=3D"Times New Roman">&nbsp; &nbsp; &nbsp; &nbsp; </f=
ont><font size=3D2 face=3D"Arial"><b>MPEG-4 Scripts Have a Rigid Representa=
tion &nbsp;</b></font>
<br><font size=3D2 face=3D"Arial">MPEG-4 scripts differ slightly from ECMA =
scripts. The most important difference is that MPEG-4 scripts are not repre=
sented textually, but are transmitted as a parse tree representation. &nbsp=
;This means that only constructs that can be represented by the MPEG-4 pars=
e grammar can be encoded and transmitted. Not all ECMA script constructs ca=
n be represented in MPEG-4 scripts. </font>
<p><font size=3D2 face=3D"Arial">The differences between ECMA scripts and s=
cripts that can be represented in MPEG-4 are given below.</font>
<p>
<br><font size=3D2 face=3D"Times New Roman">&nbsp; &nbsp; &nbsp; &nbsp; </f=
ont><font size=3D2 face=3D"Arial"><b>Keywords</b></font>
<br><font size=3D2 face=3D"Arial">MPEG-4 scripts cannot utilize the followi=
ng keywords: </font><font size=3D1 face=3D"Courier New">catch delete do fin=
ally in instanceof throw try typeof void with</font><font size=3D2 face=3D"=
Courier New"> .</font>
<p><font size=3D2 face=3D"Arial">This means that &nbsp;</font><font size=3D=
2 face=3D"Courier New">do &#8211; </font><font size=3D1 face=3D"Courier New=
">while</font><font size=3D2 face=3D"Arial"> loops and </font><font size=3D=
2 face=3D"Courier New">&nbsp;</font><font size=3D1 face=3D"Courier New">for=
</font><font size=3D2 face=3D"Courier New"> &#8211; </font><font size=3D1 f=
ace=3D"Courier New">in</font><font size=3D2 face=3D"Arial"> loops are not p=
ossible.</font>
<p><font size=3D2 face=3D"Arial">&nbsp; &nbsp; &nbsp; &nbsp; </font><font s=
ize=3D2 face=3D"Arial"><b>Relational operators</b></font>
<p><font size=3D2 face=3D"Arial">The relational operators &nbsp;"</font><fo=
nt size=3D1 face=3D"Courier New">=3D=3D=3D</font><font size=3D2 face=3D"Ari=
al">" and "</font><font size=3D1 face=3D"Courier New">!=3D=3D</font><font s=
ize=3D2 face=3D"Arial">" cannot be included in MPEG-4 Scripts. </font>
<p>
<p><font size=3D2 face=3D"Times New Roman">&nbsp; &nbsp; &nbsp; &nbsp; </fo=
nt><font size=3D2 face=3D"Arial"><b>Labeled statements</b></font>
<p><font size=3D2 face=3D"Arial">In MPEG-4 scripts it is impossible to labe=
l statements and to</font><font size=3D1 face=3D"Courier New"> break</font>=
<font size=3D2 face=3D"Arial"> or </font><font size=3D1 face=3D"Courier New=
">continue</font><font size=3D2 face=3D"Arial"> to labeled statements.</fon=
t>
<p>
<p><font size=3D2 face=3D"Arial">&nbsp; &nbsp; &nbsp; &nbsp; </font><font s=
ize=3D2 face=3D"Arial"><b>Switch statement restriction</b></font>
<p><font size=3D2 face=3D"Arial">MPEG-4 scripts with </font><font size=3D1 =
face=3D"Courier New">switch</font><font size=3D2 face=3D"Arial"> statements=
 can only take numerical </font><font size=3D1 face=3D"Courier New">case</f=
ont><font size=3D2 face=3D"Arial"> expressions and always must have at leas=
t one </font><font size=3D1 face=3D"Courier New">case</font><font size=3D2 =
face=3D"Arial"> statement.</font>
<p><font size=3D2 face=3D"Arial">In particular this means that </font>
<p><font size=3D1 face=3D"Courier New">switch {</font>
<p><font size=3D1 face=3D"Courier New">&nbsp; &nbsp; &nbsp; &nbsp; case (x+=
1): &nbsp;&#8230;.</font>
<p><font size=3D1 face=3D"Courier New">}</font>
<p>
<p><font size=3D2 face=3D"Arial">is not possible, while</font>
<p><font size=3D1 face=3D"Courier New">Switch {</font>
<p><font size=3D1 face=3D"Courier New">&nbsp; &nbsp; &nbsp; &nbsp; case 1: =
&nbsp;&#8230;.</font>
<p><font size=3D1 face=3D"Courier New">}</font>
<p><font size=3D2 face=3D"Arial">is okay.</font>
<p><font size=3D2 face=3D"Arial">&nbsp; &nbsp; &nbsp; &nbsp; </font><font s=
ize=3D2 face=3D"Arial"><b>Functions, not programs</b></font>
<p><font size=3D2 face=3D"Arial">The MPEG-4 event driven script model allow=
s only functions to be called in response to events. </font>
<p>
<p><font size=3D2 face=3D"Times New Roman">&nbsp; &nbsp; &nbsp; &nbsp; </fo=
nt><font size=3D2 face=3D"Arial"><b>Expressions </b></font>
<p><font size=3D2 face=3D"Arial">Statements that include statement blocks, =
such as for are represented in the parse tree as having an empty statement =
block, where as in ECMA script they can omit this block. Functionally, the =
statements behave identically. For example, the expression: </font>
<p><font size=3D1 face=3D"Times New Roman">&nbsp; &nbsp; &nbsp; &nbsp; </fo=
nt><font size=3D1 face=3D"Courier New">for ( &lt;expr&gt;; &lt;expr&gt;; &l=
t;expr&gt;) </font>
<br>
<br><font size=3D2 face=3D"Arial">must be represented as</font>
<p>
<br><font size=3D1 face=3D"Times New Roman">&nbsp; &nbsp; &nbsp; &nbsp; </f=
ont><font size=3D1 face=3D"Courier New">for ( &lt;expr&gt;; &lt;expr&gt;; &=
lt;expr&gt;) {}</font>
<br>
<br><font size=3D2 face=3D"Courier New">&nbsp; &nbsp; &nbsp; &nbsp; </font>=
<font size=3D2 face=3D"Arial"><b>Array and Object Literals</b></font>
<br><font size=3D2 face=3D"Arial">Array and object literals of the form </f=
ont><font size=3D1 face=3D"Courier New">[<i>value1</i>, <i>value2</i>, .., =
<i>valueN</i>]</font><font size=3D2 face=3D"Arial"> and </font><font size=
=3D1 face=3D"Courier New">{<i>property1</i>:<i>value1</i>, <i>property2</i>=
: <i>value2</i>, .. <i>propertyN</i>:<i>valueN</i>}</font><font size=3D2 fa=
ce=3D"Arial"> cannot be used in MPEG-4 scripts. </font>
<p><font size=3D2 face=3D"sans-serif">*******end of MPEG-4 system annex****=
***</font>
<p>
<br><font size=3D2 face=3D"sans-serif"><br>
</font>
<br>
<br>
<br>
<table width=3D100%>
<tr valign=3Dtop>
<td>
<td>
<br>
<br>
<br>
<br>
<br><font size=3D1 face=3D"sans-serif"><b>Allison Mankin &lt;mankin@psg.com=
&gt;</b></font>
<p><font size=3D1 face=3D"sans-serif">2003-06-27 06:19 AM</font>
<br><font size=3D1 face=3D"sans-serif">Please respond to mankin</font>
<br>
<td><font size=3D1 face=3D"Arial">&nbsp; &nbsp; &nbsp; &nbsp; </font>
<br><font size=3D1 face=3D"sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; To: &nbs=
p; &nbsp; &nbsp; &nbsp;Jan vanderMeer/EHV/CE/PHILIPS@EMEA3<br>
dmackie@apple.com<br>
viswanathan.swaminathan@sun.com<br>
singer@apple.com<br>
Philippe Gentric/MP4-SUR/CE/PHILIPS@EMEA1</font>
<br><font size=3D1 face=3D"sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; cc: &nbs=
p; &nbsp; &nbsp; &nbsp;casner@acm.org<br>
magnus.westerlund@ericsson.com<br>
csp@csperkins.org<br>
smb@research.att.com<br>
ned.freed@mrochek.com<br>
mankin@psg.com<br>
avt@ietf.org</font>
<br><font size=3D1 face=3D"sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; Subject:=
 &nbsp; &nbsp; &nbsp; &nbsp;IESG Review of draft-ietf-avt-mpeg4-simple-07.t=
xt - Discuss Comments</font>
<p><font size=3D1 face=3D"sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; Classific=
ation: &nbsp; &nbsp; &nbsp; &nbsp;</font>
<br></table>
<br>
<br>
<br><font size=3D2 face=3D"Courier New">The IESG reviewed draft-ietf-avt-mp=
eg4-simple-07.txt and had a few concerns<br>
that should be addressed before the draft can advance. &nbsp;Editorially:<b=
r>
<br>
 &nbsp; Comment: &nbsp;It is strange to have more than one section labeled<=
br>
 &nbsp;&quot;Introduction.&quot; &nbsp;Please pick a new label for section =
2.1.<br>
<br>
Steve Bellovin and Ned Freed both request that a reference be given in<br>
the Security Considerations for security model for ECMAscript. &nbsp;Here<b=
r>
is Steve's Discuss comment:<br>
<br>
 &nbsp; Where is the security model defined for ECMAScript in this context?=
 &nbsp;<br>
 &nbsp; (Problems with the model have been part of the Javascript security =
<br>
 &nbsp; problem for Web browsers.)<br>
<br>
Steve also asked:<br>
<br>
 &nbsp; 2.4: &nbsp;Why is this form of application-level fragmentation bett=
er than<br>
 &nbsp; IP fragmentation?<br>
<br>
Please discuss the issues in email and we'll see if the fixes to the draft<=
br>
can be done as notes rather than a revised i-d, to be quicker.<br>
<br>
Allison<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
</font>
<br>
<br>
--=_alternative 005C46FCC1256D59_=--

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Mon Jul  7 10:01:57 2003
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From: "Jose Rey" <rey@panasonic.de>
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        "Magnus Westerlund" <magnus.westerlund@era.ericsson.se>,
        "Y. Matsui" <matsui.yoshinori@jp.panasonic.com>,
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Date: Mon, 7 Jul 2003 15:58:40 +0200
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Hi all,

here is a quick summary of the draft and a list of the issues we want to
start discussion on in Vienna.

This draft specifies an RTP payload format for the 3GPP timed text
format. 3GPP timed text is currently only conveyed via HTTP. For
upcoming services like streaming or multimedia broadcast multicast, an
RTP payload format is needed.

The main motivation is therefore to have an RTP payload format *ready*
for Rel6 (maybe March'04, probably June'04). Envisioned areas of
application are 3GPP's MBMS (Multimedia Broadcast/Multicast Services)
and PSS (Packet Switched Services) streaming.

The open issues/TODO list includes:

- resilient transport: repetition, fragmentation, retransmission, FEC
(other?). It is our opinion that repetition should be included at the
very least.  Fragmentation is reasonable when the packets get big and
should be documented as well. Retransmission and FEC can be used, if it
makes sense for the particular use case.

- Transport of sample description information: the sample description
box (see draft for more on this) is information which is needed by many
different text samples. This amount of information might become too big
to be transported out-of-band using SDP. What are the other options?
What is the threshold to decide for in- or out-of-band transport? Should
both transport options be documented?

looking forward to the meeting in Vienna,


Jos=E9 Rey____ mailto:rey@panasonic.de
Mobile Multimedia Team -  MCOM Group
Panasonic European Laboratories GmbH
Tel: +49(0)6103-766-134
Fax: +49(0)6103-766-166
http://www.pel.panasonic.de
____________________________________


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Greetings,

The Internet-Draft - draft-ietf-avt-rtp-jpeg2000-03.txt - has been posted.  

Some time passed before this version was submitted.  Most of the time was
spent discussing if potential reconciliation with work that was taking place
in ISO/IEC JTC1 SC 29 WG 01 (JPEG) was in order.  After lengthy discussions
it was agreed by the experts in WG 01 that there were different goals and
application areas targeted for our respective work and that the JP2/RTP
proposal (draft-ietf-avt-rtp-jpeg2000-03.txt) should move forward in IETF as
is.  

So at this stage we are ready to move to Last Call if the AVT experts have
no objections to the current draft.  

Please advise and comment.

Best regards,

Eric









_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: RE: [AVT] Agenda for Vienna
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From: "Ash, Gerald R (Jerry), ALABS" <gash@att.com>
To: "Colin Perkins" <csp@csperkins.org>, <avt@ietf.org>
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All,

Here is a brief summary regarding the following agenda item:

> CRTP over MPLS 						(Ash)		20
>	draft-ash-e2e-crtp-hdr-compress-01.txt
>	draft-ash-e2e-vompls-hdr-compress-01.txt
>	draft-ash-e2e-voip-hdr-comp-rqmts-00.txt

Background:

Some service providers (such as AT&T) have plans to migrate large =
amounts of legacy voice traffic to VoIP, as well as to provide VPN =
services enabling VoIP.  VoIP over MPLS ('VoMPLS') typically uses the =
encapsulation voice/RTP/UDP/IP/MPLS.  For an MPLS VPN, the packet header =
is at least 48 bytes, while the voice payload is typically no more than =
30 bytes.  VoIP over MPLS header compression can significantly reduce =
the VoIP overhead through various compression mechanisms.  This is =
important on access links where bandwidth is scarce, and can be =
important on backbone facilities, especially where costs are high (e.g., =
some global cross-sections).
 =20
VoIP over MPLS header compression methods have been proposed earlier =
(March, 2000) in the MPLS working group, however the relevant drafts =
have expired.  In particular, George Swallow and Lou Berger proposed a =
'simple' end-to-end compression scheme in which all first-order =
differences in the RTP/UDP/IP headers were transmitted, and in which the =
header compression context was established through RSVP signaling.  The =
work was accepted into the MPLS WG pending an addition to the MPLS WG =
charter.

In the I-D's we propose 3 approaches to VoMPLS header compression: a) =
re-use the methods in cRTP to determine the context and MPLS to route =
packets, b) re-use the methods in Swallow's and Berger's 'simple' =
approach to determine the context and MPLS to route packets, and c) =
re-use the methods in cRTP to determine the context and the SCID =
(session context ID) to route packets.

The work was presented at the IETF-56 MPLS WG meeting (slides and an =
excellent set of meetings notes are available at =
http://ietf.org/proceedings/03mar/index.html.

Issues to discuss:

1. WG charter -- The VoMPLS header compression work is not within the =
current charter of the AVT WG, so a charter extension is needed.  A =
discussion among the Transport Area Directors and Sub-IP Area Directors =
suggested that the work fits most closely within the Transport Area/AVT =
WG.

2. Protocol extensions -- As detailed in the I-Ds, extensions are =
proposed to [cRTP] and [cRTP-ENHANCE], which involve a new packet type =
field to identify FULL_HEADER, CONTEXT_STATE, etc. packets.  New objects =
are defined for [RSVP-TE].  Extensions are also proposed to RFC2547 =
VPNs, which create 'SCID routing tables' to allow routing based on the =
session context ID (SCID).  These extensions need coordination with =
other WGs (e.g., MPLS).

3. Resynchronization -- E2E VoMPLS using cRTP header compression might =
not perform well with frequent resynchronizations.  [cRTP-ENHANCE] seems =
appropriate in this context.  The 'simple' header compression approach =
would avoid the need for resynchronization, but achieves less efficiency =
than cRTP.

4. Scalability -- E2E VoMPLS applied between CE-CE would perhaps require =
a large number of LSPs to be created.  There is concern for CE ability =
to do the necessary processing and the scalability of the architecture.

4. LDP application -- It would be desirable to signal the VoMPLS tunnels =
with LDP, since many RFC2547 VPN implementations use LDP as the =
underlying LSP signaling mechanism.
Please review the documents and raise any issues/comments to the list.

Please review the documents and raise any issues/comments to the list.

Thanks,
Jerry Ash

_______________________________________________
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avt@ietf.org
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Dear all,

Last month (11th June) an -02 update to draft-kerr-avt-vorbis-rtp was announced
on the AVT list and a slot in Vienna has been booked for discussion.

The draft details the RTP delivery of Vorbis audio, SDP values and delivery of
the decoders static probability model, 'codebooks'.  Examples of Vorbis frame
packing and fragmented packet handling are also covered.

Open issues:

Congestion control.  Under development is a mechanism for dynamic bitrate
reduction called 'bitrate peeling'.  When this feature becomes available it will
impact on how congestion is dealt with and a mechanism allowing its use within a
 multicast environment needs to be explored.

Codebook delivery.  There are two mechanisms outlined for the delivery of the
codebooks, SDP and RTCP.  The codebooks may change mid-stream (at the boundary
between two tracks) as well as being static for the session lifetime.  As the
codebooks are needed for the successful decoding of the Vorbis stream a reliable
mechanism to handle their delivery of mid-session codebook changes should be
investigated (TCP over RTP?) in preference to using RTCP messages.

Path MTU.  There is a mechanism for splitting large Vorbis frames across
multiple RTP packets but a reliable mechanism for path-MTU discovery needs to be
outlined to prevent packet fragmentation not dealt with by the above.

I've also requested WG status for the I-D.


Comments and feedback welcomed.

Regards

Phil



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>  jan.vandermeer@philips.com writes:
>
> As far as I understand these constraints do not allow dangerous ECMA
> script constructs, which means there is no need for an ECMA script
> security model in this context. Below I attached the MPEG-4 annex that
> describes the differences.

Do "MPEG-4 scripts" give the script control over audio volume of the  
presentation, in a way that could override the levels manually set
by the human listening to the decoder?  A rogue program that blows
out the speakers of the victim's terminal seems like a security
risk of some sort, although not of the classical kind since data
and CPU and network are not compromised ... if this is an actual
risk, might be worth warning implementors in the Security
Considerations section ...

-------------------------------------------------------------------------
John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley
lazzaro [at] cs [dot] berkeley [dot] edu     www.cs.berkeley.edu/~lazzaro
-------------------------------------------------------------------------

_______________________________________________
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https://www1.ietf.org/mailman/listinfo/avt



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>  "Jose Rey" <rey@panasonic.de> writes:
>
> The open issues/TODO list includes:
>
> - resilient transport: repetition, fragmentation, retransmission, FEC
> (other?). It is our opinion that repetition should be included at the
> very least.

Is RTCP expected to be normally used with this payload format?  If so,
maybe the core idea behind the RTP MIDI resiliency -- a "recovery journal"
in every packet coding the recent past, whose contents grow and shrink
as the sender examines RTCP RR reports which indicate the highest 
sequence number the receiver has received -- could be the starting
point of a timed text resiliency scheme.  For the default 5 second
RTCP sending rate, and text that is not sent too frequently, one
could imagine a recovery journal approach would be more efficient
that simple repetition.  But you need RTCP to make it work ...

-------------------------------------------------------------------------
John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley
lazzaro [at] cs [dot] berkeley [dot] edu     www.cs.berkeley.edu/~lazzaro
-------------------------------------------------------------------------

_______________________________________________
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https://www1.ietf.org/mailman/listinfo/avt



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Subject: [AVT] Requirements draft on video control
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Attached you will find draft-basso-avt-videoconreq-00.txt

Abstract:

A variety of communication services such as video
conferencing and video messaging rely on the capability of video
encoders and decoders to exchange control commands. This document
outlines this set of commands as well as the requirements for
their transport.

Comments are more than welcome.





Andrea Basso
NMS Communications
(732) 936-2118
(See attached file: draft-basso-avt-videoconreq-00.zip)

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_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: [AVT] Summary for draft-ietf-avt-uncomp-video-03
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Dear all:

The following is a list of changes made to draft
draft-ietf-avt-uncomp-video-03.txt

  - the tables tabulating pgroup values have been replaced by 
    explanatory text 
  - pgroup is not longer signaled out of band
  - SDP parameters color-mode and sub-sampling have been merged
    into a single parameter called sampling. 
  - added text detailing relation to RFC 3497
  - added BGR color sub-sampling

The only remaining open issue left on the draft is:
  - are the values listed for the SDP colorimetry parameters 
    sufficient?
    these values are:  BT601-5, BT709-2, SMPTE240M  


thank you,
Ladan





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From: "Claudio Thomas" <claudio.thomas@kako.de>
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Date: Tue, 8 Jul 2003 09:54:58 +0200
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Subject: [AVT] possible error on testvectors of draft "draft-ietf-avt-srtp-08.txt"
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This is a multi-part message in MIME format.

------=_NextPart_000_000D_01C34536.FE6A0F60
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	charset="us-ascii"
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Actual link:
<http://www.ietf.org/internet-drafts/draft-ietf-avt-srtp-08.txt>
http://www.ietf.org/internet-drafts/draft-ietf-avt-srtp-08.txt

=20

Chapter B.1 AES-f8 Test Vectors (page 52)

=20

Position: j=3D2

=20

IV' XOR j : 595b699bbd3bc0df26062093c1ad8f70

Should be

IV' XOR j : 595b699bbd3bc0df26062093c1ad8f71

=20

Reason:

Last byte of IV' =3D 73, j=3D02:  73 xor 02 =3D 71 (0011 xor 0010 =3D =
0001)

=20

If my calculator is right, then the values after this position also =
can't be
correct.

=20

Do someone have already implemented it to give me working vextors? Or =
can
somebody explain me what is wrong on my calc?

=20

Thanks,

=20

Claudio

=20

=20


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<div class=3DSection1>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
lang=3DEN-GB style=3D'font-size:10.0pt;font-family:"Courier New"'>Actual =
link: </span></font><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'><a
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-avt-srtp-08.txt"><=
span
lang=3DEN-GB>http://www.ietf.org/internet-drafts/draft-ietf-avt-srtp-08.t=
xt</span></a></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
lang=3DEN-GB style=3D'font-size:10.0pt;font-family:"Courier =
New"'>&nbsp;</span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
lang=3DEN-GB style=3D'font-size:10.0pt;font-family:"Courier =
New"'>Chapter B.1
AES-f8 Test Vectors (page 52)</span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
lang=3DEN-GB style=3D'font-size:10.0pt;font-family:"Courier =
New"'>&nbsp;</span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
lang=3DEN-GB style=3D'font-size:10.0pt;font-family:"Courier =
New"'>Position: j=3D2</span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
lang=3DEN-GB style=3D'font-size:10.0pt;font-family:"Courier =
New"'>&nbsp;</span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
lang=3DEN-GB style=3D'font-size:10.0pt;font-family:"Courier New"'>IV' =
XOR j :
595b699bbd3bc0df26062093c1ad8f70</span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
lang=3DEN-GB style=3D'font-size:10.0pt;font-family:"Courier New"'>Should =
be</span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
lang=3DEN-GB style=3D'font-size:10.0pt;font-family:"Courier New"'>IV' =
XOR j :
595b699bbd3bc0df26062093c1ad8f71</span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
lang=3DEN-GB style=3D'font-size:10.0pt;font-family:"Courier =
New"'>&nbsp;</span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
lang=3DEN-GB style=3D'font-size:10.0pt;font-family:"Courier =
New"'>Reason:</span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
lang=3DEN-GB style=3D'font-size:10.0pt;font-family:"Courier New"'>Last =
byte of IV'
=3D 73, j=3D02:&nbsp; 73 xor 02 =3D 71 (0011 xor 0010 =3D =
0001)</span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
lang=3DEN-GB style=3D'font-size:10.0pt;font-family:"Courier =
New"'>&nbsp;</span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
lang=3DEN-GB style=3D'font-size:10.0pt;font-family:"Courier New"'>If my =
calculator
is right, then the values after this position also can't be =
correct.</span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
lang=3DEN-GB style=3D'font-size:10.0pt;font-family:"Courier =
New"'>&nbsp;</span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
lang=3DEN-GB style=3D'font-size:10.0pt;font-family:"Courier New"'>Do =
someone have
already implemented it to give me working vextors? Or can somebody =
explain me
what is wrong on my calc?</span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
lang=3DEN-GB style=3D'font-size:10.0pt;font-family:"Courier =
New"'>&nbsp;</span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New"'>Thanks,</span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New"'>&nbsp;</span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New"'>Claudio</span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New"'>&nbsp;</span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>&nbsp;</span></font></p>

</div>

</body>

</html>

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_______________________________________________
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Hi Phil,

I spent sometime yesterday to read your draft. I have a number of 
comments that needs to be addressed in the next version of the draft.

1. Section 2.1, Padding bit definition. I don't understand why it is 
required to use the Padding if one is below the MTU, we should not waste 
bandwidth unnecessary. I would suggest that the last sentence be removed 
and replaced with something like: "Padding MAY be used with this payload 
format according to section x.x of [RFC1889]."

2. Section 2.1, Extension bit. Please don't redefine the extension bit 
or other fields in the RTP header. An informative sentence or two is 
fine, however please just say that extension header (field) MAY be used 
according to RFC 1889. This comment is valid also for CSRC count.

3. Section 2.1, Marker bit. I think that better language in this 
explanation would make it easier to understand. Suggestion: "Always set 
to 0, as audio silence suppression is not used by the Vorbis codec."

4. Section 2.1, Timestamp. How many different rates does Vorbis use? 
Will multiple rates be used in the same session requiring multiple 
payload types to be defined for this? As RTP has some problem with 
switching rate within a session, would it possible to use a common 
denominatior rate for different rates and convey the information 
somewhere else?

5. Section 2.2, I think this payload header section is coming to early. 
I would suggest that a new section is created before the current 2.2 to 
explain the capabilities and basic structure of the payload format. When 
I read it I asked myself a lot of question that was answered on the 
following pages. I would suggest that you explain the single packet, 
fragmentation and aggregation type of packets.

6. Section 2.2: Last paragraph of page 4: "If C is set to one, this 
number SHOULD be be 0." Based on the following text this is a MUST, or 
otherwise you must define how one can include fragments and other packets.

7. Section 2.3: Third paragraph, page 5: The limitation that a Vorbis 
packet shall not be larger than 64k bytes. What about IPv6 Jumbograms 
and the fragmentation process? It seem possible to have larger packets 
although it would be a bad idea.

8. Section 2.3, fifth paragraph: Why it the length field only a single 
byte? In some use cases it seems that having aggregation only for 
packets smaller than 256 bytes may be limiting. Have you studied what 
the frequency different packets are created. If you have a lot of 
packets in the size of 257-500 bytes they may be advantageous to be able 
to aggregate.

9. Section 2.3, sixth paragraph: Please clarify that the packet length 
is counted in octets.

10. Section 2.3: How does one perform RTP TS recovery for the individual 
Vorbis packets in an aggregated packet? Are the RTP TS not needed or are 
there some implicit method. Please clarify.

11. Path MTU. I think that you should have a separate chapter on Path 
MTU discovery advice. I would also try to not write 1500 bytes as a 
common size. However, it might be common, the restraining of thinking is 
maybe one of the restricting factor that have prevented the usage of 
larger MTUs. I would recommend you to look at RFC 1191, and RFC1981. It 
would also be useful to make informative reference to the work in the 
newly formed PMTUD WG, a proposal draft does exist 
"draft-mathis-plpmtud-00.txt".

12. Section 3, Some parts of this section should be moved to before 2.2.

13. Section 3, first paragraph: This contradicts the SHOULD in section 
2.2. regarding fragmentation.

14. Section 3, third paragraph: This section is mentioning final 
fragment (F) which is not consistent with the field name in section 2.2.

15. Section 3: The fragmentation rules are lacking. To make the 
fragmentation proposal work you need to require that all fragments are 
sent in consecutive RTP packets. And that this schemes does not allow 
recovery of the fragment unless all packets belonging to the fragment 
arrives. The TS requirement is not enough, it is the RTP sequence number 
you must rely on for recovery.

16. Section 3.2: I think it is important to be clear that receiver 
dropped packets should not be counted or reported as lost in the RTCP 
statistics. IF one uses congestion control as one should based on RTCP 
reports the receiver side dropped packets will be counted as lost in the 
network and as being victim of congestion, that way resulting in 
reduction of rate. I would also not be so fast to recommend dropping. I 
would rather say that a receiver should buffer packets (reordering, 
jitter), and MUST not use unless all fragments arrive.

17. Section 4.1: Use of RTCP padding flag. This MAY be used but has no 
reason that it is required to use. I don't understand the reference to 
section 6.6 of RFC1889. Please review padding in RTP specification.

18. Section 4.1: Second paragraph. "VORB RTCP packets MUST be sent just 
ahead of the change ...". I have two comments about the sentence, first 
this can't be a MUST as it isn't possible to enforce and guarantee, so 
make it a SHOULD. Secondly, the "just ahead of" is very hard to 
understand what it really means. I thing that having a sentence saying 
that the RTCP packet SHOULD be sent ahead of the change is good. Then 
further motivate that it is crucial that it arrives before the change.

19. Section 4.1, Second paragraph, The last MUST in the paragraph is 
also one of this that can't be enforced by sender behaviour. Make it a 
SHOULD.

20. Section 4.1, If these codebooks, necessary information is so crucial 
  why not use an ACK packet to ACK when it has been received. This way 
the sender can utilize the RTCP bandwidth as efficient as possible. I 
would implement it using another SUBtype of the VORB APP packet.

21. Section 4.1, how to utilize the subtype in this APP packet is not 
defined.

22. Section 4.1 Why have have flags for the different parts when you are 
required to send all of the fields?

23. Section 4.1, the o flag: I think this flag need more explanation. 
Also further information on how to handle the situation when the APP 
packet brings the RTCP packet over the MTU.

24. Section 4.1 Second last paragraph on page 10: "This setup 
information ... " what does setup information refer to. I don't think 
this is an explained term.

25. Section 4.1, First paragraph, page 11: First sentence say that a 
receiver failing to receive the code book should obtain it from SDP. Is 
this a mistake, or simply lacking explanation on how that would work?

26. Section 4.1, Second paragraph page 11: "This message MUST be sent 
..." Why have flags if this MUST exist, secondly why having the 
enforcement of including a empty header, when it does not contain any 
information?

27. Section 4.1, what is one expected to use the URI and what types of 
URIs is allowed?

28. Section 4.2, Codebook caching needs more explanation.

29. Section 5: The usage of "u=", I don't think it is appropriate to use 
"u=" to include an URI to the codebooks and other stream information.

30. Section 5: The usage of "c=", The Vorbis payload format can't 
redefine the usage of "c=" field or require it to be used. SIP use the 
field to declare the receiver address, while RTSP does not use it at all.

31. Section 5.1, I think that all Vorbis specific SDP attributes should 
be sent in the a=fmtp line and be declared as required MIME parameters 
if they always must be present. Why are there no parameter for the 
codebook itself? Also an URI to this information could be included here. 
However the implications that a client must follow a link to fetch this 
information must be further explained.

32. Section 6, MIME registration: Parameters optional and required seems 
to exist and needs to be defined properly.

33. Section 6, Reference to RFCXXXX and a RFC-editor note to ask him to 
replace the number with the specifications assigned RFC number, is how 
one usually solves the references in the MIME registration.

34. If this becomes a WG draft AVT needs to be added to the Change 
control section.

35. IPR statement section is missing, please review ID-Nits. 
http://www.ietf.org/ID-nits.html


This was my comment for this revision. There are a number of serious 
shortcomings that needs to be addressed. See further comments to your 
open issues below.


philkerr@elec.gla.ac.uk wrote:
> Dear all,
> 
> Last month (11th June) an -02 update to draft-kerr-avt-vorbis-rtp was announced
> on the AVT list and a slot in Vienna has been booked for discussion.
> 
> The draft details the RTP delivery of Vorbis audio, SDP values and delivery of
> the decoders static probability model, 'codebooks'.  Examples of Vorbis frame
> packing and fragmented packet handling are also covered.
> 
> Open issues:
> 
> Congestion control.  Under development is a mechanism for dynamic bitrate
> reduction called 'bitrate peeling'.  When this feature becomes available it will
> impact on how congestion is dealt with and a mechanism allowing its use within a
>  multicast environment needs to be explored.

I think we also need to consider the IAB's statement about congestion 
control in this draft. However you can separate the congestion control 
issues for multicast and unicast. The unicast issue must be solved.

> 
> Codebook delivery.  There are two mechanisms outlined for the delivery of the
> codebooks, SDP and RTCP.  The codebooks may change mid-stream (at the boundary
> between two tracks) as well as being static for the session lifetime.  As the
> codebooks are needed for the successful decoding of the Vorbis stream a reliable
> mechanism to handle their delivery of mid-session codebook changes should be
> investigated (TCP over RTP?) in preference to using RTCP messages.

I think there is in fact a third alternative hinted, which is out of 
band fetching using HTTP, etc. I also think one should consider sessions 
where RTP with retransmission is used. This can ensure that a client 
reports if it does receive all fragments of the the code books. I also 
think that more clear guidelines that having a session long static book 
is to be preferred.

> 
> Path MTU.  There is a mechanism for splitting large Vorbis frames across
> multiple RTP packets but a reliable mechanism for path-MTU discovery needs to be
> outlined to prevent packet fragmentation not dealt with by the above.
> 

I think that you can lean on the already, not functioning so well 
methods, plus the newly initiated work on MTU discovery. I think simply 
having more information on where to go to find solutions is appropriate 
for this document.


Best Regards

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Tue Jul  8 05:47:32 2003
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Date: Tue, 08 Jul 2003 11:41:06 +0200
X-Sybari-Space: 00000000 00000000 00000000 00000000
From: =?ISO-8859-1?Q?Mats_N=E4slund?= <mats.naslund@ericsson.com>
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To: Claudio Thomas <claudio.thomas@kako.de>
CC: avt@ietf.org
Subject: Re: [AVT] possible error on testvectors of draft "draft-ietf-avt-srtp-08.txt"
References: <000c01c34526$3ae13f60$1e0ba8c0@kako.de>
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Dear Claudio,

Your observation appears correct. Unfortunately, the person
coding our f8 test code is on vacation, so it will take
some time before we can explain what the actual error is.
(A *hypothesis* is that our code has side-effects on the
variable IV', so that IV' is not kept fixed, but rather
updated "by accident".)

Thanks for being observant, we will get back as soon as
we can.

/Mats

Claudio Thomas wrote:
> Actual link:  <http://www.ietf.org/internet-drafts/draft-ietf-avt-srtp-08.txt> http://www.ietf.org/internet-drafts/draft-ietf-avt-srtp-08.txt
> 
>  
> 
> Chapter B.1 AES-f8 Test Vectors (page 52)
> 
>  
> 
> Position: j=2
> 
>  
> 
> IV' XOR j : 595b699bbd3bc0df26062093c1ad8f70
> 
> Should be
> 
> IV' XOR j : 595b699bbd3bc0df26062093c1ad8f71
> 
>  
> 
> Reason:
> 
> Last byte of IV' = 73, j=02:  73 xor 02 = 71 (0011 xor 0010 = 0001)
> 
>  
> 
> If my calculator is right, then the values after this position also can't be correct.
> 
>  
> 
> Do someone have already implemented it to give me working vextors? Or can somebody explain me what is wrong on my calc?
> 
>  
> 
> Thanks,
> 
>  
> 
> Claudio
> 


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: RE: [AVT] Summary of 3GPP timed text draft 
Date: Tue, 8 Jul 2003 13:57:20 +0200
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Hi John,

thanks for the comments. Please see below.

> -----Original Message-----
> From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of John
> Lazzaro
> Sent: Monday, July 07, 2003 9:56 PM
> To: avt@ietf.org
> Subject: Re: [AVT] Summary of 3GPP timed text draft
>
>
>
> >  "Jose Rey" <rey@panasonic.de> writes:
> >
> > The open issues/TODO list includes:
> >
> > - resilient transport: repetition, fragmentation,
> retransmission, FEC
> > (other?). It is our opinion that repetition should be
> included at the
> > very least.
>
> Is RTCP expected to be normally used with this payload format?

Not necessarily. Yes for unicast, but there's no plan to use RTCP
feedback in a first stage of the MBMS Service, that's why repetition is
important as a base mechanism.

cheers,

Jos=E9

> If so,
> maybe the core idea behind the RTP MIDI resiliency -- a
> "recovery journal"
> in every packet coding the recent past, whose contents grow and shrink
> as the sender examines RTCP RR reports which indicate the highest
> sequence number the receiver has received -- could be the starting
> point of a timed text resiliency scheme.  For the default 5 second
> RTCP sending rate, and text that is not sent too frequently, one
> could imagine a recovery journal approach would be more efficient
> that simple repetition.  But you need RTCP to make it work ...
>
> --------------------------------------------------------------
> -----------
> John Lazzaro -- Research Specialist -- CS Division -- EECS --
> UC Berkeley
> lazzaro [at] cs [dot] berkeley [dot] edu
> www.cs.berkeley.edu/~lazzaro
> --------------------------------------------------------------
> -----------
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
>


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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To: John Lazzaro <lazzaro@CS.Berkeley.EDU>
Cc: avt@ietf.org, avt-admin@ietf.org
Subject: Re: [AVT] Re: IESG Review of draft-ietf-avt-mpeg4-simple-07.txt - Discuss
 Comments
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Hello John,

Thanks for commenting. 

> Do "MPEG-4 scripts" give the script control over audio volume of the 
> presentation, in a way that could override the levels manually set
> by the human listening to the decoder?

In my understanding, MPEG-4 only provides the capability to control the 
presence of audio objects (streams) in a scene, hence the audio level can 
be controlled from zero up to a certain "nominal" level. MPEG-4 does not 
provide the capability to control the volume level at which the user 
"enjoys" the scene. In other words, in my understanding the problem that 
you mention does not exist in MPEG-4.

Best regards,

Jan










John Lazzaro <lazzaro@CS.Berkeley.EDU>
Sent by: 
avt-admin@ietf.org
2003-07-07 09:33 PM

 
        To:     avt@ietf.org
        cc:     (bcc: Jan vanderMeer/EHV/CE/PHILIPS)
        Subject:        [AVT] Re: IESG Review of draft-ietf-avt-mpeg4-simple-07.txt - Discuss 
Comments
        Classification: 




>  jan.vandermeer@philips.com writes:
>
> As far as I understand these constraints do not allow dangerous ECMA
> script constructs, which means there is no need for an ECMA script
> security model in this context. Below I attached the MPEG-4 annex that
> describes the differences.

Do "MPEG-4 scripts" give the script control over audio volume of the 
presentation, in a way that could override the levels manually set
by the human listening to the decoder?  A rogue program that blows
out the speakers of the victim's terminal seems like a security
risk of some sort, although not of the classical kind since data
and CPU and network are not compromised ... if this is an actual
risk, might be worth warning implementors in the Security
Considerations section ...

-------------------------------------------------------------------------
John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley
lazzaro [at] cs [dot] berkeley [dot] edu     www.cs.berkeley.edu/~lazzaro
-------------------------------------------------------------------------

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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<br><font size=2 face="sans-serif">Hello John,</font>
<br>
<br><font size=2 face="sans-serif">Thanks for commenting. </font>
<br>
<br><font size=2 face="Courier New">&gt; Do &quot;MPEG-4 scripts&quot; give the script control over audio volume of the &nbsp;<br>
&gt; presentation, in a way that could override the levels manually set<br>
&gt; by the human listening to the decoder?</font><font size=2 face="sans-serif"><br>
</font>
<br><font size=2 face="sans-serif">In my understanding, MPEG-4 only provides the capability to control the presence of audio objects (streams) in a scene, hence the audio level can be controlled from zero up to a certain &quot;nominal&quot; level. MPEG-4 does not provide the capability to control the volume level at which the user &quot;enjoys&quot; the scene. In other words, in my understanding the problem that you mention does not exist in MPEG-4.</font>
<br><font size=2 face="sans-serif"><br>
Best regards,<br>
<br>
Jan<br>
</font>
<br>
<br>
<br>
<table width=100%>
<tr valign=top>
<td>
<td>
<br>
<br>
<br>
<br>
<br><font size=1 face="sans-serif"><b>John Lazzaro &lt;lazzaro@CS.Berkeley.EDU&gt;</b></font>
<p><font size=1 face="sans-serif">Sent by: </font>
<br><font size=1 face="sans-serif">avt-admin@ietf.org</font>
<p><font size=1 face="sans-serif">2003-07-07 09:33 PM</font>
<br>
<td><font size=1 face="Arial">&nbsp; &nbsp; &nbsp; &nbsp; </font>
<br><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; To: &nbsp; &nbsp; &nbsp; &nbsp;avt@ietf.org</font>
<br><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; cc: &nbsp; &nbsp; &nbsp; &nbsp;(bcc: Jan vanderMeer/EHV/CE/PHILIPS)</font>
<br><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; Subject: &nbsp; &nbsp; &nbsp; &nbsp;[AVT] Re: IESG Review of draft-ietf-avt-mpeg4-simple-07.txt - Discuss Comments</font>
<p><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; Classification: &nbsp; &nbsp; &nbsp; &nbsp;</font>
<br></table>
<br>
<br>
<br><font size=2 face="Courier New"><br>
&gt; &nbsp;jan.vandermeer@philips.com writes:<br>
&gt;<br>
&gt; As far as I understand these constraints do not allow dangerous ECMA<br>
&gt; script constructs, which means there is no need for an ECMA script<br>
&gt; security model in this context. Below I attached the MPEG-4 annex that<br>
&gt; describes the differences.<br>
<br>
Do &quot;MPEG-4 scripts&quot; give the script control over audio volume of the &nbsp;<br>
presentation, in a way that could override the levels manually set<br>
by the human listening to the decoder? &nbsp;A rogue program that blows<br>
out the speakers of the victim's terminal seems like a security<br>
risk of some sort, although not of the classical kind since data<br>
and CPU and network are not compromised ... if this is an actual<br>
risk, might be worth warning implementors in the Security<br>
Considerations section ...<br>
<br>
-------------------------------------------------------------------------<br>
John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley<br>
lazzaro [at] cs [dot] berkeley [dot] edu &nbsp; &nbsp; www.cs.berkeley.edu/~lazzaro<br>
-------------------------------------------------------------------------<br>
<br>
_______________________________________________<br>
Audio/Video Transport Working Group<br>
avt@ietf.org<br>
https://www1.ietf.org/mailman/listinfo/avt<br>
</font>
<br>
<br>
--=_alternative 005290E3C1256D5D_=--

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Tue Jul  8 15:52:37 2003
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	(envelope-from csp@csperkins.org)
Date: Tue, 8 Jul 2003 19:27:30 +0200
From: Colin Perkins <csp@csperkins.org>
To: "Edwards, Eric" <Eric.Edwards@am.sony.com>
Cc: avt@ietf.org, casner@acm.org, magnus.westerlund@ericsson.com
Message-Id: <20030708192730.6ea0d380.csp@csperkins.org>
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Subject: [AVT] Re: draft-ietf-avt-rtp-jpeg2000-03.txt
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Eric,

--> "Edwards, Eric" <Eric.Edwards@am.sony.com> writes:
> The Internet-Draft - draft-ietf-avt-rtp-jpeg2000-03.txt - has been
> posted.  
> 
> Some time passed before this version was submitted.  Most of the time was
> spent discussing if potential reconciliation with work that was taking
> place in ISO/IEC JTC1 SC 29 WG 01 (JPEG) was in order.  After lengthy
> discussions it was agreed by the experts in WG 01 that there were
> different goals and application areas targeted for our respective work
> and that the JP2/RTP proposal (draft-ietf-avt-rtp-jpeg2000-03.txt) should
> move forward in IETF as is.  
> 
> So at this stage we are ready to move to Last Call if the AVT experts
> have no objections to the current draft.  
> 
> Please advise and comment.

I have a number of comments with regards to this draft:

 - Section 5.1 seems to imply that JPEG 2000 error resilience markers are
   only used with non-intelligent packetization. Is this correct? Are the
   error resilience markers not useful if the JPEG stream is intelligently
   packetized?

 - Is non-intelligent packetization a requirement? The situations where it
   will be useful seem very limited.

 - The extension mechanism in Section 8 is not used. I suggest removing it 
   and marking the X bit in the payload header as "reserved". 

 - The second paragraph of Section 9 is not appropriate, since the payload
   format cannot specify the type of video content to be transported. Are
   there no ways in which a malicious sender can disrupt the receiver?

 - In addition to the MIME type registration, the draft should include a 
   description of how the MIME type is mapped to SDP (Section 8.2 of the
   H.264 payload format is a good example).

 - Section 10 may be better written as an informative appendix? 

 - It may be useful to include some examples, showing the complete packets
   with all headers and data (including the RTP headers).

I would also urge a careful proof-reading of the draft for both quality of
English, clarity of specification, and appropriate use of the requirements
language in RFC 2119. 

-- 
Colin Perkins
csp@csperkins.org

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: RE: [AVT] Requirements draft on video control
Date: Tue, 8 Jul 2003 17:03:29 -0700
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Thread-Topic: [AVT] Requirements draft on video control
Thread-Index: AcNEz9VQ6y2SEFg1QmuFNQm9pmX67wA1yBiA
From: "Gary Sullivan" <garysull@windows.microsoft.com>
To: "Andrea Basso" <Andrea_Basso@nmss.com>, <avt@ietf.org>,
        <avt-admin@ietf.org>, <mmusic@ietf.org>
Cc: <casner@acm.org>, <jo@tzi.uni-bremen.de>, <magnus.westerlund@ericsson.com>,
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Much of the content of this ID will look familiar to those with
experience with ITU-T systems control specifications (such as H.245).

Some of it seems like it could use a bit of updating in that regard
(e.g., videofastupdateGOB).
Draft Recommendation H.241 is one good place to look for such
information.

Section 3 is a bit confused in its terminology.  For example it seems to
say that intra frames are also known as inter frames, which is
definitely not the case.  It mixes the terms "predictive" and
"prediction" and seems to consider a "reference" frame to be synonymous
with an intra frame.  The terms "picture" and "frame" are also kind of
mixed up.

A more consistent terminology would be as follows:

An intra picture is a picture that can be decoded without first decoding
any other picture.

A predicted (or non-intra) picture is a picture that may require data
from one or more previously-decoded pictures in order to be properly
decoded.

A reference picture is a picture that is stored in the decoder for use
as a reference in the decoding process of some subsequent picture in the
bitstream.

A non-reference picture is a picture that is not used as a reference for
the decoding process of any other picture in the bitstream.

The concepts of intra versus non-Intra and reference versus
non-reference are independent.  A particular picture can (in general) be
any one of the four types (intra reference, non-intra reference, intra
non-reference, non-intra non-reference).

A "fast update", also known as an "instantaneous decoder refresh",
involves sending an intra picture and thereafter refraining from using
any picture sent prior to that intra picture as a reference for the
decoding process of any subsequent picture sent in the stream.

In particular it is worth noting that some concepts that applied to some
older codecs are not really valid in general and should be avoided
(e.g., that a non-reference picture is called a B picture or that an
intra picture is the same thing as a fast update or that an intra
picture is always a reference picture).

The characterization of a "slice" or "GOB" as being a "stripe" is not
really quite right.  For example, a single slice in MPEG-1, H.263 Annex
O, or MPEG-4 can consist of a few macroblocks at the right edge of one
row and a few macroblocks at the left edge of the next row (i.e., two
distinct rectangular regions quite far apart from each other in the
picture can be in a single slice).  A GOB in H.261 CIF consists of THREE
partial rows of macroblocks extending only halfway across a picture.
And in H.264/AVC, a slice can be ANY collection of macroblocks scattered
around absolutely anywhere in the picture (by use of a feature known as
a "slice group map").

In regard to section 5.1 VideoFreezePicture, I believe H.262 (also known
as MPEG-2 Video) also supports a freeze picture release within the ITU-T
extension.

It is also possible to obtain random access recovery without a fast
update.  This is sometimes called "gradual decoder refresh".  See for
example the recovery point SEI message in H.264/AVC.

Best Regards,

Gary Sullivan


+> -----Original Message-----
+> From: avt-admin@ietf.org [mailto:avt-admin@ietf.org] On=20
+> Behalf Of Andrea Basso
+> Sent: Monday, July 07, 2003 2:34 PM
+> To: avt@ietf.org; avt-admin@ietf.org; mmusic@ietf.org
+> Cc: casner@acm.org; jo@tzi.uni-bremen.de;=20
+> magnus.westerlund@ericsson.com; csp@csperkins.org; Orit=20
+> Levin; Roni Even; nismail@cisco.com
+> Subject: [AVT] Requirements draft on video control
+>=20
+>=20
+> Attached you will find draft-basso-avt-videoconreq-00.txt
+>=20
+> Abstract:
+>=20
+> A variety of communication services such as video
+> conferencing and video messaging rely on the capability of video
+> encoders and decoders to exchange control commands. This document
+> outlines this set of commands as well as the requirements for
+> their transport.
+>=20
+> Comments are more than welcome.
+>=20
+>=20
+>=20
+>=20
+>=20
+> Andrea Basso
+> NMS Communications
+> (732) 936-2118
+> (See attached file: draft-basso-avt-videoconreq-00.zip)
+>=20

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Wed Jul  9 02:39:55 2003
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From: "Even, Roni" <roni.even@polycom.co.il>
To: "'Gary Sullivan'" <garysull@windows.microsoft.com>,
        Andrea Basso
	 <Andrea_Basso@nmss.com>, avt@ietf.org,
        avt-admin@ietf.org, mmusic@ietf.org
Cc: casner@acm.org, jo@tzi.uni-bremen.de, magnus.westerlund@ericsson.com,
        csp@csperkins.org, Orit Levin <orit@radvision.com>,
        "Even, Roni"
	 <roni.even@polycom.co.il>, nismail@cisco.com
Subject: RE: [AVT] Requirements draft on video control
Date: Wed, 9 Jul 2003 08:46:19 +0300 
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Hi Gary

About
"It is also possible to obtain random access recovery without a fast
update.  This is sometimes called "gradual decoder refresh".  See for
example the recovery point SEI message in H.264/AVC."

In case of a video switch in a multipoint conference there is a need to get
a new recovery point since the decoder will receive a stream without a
previous recovery point. The procedure in H.241 allows for either a full
picture at once or gradual recovery based on an SEI recovery point message.
Both method are fine as an encoder response to a fast update request.

Regards
Roni Even

-----Original Message-----
From: Gary Sullivan [mailto:garysull@windows.microsoft.com]
Sent: Wednesday, July 09, 2003 3:03 AM
To: Andrea Basso; avt@ietf.org; avt-admin@ietf.org; mmusic@ietf.org
Cc: casner@acm.org; jo@tzi.uni-bremen.de; magnus.westerlund@ericsson.com;
csp@csperkins.org; Orit Levin; Roni Even; nismail@cisco.com
Subject: RE: [AVT] Requirements draft on video control



Much of the content of this ID will look familiar to those with
experience with ITU-T systems control specifications (such as H.245).

Some of it seems like it could use a bit of updating in that regard
(e.g., videofastupdateGOB).
Draft Recommendation H.241 is one good place to look for such
information.

Section 3 is a bit confused in its terminology.  For example it seems to
say that intra frames are also known as inter frames, which is
definitely not the case.  It mixes the terms "predictive" and
"prediction" and seems to consider a "reference" frame to be synonymous
with an intra frame.  The terms "picture" and "frame" are also kind of
mixed up.

A more consistent terminology would be as follows:

An intra picture is a picture that can be decoded without first decoding
any other picture.

A predicted (or non-intra) picture is a picture that may require data
from one or more previously-decoded pictures in order to be properly
decoded.

A reference picture is a picture that is stored in the decoder for use
as a reference in the decoding process of some subsequent picture in the
bitstream.

A non-reference picture is a picture that is not used as a reference for
the decoding process of any other picture in the bitstream.

The concepts of intra versus non-Intra and reference versus
non-reference are independent.  A particular picture can (in general) be
any one of the four types (intra reference, non-intra reference, intra
non-reference, non-intra non-reference).

A "fast update", also known as an "instantaneous decoder refresh",
involves sending an intra picture and thereafter refraining from using
any picture sent prior to that intra picture as a reference for the
decoding process of any subsequent picture sent in the stream.

In particular it is worth noting that some concepts that applied to some
older codecs are not really valid in general and should be avoided
(e.g., that a non-reference picture is called a B picture or that an
intra picture is the same thing as a fast update or that an intra
picture is always a reference picture).

The characterization of a "slice" or "GOB" as being a "stripe" is not
really quite right.  For example, a single slice in MPEG-1, H.263 Annex
O, or MPEG-4 can consist of a few macroblocks at the right edge of one
row and a few macroblocks at the left edge of the next row (i.e., two
distinct rectangular regions quite far apart from each other in the
picture can be in a single slice).  A GOB in H.261 CIF consists of THREE
partial rows of macroblocks extending only halfway across a picture.
And in H.264/AVC, a slice can be ANY collection of macroblocks scattered
around absolutely anywhere in the picture (by use of a feature known as
a "slice group map").

In regard to section 5.1 VideoFreezePicture, I believe H.262 (also known
as MPEG-2 Video) also supports a freeze picture release within the ITU-T
extension.

It is also possible to obtain random access recovery without a fast
update.  This is sometimes called "gradual decoder refresh".  See for
example the recovery point SEI message in H.264/AVC.

Best Regards,

Gary Sullivan


+> -----Original Message-----
+> From: avt-admin@ietf.org [mailto:avt-admin@ietf.org] On 
+> Behalf Of Andrea Basso
+> Sent: Monday, July 07, 2003 2:34 PM
+> To: avt@ietf.org; avt-admin@ietf.org; mmusic@ietf.org
+> Cc: casner@acm.org; jo@tzi.uni-bremen.de; 
+> magnus.westerlund@ericsson.com; csp@csperkins.org; Orit 
+> Levin; Roni Even; nismail@cisco.com
+> Subject: [AVT] Requirements draft on video control
+> 
+> 
+> Attached you will find draft-basso-avt-videoconreq-00.txt
+> 
+> Abstract:
+> 
+> A variety of communication services such as video
+> conferencing and video messaging rely on the capability of video
+> encoders and decoders to exchange control commands. This document
+> outlines this set of commands as well as the requirements for
+> their transport.
+> 
+> Comments are more than welcome.
+> 
+> 
+> 
+> 
+> 
+> Andrea Basso
+> NMS Communications
+> (732) 936-2118
+> (See attached file: draft-basso-avt-videoconreq-00.zip)
+> 

_______________________________________________
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Date: Wed,  9 Jul 2003 09:53:47 +0100
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Cc: avt@ietf.org
Subject: Re: [AVT] Vorbis RTP issues list for Vienna
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Hi Magnus,

Thanks for a comprehensive review, comments in-line.


Quoting Magnus Westerlund <magnus.westerlund@ericsson.com>:

> Hi Phil,
> 
> I spent sometime yesterday to read your draft. I have a number of 
> comments that needs to be addressed in the next version of the draft.
> 
> 1. Section 2.1, Padding bit definition. I don't understand why it is 
> required to use the Padding if one is below the MTU, we should not waste 
> bandwidth unnecessary. I would suggest that the last sentence be removed 
> and replaced with something like: "Padding MAY be used with this payload 
> format according to section x.x of [RFC1889]."

Agreed.

> 
> 2. Section 2.1, Extension bit. Please don't redefine the extension bit 
> or other fields in the RTP header. An informative sentence or two is 
> fine, however please just say that extension header (field) MAY be used 
> according to RFC 1889. This comment is valid also for CSRC count.
> 
> 3. Section 2.1, Marker bit. I think that better language in this 
> explanation would make it easier to understand. Suggestion: "Always set 
> to 0, as audio silence suppression is not used by the Vorbis codec."

Agreed.

> 
> 4. Section 2.1, Timestamp. How many different rates does Vorbis use? 
> Will multiple rates be used in the same session requiring multiple 
> payload types to be defined for this? As RTP has some problem with 
> switching rate within a session, would it possible to use a common 
> denominatior rate for different rates and convey the information 
> somewhere else?
> 
> 5. Section 2.2, I think this payload header section is coming to early. 
> I would suggest that a new section is created before the current 2.2 to 
> explain the capabilities and basic structure of the payload format. When 
> I read it I asked myself a lot of question that was answered on the 
> following pages. I would suggest that you explain the single packet, 
> fragmentation and aggregation type of packets.

Agreed.  Section 2.0 should be expanded to outline the basic packet structure. 
This will make section 2.2 clearer.

> 
> 6. Section 2.2: Last paragraph of page 4: "If C is set to one, this 
> number SHOULD be be 0." Based on the following text this is a MUST, or 
> otherwise you must define how one can include fragments and other packets.

Yes, this should be a MUST.

> 
> 7. Section 2.3: Third paragraph, page 5: The limitation that a Vorbis 
> packet shall not be larger than 64k bytes. What about IPv6 Jumbograms 
> and the fragmentation process? It seem possible to have larger packets 
> although it would be a bad idea.

The codebooks can be far greater than 64k in size but a limit was imposed to
allow it to be transported inside a RTCP packet and to reduce the delay a client
may have in waiting for a large codebook to be received.   Perhaps a note
detailing that transport over IPv6 allows greater packet size would cover this
scenario.

> 
> 8. Section 2.3, fifth paragraph: Why it the length field only a single 
> byte? In some use cases it seems that having aggregation only for 
> packets smaller than 256 bytes may be limiting. Have you studied what 
> the frequency different packets are created. If you have a lot of 
> packets in the size of 257-500 bytes they may be advantageous to be able 
> to aggregate.

The length field size was specified in the original draft-moffitt-vorbis-rtp I-D
and it's been kept as it seems a fair limit, losing a packet with a 64 or 128
Vorbis frames will cause a greater skip than 32.  Aggregating packets larger
than 256 bytes is allowed so your scenario is catered for.  When you read this
section did you get the impression that this was not allowed?  If so it can be
altered to make it clearer.

> 
> 9. Section 2.3, sixth paragraph: Please clarify that the packet length 
> is counted in octets.

Yes, this will be fixed.

> 
> 10. Section 2.3: How does one perform RTP TS recovery for the individual 
> Vorbis packets in an aggregated packet? Are the RTP TS not needed or are 
> there some implicit method. Please clarify.

This needs further investigation.

> 
> 11. Path MTU. I think that you should have a separate chapter on Path 
> MTU discovery advice. I would also try to not write 1500 bytes as a 
> common size. However, it might be common, the restraining of thinking is 
> maybe one of the restricting factor that have prevented the usage of 
> larger MTUs. I would recommend you to look at RFC 1191, and RFC1981. It 
> would also be useful to make informative reference to the work in the 
> newly formed PMTUD WG, a proposal draft does exist 
> "draft-mathis-plpmtud-00.txt".

Agreed.

> 
> 12. Section 3, Some parts of this section should be moved to before 2.2.
> 
> 13. Section 3, first paragraph: This contradicts the SHOULD in section 
> 2.2. regarding fragmentation.

This will be fixed.

> 
> 14. Section 3, third paragraph: This section is mentioning final 
> fragment (F) which is not consistent with the field name in section 2.2.

Yes, the naming will be changed for consistency.

> 
> 15. Section 3: The fragmentation rules are lacking. To make the 
> fragmentation proposal work you need to require that all fragments are 
> sent in consecutive RTP packets. And that this schemes does not allow 
> recovery of the fragment unless all packets belonging to the fragment 
> arrives. The TS requirement is not enough, it is the RTP sequence number 
> you must rely on for recovery.

Section 3.1 describes this, focusing on the sequence number, but it will be
expanded for clarity.

> 
> 16. Section 3.2: I think it is important to be clear that receiver 
> dropped packets should not be counted or reported as lost in the RTCP 
> statistics. IF one uses congestion control as one should based on RTCP 
> reports the receiver side dropped packets will be counted as lost in the 
> network and as being victim of congestion, that way resulting in 
> reduction of rate. I would also not be so fast to recommend dropping. I 
> would rather say that a receiver should buffer packets (reordering, 
> jitter), and MUST not use unless all fragments arrive.

Agreed.

> 
> 17. Section 4.1: Use of RTCP padding flag. This MAY be used but has no 
> reason that it is required to use. I don't understand the reference to 
> section 6.6 of RFC1889. Please review padding in RTP specification.

Will do.

> 
> 18. Section 4.1: Second paragraph. "VORB RTCP packets MUST be sent just 
> ahead of the change ...". I have two comments about the sentence, first 
> this can't be a MUST as it isn't possible to enforce and guarantee, so 
> make it a SHOULD. Secondly, the "just ahead of" is very hard to 
> understand what it really means. I thing that having a sentence saying 
> that the RTCP packet SHOULD be sent ahead of the change is good. Then 
> further motivate that it is crucial that it arrives before the change.

Agreed, this section will be re-written to clarify 

> 
> 19. Section 4.1, Second paragraph, The last MUST in the paragraph is 
> also one of this that can't be enforced by sender behaviour. Make it a 
> SHOULD.

Yes, this will be fixed.

> 
> 20. Section 4.1, If these codebooks, necessary information is so crucial 
>   why not use an ACK packet to ACK when it has been received. This way 
> the sender can utilize the RTCP bandwidth as efficient as possible. I 
> would implement it using another SUBtype of the VORB APP packet.

This is a good idea and will be explored. 

> 
> 21. Section 4.1, how to utilize the subtype in this APP packet is not 
> defined.
> 
> 22. Section 4.1 Why have have flags for the different parts when you are 
> required to send all of the fields?

All of the header information is required for the initial setup of the clients
decoder but the codebook and metadata may change mid-session.  The meta-data
will probably be the most dynamic (track number/title/artist changing) and the
flags are used to indicate the payload sub-type.  The VORB RTCP packet can
contain one, or all of the sub headers components.

> 
> 23. Section 4.1, the o flag: I think this flag need more explanation. 
> Also further information on how to handle the situation when the APP 
> packet brings the RTCP packet over the MTU.
> 
> 24. Section 4.1 Second last paragraph on page 10: "This setup 
> information ... " what does setup information refer to. I don't think 
> this is an explained term.

This should link back to section 4, and the word setup will be changed to
configuration.

> 
> 25. Section 4.1, First paragraph, page 11: First sentence say that a 
> receiver failing to receive the code book should obtain it from SDP. Is 
> this a mistake, or simply lacking explanation on how that would work?

This needs to be explained better and I'll re-write this section.

> 
> 26. Section 4.1, Second paragraph page 11: "This message MUST be sent 
> ..." Why have flags if this MUST exist, secondly why having the 
> enforcement of including a empty header, when it does not contain any 
> information?

See comment above regarding point 22.

> 
> 27. Section 4.1, what is one expected to use the URI and what types of 
> URIs is allowed?
> 
> 28. Section 4.2, Codebook caching needs more explanation.

This will be expanded.

> 
> 29. Section 5: The usage of "u=", I don't think it is appropriate to use 
> "u=" to include an URI to the codebooks and other stream information.
> 
> 30. Section 5: The usage of "c=", The Vorbis payload format can't 
> redefine the usage of "c=" field or require it to be used. SIP use the 
> field to declare the receiver address, while RTSP does not use it at all.
> 
> 31. Section 5.1, I think that all Vorbis specific SDP attributes should 
> be sent in the a=fmtp line and be declared as required MIME parameters 
> if they always must be present. Why are there no parameter for the 
> codebook itself? Also an URI to this information could be included here. 
> However the implications that a client must follow a link to fetch this 
> information must be further explained.

I'll look into this further.

> 
> 32. Section 6, MIME registration: Parameters optional and required seems 
> to exist and needs to be defined properly.
> 
> 33. Section 6, Reference to RFCXXXX and a RFC-editor note to ask him to 
> replace the number with the specifications assigned RFC number, is how 
> one usually solves the references in the MIME registration.
> 
> 34. If this becomes a WG draft AVT needs to be added to the Change 
> control section.
> 
> 35. IPR statement section is missing, please review ID-Nits. 
> http://www.ietf.org/ID-nits.html

These will be addressed.

> 
> 
> This was my comment for this revision. There are a number of serious 
> shortcomings that needs to be addressed. See further comments to your 
> open issues below.
> 
> 
> philkerr@elec.gla.ac.uk wrote:
> > Dear all,
> > 
> > Last month (11th June) an -02 update to draft-kerr-avt-vorbis-rtp was
> announced
> > on the AVT list and a slot in Vienna has been booked for discussion.
> > 
> > The draft details the RTP delivery of Vorbis audio, SDP values and delivery
> of
> > the decoders static probability model, 'codebooks'.  Examples of Vorbis
> frame
> > packing and fragmented packet handling are also covered.
> > 
> > Open issues:
> > 
> > Congestion control.  Under development is a mechanism for dynamic bitrate
> > reduction called 'bitrate peeling'.  When this feature becomes available it
> will
> > impact on how congestion is dealt with and a mechanism allowing its use
> within a
> >  multicast environment needs to be explored.
> 
> I think we also need to consider the IAB's statement about congestion 
> control in this draft. However you can separate the congestion control 
> issues for multicast and unicast. The unicast issue must be solved.

Unicast congestion control should be the easier of the two to solve as the
client may send an explicit request to downgrade the stream, or the server may
do this automatically from the RR's.

> 
> > 
> > Codebook delivery.  There are two mechanisms outlined for the delivery of
> the
> > codebooks, SDP and RTCP.  The codebooks may change mid-stream (at the
> boundary
> > between two tracks) as well as being static for the session lifetime.  As
> the
> > codebooks are needed for the successful decoding of the Vorbis stream a
> reliable
> > mechanism to handle their delivery of mid-session codebook changes should
> be
> > investigated (TCP over RTP?) in preference to using RTCP messages.
> 
> I think there is in fact a third alternative hinted, which is out of 
> band fetching using HTTP, etc. I also think one should consider sessions 
> where RTP with retransmission is used. This can ensure that a client 
> reports if it does receive all fragments of the the code books. I also 
> think that more clear guidelines that having a session long static book 
> is to be preferred.

Agreed, but in certain usage scenarios the codebooks may change quite frequently
such as jingle breaks between two tracks. 

> 
> > 
> > Path MTU.  There is a mechanism for splitting large Vorbis frames across
> > multiple RTP packets but a reliable mechanism for path-MTU discovery needs
> to be
> > outlined to prevent packet fragmentation not dealt with by the above.
> > 
> 
> I think that you can lean on the already, not functioning so well 
> methods, plus the newly initiated work on MTU discovery. I think simply 
> having more information on where to go to find solutions is appropriate 
> for this document.

This sounds like the best solution.


Many thanks

Phil

> 
> 
> Best Regards
> 
> Magnus Westerlund
> 
> Multimedia Technologies, Ericsson Research EAB/TVA/A
> ----------------------------------------------------------------------
> Ericsson AB                | Phone +46 8 4048287
> Torshamsgatan 23           | Fax   +46 8 7575550
> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> 
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
> 




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Date: Wed, 9 Jul 2003 11:05:22 +0200
From: Colin Perkins <csp@csperkins.org>
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Subject: Re: [AVT] draft-herlein-speex-rtp-profile-01
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--> philkerr@elec.gla.ac.uk writes:
> Dear all,
> 
> An -01 update to draft-herlein-speex-rtp-profile is now available:
> 
> http://www.ietf.org/internet-drafts/draft-herlein-speex-rtp-profile-01.txt
> 
> Comments and feedback welcomed.

I have a number of questions and comments:

- It's not appropriate for the draft to claim that the Speex codec is free
  from patents and/or royalties.  If you wish to make an IPR statement, it
  should be separately submitted to the IESG Secretary for inclusion on the
  IPR-claims page the IETF maintains. See also section 10 of RFC 2026.

- The draft would benefit from a little more introduction to Speex at the
  beginning.

- The definition of the Padding bit in the RTP header is unusual. It may be
  better to state that padding MAY be used in the usual RTP manner, unless
  there is a specific reason why a different padding scheme is needed?

- The use of the M bit in the RTP header to signify comfort noise is also
  unusual. This is not necessarily inappropriate, but it is important to
  consider if the advantages of this use outweigh the costs of being
  different to other audio payload formats (which use the M bit to signal
  the first packet after a silent period).

- The definition of the timestamp says "Speex uses 20 msec frames and a
  variable sampling rate clock". Does this mean that the sample rate can
  change at anytime within a session? Or is the sample rate chosen from
  one of a number of values, and then fixed within the session?

- Section 3.2 is unclear if the bit rate and sample rate are coupled. Can
  Speex vary the compression ratio to change the bit rate without changing
  the sampling rate?

- In Section 3.2, it is important to be clear that the padding needed to
  ensure that the payload is octet aligned is independent of the padding
  at the RTP level (i.e. the P bit in the RTP header).

- The examples in Section 3.3 and 3.4 might be clearer if they show the
  actual length of the Speex data frames. An example showing two Speex
  frames in an RTP packet, with the first Speex frame being non-octet
  aligned, would be useful (to make it clear how to handle padding in
  this case).

- Section 4 "MIME registration of Speex" should be renamed to "IANA 
  considerations", and needs an introductory sentence saying something 
  like "The following MIME type and associated RTP payload format are 
  to be registered".

- The usual way in which a MIME type is registered only for particular
  use is to define the use in the "Encoding considerations" section of
  the MIME registration form. In this case, you might say something like:

        This type is defined only for transfer within RTP as specified in
        Section 4 of RFC XXXX. For file storage, or for transmission over
        lossless channels (such as TCP), Speex data is transported within
        an Ogg bitstream [RFC 3534].

  This would replace the introductory sentence and last paragraph of the
  current section 4. If there is a need to register parameters of the Ogg
  bitstream, that should be done as a separate subsection under section 4.

- The "a=fmtp:" parameters from Section 5 should be defined in Section 4,
  as Optional Parameters to the audio/speex type, and section 5 should be
  an explanation of how the parameters are mapped onto SDP.

- The "ebw" parameter may be redundant, since the clock rate is signalled
  in the "a=rtpmap:" line?

- It may be clearer to separate out the explanation of how parameters are
  mapped to SDP from the description of how the SDP is used with the offer/
  answer model (as two subsections). There are uses of SDP that don't use
  offer/answer.

- Section 6 (use with H.323) is not appropriate in an IETF document. You
  can include a reference to an ITU - or other - document explaining how
  to use Speex with H.323, but the IETF can't mandate use of H.245
  parameters.

- Section 10: RFCs are published using strict US ASCII. If there is an
  acceptable transliteration of the non-ASCII characters in Jean-Marc
  Valin's address, it will save a "discussion" with the RFC Editor. See
  draft-rfc-editor-rfc2223bis-06.txt.

- 
Colin Perkins
csp@csperkins.org

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Hi Phil,

Here are a few more comments and some pointers worth checking out.


philkerr@elec.gla.ac.uk wrote:
> Quoting Magnus Westerlund <magnus.westerlund@ericsson.com>:


>>7. Section 2.3: Third paragraph, page 5: The limitation that a Vorbis 
>>packet shall not be larger than 64k bytes. What about IPv6 Jumbograms 
>>and the fragmentation process? It seem possible to have larger packets 
>>although it would be a bad idea.
> 
> 
> The codebooks can be far greater than 64k in size but a limit was imposed to
> allow it to be transported inside a RTCP packet and to reduce the delay a client
> may have in waiting for a large codebook to be received.   Perhaps a note
> detailing that transport over IPv6 allows greater packet size would cover this
> scenario.
> 

I think that we need to clear consider how to get these codebooks 
transported. A larger than 64kb codebook should neither be transported 
in SDP or in RTCP. I think that using HTTP can clearly be an 
alternative, and only provide the URI to the codebook.

I wouldn't try to transport a codebook over RTCP that results in a RTCP 
packet many time the MTU due to the high number of IP fragments. Further 
such a packet will severely reduce the RTCP report rate in both directions.

> 
>>8. Section 2.3, fifth paragraph: Why it the length field only a single 
>>byte? In some use cases it seems that having aggregation only for 
>>packets smaller than 256 bytes may be limiting. Have you studied what 
>>the frequency different packets are created. If you have a lot of 
>>packets in the size of 257-500 bytes they may be advantageous to be able 
>>to aggregate.
> 
> 
> The length field size was specified in the original draft-moffitt-vorbis-rtp I-D
> and it's been kept as it seems a fair limit, losing a packet with a 64 or 128
> Vorbis frames will cause a greater skip than 32.  Aggregating packets larger
> than 256 bytes is allowed so your scenario is catered for.  When you read this
> section did you get the impression that this was not allowed?  If so it can be
> altered to make it clearer.

The first indication is the second paragraph in section 3: "Any Vorbis 
packet that is larger than 256 octets and less than the path-MTU MUST be 
placed in a RTP packet by itself.". Secondly the payload format does not 
describe a mechanism to allow it.

> 
>>10. Section 2.3: How does one perform RTP TS recovery for the individual 
>>Vorbis packets in an aggregated packet? Are the RTP TS not needed or are 
>>there some implicit method. Please clarify.
> 
> 
> This needs further investigation.

This is probably one of the most important things as most decoders and 
error recovery mechanism needs to receive the frames/packets in timing 
order plus with detection of lost frames.

> 
>>21. Section 4.1, how to utilize the subtype in this APP packet is not 
>>defined.
>>
>>22. Section 4.1 Why have have flags for the different parts when you are 
>>required to send all of the fields?
> 
> 
> All of the header information is required for the initial setup of the clients
> decoder but the codebook and metadata may change mid-session.  The meta-data
> will probably be the most dynamic (track number/title/artist changing) and the
> flags are used to indicate the payload sub-type.  The VORB RTCP packet can
> contain one, or all of the sub headers components.

That was not how I read the text. See section 4.1 paragraph 2 on page 
11. There is a MUST there in the third sentence that I interpreted as 
requiring all as needed.

I think that sending this meta information in RTCP can be appropriate, 
or in the packet stream if RTP retransmission is used. However the 
codebooks will most probably need special treatment, unless they are no 
bigger than a 1 kb.

> 
>>29. Section 5: The usage of "u=", I don't think it is appropriate to use 
>>"u=" to include an URI to the codebooks and other stream information.
>>
>>30. Section 5: The usage of "c=", The Vorbis payload format can't 
>>redefine the usage of "c=" field or require it to be used. SIP use the 
>>field to declare the receiver address, while RTSP does not use it at all.
>>
>>31. Section 5.1, I think that all Vorbis specific SDP attributes should 
>>be sent in the a=fmtp line and be declared as required MIME parameters 
>>if they always must be present. Why are there no parameter for the 
>>codebook itself? Also an URI to this information could be included here. 
>>However the implications that a client must follow a link to fetch this 
>>information must be further explained.
> 
> 
> I'll look into this further.

I would suggest that you take a look at the MIME and SDP sections of the 
  H264 payload format (draft-ietf-avt-h264-02.txt) or the AMR payload 
format (RFC 3267).

Cheers


Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com


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> Magnus Westerlund <magnus.westerlund@ericsson.com> writes:
>
> I think that we need to clear consider how to get these codebooks
> transported. A larger than 64kb codebook should neither be transported
> in SDP or in RTCP. I think that using HTTP can clearly be an
> alternative, and only provide the URI to the codebook.
>
> I wouldn't try to transport a codebook over RTCP that results in a
> RTCP packet many time the MTU due to the high number of IP
> fragments. Further such a packet will severely reduce the RTCP report
> rate in both directions.

I may be incorrect here, but from my discussions with Phil earlier
in the year, the codebook issue has these requirements:

 -- May change every few minutes (say, once per song when used for 
    Internet radio apps).
 -- Tightly synchronized with the stream: if you don't have the new
    codebook in time for the first packet that needs it, its a 
    catastrophe.
 -- Reliable reception a must, no built in resiliency if it is
    sent as a stream of UDP packets and some but not all arrive.

One way to handle this is to have two synchronized streams in the
media session: a UDP RTP/AVP stream for the media data, and a
TCP stream (perhaps an RTP/AVP stream if a solution like
draft-lazzaro-avt-rtp-framing-contrans happens, or perhaps an
opaque TCP stream) for the codebooks.  In either case, the
synchronization would happen as lip-sync synchronization happens.

-------------------------------------------------------------------------
John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley
lazzaro [at] cs [dot] berkeley [dot] edu     www.cs.berkeley.edu/~lazzaro
-------------------------------------------------------------------------

_______________________________________________
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Date: Wed, 09 Jul 2003 15:16:55 -0400
To: Colin Perkins <csp@csperkins.org>, philkerr@elec.gla.ac.uk
From: "Michael A. Ramalho" <mramalho@cisco.com>
Subject: Re: [AVT] draft-herlein-speex-rtp-profile-01
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Comment on M bit below.

At 11:05 AM 7/9/2003 +0200, Colin Perkins wrote:
>--> philkerr@elec.gla.ac.uk writes:
> > Dear all,
> >
> > An -01 update to draft-herlein-speex-rtp-profile is now available:
> >
> > http://www.ietf.org/internet-drafts/draft-herlein-speex-rtp-profile-01.txt
> >
> > Comments and feedback welcomed.
>
>I have a number of questions and comments:
>
>- It's not appropriate for the draft to claim that the Speex codec is free
>   from patents and/or royalties.  If you wish to make an IPR statement, it
>   should be separately submitted to the IESG Secretary for inclusion on the
>   IPR-claims page the IETF maintains. See also section 10 of RFC 2026.
>
>- The draft would benefit from a little more introduction to Speex at the
>   beginning.
>
>- The definition of the Padding bit in the RTP header is unusual. It may be
>   better to state that padding MAY be used in the usual RTP manner, unless
>   there is a specific reason why a different padding scheme is needed?
>
>- The use of the M bit in the RTP header to signify comfort noise is also
>   unusual. This is not necessarily inappropriate, but it is important to
>   consider if the advantages of this use outweigh the costs of being
>   different to other audio payload formats (which use the M bit to signal
>   the first packet after a silent period).

Colin,

This may be contrary to the guidance I got from Steve a while back.

A little background first ..

 From RFC 1889:

"marker (M): 1 bit The interpretation of the marker is defined by a 
profile. It
is intended to allow significant events such as frame boundaries to be marked
in the packet stream. A profile may define additional marker bits or specify
that there is no marker bit by changing the number of bits in the payload type
field (see Section 5.3)."

But from RFC 1890 (Audio and Video Conferences with Minimal Control
that can be used with G.711):

"For applications which send no packets during silence, the first packet of
a talk spurt (first packet after a silence period) is distinguished by setting
the marker bit in the RTP data header. Applications without silence
suppression set the bit to zero."

Then Steve said on 2/2/2003 (in regards to the RGL codec) ...

"I believe you should follow the marker bit convention as specified in
RFC 1890. The RGL codec is likely to be used in implementations that
also use other payload formats (such as PCMU) specified in RFC 1890,
and the behavior should be consistent."

Thus, since I think that speex will also be used in similar
deployments (i.e., packet voice) ... I would think Steve's
guidance should apply here as well.

My question to the speex authors is that if they simply define
the periods in which they desire to send comfort noise to be
*identically* the same definition as a RFC 1890 "beginning or
ending of a talk spurt", then there would be no inconsistency
in the bit's meaning.

My $0.02,

Michael Ramalho


Michael A. Ramalho, Ph.D.
Office email: mramalho@cisco.com
Personal email: mar42@cornell.edu
Office: +1.941.708.4650
Cell: +1.941.544.2844


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>I may be incorrect here, but from my discussions with Phil earlier
>in the year, the codebook issue has these requirements:
>
>  -- May change every few minutes (say, once per song when used for
>     Internet radio apps).

IMHO, this is a flaw in the "Vorbis" architecture.  It would be much better 
to have a fixed, predefined set of codebooks that all receivers would know 
in advance.  Then you'd need to send only a 'codebook id', which should be 
small enough to include in the Vorbis RTP payload format header.

Having to send a complete codebook for each song partially defeats the 
purpose of audio compression.

The Vorbis designers should just figure out a complete set of codebooks 
that are likely to be useful, and settle on those.

         Ross.


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: Aaron Colwell <acolwell@real.com>
To: Ross Finlayson <finlayson@live.com>
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Subject: Re: [AVT] Vorbis RTP issues list for Vienna
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I agree with Ross on this. It seems that these codebooks are causing too
many problems that aren't worth dealing with. People have been suggesting
getting the codebooks via HTTP, alternate TCP links, and all sorts of
other methods. If your going to go that far why not just use HTTP to
deliver the data. That way you are guaranteed to get everything in the
right order.

My guess is that other video and audio codecs use predefined tables to
avoid this very problem. Vorbis should do the same. The Vorbis folks
should come up with a "streamable" profile that encodes the audio using
predefined codebooks. Doing this makes all of the complicated codebook
issues go away. They could also create an offline tool that will translate
any Vorbis file into a "streamable" Vorbis file. This idea is similar to
hinting tools used for MPEG4 and 3GPP files.

Aaron

On Wed, 9 Jul 2003, Ross Finlayson wrote:

>
> >I may be incorrect here, but from my discussions with Phil earlier
> >in the year, the codebook issue has these requirements:
> >
> >  -- May change every few minutes (say, once per song when used for
> >     Internet radio apps).
>
> IMHO, this is a flaw in the "Vorbis" architecture.  It would be much better
> to have a fixed, predefined set of codebooks that all receivers would know
> in advance.  Then you'd need to send only a 'codebook id', which should be
> small enough to include in the Vorbis RTP payload format header.
>
> Having to send a complete codebook for each song partially defeats the
> purpose of audio compression.
>
> The Vorbis designers should just figure out a complete set of codebooks
> that are likely to be useful, and settle on those.
>
>          Ross.
>
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
>

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: RE: [AVT] Requirements draft on video control
Date: Wed, 9 Jul 2003 13:39:46 -0700
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From: "Gary Sullivan" <garysull@windows.microsoft.com>
To: "Andrea Basso" <Andrea_Basso@nmss.com>
Cc: <avt@ietf.org>, <casner@acm.org>, <csp@csperkins.org>,
        <jo@tzi.uni-bremen.de>, <magnus.westerlund@ericsson.com>,
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Andrea et al,

Perhaps I was having some trouble understanding the purpose of this ID.
I thought it was some sort of draft of a formal specification, rather
than just an expression of needs to be taken as input by the group.  So
I criticized the language more heavily than necessary.

In regard to the "intra frames" =3D "inter frames" mix-up:

+> >intra frames are also known as inter frames, which is
+> >definitely not the case
+>=20
+>  I do not think the text says this.  Here is the sentence:
+>=20
+>      Current coding schemes such as H.261 [2], H.263 [3], MPEG-1, 2,4
+>       [5], H.264 [6] can encode video pictures as reference=20
+> frames, also
+>       known as intra frames or predicted frames, also known as inter
+>       frames.

It seems like what we have is a scoping problem.  To what set of words
does the first "also known as" phrase apply?  To me, when you said "A,
also known as B or C, also known as D", I interpreted this as saying
that A, B, C, and D are all synonyms (with the first "also known as"
applying to the "B or C").  Apparently that's not what you meant.

Also, the way the term "reference frame" is used in this document may
have contributed to my confusion.  As I noted in my original message,
that term is used differently here than it is in some other video coding
discussions.

Best Regards,

Gary Sullivan


+> -----Original Message-----
+> From: Andrea Basso [mailto:Andrea_Basso@nmss.com]=20
+> Sent: Wednesday, July 09, 2003 12:17 PM
+> To: Gary Sullivan
+> Cc: Andrea Basso; avt@ietf.org; casner@acm.org;=20
+> csp@csperkins.org; jo@tzi.uni-bremen.de;=20
+> magnus.westerlund@ericsson.com; mmusic@ietf.org;=20
+> nismail@cisco.com; Orit Levin; Roni Even
+> Subject: RE: [AVT] Requirements draft on video control
+>=20
+>=20
+>=20
+> Gary,
+> here are some comments.
+>=20
+>=20
+> > Some of it seems like it could use a bit of updating in that regard
+> > (e.g., videofastupdateGOB).
+> > Draft Recommendation H.241 is one good place to look for such
+> > information.
+>=20
+> H.241 is very focused on H.264 and in very early stages in addressing
+> Commands and Indications applicable to all codecs.
+>  If you look at the H.241 draft section 6 says:
+>=20
+>=20
+>=20
+> 6      Commands and Indications
+>=20
+>=20
+>=20
+> 6.1    C&I applicable to all video codecs
+>=20
+> For further study.
+>=20
+>=20
+> We plan to monitor the evolution the H.241 and to update the=20
+> ID accordingly
+> if needed.
+>=20
+>=20
+> >intra frames are also known as inter frames, which is
+> >definitely not the case
+>=20
+>  I do not think the text says this.  Here is the sentence:
+>=20
+>      Current coding schemes such as H.261 [2], H.263 [3], MPEG-1, 2,4
+>       [5], H.264 [6] can encode video pictures as reference=20
+> frames, also
+>       known as intra frames or predicted frames, also known as inter
+>       frames.
+>=20
+>=20
+> >The terms "picture" and "frame" are also kind of
+> >mixed up.
+>=20
+> For progressive video, a picture is identical to a frame.=20
+> The difference
+> appears for  interlaced video only.
+> The new text will use the term  'picture' only.
+>=20
+>=20
+> >The characterization of a "slice" or "GOB" as being a=20
+> "stripe" is not
+> >really quite right.
+>=20
+> The text does not say "image stripe" says "stripe" and in=20
+> particular it
+> does not imply that such stripe goes border-to-border in the=20
+> image (i.e.
+> can cover a partial macroblock row) nor that is should only include a
+> single  macroblock row (i.e., can cover multiple partial=20
+> rows as in H.261).
+> Clarifications will be addressed in the new version of the draft.
+>=20
+>=20
+> -A
+>=20
+> Andrea Basso
+> NMS Communications
+> (732) 936-2118
+>=20
+>=20
+>=20
+>                                                             =20
+>                                                                  =20
+>                       "Gary Sullivan"                       =20
+>                                                                  =20
+>                       <garysull@windows.mic        To:      =20
+> "Andrea Basso" <Andrea_Basso@nmss.com>, <avt@ietf.org>,          =20
+>                       rosoft.com>                  =20
+> <avt-admin@ietf.org>, <mmusic@ietf.org>                     =20
+>              =20
+>                                                    cc:      =20
+> <casner@acm.org>, <jo@tzi.uni-bremen.de>,                        =20
+>                       07/08/2003 08:03 PM          =20
+> <magnus.westerlund@ericsson.com>, <csp@csperkins.org>, "Orit=20
+> Levin"       =20
+>                                                    =20
+> <orit@radvision.com>, "Roni Even" <roni.even@polycom.co.il>,=20
+>              =20
+>                                                    =20
+> <nismail@cisco.com>                                         =20
+>              =20
+>                                                    Subject: =20
+> RE: [AVT] Requirements draft on video control                    =20
+>                                                             =20
+>                                                                  =20
+>=20
+>=20
+>=20
+>=20
+>=20
+> Much of the content of this ID will look familiar to those with
+> experience with ITU-T systems control specifications (such as H.245).
+>=20
+> Some of it seems like it could use a bit of updating in that regard
+> (e.g., videofastupdateGOB).
+> Draft Recommendation H.241 is one good place to look for such
+> information.
+>=20
+> Section 3 is a bit confused in its terminology.  For example=20
+> it seems to
+> say that intra frames are also known as inter frames, which is
+> definitely not the case.  It mixes the terms "predictive" and
+> "prediction" and seems to consider a "reference" frame to be=20
+> synonymous
+> with an intra frame.  The terms "picture" and "frame" are=20
+> also kind of
+> mixed up.
+>=20
+> A more consistent terminology would be as follows:
+>=20
+> An intra picture is a picture that can be decoded without=20
+> first decoding
+> any other picture.
+>=20
+> A predicted (or non-intra) picture is a picture that may require data
+> from one or more previously-decoded pictures in order to be properly
+> decoded.
+>=20
+> A reference picture is a picture that is stored in the=20
+> decoder for use
+> as a reference in the decoding process of some subsequent=20
+> picture in the
+> bitstream.
+>=20
+> A non-reference picture is a picture that is not used as a=20
+> reference for
+> the decoding process of any other picture in the bitstream.
+>=20
+> The concepts of intra versus non-Intra and reference versus
+> non-reference are independent.  A particular picture can (in=20
+> general) be
+> any one of the four types (intra reference, non-intra=20
+> reference, intra
+> non-reference, non-intra non-reference).
+>=20
+> A "fast update", also known as an "instantaneous decoder refresh",
+> involves sending an intra picture and thereafter refraining=20
+> from using
+> any picture sent prior to that intra picture as a reference for the
+> decoding process of any subsequent picture sent in the stream.
+>=20
+> In particular it is worth noting that some concepts that=20
+> applied to some
+> older codecs are not really valid in general and should be avoided
+> (e.g., that a non-reference picture is called a B picture or that an
+> intra picture is the same thing as a fast update or that an intra
+> picture is always a reference picture).
+>=20
+> The characterization of a "slice" or "GOB" as being a "stripe" is not
+> really quite right.  For example, a single slice in MPEG-1,=20
+> H.263 Annex
+> O, or MPEG-4 can consist of a few macroblocks at the right=20
+> edge of one
+> row and a few macroblocks at the left edge of the next row (i.e., two
+> distinct rectangular regions quite far apart from each other in the
+> picture can be in a single slice).  A GOB in H.261 CIF=20
+> consists of THREE
+> partial rows of macroblocks extending only halfway across a picture.
+> And in H.264/AVC, a slice can be ANY collection of=20
+> macroblocks scattered
+> around absolutely anywhere in the picture (by use of a=20
+> feature known as
+> a "slice group map").
+>=20
+> In regard to section 5.1 VideoFreezePicture, I believe H.262=20
+> (also known
+> as MPEG-2 Video) also supports a freeze picture release=20
+> within the ITU-T
+> extension.
+>=20
+> It is also possible to obtain random access recovery without a fast
+> update.  This is sometimes called "gradual decoder refresh".  See for
+> example the recovery point SEI message in H.264/AVC.
+>=20
+> Best Regards,
+>=20
+> Gary Sullivan
+>=20
+>=20
+> +> -----Original Message-----
+> +> From: avt-admin@ietf.org [mailto:avt-admin@ietf.org] On
+> +> Behalf Of Andrea Basso
+> +> Sent: Monday, July 07, 2003 2:34 PM
+> +> To: avt@ietf.org; avt-admin@ietf.org; mmusic@ietf.org
+> +> Cc: casner@acm.org; jo@tzi.uni-bremen.de;
+> +> magnus.westerlund@ericsson.com; csp@csperkins.org; Orit
+> +> Levin; Roni Even; nismail@cisco.com
+> +> Subject: [AVT] Requirements draft on video control
+> +>
+> +>
+> +> Attached you will find draft-basso-avt-videoconreq-00.txt
+> +>
+> +> Abstract:
+> +>
+> +> A variety of communication services such as video
+> +> conferencing and video messaging rely on the capability of video
+> +> encoders and decoders to exchange control commands. This document
+> +> outlines this set of commands as well as the requirements for
+> +> their transport.
+> +>
+> +> Comments are more than welcome.
+> +>
+> +>
+> +>
+> +>
+> +>
+> +> Andrea Basso
+> +> NMS Communications
+> +> (732) 936-2118
+> +> (See attached file: draft-basso-avt-videoconreq-00.zip)
+> +>
+>=20
+>=20
+>=20
+>=20
+>=20
+>=20

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: Stephen Casner <casner@acm.org>
To: Aaron Colwell <acolwell@real.com>
cc: avt@ietf.org
Subject: Re: [AVT] draft-ietf-avt-rtcp-bw-05.txt question
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Aaron,

> In the case where b=RR is specified that would mean that no sender reports
> or BYE packets could be sent. Is that allowed? In the A/V case the client
> would not be able to sync the streams because it would never get their
> CNAME. I suppose that is the server's fault for not specifying a value
> for the sender reports. I guess this brings up another question. Does it
> make sense for the RS value to be 0?

We are talking here about a case where b=RR is specified to be > 5%,
and b=RS is not specified, so RS would be taken to be zero.  However,
your interpretation of the effect of RS=0 is incorrect.  First, both
RS and RR apply to all RTCP packet types that a participant might
send.  Second, as Magnus and I have been discussing on the list, the
"1/4 rule" still applies when RS and/or RR are/is specified, except
that we have corrected it to be the "RS/(RS+RR) rule".  If RS=0, that
simply means that senders do not have any RTCP bandwidth dedicated to
them, instead they share RTCP bandwith equally with all participants.

I have clarified this in the draft now, during the final stages with
the RFC editor.

                                                        -- Steve



_______________________________________________
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https://www1.ietf.org/mailman/listinfo/avt



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From: Aaron Colwell <acolwell@real.com>
To: Stephen Casner <casner@acm.org>
cc: avt@ietf.org
Subject: Re: [AVT] draft-ietf-avt-rtcp-bw-05.txt question
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That makes sense to me. Thanks for the clarifications.

Aaron

On Wed, 9 Jul 2003, Stephen Casner wrote:

> Aaron,
>
> > In the case where b=RR is specified that would mean that no sender reports
> > or BYE packets could be sent. Is that allowed? In the A/V case the client
> > would not be able to sync the streams because it would never get their
> > CNAME. I suppose that is the server's fault for not specifying a value
> > for the sender reports. I guess this brings up another question. Does it
> > make sense for the RS value to be 0?
>
> We are talking here about a case where b=RR is specified to be > 5%,
> and b=RS is not specified, so RS would be taken to be zero.  However,
> your interpretation of the effect of RS=0 is incorrect.  First, both
> RS and RR apply to all RTCP packet types that a participant might
> send.  Second, as Magnus and I have been discussing on the list, the
> "1/4 rule" still applies when RS and/or RR are/is specified, except
> that we have corrected it to be the "RS/(RS+RR) rule".  If RS=0, that
> simply means that senders do not have any RTCP bandwidth dedicated to
> them, instead they share RTCP bandwith equally with all participants.
>
> I have clarified this in the draft now, during the final stages with
> the RFC editor.
>
>                                                         -- Steve
>
>
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
>

_______________________________________________
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avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: John Lazzaro <lazzaro@CS.Berkeley.EDU>
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Subject: Re: [AVT] Vorbis RTP issues list for Vienna 
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> Aaron Colwell <acolwell@real.com> writes:
> I agree with Ross on this. It seems that these codebooks are causing too
> many problems that aren't worth dealing with. People have been suggesting
> getting the codebooks via HTTP, alternate TCP links, and all sorts of
> other methods. If your going to go that far why not just use HTTP to
> deliver the data. That way you are guaranteed to get everything in the
> right order.

I think its just an instance of a general problem -- transport of
hybrid streams with two components:

  -- A steady flow of media information that can handle the 
     occassional packet loss episode gracefully and without
     retransmission

  -- Occassional chunks of data, which can be sent well ahead of
     its need, which needs reliability, and which is synchronous
     to the media flow.

So, I think its a good idea for RTP to offer solutions that work
in this domain.  There are some studio-oriented uses of RTP MIDI
which fit into the model, and my motivation for doing 
draft-lazzaro-avt-rtp-framing-contrans-01 is to support those in
a general way, so that when things like Vorbis codebooks come
along, RTP has a way to support them.  I really don't think that
posing the question as all-or-none -- RTP streaming or Shoutcast-esqe
pseudo-streaming -- benefits AVT, it will just spawn more HTTP
streaming implementations.

That said:

> The Vorbis folks
> should come up with a "streamable" profile that encodes the audio using
> predefined codebooks.

This is a good option too -- my guess is that most Vorbis setups
will only need one codebook for the life of the stream, and that
need will be known in advance.  A URL in the session description
to support HTTP download of that one codebook would be a way to
go, with perhaps a known set of "standard codebooks" codable by
speak tokens or by GUIDs for the MIME HTTP codebook object.

-------------------------------------------------------------------------
John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley
lazzaro [at] cs [dot] berkeley [dot] edu     www.cs.berkeley.edu/~lazzaro
-------------------------------------------------------------------------

_______________________________________________
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To: Colin Perkins <csp@csperkins.org>
Cc: avt@ietf.org
Subject: Re: [AVT] draft-herlein-speex-rtp-profile-01
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Hi Colin,

Thanks for the feedback, comments in-line:


Quoting Colin Perkins <csp@csperkins.org>:

> --> philkerr@elec.gla.ac.uk writes:
> > Dear all,
> > 
> > An -01 update to draft-herlein-speex-rtp-profile is now available:
> > 
> > http://www.ietf.org/internet-drafts/draft-herlein-speex-rtp-profile-01.txt
> > 
> > Comments and feedback welcomed.
> 
> I have a number of questions and comments:
> 
> - It's not appropriate for the draft to claim that the Speex codec is free
>   from patents and/or royalties.  If you wish to make an IPR statement, it
>   should be separately submitted to the IESG Secretary for inclusion on the
>   IPR-claims page the IETF maintains. See also section 10 of RFC 2026.

Submitting an IPR statement should be no problem.

> 
> - The draft would benefit from a little more introduction to Speex at the
>   beginning.

Agreed.  This section can be expanded.

> 
> - The definition of the Padding bit in the RTP header is unusual. It may be
>   better to state that padding MAY be used in the usual RTP manner, unless
>   there is a specific reason why a different padding scheme is needed?

Changing the Padding bit to MAY should be fine.

> 
> - The use of the M bit in the RTP header to signify comfort noise is also
>   unusual. This is not necessarily inappropriate, but it is important to
>   consider if the advantages of this use outweigh the costs of being
>   different to other audio payload formats (which use the M bit to signal
>   the first packet after a silent period).

Its use was suggested by Henning Schulzrinne to the other Speex draft authors
and I've based it on their notes and Section 4.1 of RFC 1890.  If there is some
discrepency in its use, as highlighted in the feedback by  Michael Ramalho, then
we may need to look at this section again to determine best practice.

> 
> - The definition of the timestamp says "Speex uses 20 msec frames and a
>   variable sampling rate clock". Does this mean that the sample rate can
>   change at anytime within a session? Or is the sample rate chosen from
>   one of a number of values, and then fixed within the session?

There is a fixed range of values for the sample rate but the rate may change mid
session.  

> 
> - Section 3.2 is unclear if the bit rate and sample rate are coupled. Can
>   Speex vary the compression ratio to change the bit rate without changing
>   the sampling rate?

I will need to check this just to make sure.

> 
> - In Section 3.2, it is important to be clear that the padding needed to
>   ensure that the payload is octet aligned is independent of the padding
>   at the RTP level (i.e. the P bit in the RTP header).

This can be made clearer.

> 
> - The examples in Section 3.3 and 3.4 might be clearer if they show the
>   actual length of the Speex data frames. An example showing two Speex
>   frames in an RTP packet, with the first Speex frame being non-octet
>   aligned, would be useful (to make it clear how to handle padding in
>   this case).
> 
> - Section 4 "MIME registration of Speex" should be renamed to "IANA 
>   considerations", and needs an introductory sentence saying something 
>   like "The following MIME type and associated RTP payload format are 
>   to be registered".
> 
> - The usual way in which a MIME type is registered only for particular
>   use is to define the use in the "Encoding considerations" section of
>   the MIME registration form. In this case, you might say something like:
> 
>         This type is defined only for transfer within RTP as specified in
>         Section 4 of RFC XXXX. For file storage, or for transmission over
>         lossless channels (such as TCP), Speex data is transported within
>         an Ogg bitstream [RFC 3534].
> 
>   This would replace the introductory sentence and last paragraph of the
>   current section 4. If there is a need to register parameters of the Ogg
>   bitstream, that should be done as a separate subsection under section 4.

The text can be amended as suggested.

> 
> - The "a=fmtp:" parameters from Section 5 should be defined in Section 4,
>   as Optional Parameters to the audio/speex type, and section 5 should be
>   an explanation of how the parameters are mapped onto SDP.

Agreed, this will be changed.

> 
> - The "ebw" parameter may be redundant, since the clock rate is signalled
>   in the "a=rtpmap:" line?

I'll need to check this, but if it is redundant it can be removed.

> 
> - It may be clearer to separate out the explanation of how parameters are
>   mapped to SDP from the description of how the SDP is used with the offer/
>   answer model (as two subsections). There are uses of SDP that don't use
>   offer/answer.

Agreed, this section will be split into two.

> 
> - Section 6 (use with H.323) is not appropriate in an IETF document. You
>   can include a reference to an ITU - or other - document explaining how
>   to use Speex with H.323, but the IETF can't mandate use of H.245
>   parameters.

Agreed.  I presume we can also link back to a Speex document if further
explanation is required, with the usual proviso about linking to URLs as
mentioned in the ID-nits.

> 
> - Section 10: RFCs are published using strict US ASCII. If there is an
>   acceptable transliteration of the non-ASCII characters in Jean-Marc
>   Valin's address, it will save a "discussion" with the RFC Editor. See
>   draft-rfc-editor-rfc2223bis-06.txt.

I will check with Jean-Marc to see if his address can be written in US ASCII.

> 
> - 
> Colin Perkins
> csp@csperkins.org
> 

Thanks

Phil


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Date: Thu, 10 Jul 2003 00:05:04 +0100
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Cc: avt@ietf.org
Subject: Re: [AVT] Vorbis RTP issues list for Vienna
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Hi Magnus,


Quoting Magnus Westerlund <magnus.westerlund@ericsson.com>:

> Hi Phil,
> 
> Here are a few more comments and some pointers worth checking out.
> 
> 
> philkerr@elec.gla.ac.uk wrote:
> > Quoting Magnus Westerlund <magnus.westerlund@ericsson.com>:
> 
> 
> >>7. Section 2.3: Third paragraph, page 5: The limitation that a Vorbis 
> >>packet shall not be larger than 64k bytes. What about IPv6 Jumbograms 
> >>and the fragmentation process? It seem possible to have larger packets 
> >>although it would be a bad idea.
> > 
> > 
> > The codebooks can be far greater than 64k in size but a limit was imposed
> to
> > allow it to be transported inside a RTCP packet and to reduce the delay a
> client
> > may have in waiting for a large codebook to be received.   Perhaps a note
> > detailing that transport over IPv6 allows greater packet size would cover
> this
> > scenario.
> > 
> 
> I think that we need to clear consider how to get these codebooks 
> transported. A larger than 64kb codebook should neither be transported 
> in SDP or in RTCP. I think that using HTTP can clearly be an 
> alternative, and only provide the URI to the codebook.
> 
> I wouldn't try to transport a codebook over RTCP that results in a RTCP 
> packet many time the MTU due to the high number of IP fragments. Further 
> such a packet will severely reduce the RTCP report rate in both directions.

Agreed.  The size should be less than 64k, even if IPv6 can accomodate it and in
normal use they are typically under 15k.  Having very large codebooks wizzing
around is less than optimal.  

> 
> > 
> >>8. Section 2.3, fifth paragraph: Why it the length field only a single 
> >>byte? In some use cases it seems that having aggregation only for 
> >>packets smaller than 256 bytes may be limiting. Have you studied what 
> >>the frequency different packets are created. If you have a lot of 
> >>packets in the size of 257-500 bytes they may be advantageous to be able 
> >>to aggregate.
> > 
> > 
> > The length field size was specified in the original
> draft-moffitt-vorbis-rtp I-D
> > and it's been kept as it seems a fair limit, losing a packet with a 64 or
> 128
> > Vorbis frames will cause a greater skip than 32.  Aggregating packets
> larger
> > than 256 bytes is allowed so your scenario is catered for.  When you read
> this
> > section did you get the impression that this was not allowed?  If so it can
> be
> > altered to make it clearer.
> 
> The first indication is the second paragraph in section 3: "Any Vorbis 
> packet that is larger than 256 octets and less than the path-MTU MUST be 
> placed in a RTP packet by itself.". Secondly the payload format does not 
> describe a mechanism to allow it.

I have a feeling that this limit is artifically low and I'll check the to see if
it can be raised to allow larger Vorbis frames to be aggregated as you outlined
in your previous feedback.

> 
> > 
> >>10. Section 2.3: How does one perform RTP TS recovery for the individual 
> >>Vorbis packets in an aggregated packet? Are the RTP TS not needed or are 
> >>there some implicit method. Please clarify.
> > 
> > 
> > This needs further investigation.
> 
> This is probably one of the most important things as most decoders and 
> error recovery mechanism needs to receive the frames/packets in timing 
> order plus with detection of lost frames.

Agreed.

> 
> > 
> >>21. Section 4.1, how to utilize the subtype in this APP packet is not 
> >>defined.
> >>
> >>22. Section 4.1 Why have have flags for the different parts when you are 
> >>required to send all of the fields?
> > 
> > 
> > All of the header information is required for the initial setup of the
> clients
> > decoder but the codebook and metadata may change mid-session.  The
> meta-data
> > will probably be the most dynamic (track number/title/artist changing) and
> the
> > flags are used to indicate the payload sub-type.  The VORB RTCP packet can
> > contain one, or all of the sub headers components.
> 
> That was not how I read the text. See section 4.1 paragraph 2 on page 
> 11. There is a MUST there in the third sentence that I interpreted as 
> requiring all as needed.

I'll review this section and alter to remove any ambiguity.

> 
> I think that sending this meta information in RTCP can be appropriate, 
> or in the packet stream if RTP retransmission is used. However the 
> codebooks will most probably need special treatment, unless they are no 
> bigger than a 1 kb.
> 
> > 
> >>29. Section 5: The usage of "u=", I don't think it is appropriate to use 
> >>"u=" to include an URI to the codebooks and other stream information.
> >>
> >>30. Section 5: The usage of "c=", The Vorbis payload format can't 
> >>redefine the usage of "c=" field or require it to be used. SIP use the 
> >>field to declare the receiver address, while RTSP does not use it at all.
> >>
> >>31. Section 5.1, I think that all Vorbis specific SDP attributes should 
> >>be sent in the a=fmtp line and be declared as required MIME parameters 
> >>if they always must be present. Why are there no parameter for the 
> >>codebook itself? Also an URI to this information could be included here. 
> >>However the implications that a client must follow a link to fetch this 
> >>information must be further explained.
> > 
> > 
> > I'll look into this further.
> 
> I would suggest that you take a look at the MIME and SDP sections of the 
>   H264 payload format (draft-ietf-avt-h264-02.txt) or the AMR payload 
> format (RFC 3267).

Will do.

Cheers

Phil

> 
> Cheers
> 
> 
> Magnus Westerlund
> 
> Multimedia Technologies, Ericsson Research EAB/TVA/A
> ----------------------------------------------------------------------
> Ericsson AB                | Phone +46 8 4048287
> Torshamsgatan 23           | Fax   +46 8 7575550
> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> 
> 




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Date: Wed, 09 Jul 2003 16:25:05 -0700
To: "Gary Sullivan" <garysull@windows.microsoft.com>
From: Nermeen Ismail <nismail@cisco.com>
Subject: RE: [AVT] Requirements draft on video control
Cc: "Andrea Basso" <Andrea_Basso@nmss.com>, <avt@ietf.org>, <casner@acm.org>,
        <csp@csperkins.org>, <jo@tzi.uni-bremen.de>,
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Gary,

Thanks for the comments. Definitely we need to add consistency in the terms 
we are using in the draft.

>Andrea et al,
>
>Perhaps I was having some trouble understanding the purpose of this ID.
>I thought it was some sort of draft of a formal specification, rather
>than just an expression of needs to be taken as input by the group.  So
>I criticized the language more heavily than necessary.


So now that this is clarified we need to discuss the requirements that are 
specified in that draft
and see if people think that these are legitimate requirements that need to 
be addressed or not.

nermeen




>In regard to the "intra frames" = "inter frames" mix-up:
>
>+> >intra frames are also known as inter frames, which is
>+> >definitely not the case
>+>
>+>  I do not think the text says this.  Here is the sentence:
>+>
>+>      Current coding schemes such as H.261 [2], H.263 [3], MPEG-1, 2,4
>+>       [5], H.264 [6] can encode video pictures as reference
>+> frames, also
>+>       known as intra frames or predicted frames, also known as inter
>+>       frames.
>
>It seems like what we have is a scoping problem.  To what set of words
>does the first "also known as" phrase apply?  To me, when you said "A,
>also known as B or C, also known as D", I interpreted this as saying
>that A, B, C, and D are all synonyms (with the first "also known as"
>applying to the "B or C").  Apparently that's not what you meant.
>
>Also, the way the term "reference frame" is used in this document may
>have contributed to my confusion.  As I noted in my original message,
>that term is used differently here than it is in some other video coding
>discussions.
>
>Best Regards,
>
>Gary Sullivan
>
>
>+> -----Original Message-----
>+> From: Andrea Basso [mailto:Andrea_Basso@nmss.com]
>+> Sent: Wednesday, July 09, 2003 12:17 PM
>+> To: Gary Sullivan
>+> Cc: Andrea Basso; avt@ietf.org; casner@acm.org;
>+> csp@csperkins.org; jo@tzi.uni-bremen.de;
>+> magnus.westerlund@ericsson.com; mmusic@ietf.org;
>+> nismail@cisco.com; Orit Levin; Roni Even
>+> Subject: RE: [AVT] Requirements draft on video control
>+>
>+>
>+>
>+> Gary,
>+> here are some comments.
>+>
>+>
>+> > Some of it seems like it could use a bit of updating in that regard
>+> > (e.g., videofastupdateGOB).
>+> > Draft Recommendation H.241 is one good place to look for such
>+> > information.
>+>
>+> H.241 is very focused on H.264 and in very early stages in addressing
>+> Commands and Indications applicable to all codecs.
>+>  If you look at the H.241 draft section 6 says:
>+>
>+>
>+>
>+> 6      Commands and Indications
>+>
>+>
>+>
>+> 6.1    C&I applicable to all video codecs
>+>
>+> For further study.
>+>
>+>
>+> We plan to monitor the evolution the H.241 and to update the
>+> ID accordingly
>+> if needed.
>+>
>+>
>+> >intra frames are also known as inter frames, which is
>+> >definitely not the case
>+>
>+>  I do not think the text says this.  Here is the sentence:
>+>
>+>      Current coding schemes such as H.261 [2], H.263 [3], MPEG-1, 2,4
>+>       [5], H.264 [6] can encode video pictures as reference
>+> frames, also
>+>       known as intra frames or predicted frames, also known as inter
>+>       frames.
>+>
>+>
>+> >The terms "picture" and "frame" are also kind of
>+> >mixed up.
>+>
>+> For progressive video, a picture is identical to a frame.
>+> The difference
>+> appears for  interlaced video only.
>+> The new text will use the term  'picture' only.
>+>
>+>
>+> >The characterization of a "slice" or "GOB" as being a
>+> "stripe" is not
>+> >really quite right.
>+>
>+> The text does not say "image stripe" says "stripe" and in
>+> particular it
>+> does not imply that such stripe goes border-to-border in the
>+> image (i.e.
>+> can cover a partial macroblock row) nor that is should only include a
>+> single  macroblock row (i.e., can cover multiple partial
>+> rows as in H.261).
>+> Clarifications will be addressed in the new version of the draft.
>+>
>+>
>+> -A
>+>
>+> Andrea Basso
>+> NMS Communications
>+> (732) 936-2118
>+>
>+>
>+>
>+>
>+>
>+>                       "Gary Sullivan"
>+>
>+>                       <garysull@windows.mic        To:
>+> "Andrea Basso" <Andrea_Basso@nmss.com>, <avt@ietf.org>,
>+>                       rosoft.com>
>+> <avt-admin@ietf.org>, <mmusic@ietf.org>
>+>
>+>                                                    cc:
>+> <casner@acm.org>, <jo@tzi.uni-bremen.de>,
>+>                       07/08/2003 08:03 PM
>+> <magnus.westerlund@ericsson.com>, <csp@csperkins.org>, "Orit
>+> Levin"
>+>
>+> <orit@radvision.com>, "Roni Even" <roni.even@polycom.co.il>,
>+>
>+>
>+> <nismail@cisco.com>
>+>
>+>                                                    Subject:
>+> RE: [AVT] Requirements draft on video control
>+>
>+>
>+>
>+>
>+>
>+>
>+>
>+> Much of the content of this ID will look familiar to those with
>+> experience with ITU-T systems control specifications (such as H.245).
>+>
>+> Some of it seems like it could use a bit of updating in that regard
>+> (e.g., videofastupdateGOB).
>+> Draft Recommendation H.241 is one good place to look for such
>+> information.
>+>
>+> Section 3 is a bit confused in its terminology.  For example
>+> it seems to
>+> say that intra frames are also known as inter frames, which is
>+> definitely not the case.  It mixes the terms "predictive" and
>+> "prediction" and seems to consider a "reference" frame to be
>+> synonymous
>+> with an intra frame.  The terms "picture" and "frame" are
>+> also kind of
>+> mixed up.
>+>
>+> A more consistent terminology would be as follows:
>+>
>+> An intra picture is a picture that can be decoded without
>+> first decoding
>+> any other picture.
>+>
>+> A predicted (or non-intra) picture is a picture that may require data
>+> from one or more previously-decoded pictures in order to be properly
>+> decoded.
>+>
>+> A reference picture is a picture that is stored in the
>+> decoder for use
>+> as a reference in the decoding process of some subsequent
>+> picture in the
>+> bitstream.
>+>
>+> A non-reference picture is a picture that is not used as a
>+> reference for
>+> the decoding process of any other picture in the bitstream.
>+>
>+> The concepts of intra versus non-Intra and reference versus
>+> non-reference are independent.  A particular picture can (in
>+> general) be
>+> any one of the four types (intra reference, non-intra
>+> reference, intra
>+> non-reference, non-intra non-reference).
>+>
>+> A "fast update", also known as an "instantaneous decoder refresh",
>+> involves sending an intra picture and thereafter refraining
>+> from using
>+> any picture sent prior to that intra picture as a reference for the
>+> decoding process of any subsequent picture sent in the stream.
>+>
>+> In particular it is worth noting that some concepts that
>+> applied to some
>+> older codecs are not really valid in general and should be avoided
>+> (e.g., that a non-reference picture is called a B picture or that an
>+> intra picture is the same thing as a fast update or that an intra
>+> picture is always a reference picture).
>+>
>+> The characterization of a "slice" or "GOB" as being a "stripe" is not
>+> really quite right.  For example, a single slice in MPEG-1,
>+> H.263 Annex
>+> O, or MPEG-4 can consist of a few macroblocks at the right
>+> edge of one
>+> row and a few macroblocks at the left edge of the next row (i.e., two
>+> distinct rectangular regions quite far apart from each other in the
>+> picture can be in a single slice).  A GOB in H.261 CIF
>+> consists of THREE
>+> partial rows of macroblocks extending only halfway across a picture.
>+> And in H.264/AVC, a slice can be ANY collection of
>+> macroblocks scattered
>+> around absolutely anywhere in the picture (by use of a
>+> feature known as
>+> a "slice group map").
>+>
>+> In regard to section 5.1 VideoFreezePicture, I believe H.262
>+> (also known
>+> as MPEG-2 Video) also supports a freeze picture release
>+> within the ITU-T
>+> extension.
>+>
>+> It is also possible to obtain random access recovery without a fast
>+> update.  This is sometimes called "gradual decoder refresh".  See for
>+> example the recovery point SEI message in H.264/AVC.
>+>
>+> Best Regards,
>+>
>+> Gary Sullivan
>+>
>+>
>+> +> -----Original Message-----
>+> +> From: avt-admin@ietf.org [mailto:avt-admin@ietf.org] On
>+> +> Behalf Of Andrea Basso
>+> +> Sent: Monday, July 07, 2003 2:34 PM
>+> +> To: avt@ietf.org; avt-admin@ietf.org; mmusic@ietf.org
>+> +> Cc: casner@acm.org; jo@tzi.uni-bremen.de;
>+> +> magnus.westerlund@ericsson.com; csp@csperkins.org; Orit
>+> +> Levin; Roni Even; nismail@cisco.com
>+> +> Subject: [AVT] Requirements draft on video control
>+> +>
>+> +>
>+> +> Attached you will find draft-basso-avt-videoconreq-00.txt
>+> +>
>+> +> Abstract:
>+> +>
>+> +> A variety of communication services such as video
>+> +> conferencing and video messaging rely on the capability of video
>+> +> encoders and decoders to exchange control commands. This document
>+> +> outlines this set of commands as well as the requirements for
>+> +> their transport.
>+> +>
>+> +> Comments are more than welcome.
>+> +>
>+> +>
>+> +>
>+> +>
>+> +>
>+> +> Andrea Basso
>+> +> NMS Communications
>+> +> (732) 936-2118
>+> +> (See attached file: draft-basso-avt-videoconreq-00.zip)
>+> +>
>+>
>+>
>+>
>+>
>+>
>+>
>
>_______________________________________________
>Audio/Video Transport Working Group
>avt@ietf.org
>https://www1.ietf.org/mailman/listinfo/avt


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Thu, 10 Jul 2003 00:25:35 +0100
To: John Lazzaro <lazzaro@CS.Berkeley.EDU>
Cc: avt@ietf.org
Subject: Re: [AVT] Vorbis RTP issues list for Vienna 
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Hi John / Aaron / Ross,

John, yes you are right when you state that the codebooks can change quite
frequently, between tracks, and this is a function of the codec which the RTP
stream must cope with.  Your draft-lazzaro-avt-rtp-framing-contrans-01 looks
like it could offer a good solution and I should track its development more
closely. 

Ross / Aaron, the number of codebooks that can be found 'in the wild' is a
reasonably large, but not infinite, number and the caching mechanism is meant to
reduce the need to retransmit them each time.  Having a general purpose
streaming codebook is a reasonable suggestion, John and I discussed when we met,
but it is uncertain if a fully generic one can be produced which will cater for
all scenarios.

Regards

Phil

Quoting John Lazzaro <lazzaro@CS.Berkeley.EDU>:

> 
> > Aaron Colwell <acolwell@real.com> writes:
> > I agree with Ross on this. It seems that these codebooks are causing too
> > many problems that aren't worth dealing with. People have been suggesting
> > getting the codebooks via HTTP, alternate TCP links, and all sorts of
> > other methods. If your going to go that far why not just use HTTP to
> > deliver the data. That way you are guaranteed to get everything in the
> > right order.
> 
> I think its just an instance of a general problem -- transport of
> hybrid streams with two components:
> 
>   -- A steady flow of media information that can handle the 
>      occassional packet loss episode gracefully and without
>      retransmission
> 
>   -- Occassional chunks of data, which can be sent well ahead of
>      its need, which needs reliability, and which is synchronous
>      to the media flow.
> 
> So, I think its a good idea for RTP to offer solutions that work
> in this domain.  There are some studio-oriented uses of RTP MIDI
> which fit into the model, and my motivation for doing 
> draft-lazzaro-avt-rtp-framing-contrans-01 is to support those in
> a general way, so that when things like Vorbis codebooks come
> along, RTP has a way to support them.  I really don't think that
> posing the question as all-or-none -- RTP streaming or Shoutcast-esqe
> pseudo-streaming -- benefits AVT, it will just spawn more HTTP
> streaming implementations.
> 
> That said:
> 
> > The Vorbis folks
> > should come up with a "streamable" profile that encodes the audio using
> > predefined codebooks.
> 
> This is a good option too -- my guess is that most Vorbis setups
> will only need one codebook for the life of the stream, and that
> need will be known in advance.  A URL in the session description
> to support HTTP download of that one codebook would be a way to
> go, with perhaps a known set of "standard codebooks" codable by
> speak tokens or by GUIDs for the MIME HTTP codebook object.
> 
> -------------------------------------------------------------------------
> John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley
> lazzaro [at] cs [dot] berkeley [dot] edu     www.cs.berkeley.edu/~lazzaro
> -------------------------------------------------------------------------
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
> 




-------------------------------------------------
This mail sent through IMP: http://horde.org/imp/

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Wed Jul  9 19:46:03 2003
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From: Aaron Colwell <acolwell@real.com>
To: John Lazzaro <lazzaro@CS.Berkeley.EDU>
cc: avt@ietf.org
Subject: Re: [AVT] Vorbis RTP issues list for Vienna 
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On Wed, 9 Jul 2003, John Lazzaro wrote:

>
> > Aaron Colwell <acolwell@real.com> writes:
> > I agree with Ross on this. It seems that these codebooks are causing too
> > many problems that aren't worth dealing with. People have been suggesting
> > getting the codebooks via HTTP, alternate TCP links, and all sorts of
> > other methods. If your going to go that far why not just use HTTP to
> > deliver the data. That way you are guaranteed to get everything in the
> > right order.
>
> I think its just an instance of a general problem -- transport of
> hybrid streams with two components:
>
>   -- A steady flow of media information that can handle the
>      occassional packet loss episode gracefully and without
>      retransmission
>
>   -- Occassional chunks of data, which can be sent well ahead of
>      its need, which needs reliability, and which is synchronous
>      to the media flow.
>
> So, I think its a good idea for RTP to offer solutions that work
> in this domain.  There are some studio-oriented uses of RTP MIDI
> which fit into the model, and my motivation for doing
> draft-lazzaro-avt-rtp-framing-contrans-01 is to support those in
> a general way, so that when things like Vorbis codebooks come
> along, RTP has a way to support them.  I really don't think that
> posing the question as all-or-none -- RTP streaming or Shoutcast-esqe
> pseudo-streaming -- benefits AVT, it will just spawn more HTTP
> streaming implementations.

When stated in these more general terms I can see your point about how
this scenario could be useful. Another use may be a key stream for
encrypted content that changes encryption keys often. You would want this
data to be transmitted reliably and ensure that it gets there before the
data that depends on it arrives.

I did not mean to actually suggest the use of HTTP as a viable delivery
mechanism instead of RTP. I don't really want more "Shoutcast-esqe
pseudo-streaming" solutions either. I'll be a little more careful with
comments like this.

I'll take a look at the draft that you mentioned and send you any comments
I may have.

>
> That said:
>
> > The Vorbis folks
> > should come up with a "streamable" profile that encodes the audio using
> > predefined codebooks.
>
> This is a good option too -- my guess is that most Vorbis setups
> will only need one codebook for the life of the stream, and that
> need will be known in advance.  A URL in the session description
> to support HTTP download of that one codebook would be a way to
> go, with perhaps a known set of "standard codebooks" codable by
> speak tokens or by GUIDs for the MIME HTTP codebook object.

Hopefully they will embrace these ideas. I think it could simplify client
implementations a bit. If they used just predefined codebooks or only used
one codebook and specified it via SDP, Vorbis would be just like many
other datatypes and would not require a lot of extra work to support in
existing players.

Aaron

>
> -------------------------------------------------------------------------
> John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley
> lazzaro [at] cs [dot] berkeley [dot] edu     www.cs.berkeley.edu/~lazzaro
> -------------------------------------------------------------------------
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
>

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Wed Jul  9 20:06:07 2003
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Date: Wed, 09 Jul 2003 17:03:03 -0700
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From: Ross Finlayson <finlayson@live.com>
Subject: Re: [AVT] Vorbis RTP issues list for Vienna 
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>Having a general purpose streaming codebook is a reasonable suggestion, 
>John and I discussed when we met, but it is uncertain if a fully generic 
>one can be produced which will cater for all scenarios.

There doesn't have to be just one 'codebook for streaming'.  Having a small 
number of pre-defined ones would be OK also.  E.g., if there were <= 256 
pre-defined 'codebooks for streaming', then you could use a 1-byte field 
within the RTP payload format to indicate which one.  But I suspect that <= 
16 pre-defined codebooks (i.e., 4 bits) would be enough.

         Ross.


_______________________________________________
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https://www1.ietf.org/mailman/listinfo/avt



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From: Aaron Colwell <acolwell@real.com>
To: philkerr@elec.gla.ac.uk
cc: John Lazzaro <lazzaro@CS.Berkeley.EDU>, avt@ietf.org
Subject: Re: [AVT] Vorbis RTP issues list for Vienna 
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On Thu, 10 Jul 2003 philkerr@elec.gla.ac.uk wrote:

> Hi John / Aaron / Ross,
>
> John, yes you are right when you state that the codebooks can change quite
> frequently, between tracks, and this is a function of the codec which the RTP
> stream must cope with.  Your draft-lazzaro-avt-rtp-framing-contrans-01 looks
> like it could offer a good solution and I should track its development more
> closely.
>
> Ross / Aaron, the number of codebooks that can be found 'in the wild' is a
> reasonably large, but not infinite, number and the caching mechanism is
> meant to
> reduce the need to retransmit them each time.  Having a general purpose
> streaming codebook is a reasonable suggestion, John and I discussed
> when we met,
> but it is uncertain if a fully generic one can be produced which will
> cater for all scenarios.

Could a set of codebooks help? Perhaps you could predefine a set of
standard codebooks that are close to ones that are in the wild. For any
codebooks that don't match one of these, a predefined codebook could be
specified, plus some sort of codebook diff. My guess is that if the
predefined codebooks are pretty representative of codebooks in the wild,
the diff should be rather small.

There is still the problem of sending the diff when codebook changes
mid-stream. I can't see a way around this other than either using a
reliable transport for transmitting it or sending the diff's periodically.
The diffs should be much smaller than the actual codebook so periodic
sending of diffs won't be as bad as doing the same with the actual
codebooks. I'll have to think about this a little more.

I think my idea of a codebook translation tool could also help deal with
all the files that are out in the wild. Whoever wants to stream the file
can convert any existing file to one that uses one of the predefined
codebooks.

Aaron

>
> Regards
>
> Phil
>
> Quoting John Lazzaro <lazzaro@CS.Berkeley.EDU>:
>
> >
> > > Aaron Colwell <acolwell@real.com> writes:
> > > I agree with Ross on this. It seems that these codebooks are causing too
> > > many problems that aren't worth dealing with. People have been suggesting
> > > getting the codebooks via HTTP, alternate TCP links, and all sorts of
> > > other methods. If your going to go that far why not just use HTTP to
> > > deliver the data. That way you are guaranteed to get everything in the
> > > right order.
> >
> > I think its just an instance of a general problem -- transport of
> > hybrid streams with two components:
> >
> >   -- A steady flow of media information that can handle the
> >      occassional packet loss episode gracefully and without
> >      retransmission
> >
> >   -- Occassional chunks of data, which can be sent well ahead of
> >      its need, which needs reliability, and which is synchronous
> >      to the media flow.
> >
> > So, I think its a good idea for RTP to offer solutions that work
> > in this domain.  There are some studio-oriented uses of RTP MIDI
> > which fit into the model, and my motivation for doing
> > draft-lazzaro-avt-rtp-framing-contrans-01 is to support those in
> > a general way, so that when things like Vorbis codebooks come
> > along, RTP has a way to support them.  I really don't think that
> > posing the question as all-or-none -- RTP streaming or Shoutcast-esqe
> > pseudo-streaming -- benefits AVT, it will just spawn more HTTP
> > streaming implementations.
> >
> > That said:
> >
> > > The Vorbis folks
> > > should come up with a "streamable" profile that encodes the audio using
> > > predefined codebooks.
> >
> > This is a good option too -- my guess is that most Vorbis setups
> > will only need one codebook for the life of the stream, and that
> > need will be known in advance.  A URL in the session description
> > to support HTTP download of that one codebook would be a way to
> > go, with perhaps a known set of "standard codebooks" codable by
> > speak tokens or by GUIDs for the MIME HTTP codebook object.
> >
> > -------------------------------------------------------------------------
> > John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley
> > lazzaro [at] cs [dot] berkeley [dot] edu     www.cs.berkeley.edu/~lazzaro
> > -------------------------------------------------------------------------
> >
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www1.ietf.org/mailman/listinfo/avt
> >
>
>
>
>
> -------------------------------------------------
> This mail sent through IMP: http://horde.org/imp/
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
>

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: RE: [AVT] Requirements draft on video control
To: "Gary Sullivan" <garysull@windows.microsoft.com>
Cc: "Andrea Basso" <Andrea_Basso@nmss.com>, avt@ietf.org, casner@acm.org,
        csp@csperkins.org, jo@tzi.uni-bremen.de,
        magnus.westerlund@ericsson.com, mmusic@ietf.org, nismail@cisco.com,
        "Orit Levin" <orit@radvision.com>,
        "Roni Even" <roni.even@polycom.co.il>
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From: "Andrea Basso" <Andrea_Basso@nmss.com>
Date: Wed, 9 Jul 2003 15:17:29 -0400
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Gary,
here are some comments.


> Some of it seems like it could use a bit of updating in that regard
> (e.g., videofastupdateGOB).
> Draft Recommendation H.241 is one good place to look for such
> information.

H.241 is very focused on H.264 and in very early stages in addressing
Commands and Indications applicable to all codecs.
 If you look at the H.241 draft section 6 says:



6      Commands and Indications



6.1    C&I applicable to all video codecs

For further study.


We plan to monitor the evolution the H.241 and to update the ID accordingly
if needed.


>intra frames are also known as inter frames, which is
>definitely not the case

 I do not think the text says this.  Here is the sentence:

     Current coding schemes such as H.261 [2], H.263 [3], MPEG-1, 2,4
      [5], H.264 [6] can encode video pictures as reference frames, also
      known as intra frames or predicted frames, also known as inter
      frames.


>The terms "picture" and "frame" are also kind of
>mixed up.

For progressive video, a picture is identical to a frame. The difference
appears for  interlaced video only.
The new text will use the term  'picture' only.


>The characterization of a "slice" or "GOB" as being a "stripe" is not
>really quite right.

The text does not say "image stripe" says "stripe" and in particular it
does not imply that such stripe goes border-to-border in the image (i.e.
can cover a partial macroblock row) nor that is should only include a
single  macroblock row (i.e., can cover multiple partial rows as in H.261).
Clarifications will be addressed in the new version of the draft.


-A

Andrea Basso
NMS Communications
(732) 936-2118



                                                                                                                               
                      "Gary Sullivan"                                                                                          
                      <garysull@windows.mic        To:       "Andrea Basso" <Andrea_Basso@nmss.com>, <avt@ietf.org>,           
                      rosoft.com>                   <avt-admin@ietf.org>, <mmusic@ietf.org>                                    
                                                   cc:       <casner@acm.org>, <jo@tzi.uni-bremen.de>,                         
                      07/08/2003 08:03 PM           <magnus.westerlund@ericsson.com>, <csp@csperkins.org>, "Orit Levin"        
                                                    <orit@radvision.com>, "Roni Even" <roni.even@polycom.co.il>,               
                                                    <nismail@cisco.com>                                                        
                                                   Subject:  RE: [AVT] Requirements draft on video control                     
                                                                                                                               





Much of the content of this ID will look familiar to those with
experience with ITU-T systems control specifications (such as H.245).

Some of it seems like it could use a bit of updating in that regard
(e.g., videofastupdateGOB).
Draft Recommendation H.241 is one good place to look for such
information.

Section 3 is a bit confused in its terminology.  For example it seems to
say that intra frames are also known as inter frames, which is
definitely not the case.  It mixes the terms "predictive" and
"prediction" and seems to consider a "reference" frame to be synonymous
with an intra frame.  The terms "picture" and "frame" are also kind of
mixed up.

A more consistent terminology would be as follows:

An intra picture is a picture that can be decoded without first decoding
any other picture.

A predicted (or non-intra) picture is a picture that may require data
from one or more previously-decoded pictures in order to be properly
decoded.

A reference picture is a picture that is stored in the decoder for use
as a reference in the decoding process of some subsequent picture in the
bitstream.

A non-reference picture is a picture that is not used as a reference for
the decoding process of any other picture in the bitstream.

The concepts of intra versus non-Intra and reference versus
non-reference are independent.  A particular picture can (in general) be
any one of the four types (intra reference, non-intra reference, intra
non-reference, non-intra non-reference).

A "fast update", also known as an "instantaneous decoder refresh",
involves sending an intra picture and thereafter refraining from using
any picture sent prior to that intra picture as a reference for the
decoding process of any subsequent picture sent in the stream.

In particular it is worth noting that some concepts that applied to some
older codecs are not really valid in general and should be avoided
(e.g., that a non-reference picture is called a B picture or that an
intra picture is the same thing as a fast update or that an intra
picture is always a reference picture).

The characterization of a "slice" or "GOB" as being a "stripe" is not
really quite right.  For example, a single slice in MPEG-1, H.263 Annex
O, or MPEG-4 can consist of a few macroblocks at the right edge of one
row and a few macroblocks at the left edge of the next row (i.e., two
distinct rectangular regions quite far apart from each other in the
picture can be in a single slice).  A GOB in H.261 CIF consists of THREE
partial rows of macroblocks extending only halfway across a picture.
And in H.264/AVC, a slice can be ANY collection of macroblocks scattered
around absolutely anywhere in the picture (by use of a feature known as
a "slice group map").

In regard to section 5.1 VideoFreezePicture, I believe H.262 (also known
as MPEG-2 Video) also supports a freeze picture release within the ITU-T
extension.

It is also possible to obtain random access recovery without a fast
update.  This is sometimes called "gradual decoder refresh".  See for
example the recovery point SEI message in H.264/AVC.

Best Regards,

Gary Sullivan


+> -----Original Message-----
+> From: avt-admin@ietf.org [mailto:avt-admin@ietf.org] On
+> Behalf Of Andrea Basso
+> Sent: Monday, July 07, 2003 2:34 PM
+> To: avt@ietf.org; avt-admin@ietf.org; mmusic@ietf.org
+> Cc: casner@acm.org; jo@tzi.uni-bremen.de;
+> magnus.westerlund@ericsson.com; csp@csperkins.org; Orit
+> Levin; Roni Even; nismail@cisco.com
+> Subject: [AVT] Requirements draft on video control
+>
+>
+> Attached you will find draft-basso-avt-videoconreq-00.txt
+>
+> Abstract:
+>
+> A variety of communication services such as video
+> conferencing and video messaging rely on the capability of video
+> encoders and decoders to exchange control commands. This document
+> outlines this set of commands as well as the requirements for
+> their transport.
+>
+> Comments are more than welcome.
+>
+>
+>
+>
+>
+> Andrea Basso
+> NMS Communications
+> (732) 936-2118
+> (See attached file: draft-basso-avt-videoconreq-00.zip)
+>






_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Thu Jul 10 03:25:29 2003
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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [AVT] Requirements draft on video control
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Hi,

I have some remarks around the posting of these messages. First is the 
receipients list, its way to long. I would recommed that we drop it to 
only the mailing lists. I assume that all persons being directly 
addressed are in fact part at least one of the lists. The list is in 
fact so long that it get caugth in the mailing list software and needs 
approval for every mail.

The second thing is the crosspostings. We should move to a single list, 
I propose AVT as it has the most subscribers while I know there is need 
to include both communities. Due to the crosspostings, if a person is 
not memeber of both list it will get caught in that lists filter and 
need manual approval by the chairs. If you need to inform some other 
list about a subject, the recommended way is to post to each list 
individually and state that any discussion will take place on one of 
these list.

Third, don't post to avt-admin@ietf.org unless it has something to do 
with administrative matters regarding the mailing list. We chairs are 
members of the list and don't need to get a second copy through the 
avt-admin@ietf.org address. In fact before trimming this receipient list 
I should have received 4 copies of each mail.

So please follow this recommendations any you will make us chairs life 
easier.


Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Thu, 10 Jul 2003 06:46:56 -0700
To: Aaron Colwell <acolwell@real.com>
From: Mark Baugher <mbaugher@cisco.com>
Subject: Re: [AVT] Vorbis RTP issues list for Vienna 
Cc: John Lazzaro <lazzaro@CS.Berkeley.EDU>, avt@ietf.org, mcgrew@cisco.com
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At 04:51 PM 7/9/2003 -0700, Aaron Colwell wrote:


>On Wed, 9 Jul 2003, John Lazzaro wrote:
>
> >
> > > Aaron Colwell <acolwell@real.com> writes:
> > > I agree with Ross on this. It seems that these codebooks are causing too
> > > many problems that aren't worth dealing with. People have been suggesting
> > > getting the codebooks via HTTP, alternate TCP links, and all sorts of
> > > other methods. If your going to go that far why not just use HTTP to
> > > deliver the data. That way you are guaranteed to get everything in the
> > > right order.
> >
> > I think its just an instance of a general problem -- transport of
> > hybrid streams with two components:
> >
> >   -- A steady flow of media information that can handle the
> >      occassional packet loss episode gracefully and without
> >      retransmission
> >
> >   -- Occassional chunks of data, which can be sent well ahead of
> >      its need, which needs reliability, and which is synchronous
> >      to the media flow.
> >
> > So, I think its a good idea for RTP to offer solutions that work
> > in this domain.  There are some studio-oriented uses of RTP MIDI
> > which fit into the model, and my motivation for doing
> > draft-lazzaro-avt-rtp-framing-contrans-01 is to support those in
> > a general way, so that when things like Vorbis codebooks come
> > along, RTP has a way to support them.  I really don't think that
> > posing the question as all-or-none -- RTP streaming or Shoutcast-esqe
> > pseudo-streaming -- benefits AVT, it will just spawn more HTTP
> > streaming implementations.
>
>When stated in these more general terms I can see your point about how
>this scenario could be useful. Another use may be a key stream for
>encrypted content that changes encryption keys often. You would want this
>data to be transmitted reliably and ensure that it gets there before the
>data that depends on it arrives.

The keys to the data need to be delivered "at least as reliably" as the 
data but not necessarily "reliably."  Putting keys in the data packet is 
one such method, but we are not likely to use this method in RTP (poor 
separation of functions, no third-party key mgt, etc).  But there have been 
proposals for using RTCP or SRTCP.  Rather than extend RTCP or SRTCP for 
this purpose, however, a better approach IMHO is to put the key in the SDP 
and send it "at least as reliably" as the SRTP.  The SDP must of course be 
authenticated/integrity-checked and encrypted.


>I did not mean to actually suggest the use of HTTP as a viable delivery
>mechanism instead of RTP. I don't really want more "Shoutcast-esqe
>pseudo-streaming" solutions either. I'll be a little more careful with
>comments like this.

It's hard to engineer a network where media flows masquerade as something else.

Mark


>I'll take a look at the draft that you mentioned and send you any comments
>I may have.
>
> >
> > That said:
> >
> > > The Vorbis folks
> > > should come up with a "streamable" profile that encodes the audio using
> > > predefined codebooks.
> >
> > This is a good option too -- my guess is that most Vorbis setups
> > will only need one codebook for the life of the stream, and that
> > need will be known in advance.  A URL in the session description
> > to support HTTP download of that one codebook would be a way to
> > go, with perhaps a known set of "standard codebooks" codable by
> > speak tokens or by GUIDs for the MIME HTTP codebook object.
>
>Hopefully they will embrace these ideas. I think it could simplify client
>implementations a bit. If they used just predefined codebooks or only used
>one codebook and specified it via SDP, Vorbis would be just like many
>other datatypes and would not require a lot of extra work to support in
>existing players.
>
>Aaron
>
> >
> > -------------------------------------------------------------------------
> > John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley
> > lazzaro [at] cs [dot] berkeley [dot] edu     www.cs.berkeley.edu/~lazzaro
> > -------------------------------------------------------------------------
> >
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www1.ietf.org/mailman/listinfo/avt
> >
>
>_______________________________________________
>Audio/Video Transport Working Group
>avt@ietf.org
>https://www1.ietf.org/mailman/listinfo/avt



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Alan Duric" <alan.duric@globalipsound.com>
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Subject: [AVT] Agenda item RTP Payload Format for iLBC Speech/iLBC codec
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Hi all,

here is some info about the agenda item:

RTP Payload Format for iLBC Speech/iLBC codec 	(Duric)     15min
	draft-ietf-avt-rtp-ilbc-02.txt
	draft-ietf-avt-ilbc-codec-02.txt

and the issues that will be addressed/discussed at the meeting on
Monday, July the 14th:

- details about the changes made from 01 versions 
- feedback and results of the iLBC codec spec review 
- resolve potential issue with the use of "mode" parameter in SDP
"a=fmtp" attribute, for the RTP Payload Format draft
- next steps 
- time slot for Q&A 


Drafts are available under the following links: 

http://www.ietf.org/internet-drafts/draft-ietf-avt-ilbc-codec-02.txt -
iLBC codec spec 

http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-ilbc-02.txt - RTP
payload profile for iLBC


Best regards,
Alan






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From: Colin Perkins <csp@csperkins.org>
To: Mitsuyuki Hatanaka <hatanaka@av.crl.sony.co.jp>
Cc: avt@ietf.org
Subject: Re: [AVT] Revised internet draft of RTP Payload Format for ATRAC-X
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--> Mitsuyuki Hatanaka <hatanaka@av.crl.sony.co.jp> writes:
> We have submitted our revised internet draft as
> "draft-hatanaka-avt-rtp-atracx-02.txt". You will be able to get it from
> the on-line Internet-Drafts directories.
> 
> We do not have a plan to discuss in AVT group at Vienna meeting,
>  because this is slight changing version.
>  
> The revised items are as follows.
> (1) Changing a terminology of "StreamID" to "ElementID" in section 2.3.
> (2) Changing a unit of LENGTH "byte" to "bit" in section 2.3.
> (3) Changing a unit of time stamp "millisecond of a presentation time"
>  to "sample number of a presentation time" in section 7.
> (4) Addition of the description about multicasting consideration in
>  section 8.
>  
> We will evaluate the defined functions of this payload with the practical
>  test application until next meeting, and we would like to reflect the
>  results in next document.

Thanks for submitting the update, I see that some of the concerns we
discussed at the San Francisco IETF meeting have been addressed (in
particular, the use of the sample clock for presentation time rather
than milliseconds). 

In addition to including the results of your practical tests, do you have
plans to further update the draft? If I recall correctly, there was some
confusion about the terminology used, which it would be useful to clarify.

Regards,
-- 
Colin Perkins
csp@csperkins.org

_______________________________________________
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Date: Fri, 11 Jul 2003 21:57:19 +0200
From: Colin Perkins <csp@csperkins.org>
To: "Michael A. Ramalho" <mramalho@cisco.com>
Cc: philkerr@elec.gla.ac.uk, avt@ietf.org
Subject: Re: [AVT] draft-herlein-speex-rtp-profile-01
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Michael,

--> "Michael A. Ramalho" <mramalho@cisco.com> writes:
>--> Colin Perkins writes:
>>- The use of the M bit in the RTP header to signify comfort noise is also
>>   unusual. This is not necessarily inappropriate, but it is important
>>   to consider if the advantages of this use outweigh the costs of being
>>   different to other audio payload formats (which use the M bit to
>>   signal the first packet after a silent period).
> 
> This may be contrary to the guidance I got from Steve a while back.
> 
> A little background first ..
> 
>  From RFC 1889:
> 
> "marker (M): 1 bit The interpretation of the marker is defined by a 
> profile. It
> is intended to allow significant events such as frame boundaries to be
> marked in the packet stream. A profile may define additional marker bits
> or specify that there is no marker bit by changing the number of bits in
> the payload type field (see Section 5.3)."
> 
> But from RFC 1890 (Audio and Video Conferences with Minimal Control
> that can be used with G.711):
> 
> "For applications which send no packets during silence, the first packet
> of a talk spurt (first packet after a silence period) is distinguished by
> setting the marker bit in the RTP data header. Applications without
> silence suppression set the bit to zero."
> 
> Then Steve said on 2/2/2003 (in regards to the RGL codec) ...
> 
> "I believe you should follow the marker bit convention as specified in
> RFC 1890. The RGL codec is likely to be used in implementations that
> also use other payload formats (such as PCMU) specified in RFC 1890,
> and the behavior should be consistent."
> 
> Thus, since I think that speex will also be used in similar
> deployments (i.e., packet voice) ... I would think Steve's
> guidance should apply here as well.
> 
> My question to the speex authors is that if they simply define
> the periods in which they desire to send comfort noise to be
> *identically* the same definition as a RFC 1890 "beginning or
> ending of a talk spurt", then there would be no inconsistency
> in the bit's meaning.

I'm not sure I understand your suggestion. I was expecting a period of
comfort noise to be treated identically to a silent period.  Hence the
M bit would be set on the first packet after the comfort noise (and not
set on any of the comfort noise packet).

-- 
Colin Perkins
csp@csperkins.org

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Date: Fri, 11 Jul 2003 22:20:03 +0200
From: Colin Perkins <csp@csperkins.org>
To: philkerr@elec.gla.ac.uk
Cc: avt@ietf.org
Subject: Re: [AVT] draft-herlein-speex-rtp-profile-01
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Hi,

--> philkerr@elec.gla.ac.uk writes:
...
> Quoting Colin Perkins <csp@csperkins.org>:
> > --> philkerr@elec.gla.ac.uk writes:
> > > Dear all,
> > > 
> > > An -01 update to draft-herlein-speex-rtp-profile is now available:
> > > 
> > > http://www.ietf.org/internet-drafts/draft-herlein-speex-rtp-profile-01.txt
> > > 
> > > Comments and feedback welcomed.
> > 
> > I have a number of questions and comments:
...
> > - The definition of the Padding bit in the RTP header is unusual. It
> >   may be better to state that padding MAY be used in the usual RTP
> >   manner, unless there is a specific reason why a different padding
> >   scheme is needed?
> 
> Changing the Padding bit to MAY should be fine.

The main issue is the text "P is set if the total packet size is less than
the MTU" which implies that padding is used to bring the packet size up to
the MTU. I'm assuming that is not intended!

> > - The use of the M bit in the RTP header to signify comfort noise is
> > also
> >   unusual. This is not necessarily inappropriate, but it is important
> >   to consider if the advantages of this use outweigh the costs of being
> >   different to other audio payload formats (which use the M bit to
> >   signal the first packet after a silent period).
> 
> Its use was suggested by Henning Schulzrinne to the other Speex draft
> authors and I've based it on their notes and Section 4.1 of RFC 1890.  If
> there is some discrepency in its use, as highlighted in the feedback by 
> Michael Ramalho, then we may need to look at this section again to
> determine best practice.

Does my reply to Michael clarify?

> > - The definition of the timestamp says "Speex uses 20 msec frames and a
> >   variable sampling rate clock". Does this mean that the sample rate
> >   can change at anytime within a session? Or is the sample rate chosen
> >   from one of a number of values, and then fixed within the session?
> 
> There is a fixed range of values for the sample rate but the rate may
> change mid session.  

It's easier from the point of view of RTP and signalling in SDP if each
different sample rate is sent using a different payload type number. In
SDP, something like:

	m=audio 49170 RTP/AVP 97 98
	a=rtpmap:97 speex/8000
	a=rtpmap:98 speex/16000

Might be useful to make this mandatory in the draft?

> > - Section 3.2 is unclear if the bit rate and sample rate are coupled.
> > Can
> >   Speex vary the compression ratio to change the bit rate without
> >   changing the sampling rate?
> 
> I will need to check this just to make sure.
> 
...
> > - Section 6 (use with H.323) is not appropriate in an IETF document.
> > You
> >   can include a reference to an ITU - or other - document explaining
> >   how to use Speex with H.323, but the IETF can't mandate use of H.245
> >   parameters.
> 
> Agreed.  I presume we can also link back to a Speex document if further
> explanation is required, with the usual proviso about linking to URLs as
> mentioned in the ID-nits.

Correct. How is Speex itself being referenced?

Cheers,
-- 
Colin Perkins
csp@csperkins.org

_______________________________________________
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avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Fri, 11 Jul 2003 21:54:40 +0100
To: Colin Perkins <csp@csperkins.org>
Cc: avt@ietf.org
Subject: Re: [AVT] draft-herlein-speex-rtp-profile-01
References: <1057175019.3f0335eb5bf7c@dlana.elec.gla.ac.uk> <20030709110522.1217e183.csp@csperkins.org> <1057791436.3f0c9dcc90ee7@dlana.elec.gla.ac.uk> <20030711222003.5ca749fe.csp@csperkins.org>
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Thanks Colin,

Quoting Colin Perkins <csp@csperkins.org>:

> Hi,
> 
> --> philkerr@elec.gla.ac.uk writes:
> ...
> > Quoting Colin Perkins <csp@csperkins.org>:
> > > --> philkerr@elec.gla.ac.uk writes:
> > > > Dear all,
> > > > 
> > > > An -01 update to draft-herlein-speex-rtp-profile is now available:
> > > > 
> > > >
> http://www.ietf.org/internet-drafts/draft-herlein-speex-rtp-profile-01.txt
> > > > 
> > > > Comments and feedback welcomed.
> > > 
> > > I have a number of questions and comments:
> ...
> > > - The definition of the Padding bit in the RTP header is unusual. It
> > >   may be better to state that padding MAY be used in the usual RTP
> > >   manner, unless there is a specific reason why a different padding
> > >   scheme is needed?
> > 
> > Changing the Padding bit to MAY should be fine.
> 
> The main issue is the text "P is set if the total packet size is less than
> the MTU" which implies that padding is used to bring the packet size up to
> the MTU. I'm assuming that is not intended!

It was, but can be changed so it is not intended.

> 
> > > - The use of the M bit in the RTP header to signify comfort noise is
> > > also
> > >   unusual. This is not necessarily inappropriate, but it is important
> > >   to consider if the advantages of this use outweigh the costs of being
> > >   different to other audio payload formats (which use the M bit to
> > >   signal the first packet after a silent period).
> > 
> > Its use was suggested by Henning Schulzrinne to the other Speex draft
> > authors and I've based it on their notes and Section 4.1 of RFC 1890.  If
> > there is some discrepency in its use, as highlighted in the feedback by 
> > Michael Ramalho, then we may need to look at this section again to
> > determine best practice.
> 
> Does my reply to Michael clarify?

Yes.

> 
> > > - The definition of the timestamp says "Speex uses 20 msec frames and a
> > >   variable sampling rate clock". Does this mean that the sample rate
> > >   can change at anytime within a session? Or is the sample rate chosen
> > >   from one of a number of values, and then fixed within the session?
> > 
> > There is a fixed range of values for the sample rate but the rate may
> > change mid session.  
> 
> It's easier from the point of view of RTP and signalling in SDP if each
> different sample rate is sent using a different payload type number. In
> SDP, something like:
> 
> 	m=audio 49170 RTP/AVP 97 98
> 	a=rtpmap:97 speex/8000
> 	a=rtpmap:98 speex/16000
> 
> Might be useful to make this mandatory in the draft?

Agreed.

> 
> > > - Section 3.2 is unclear if the bit rate and sample rate are coupled.
> > > Can
> > >   Speex vary the compression ratio to change the bit rate without
> > >   changing the sampling rate?
> > 
> > I will need to check this just to make sure.
> > 
> ...
> > > - Section 6 (use with H.323) is not appropriate in an IETF document.
> > > You
> > >   can include a reference to an ITU - or other - document explaining
> > >   how to use Speex with H.323, but the IETF can't mandate use of H.245
> > >   parameters.
> > 
> > Agreed.  I presume we can also link back to a Speex document if further
> > explanation is required, with the usual proviso about linking to URLs as
> > mentioned in the ID-nits.
> 
> Correct. How is Speex itself being referenced?

A reference needs to be added pointing back to the website.

Cheers

Phil

> 
> Cheers,
> -- 
> Colin Perkins
> csp@csperkins.org
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
> 




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_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Edwards, Eric" <Eric.Edwards@am.sony.com>
To: "'Colin Perkins'" <csp@csperkins.org>,
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	 <Eric.Edwards@am.sony.com>
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Subject: [AVT] RE: draft-ietf-avt-rtp-jpeg2000-03.txt
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Dear Colin and all,

Sorry for the delay.  Unfortunately, my Sony colleagues and I are tied up
for the next few weeks.  So, we will not be able to make all necessary
updates to the jp2/rtp draft for the Vienna meeting. We should be able to
post an updated version in about two weeks.

As to your questions, my colleague has forwarded these answers - 

> 
>  - Section 5.1 seems to imply that JPEG 2000 error resilience markers are
>    only used with non-intelligent packetization. Is this correct? Are the
>    error resilience markers not useful if the JPEG stream is intelligently
>    packetized?

The first sentence is not true.

In non-intelligent packetization, a serious quality distortion will be 
occured without JPEG 2000 error resilience markers, 
but it does not imply that error resilience markers are only used 
with non-intelligent packetization. 

The error resilient markers are also useful with intelligent 
packetization.


>  - Is non-intelligent packetization a requirement? The situations where it
>    will be useful seem very limited.

Non-intelligent packetization is not always required (optional) at a sender,

but is required at a receiver.

It is true that the situations where non-intelligent packetization will 
be used are limited, but a receiver who received the non-intelligent 
packets must process the packets properly. 

>  - The extension mechanism in Section 8 is not used. I suggest removing it

>    and marking the X bit in the payload header as "reserved". 

We agree.  -> it is an item to change in the next version.

>  - The second paragraph of Section 9 is not appropriate, since the payload
>    format cannot specify the type of video content to be transported. Are
>    there no ways in which a malicious sender can disrupt the receiver?

It is entirely correct.
I will refer to other drafts and find the better expressions.

>  - In addition to the MIME type registration, the draft should include a 
>    description of how the MIME type is mapped to SDP (Section 8.2 of the
>    H.264 payload format is a good example).

We agree.  -> it is another item to change in the next version.


>  - Section 10 may be better written as an informative appendix? 

We agree.  -> it is also an item to change in the next version.

>  - It may be useful to include some examples, showing the complete packets
>    with all headers and data (including the RTP headers).

We agree. -> it is also an item to change in the next version



I will proof-read for clarity before we post the next draft.

Best regards,

Eric



-----Original Message-----
From: Colin Perkins [mailto:csp@csperkins.org]
Sent: Tuesday, July 08, 2003 10:28 AM
To: Edwards, Eric
Cc: avt@ietf.org; casner@acm.org; magnus.westerlund@ericsson.com
Subject: Re: draft-ietf-avt-rtp-jpeg2000-03.txt


Eric,

--> "Edwards, Eric" <Eric.Edwards@am.sony.com> writes:
> The Internet-Draft - draft-ietf-avt-rtp-jpeg2000-03.txt - has been
> posted.  
> 
> Some time passed before this version was submitted.  Most of the time was
> spent discussing if potential reconciliation with work that was taking
> place in ISO/IEC JTC1 SC 29 WG 01 (JPEG) was in order.  After lengthy
> discussions it was agreed by the experts in WG 01 that there were
> different goals and application areas targeted for our respective work
> and that the JP2/RTP proposal (draft-ietf-avt-rtp-jpeg2000-03.txt) should
> move forward in IETF as is.  
> 
> So at this stage we are ready to move to Last Call if the AVT experts
> have no objections to the current draft.  
> 
> Please advise and comment.

I have a number of comments with regards to this draft:

 - Section 5.1 seems to imply that JPEG 2000 error resilience markers are
   only used with non-intelligent packetization. Is this correct? Are the
   error resilience markers not useful if the JPEG stream is intelligently
   packetized?

 - Is non-intelligent packetization a requirement? The situations where it
   will be useful seem very limited.

 - The extension mechanism in Section 8 is not used. I suggest removing it 
   and marking the X bit in the payload header as "reserved". 

 - The second paragraph of Section 9 is not appropriate, since the payload
   format cannot specify the type of video content to be transported. Are
   there no ways in which a malicious sender can disrupt the receiver?

 - In addition to the MIME type registration, the draft should include a 
   description of how the MIME type is mapped to SDP (Section 8.2 of the
   H.264 payload format is a good example).

 - Section 10 may be better written as an informative appendix? 

 - It may be useful to include some examples, showing the complete packets
   with all headers and data (including the RTP headers).

I would also urge a careful proof-reading of the draft for both quality of
English, clarity of specification, and appropriate use of the requirements
language in RFC 2119. 

-- 
Colin Perkins
csp@csperkins.org

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: [AVT] Proposed change to Enhanced CRTP
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Content-Type: text/plain; charset="us-ascii"; format=flowed



We'd like to propose a change to the Enhanced CRTP draft, 
draft-ietf-avt-crtp-enhance-07.txt
http://www.ietf.org/internet-drafts/draft-ietf-avt-crtp-enhance-07.txt

One of the enhancements is to add a Headers Checksum (HDRCKSUM) when UDP 
checksum is not present (see section 2.2).
The UDP checksum is used by the decompressor to periodically validate the 
reconstructed packets. When not present,
a HDRCKSUM may be inserted by the compressor replacing the missing UDP 
checksum. The decompressor removes the HDRCKSUM after validating the 
reconstructed packet.

Currently the HDRCKSUM covers the pseudo-IP header, the UDP header and the 
fixed part of the RTP header (first 12 bytes).
Those are the only parts of the reconstructed packet that can be validated 
by the HDRCKSUM.
The CSRC list is not included in the HDRCKSUM.

We propose to include the CSRC list in the HDRCKSUM since the CSRC list is 
part of the header compression context.

The CSRC list is stored at the decompressor context and inserted to the 
restored packets. If the HDRCKSUM does not cover the CSRC list and the CSRC 
list at the decompressor is errored, a bad CSRC list will be restored into 
multiple packets until the CSRC
list changes and is resent by the compressor as an update.
If the HDRCKSUM covers the CSRC list, an error in the CSRC list will be 
caught early and corrected.

When there is a change in the CSRC list, the CSRC list is sent as an update 
to the context.
The CSRC list should be treated in the same way as the other context 
updating changes and be included in the HDRCKSUM.

On the other hand, adding the CSRC list to the HDRCKSUM requires more 
processing calculating the HDRCKSUM. Also if
the accuracy of the CSRC list is not crucial and the CSRC list is included 
in the HDRCKSUM, then an error in the CSRC list
will result in dropping a few packets while the contexts are getting 
synchronized.


The proposed change to the text is as follows:

Old:

    The HDRCKSUM is calculated in the same way as a UDP checksum except
    that it does not cover all of the UDP data.  That is, the HDRCKSUM is
    the 16-bit one's complement of the one's complement sum of the
    pseudo-IP header (as defined for UDP), the UDP header, and the first
    12 bytes of the UDP data which are assumed to hold the fixed part of
    an RTP header.  The extended part of the RTP header and the RTP data
    will not be included in the HDRCKSUM.

New:

    The HDRCKSUM is calculated in the same way as a UDP checksum except
    that it does not cover all of the UDP data.  That is, the HDRCKSUM is
    the 16-bit one's complement of the one's complement sum of the
    pseudo-IP header (as defined for UDP), the UDP header,     the first
                                                           ^^^^
    12 bytes of the UDP data which are assumed to hold the fixed part of
    an RTP header, and the CSRC list.
                 ^^^^^^^^^^^^^^^^^^^
    The extended part of the RTP header beyond the CSRC list and the RTP data
                                        ^^^^^^^^^^^^^^^^^^^^
    will not be included in the HDRCKSUM.

--=====================_25267062==_.ALT
Content-Type: text/html; charset="us-ascii"

<html>
<br>
<br>
We'd like to propose a change to the Enhanced CRTP draft,
draft-ietf-avt-crtp-enhance-07.txt<br>
<a href="http://www.ietf.org/internet-drafts/draft-ietf-avt-crtp-enhance-07.txt" eudora="autourl">http://www.ietf.org/internet-drafts/draft-ietf-avt-crtp-enhance-07.txt</a><br>
<br>
One of the enhancements is to add a Headers Checksum (HDRCKSUM) when UDP
checksum is not present (see section 2.2).<br>
The UDP checksum is used by the decompressor to periodically validate the
reconstructed packets. When not present, <br>
a HDRCKSUM may be inserted by the compressor replacing the missing UDP
checksum. The decompressor removes the HDRCKSUM after validating the
reconstructed packet.<br>
<br>
Currently the HDRCKSUM covers the pseudo-IP header, the UDP header and
the fixed part of the RTP header (first 12 bytes).<br>
Those are the only parts of the reconstructed packet that can be
validated by the HDRCKSUM.<br>
The CSRC list is not included in the HDRCKSUM. <br>
<br>
We propose to include the CSRC list in the HDRCKSUM since the CSRC list
is part of the header compression context.<br>
<br>
The CSRC list is stored at the decompressor context and inserted to the
restored packets. If the HDRCKSUM does not cover the CSRC list and the
CSRC list at the decompressor is errored, a bad CSRC list will be
restored into multiple packets until the CSRC<br>
list changes and is resent by the compressor as an update.<br>
If the HDRCKSUM covers the CSRC list, an error in the CSRC list will be
caught early and corrected.<br>
<br>
When there is a change in the CSRC list, the CSRC list is sent as an
update to the context. <br>
The CSRC list should be treated in the same way as the other context
updating changes and be included in the HDRCKSUM.<br>
<br>
On the other hand, adding the CSRC list to the HDRCKSUM requires more
processing calculating the HDRCKSUM. Also if<br>
the accuracy of the CSRC list is not crucial and the CSRC list is
included in the HDRCKSUM, then an error in the CSRC list<br>
will result in dropping a few packets while the contexts are getting
synchronized.<br>
<br>
<br>
The proposed change to the text is as follows:<br>
<br>
Old:<br>
<br>
<font face="Courier New, Courier">&nbsp;&nbsp; The HDRCKSUM is calculated
in the same way as a UDP checksum except<br>
&nbsp;&nbsp; that it does not cover all of the UDP data.&nbsp; That is,
the HDRCKSUM is<br>
&nbsp;&nbsp; the 16-bit one's complement of the one's complement sum of
the<br>
&nbsp;&nbsp; pseudo-IP header (as defined for UDP), the UDP header, and
the first<br>
&nbsp;&nbsp; 12 bytes of the UDP data which are assumed to hold the fixed
part of<br>
&nbsp;&nbsp; an RTP header.&nbsp; The extended part of the RTP header and
the RTP data<br>
&nbsp;&nbsp; will not be included in the HDRCKSUM.<br>
<br>
</font>New:<br>
<br>
<font face="Courier New, Courier">&nbsp;&nbsp; The HDRCKSUM is calculated
in the same way as a UDP checksum except<br>
&nbsp;&nbsp; that it does not cover all of the UDP data.&nbsp; That is,
the HDRCKSUM is<br>
&nbsp;&nbsp; the 16-bit one's complement of the one's complement sum of
the<br>
&nbsp;&nbsp; pseudo-IP header (as defined for UDP), the UDP
header,&nbsp;&nbsp;&nbsp;&nbsp; the first<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
^^^^<br>
&nbsp;&nbsp; 12 bytes of the UDP data which are assumed to hold the fixed
part of<br>
&nbsp;&nbsp; an RTP header, and the CSRC list.<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
^^^^^^^^^^^^^^^^^^^<br>
&nbsp;&nbsp; The extended part of the RTP header beyond the CSRC list and
the RTP data<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
^^^^^^^^^^^^^^^^^^^^ <br>
&nbsp;&nbsp; will not be included in the HDRCKSUM.<br>
</font></html>

--=====================_25267062==_.ALT--


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: Henning Schulzrinne <hgs@cs.columbia.edu>
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Subject: [AVT] IMA; DVI
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The Interactive Multimedia Association has either disappeared or merged 
with another organization. (The IMTC does not seem have been involved, 
http://www.imtc.org/faqbody.asp#17). Does anybody know its whereabouts?

Suspecting its untimely ascension to the great Association of 
Federations cemetary, does anybody have access to a softcopy of the DVI 
(32 kb/s and 24 kb/s) audio spec?

Thanks.

Henning


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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                           Call for Papers

  2004 IEEE/IFIP Network Operations and Management Symposium (NOMS 2004)

      "Managing Next Generation Convergence Networks and Services"

                     Seoul Korea, 19-23, April 2004

                        http://www.noms2004.org

The 9th IEEE/IFIP Network Operations and Management Symposium (NOMS 2004) 
will be held 19-23 April, 2004 in COEX Convention Center, Seoul, Korea.
NOMS 2004 will present the latest approaches and technical solutions in
the area of network operations and management. Held in the even-numbered
years since 1988, NOMS 2004 continues the established tradition of NOMS
and IM as the primary forum for technical exchange of research, standards,
development, systems integrator, service providers, and user communities.
An exciting, peer-reviewed program of technical sessions, application
sessions, panels, tutorials, posters, and vendor exhibits will address the
ever-increasing interest in overall management solutions for all types of
communications and computing networks, systems, services and enterprise
applications. 

The symposium is sponsored by the IEEE Communications Society Technical
Committee on Network Operations and Management (CNOM), and by the
International Federation for Information Processing (IFIP) Working Group
6.6 on Management of Networks and Distributed Systems. 

The concept of network convergence has recently emerged as a new attempt
for the merging of telephony and data networks into a single,
multi-service network exploiting the ubiquity of the Internet. For this
increasingly attractive business model, in both wired and wireless
domains, strategic research is required to devise the best integration
architectures, operations and management solutions. This creates a unique
opportunity for the network operations and management community to respond
to the ever-increasing demand for network resilience, security,
quality-of-service and mobility management at unprecedented scales. NOMS
2004 provides the forum for discussing these research challenges and many
others inherent to the integrated management of next generation converged
networks and services. 

NOMS 2004 will broaden the scope of previous NOMS by expanding its program
to include a broader set of topics ranging from network operations and
management to network planning, service engineering and business processes
for network and service management. Special attention will be given to
experiences that emphasize lessons learned and reports on practice from
industry. 

The NOMS 2004 program consists of three distinct sessions: technical,
application, and panels. Technical sessions present the papers on the
latest research results in the network operations and management area.
Application sessions present the papers focusing on the experience of
IT/telecommunications industries (service providers, OSS/equipment
vendors, etc.) and customers based on their system implementation and
business practice in the operations and management. Panel sessions focus
on business implications, market trends of cutting edge technologies, and
applications with the panelists consisting of technology experts and
business executives. In addition, NOMS 2004 will feature tutorials,
posters, birds-of-a-feather sessions, and vendor exhibitions.

Authors are invited to submit complete unpublished papers, which are not
under review in any other conference or journal. Authors are also invited
to submit proposals for tutorials, panel sessions, poster demonstrations
and BoF sessions.

Important Dates:
Deadline for Technical Session Papers:          1 August 2003
Deadline for Application Session Papers:    15 September 2003
Deadline for Tutorials, Panels and Posters: 15 September 2003
Notification of Acceptance:                  15 November 2003
Deadline for Submitting Revised Papers:      15 December 2003
Final Camera Ready Papers Due:                15 January 2004

      Visit http://www.noms2004.org for the detailed information




_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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	(envelope-from csp@csperkins.org)
Date: Mon, 14 Jul 2003 00:23:09 +0200
From: Colin Perkins <csp@csperkins.org>
To: avt@ietf.org
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This is a multi-part message in MIME format.

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Attached is a draft for the revised AVT charter. This will be discussed in
the session tomorrow morning; comments are welcome.

-- 
Colin Perkins
csp@csperkins.org

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VlBGIGZvciBEcmFmdCBTdGFuZGFyZAoK

--Multipart_Mon__14_Jul_2003_00:23:09_+0200_0819d200--

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Sun Jul 13 22:25:36 2003
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Date: Sun, 13 Jul 2003 22:23:43 -0400
To: Colin Perkins <csp@csperkins.org>, avt@ietf.org
From: "Michael A. Ramalho" <mramalho@cisco.com>
Subject: Re: [AVT] Draft charter revision
Cc: casner@acm.org, magnus.westerlund@ericsson.com, mankin@psg.com
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At 12:23 AM 7/14/2003 +0200, Colin Perkins wrote:
>Attached is a draft for the revised AVT charter. This will be discussed in
>the session tomorrow morning; comments are welcome.
>
>--
>Colin Perkins
>csp@csperkins.org


Colin/Magnus/Steve,

I've got two questions regarding the following snippet from the re-charter:

"The group continues to be precluded from work on codecs themselves
because of overlap with the other standards bodies, and because the
IETF does not have the ability to effectively review new codecs. An
exception was made for the freeware iLBC codec on a highly experimental
basis, but acceptance of new codec work is unexpected and subject
to rechartering."

1) With regard to iLBC, while I understood that the issue was not fully
resolved, I thought AVT was to be the "primary repository" for iLBC's
definition (that is, not the ITU-T or other standards defining bodies).
I more or less understand what "experimental" means in the IETF - but
what does "highly experimental" mean? Is it even less probable that
iLBC will eventually attain standards status by the additional adjective?

2) With regard to RGL (for people unfamiliar to RGL, see www.vovida.org
for details), RGL is really a "lossless data compression technique" applied
to (A or mu-law) G.711. Steve had requested that "avt" be included in the
titles of RGL revisions - so I assumed it was a working group item. Is RGL
covered in the "re-chartering statement" (above)? Or is RGL, by virtue
of it having more in common with a data compression algorithm than a
(lossy) signal processing codec, exempt (no codec expertise needed - a
rational noted above)? If this is unclear, could you obtain further guidance
from the ADs (Allison) or the IESG?

Thanks,

Michael Ramalho

Michael A. Ramalho, Ph.D.
Office email: mramalho@cisco.com
Personal email: mar42@cornell.edu
Office: +1.941.708.4650
Cell: +1.941.544.2844


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Mon Jul 14 03:21:43 2003
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Copies of the slides for today's session can be found at
	http://www.east.isi.edu/~csp/avt57.zip
These will migrate onto the IETF proceedings site shortly.

-- 
Colin Perkins
csp@csperkins.org

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Mon Jul 14 08:59:41 2003
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From: "Allison, Art" <AAllison@nab.org>
To: "'Henning Schulzrinne'" <hgs@cs.columbia.edu>,
        "'avt@ietf.org'"
	 <avt@ietf.org>
Subject: RE: [AVT] IMA; DVI
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The IMA indeed has folded.  What is the date of the DVI spec?
We may have it in backups...but need a date (and Title) to search.

Art
::{)
Art Allison
Director Advanced Engineering
NAB
1771 N St NW
Washington DC 20036
202 429 5418


-----Original Message-----
From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
Sent: Saturday, July 12, 2003 11:56 AM
To: 'avt@ietf.org'
Subject: [AVT] IMA; DVI


The Interactive Multimedia Association has either disappeared or merged 
with another organization. (The IMTC does not seem have been involved, 
http://www.imtc.org/faqbody.asp#17). Does anybody know its whereabouts?

Suspecting its untimely ascension to the great Association of 
Federations cemetary, does anybody have access to a softcopy of the DVI 
(32 kb/s and 24 kb/s) audio spec?

Thanks.

Henning


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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I suspect around 1992, since it has been in RTP profile for a while. My 
citation is

@TECHREPORT{IMAD9210:Recommended,
AUTHOR="I. M. A Digital Audio Focus and Technical Working Groups",
TITLE="Recommended practices for enhancing digital audio compatibility
in multimedia systems (Version 3.00)",
INSTITUTION="Interactive Multimedia Association",
ADDRESS="Annapolis, Maryland",
MONTH=oct,
YEAR=1992,
KEYWORDS="audio; standard; ADPCM",
ANNOTE="Audio interchange formats; reference implementations",
}

Allison, Art wrote:

> The IMA indeed has folded.  What is the date of the DVI spec?
> We may have it in backups...but need a date (and Title) to search.
> 
> Art
> ::{)
> Art Allison
> Director Advanced Engineering
> NAB
> 1771 N St NW
> Washington DC 20036
> 202 429 5418
> 



_______________________________________________
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avt@ietf.org
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On Mon, 14 Jul 2003, Colin Perkins wrote:

> Copies of the slides for today's session can be found at
> 	http://www.east.isi.edu/~csp/avt57.zip
> These will migrate onto the IETF proceedings site shortly.

Colin,

Did you send these yet?  I guess we need to make up a presentation
list to go along with the files.  And the filenames should be avt0,
avt1 etc.

                                                        -- Steve


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Date: Mon, 14 Jul 2003 20:39:33 +0200
From: Colin Perkins <csp@csperkins.org>
To: "Michael A. Ramalho" <mramalho@cisco.com>
Cc: avt@ietf.org, casner@acm.org, magnus.westerlund@ericsson.com,
        mankin@psg.com
Subject: Re: [AVT] Draft charter revision
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Michael,

--> "Michael A. Ramalho" <mramalho@cisco.com> writes:
> I've got two questions regarding the following snippet from the
> re-charter:
> 
> "The group continues to be precluded from work on codecs themselves
> because of overlap with the other standards bodies, and because the
> IETF does not have the ability to effectively review new codecs. An
> exception was made for the freeware iLBC codec on a highly experimental
> basis, but acceptance of new codec work is unexpected and subject
> to rechartering."
> 
> 1) With regard to iLBC, while I understood that the issue was not fully
> resolved, I thought AVT was to be the "primary repository" for iLBC's
> definition (that is, not the ITU-T or other standards defining bodies).
> I more or less understand what "experimental" means in the IETF - but
> what does "highly experimental" mean? Is it even less probable that
> iLBC will eventually attain standards status by the additional adjective?

The iLBC codec specification is on track to be published as an Experimental
RFC. There are no plans to advance it to the standards track.

> 2) With regard to RGL (for people unfamiliar to RGL, see www.vovida.org
> for details), RGL is really a "lossless data compression technique"
> applied to (A or mu-law) G.711. Steve had requested that "avt" be
> included in the titles of RGL revisions - so I assumed it was a working
> group item. Is RGL covered in the "re-chartering statement" (above)? Or
> is RGL, by virtue of it having more in common with a data compression
> algorithm than a(lossy) signal processing codec, exempt (no codec
> expertise needed - a rational noted above)? If this is unclear, could you
> obtain further guidance from the ADs (Allison) or the IESG?

The RGL codec and payload format are not currently AVT work items.  The
codec is covered by the statement that "acceptance of new codec work is
unexpected and subject to rechartering". 

-- 
Colin Perkins
csp@csperkins.org

_______________________________________________
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avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Mark Baugher'" <mbaugher@cisco.com>, Aaron Colwell <acolwell@real.com>
Cc: John Lazzaro <lazzaro@CS.Berkeley.EDU>, avt@ietf.org, mcgrew@cisco.com
Subject: RE: [AVT] Vorbis RTP issues list for Vienna 
Date: Tue, 15 Jul 2003 07:25:42 -0400
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> The keys to the data need to be delivered "at least as 
> reliably" as the 
> data but not necessarily "reliably."  Putting keys in the 
> data packet is 
> one such method, but we are not likely to use this method in 
> RTP (poor 
> separation of functions, no third-party key mgt, etc).  But 
> there have been 
> proposals for using RTCP or SRTCP.  Rather than extend RTCP 
> or SRTCP for 
> this purpose, however, a better approach IMHO is to put the 
> key in the SDP 
> and send it "at least as reliably" as the SRTP.  The SDP must 
> of course be 
> authenticated/integrity-checked and encrypted.
I'm not sure this is a good idea.  At least in SIP, the
media path is considerably different, and in most cases,
considerably shorter than the signalling path, because
the signalling path has intermediaries that the media
path does not.  Key change is not something I would
want to see delayed by the intermediaries.  It really
is a "real time" transport problem, although somewhat
less real time than the actual media.
Passing keys may be best done in the media path.  

Brian

_______________________________________________
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From exim@www1.ietf.org  Tue Jul 15 10:12:29 2003
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        "'Mark Baugher'" <mbaugher@cisco.com>,
        Aaron Colwell <acolwell@real.com>
From: "Michael A. Ramalho" <mramalho@cisco.com>
Subject: RE: [AVT] Vorbis RTP issues list for Vienna 
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I'd like to add to Brian's comment ... but I fear it might start a
long-winded DCCP/PR-SCTP/UDP-congestion-backoff
discussion.

Here goes ...

Michael Ramalho

At 07:25 AM 7/15/2003 -0400, Rosen, Brian wrote:
> > The keys to the data need to be delivered "at least as
> > reliably" as the
> > data but not necessarily "reliably."  Putting keys in the
> > data packet is
> > one such method, but we are not likely to use this method in
> > RTP (poor
> > separation of functions, no third-party key mgt, etc).  But
> > there have been
> > proposals for using RTCP or SRTCP.  Rather than extend RTCP
> > or SRTCP for
> > this purpose, however, a better approach IMHO is to put the
> > key in the SDP
> > and send it "at least as reliably" as the SRTP.  The SDP must
> > of course be
> > authenticated/integrity-checked and encrypted.
>I'm not sure this is a good idea.  At least in SIP, the
>media path is considerably different, and in most cases,
>considerably shorter than the signalling path, because
>the signalling path has intermediaries that the media
>path does not.  Key change is not something I would
>want to see delayed by the intermediaries.  It really
>is a "real time" transport problem, although somewhat
>less real time than the actual media.
>Passing keys may be best done in the media path.
>
>Brian

I agree with Brian here - ideally the keys should be passed
reliably in the media path. But what to do?

The transport people know that we have a protocol that
can do this job ... it just not being used yet because it
new and has not gone through IETF last call.

pr-SCTP has the ability to have "reliable" and "unreliable"
streams present on the same "association". That is, you
can run RTP over a "unreliable stream" whilst sending the
keying material over a "reliable stream" to the same
(potentially set of) IP address(es).

Like DCCP, RTP over an "unreliable pr-sctp stream" has
the desired property of having "TCP-friendly" backoff under
congestion. Unlike DCCP - pr-SCTP has a "reliable
option" (and was the first offering of pr-sctp, via it's
SCTP origin).

On the other side of the same coin, DCCP has a notion
of which packets provided to it were dropped (or not
forwarded) ... so a layer above DCCP could be designed
to add "reliability" to a particular packet. However this
layer wouldn't have the advantage of being integrated
in the transport. This potentially adds more delay for
the keys (Brian's primary comment above).

As a side note, more complex codecs have the notion
of "frame/payload" importance (e.g., base frames of
a video codec having more importance than interpolated
frames). I'm waiting for the discussion of the need of
"mixed reliability/importance/persistence" payloads in
AVT payload formats for particular codecs ... and the
inevitable impact on proposed transport protocols. I
think this must be addressed with a high level of
interaction with the transport WG to get it right
(i.e., anywhere near optimal).

Michael Ramalho

Michael A. Ramalho, Ph.D.
Office email: mramalho@cisco.com
Personal email: mar42@cornell.edu
Office: +1.941.708.4650
Cell: +1.941.544.2844


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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To: "Michael A. Ramalho" <mramalho@cisco.com>
From: Mark Baugher <mbaugher@cisco.com>
Subject: RE: [AVT] Vorbis RTP issues list for Vienna 
Cc: "Rosen, Brian" <Brian.Rosen@marconi.com>,
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        John Lazzaro <lazzaro@CS.Berkeley.EDU>, avt@ietf.org, mcgrew@cisco.com
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At 10:10 AM 7/15/2003 -0400, Michael A. Ramalho wrote:
>I'd like to add to Brian's comment ... but I fear it might start a
>long-winded DCCP/PR-SCTP/UDP-congestion-backoff
>discussion.
>
>Here goes ...
>
>Michael Ramalho
>
>At 07:25 AM 7/15/2003 -0400, Rosen, Brian wrote:
>> > The keys to the data need to be delivered "at least as
>> > reliably" as the
>> > data but not necessarily "reliably."  Putting keys in the
>> > data packet is
>> > one such method, but we are not likely to use this method in
>> > RTP (poor
>> > separation of functions, no third-party key mgt, etc).  But
>> > there have been
>> > proposals for using RTCP or SRTCP.  Rather than extend RTCP
>> > or SRTCP for
>> > this purpose, however, a better approach IMHO is to put the
>> > key in the SDP
>> > and send it "at least as reliably" as the SRTP.  The SDP must
>> > of course be
>> > authenticated/integrity-checked and encrypted.
>>I'm not sure this is a good idea.  At least in SIP, the
>>media path is considerably different, and in most cases,
>>considerably shorter than the signalling path, because
>>the signalling path has intermediaries that the media
>>path does not.  Key change is not something I would
>>want to see delayed by the intermediaries.  It really
>>is a "real time" transport problem, although somewhat
>>less real time than the actual media.
>>Passing keys may be best done in the media path.
>>
>>Brian
>
>I agree with Brian here - ideally the keys should be passed
>reliably in the media path. But what to do?
>
>The transport people know that we have a protocol that
>can do this job ...

What transport protocol has been designed to send cryptographic keys?

Mark

>it just not being used yet because it
>new and has not gone through IETF last call.
>
>pr-SCTP has the ability to have "reliable" and "unreliable"
>streams present on the same "association". That is, you
>can run RTP over a "unreliable stream" whilst sending the
>keying material over a "reliable stream" to the same
>(potentially set of) IP address(es).
>
>Like DCCP, RTP over an "unreliable pr-sctp stream" has
>the desired property of having "TCP-friendly" backoff under
>congestion. Unlike DCCP - pr-SCTP has a "reliable
>option" (and was the first offering of pr-sctp, via it's
>SCTP origin).
>
>On the other side of the same coin, DCCP has a notion
>of which packets provided to it were dropped (or not
>forwarded) ... so a layer above DCCP could be designed
>to add "reliability" to a particular packet. However this
>layer wouldn't have the advantage of being integrated
>in the transport. This potentially adds more delay for
>the keys (Brian's primary comment above).
>
>As a side note, more complex codecs have the notion
>of "frame/payload" importance (e.g., base frames of
>a video codec having more importance than interpolated
>frames). I'm waiting for the discussion of the need of
>"mixed reliability/importance/persistence" payloads in
>AVT payload formats for particular codecs ... and the
>inevitable impact on proposed transport protocols. I
>think this must be addressed with a high level of
>interaction with the transport WG to get it right
>(i.e., anywhere near optimal).
>
>Michael Ramalho
>
>Michael A. Ramalho, Ph.D.
>Office email: mramalho@cisco.com
>Personal email: mar42@cornell.edu
>Office: +1.941.708.4650
>Cell: +1.941.544.2844



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Wed Jul 16 14:38:32 2003
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To: avt@ietf.org
Subject: [AVT] MUSICNETWORK Workshop & WEDELMUSIC Conference, Leeds, UK, September 2003
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Dear MUSICNETWORK and WEDELMUSIC Friend

All the most recent innovations and hot topics related to music and multimedia
will be discussed in Leeds, UK, 15-18 September, 2003, Open to all

You are invited to attend the work of the second MUSICNETWORK
open workshop. A lot of interesting topics will be addressed:
Digital right management, music notation (XML and non), optical music recognition,
restoring, audio processing, personalisation, indexing, retrieval, 
libraries and archives, music publication, internet publication, 
cross media distribution and protection, MPEG standards, music and 
standards, etc. Innovative applications, case studies tutorials,
demonstrators, etc. 

http://www.interactivemusicnetwork.org   

for the FINAL program see:

http://www.interactivemusicnetwork.org/events/Second_OpenWorkshop_2003/programme_openworkshop_2003.html

The workshop is colocated with the WEDELMUSIC conference
innovative aspects of multimedia, DRM technologies, 

http://www.wedelmusic.org/wedelmusic2003/index.html

The final program of the WEDELMUSIC conference will be ready in
the next days. The general arrangements and schedule of 
all the events is reported in the above page. 

The registration pages are available on line for both events,
the registration for the MUSICNETWORK is free of charge. 

cheers,
Paolo Nesi
(MUSICNETWORK, Thematic Network of the European Commission)
Kia Ng
(General Chair of WEDELMUSIC2003 conference, UNIV. Leeds)



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Fri Jul 18 06:45:31 2003
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From: Henning Schulzrinne <hgs@cs.columbia.edu>
Organization: Columbia University
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Subject: [AVT] DVI ADPCM documentation
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Does anybody have a copy or a reference to msdn.microsoft.com for this 
document? (The organization that published the document is defunct and 
we'd like to have a reasonable reference in the new RTP profile.)

% A specification titled ``DVI ADPCM Wave
% Type'' can also be found in the \emph{Microsoft Developer Network
% Development Library} CD ROM published quarterly by Microsoft.  The
% relevant section is found under Product Documentation, SDKs,
% Multimedia Standards Update, New Multimedia Data Types and Data
% Techniques, Revision 3.0, April 15, 1994.

Thank you.



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Kumar Premendra (IFIN DC COM)" <Premendra.Kumar@infineon.com>
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Hello,

I have some doubt while implementing RTP/RTCP....please clarify my doubt if
possible.

Q.  We must send RTCP packet  in compound packet format ,Always ?
 If not, how do we identify whether received packet is  normal RTCP packet
or compound packet.


Thanks in advance
prem


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Hi,

RTCP packets are always compound packets. A RTCP compound packet must 
consist of at least RR and SDES or SR and SDES packets. Other packet 
types may also be present.

I hope that you are using draft-ietf-avt-rtp-new-12 when implementing.

Best Regards

Magnus


Kumar Premendra (IFIN DC COM) wrote:

> 
> Hello,
> 
> I have some doubt while implementing RTP/RTCP....please clarify my doubt if
> possible.
> 
> Q.  We must send RTCP packet  in compound packet format ,Always ?
>  If not, how do we identify whether received packet is  normal RTCP packet
> or compound packet.
> 
> 
> Thanks in advance
> prem
> 
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
> 

-- 

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Prem:

see below.

- thomas



-----Original Message-----
From: Kumar Premendra (IFIN DC COM)
To: avt@ietf.org
Sent: 22.07.2003 01:20
Subject: [AVT] (no subject)



Hello,

I have some doubt while implementing RTP/RTCP....please clarify my doubt
if
possible.

Q.  We must send RTCP packet  in compound packet format ,Always ?
 If not, how do we identify whether received packet is  normal RTCP
packet
or compound packet.

[tz] based on size of each rtcp packet, as well as overall size of the UDP
packet, you can tell whether you are getting a correctly formated RTCP
compound packet. If it is not a compound RTCP packet, it is ill-formated.


Thanks in advance
prem


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Wed Jul 23 03:08:37 2003
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Date: Wed, 23 Jul 2003 00:14:43 -0700 (PDT)
From: Stephen Casner <casner@acm.org>
To: Tmima Koren <tmima@cisco.com>
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        <geevjohn@hotmail.com>, <brucet@cisco.com>, <pruddy@cisco.com>,
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Subject: [AVT] Re: Proposed change to Enhanced CRTP
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The change to the Enhanced CRTP draft proposed in the message enclosed
below has received no objections on the AVT mailing list.  The change
was also discussed in the AVT session at the IETF meeting just
concluded in Vienna, where comments in favor were received.
Therefore, we consider this change request to be appropriately
reviewed by the working group and accepted.

                                                        -- Steve

On Fri, 11 Jul 2003, Tmima Koren wrote:

> We'd like to propose a change to the Enhanced CRTP draft,
> draft-ietf-avt-crtp-enhance-07.txt
> http://www.ietf.org/internet-drafts/draft-ietf-avt-crtp-enhance-07.txt
>
> One of the enhancements is to add a Headers Checksum (HDRCKSUM) when UDP
> checksum is not present (see section 2.2).
> The UDP checksum is used by the decompressor to periodically validate the
> reconstructed packets. When not present,
> a HDRCKSUM may be inserted by the compressor replacing the missing UDP
> checksum. The decompressor removes the HDRCKSUM after validating the
> reconstructed packet.
>
> Currently the HDRCKSUM covers the pseudo-IP header, the UDP header and the
> fixed part of the RTP header (first 12 bytes).
> Those are the only parts of the reconstructed packet that can be validated
> by the HDRCKSUM.
> The CSRC list is not included in the HDRCKSUM.
>
> We propose to include the CSRC list in the HDRCKSUM since the CSRC list is
> part of the header compression context.
>
> The CSRC list is stored at the decompressor context and inserted to the
> restored packets. If the HDRCKSUM does not cover the CSRC list and the CSRC
> list at the decompressor is errored, a bad CSRC list will be restored into
> multiple packets until the CSRC
> list changes and is resent by the compressor as an update.
> If the HDRCKSUM covers the CSRC list, an error in the CSRC list will be
> caught early and corrected.
>
> When there is a change in the CSRC list, the CSRC list is sent as an update
> to the context.
> The CSRC list should be treated in the same way as the other context
> updating changes and be included in the HDRCKSUM.
>
> On the other hand, adding the CSRC list to the HDRCKSUM requires more
> processing calculating the HDRCKSUM. Also if
> the accuracy of the CSRC list is not crucial and the CSRC list is included
> in the HDRCKSUM, then an error in the CSRC list
> will result in dropping a few packets while the contexts are getting
> synchronized.
>
>
> The proposed change to the text is as follows:
>
> Old:
>
>     The HDRCKSUM is calculated in the same way as a UDP checksum except
>     that it does not cover all of the UDP data.  That is, the HDRCKSUM is
>     the 16-bit one's complement of the one's complement sum of the
>     pseudo-IP header (as defined for UDP), the UDP header, and the first
>     12 bytes of the UDP data which are assumed to hold the fixed part of
>     an RTP header.  The extended part of the RTP header and the RTP data
>     will not be included in the HDRCKSUM.
>
> New:
>
>     The HDRCKSUM is calculated in the same way as a UDP checksum except
>     that it does not cover all of the UDP data.  That is, the HDRCKSUM is
>     the 16-bit one's complement of the one's complement sum of the
>     pseudo-IP header (as defined for UDP), the UDP header,     the first
>                                                            ^^^^
>     12 bytes of the UDP data which are assumed to hold the fixed part of
>     an RTP header, and the CSRC list.
>                  ^^^^^^^^^^^^^^^^^^^
>     The extended part of the RTP header beyond the CSRC list and the RTP data
>                                         ^^^^^^^^^^^^^^^^^^^^
>     will not be included in the HDRCKSUM.
>


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: [AVT] Frame Interleaving in AMR profile->RFC 3267
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Dear All
The frame interleaving in the AMR profile (RFC 3267) states that

" The sender of the payload MUST only apply interleaving if the
   receiver has signalled its use through out-of-band means.  Since
   interleaving will increase buffering requirements at the receiver,
   the receiver uses MIME parameter "interleaving=I" to set the maximum
   number of frame-blocks allowed in an interleaving group to I.

   When performing interleaving the sender MUST use a proper number of
   frame-blocks per payload (N) and ILL so that the resulting size of an
   interleave group is less or equal to I, i.e., N*(L+1)<=I. "

ILL is the interleaving Length i.e. number of RTP packets across which 
interleaving happens.
NOTE: The value of ILL starts from 0. Hence, one needs to do ILL +1 in 
the above formula.

Nmax, the maximum value N (number of frame-blocks per payload) can take, 
is calculated by the formula ->
Nmax = (RTP MTU SIZE)/(The frame size of a frame block)

Assuming,

RTP MTU SIZE = 1500 bytes, and
The frame size of a AMR frame block = 13 octets for AMR frame type=0

The above assumption gives:
Nmax = 1500/13 = 115 packets.

The information obtained from out-of-band means, say, SDP is 
"interleaving=30", which means that an interleaving group can have a 
maximum of 30 frames per group such that:
(ILL+1) * N less than or equal to I  
        where    N varies from 0<N<Nmax.
                      ILL is the interleaving Length i.e. number of RTP 
packets across which interleaving happens.

Now, in the case taken by me above, Nmax > I   (i.e. 115>30). So, the 
condition (ILL+1)*N less than or equal to I with N=Nmax doesn't hold good.

So, how does one select the value of N and ILL such that it is less than 
I, in the above case?

Right now, I can think of only one option of handling this:
My option:  Assume the minimum value of ILL, say 1. A value of 2 makes 
sense because one needs at least 2 RTP packets to do interleaving. Based 
on this value calculate N
N = (30/2)-1
N = 14
With this value of N we satisfy both the conditions, namely,
(ILL+1) * N less than or equal to I  -> (2+1)
and
N varies from 0<N<Nmax.

I wanted to ask the others on this list whether this solution is the 
only solution? Or there are other ways of handling this?
Shouldn't the values of N and ILL be very apparent rather than AMR 
profile designer making such assumptions about it?

-- 
Prashanth B.P.
Sr. Design Engr.
NeoMagic Design Center.
Polyplex Building, First Floor
B-37 Sector 1
Noida 
INDIA 201301
E-Mail:  bprashanth@neomagic.com
Tel:     +91-120-2536208
Fax:     +91-120-2536207


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Dear All<br>
The frame interleaving in the AMR profile (RFC 3267) states that <br>
<br>
" The sender of the payload MUST only apply interleaving if the<br>
&nbsp;&nbsp; receiver has signalled its use through out-of-band means.&nbsp; Since<br>
&nbsp;&nbsp; interleaving will increase buffering requirements at the receiver,<br>
&nbsp;&nbsp; the receiver uses MIME parameter "interleaving=I" to set the maximum<br>
&nbsp;&nbsp; number of frame-blocks allowed in an interleaving group to I.<br>
<br>
&nbsp;&nbsp; When performing interleaving the sender MUST use a proper number of<br>
&nbsp;&nbsp; frame-blocks per payload (N) and ILL so that the resulting size of an<br>
&nbsp;&nbsp; interleave group is less or equal to I, i.e., N*(L+1)&lt;=I. "<br>
<br>
ILL is the interleaving Length i.e. number of RTP packets across which interleaving
happens. <br>
<u>NOTE</u>: The value of ILL starts from 0. Hence, one needs to do ILL +1
in the above formula.<br>
<br>
Nmax, the maximum value N (number of frame-blocks per payload) can take,
is calculated by the formula -&gt;<br>
Nmax = (RTP MTU SIZE)/(The frame size of a frame block)<br>
<br>
Assuming,<br>
<br>
RTP MTU SIZE = 1500 bytes, and<br>
The frame size of a AMR frame block = 13 octets for AMR frame type=0<br>
<br>
The above assumption gives: <br>
Nmax = 1500/13 = 115 packets.<br>
<br>
The information obtained from out-of-band means, say, SDP is "interleaving=30",
which means that an interleaving group can have a maximum of 30 frames per
group such that:<br>
(ILL+1) * N less than or equal to I &nbsp;<br>
&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; where&nbsp;&nbsp; &nbsp;N varies from 0&lt;N&lt;Nmax.<br>
&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp; ILL is the interleaving Length i.e. number of RTP packets
across which interleaving happens.<br>
<br>
Now, in the case taken by me above, Nmax &gt; I &nbsp; (i.e. 115&gt;30). So, the
condition (ILL+1)*N less than or equal to I with N=Nmax doesn't hold good.<br>
<br>
<u>So, how does one select the value of N and ILL such that it is less than
I, in the above case? </u><br>
<br>
Right now, I can think of only one option of handling this:<br>
<u>My option</u>: &nbsp;Assume the minimum value of ILL, say 1. A value of 2 makes
sense because one needs at least 2 RTP packets to do interleaving. Based
on this value calculate N <br>
N = (30/2)-1<br>
N = 14<br>
With this value of N we satisfy both the conditions, namely,<br>
(ILL+1) * N less than or equal to I &nbsp;-&gt; (2+1)<br>
and<br>
N varies from 0&lt;N&lt;Nmax.<br>
<br>
I wanted to ask the others on this list whether this solution is the only
solution? Or there are other ways of handling this?<br>
Shouldn't the values of N and ILL be very apparent rather than AMR profile
designer making such assumptions about it?
<pre class="moz-signature" cols="$mailwrapcol">-- 
Prashanth B.P.
Sr. Design Engr.
NeoMagic Design Center.
Polyplex Building, First Floor
B-37 Sector 1
Noida 
INDIA 201301
E-Mail:  <a class="moz-txt-link-abbreviated" href="mailto:bprashanth@neomagic.com">bprashanth@neomagic.com</a>
Tel:     +91-120-2536208
Fax:     +91-120-2536207</pre>
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To: Stephen Casner <casner@acm.org>,
        Magnus Westerlund
	 <magnus.westerlund@ericsson.com>
Cc: IETF AVT WG <avt@ietf.org>
Subject: RE: [AVT] Re: Errors in draft-ietf-avt-rtcp-bw-05.txt
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Hi,

Just for clarification, in Appendix A.7 of RFC 1889 bis Internet draft have
the following changes been agreed to (or are these changes not required per
the agreed changes to the RTCP bandwidth modifier Internet draft)?

(1) "RTCP_RCVR_BW_FRACTION = 0.25"
changed to
"RTCP_RCVR_BW_FRACTION = rtcp_bw_senders/(rtcp_bw_senders +
rtcp_bw_receivers)"

and

(2) "senders > 0" removed?

Thanks.

Regards,

Pete

-----Original Message-----
From: Jose Rey [mailto:rey@panasonic.de]
Sent: Monday, June 16, 2003 6:32 AM
To: Stephen Casner; Magnus Westerlund
Cc: IETF AVT WG
Subject: RE: [AVT] Re: Errors in draft-ietf-avt-rtcp-bw-05.txt


Hi,

> -----Original Message-----
> From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of Stephen
> Casner
> Sent: Monday, June 16, 2003 8:47 AM
> To: Magnus Westerlund; Jose Rey
> Cc: IETF AVT WG
> Subject: Re: [AVT] Re: Errors in draft-ietf-avt-rtcp-bw-05.txt
>
>
> Some more thoughts on this:
>
> To simplify the unicast streaming application, we could specify that
> when RTCP bw is specified with RS and RR those values are used
> separately for senders and non-senders without any consideration of
> treating senders and receivers together when the number of senders is
> greater than RS/(RS+RR).

I think this is the best and easiest solution. In 3GPP unicast streaming it
is
the server that sets the RS and RR values, that means the server should
consider
well how much bandwidth it is needed for own reporting and for the sender's.
There's no case where the algorithm must adapt, the scenario is rather
static.

>
> However, that would not be good for conversational applications
> (unicast or multicast) because the classification as sender changes
> over time.  At times when both ends have sent within two intervals,
> they would be using the RS bw only and ignoring the RR bw.  To get
> around this we could set RS to be the total, and RR zero, but then if
> one was sending and the other just listening for a long time, the
> listener would get nothing.  For these applications, you need the
> smooth transision that the existing algorithm was intended to
> provide.
>
I see. I was referring more to the streaming unicast scenario. I guess my
characterization was not accure enough.

cheers,

Jose

>                                                         -- Steve
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt

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

<P><FONT SIZE=3D2>Just for clarification, in Appendix A.7 of RFC 1889 =
bis Internet draft have the following changes been agreed to (or are =
these changes not required per the agreed changes to the RTCP bandwidth =
modifier Internet draft)?</FONT></P>

<P><FONT SIZE=3D2>(1) &quot;RTCP_RCVR_BW_FRACTION =3D 0.25&quot;</FONT>
<BR><FONT SIZE=3D2>changed to</FONT>
<BR><FONT SIZE=3D2>&quot;RTCP_RCVR_BW_FRACTION =3D =
rtcp_bw_senders/(rtcp_bw_senders + rtcp_bw_receivers)&quot;</FONT>
</P>

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

<P><FONT SIZE=3D2>(2) &quot;senders &gt; 0&quot; removed?</FONT>
</P>

<P><FONT SIZE=3D2>Thanks.</FONT>
</P>

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

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

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Jose Rey [<A =
HREF=3D"mailto:rey@panasonic.de">mailto:rey@panasonic.de</A>]</FONT>
<BR><FONT SIZE=3D2>Sent: Monday, June 16, 2003 6:32 AM</FONT>
<BR><FONT SIZE=3D2>To: Stephen Casner; Magnus Westerlund</FONT>
<BR><FONT SIZE=3D2>Cc: IETF AVT WG</FONT>
<BR><FONT SIZE=3D2>Subject: RE: [AVT] Re: Errors in =
draft-ietf-avt-rtcp-bw-05.txt</FONT>
</P>
<BR>

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

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: avt-admin@ietf.org [<A =
HREF=3D"mailto:avt-admin@ietf.org">mailto:avt-admin@ietf.org</A>]On =
Behalf Of Stephen</FONT>
<BR><FONT SIZE=3D2>&gt; Casner</FONT>
<BR><FONT SIZE=3D2>&gt; Sent: Monday, June 16, 2003 8:47 AM</FONT>
<BR><FONT SIZE=3D2>&gt; To: Magnus Westerlund; Jose Rey</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: IETF AVT WG</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: Re: [AVT] Re: Errors in =
draft-ietf-avt-rtcp-bw-05.txt</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; Some more thoughts on this:</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; To simplify the unicast streaming application, =
we could specify that</FONT>
<BR><FONT SIZE=3D2>&gt; when RTCP bw is specified with RS and RR those =
values are used</FONT>
<BR><FONT SIZE=3D2>&gt; separately for senders and non-senders without =
any consideration of</FONT>
<BR><FONT SIZE=3D2>&gt; treating senders and receivers together when =
the number of senders is</FONT>
<BR><FONT SIZE=3D2>&gt; greater than RS/(RS+RR).</FONT>
</P>

<P><FONT SIZE=3D2>I think this is the best and easiest solution. In =
3GPP unicast streaming it is</FONT>
<BR><FONT SIZE=3D2>the server that sets the RS and RR values, that =
means the server should consider</FONT>
<BR><FONT SIZE=3D2>well how much bandwidth it is needed for own =
reporting and for the sender's.</FONT>
<BR><FONT SIZE=3D2>There's no case where the algorithm must adapt, the =
scenario is rather static.</FONT>
</P>

<P><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; However, that would not be good for =
conversational applications</FONT>
<BR><FONT SIZE=3D2>&gt; (unicast or multicast) because the =
classification as sender changes</FONT>
<BR><FONT SIZE=3D2>&gt; over time.&nbsp; At times when both ends have =
sent within two intervals,</FONT>
<BR><FONT SIZE=3D2>&gt; they would be using the RS bw only and ignoring =
the RR bw.&nbsp; To get</FONT>
<BR><FONT SIZE=3D2>&gt; around this we could set RS to be the total, =
and RR zero, but then if</FONT>
<BR><FONT SIZE=3D2>&gt; one was sending and the other just listening =
for a long time, the</FONT>
<BR><FONT SIZE=3D2>&gt; listener would get nothing.&nbsp; For these =
applications, you need the</FONT>
<BR><FONT SIZE=3D2>&gt; smooth transision that the existing algorithm =
was intended to</FONT>
<BR><FONT SIZE=3D2>&gt; provide.</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>I see. I was referring more to the streaming unicast =
scenario. I guess my</FONT>
<BR><FONT SIZE=3D2>characterization was not accure enough.</FONT>
</P>

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

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

<P><FONT =
SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; -- =
Steve</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; Audio/Video Transport Working Group</FONT>
<BR><FONT SIZE=3D2>&gt; avt@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/avt" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/avt</A></FONT>
</P>

<P><FONT =
SIZE=3D2>_______________________________________________</FONT>
<BR><FONT SIZE=3D2>Audio/Video Transport Working Group</FONT>
<BR><FONT SIZE=3D2>avt@ietf.org</FONT>
<BR><FONT SIZE=3D2><A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/avt" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/avt</A></FONT>
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From exim@www1.ietf.org  Wed Jul 23 14:43:33 2003
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cc: Magnus Westerlund <magnus.westerlund@ericsson.com>,
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Peter,

> Just for clarification, in Appendix A.7 of RFC 1889 bis Internet draft have
> the following changes been agreed to (or are these changes not required per
> the agreed changes to the RTCP bandwidth modifier Internet draft)?
>
> (1) "RTCP_RCVR_BW_FRACTION = 0.25"
> changed to
> "RTCP_RCVR_BW_FRACTION = rtcp_bw_senders/(rtcp_bw_senders +
> rtcp_bw_receivers)"
>
> and
>
> (2) "senders > 0" removed?

Those changes have logically been agreed, but the edits are not
exactly as you state.  In Section 6.2, the proportion of senders has
been changed from 1/4 to S/(S+R).  In Section 6.3.1 item 1, the
"senders > 0" clause has been removed, and the following text added at
the end:

      As stated in Section 6.2, an RTP profile MAY specify that the
      RTCP bandwidth may be explicitly defined by two separate
      parameters (call them S and R) for those participants which are
      senders and those which are not.  In that case, the 25% fraction
      becomes S/(S+R) and the 75% fraction becomes R/(S+R).  Note that
      if R is zero, the percentage of senders is never greater than
      S/(S+R), and the implementation must avoid division by zero.

In Appendix A.7, the "senders < 0" clause has been removed and the
inequality is changed from "<" to "<=" so that the inequality is
always true when RTCP_SENDER_BW_FRACTION is 1.0:

       if (senders <= members * RTCP_SENDER_BW_FRACTION) {

However, your change (1) above has not been made because I did not
want to make modifications to the extent that would be required to add
rtcp_bw_senders and rtcp_bw_receivers as arguments to the function and
all calls to it.  Instead, I added the following paragraph just before
the rtcp_interval() function:

   These functions would have to be extended for an implementation
   that allows the RTCP bandwidth fractions for senders and
   non-senders to be specified as explict parameters rather than fixed
   values of 25% and 75%.  The extended implementation of
   rtcp_interval() would need to avoid division by zero if one of the
   parameters was zero.

The full RFCs should be released within a few days.

                                                        -- Steve


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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This draft is a work item of the Audio/Video Transport Working Group of the IETF.

	Title		: The Secure Real-time Transport Protocol
	Author(s)	: M. Baugher et al.
	Filename	: draft-ietf-avt-srtp-09.txt
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	Date		: 2003-7-23
	
This document describes the Secure Real-time Transport Protocol
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_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Thu Jul 24 03:44:10 2003
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To: B Prashanth <bprashanth@neomagic.com>
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Subject: Re: [AVT] Frame Interleaving in AMR profile->RFC 3267
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Hi,

When it comes to meeting the limitations the sender side has the freedom 
to select any combination of ILL and number of frames that satisfies the 
restrictions. Another restriction then I that may be used is "maxptime" 
which puts a limit on the number of frames per packet.

In your example with I=30 the sender has the choice to go from ILL=0 (No 
interleaving, with 30 frames in a single RTP packet, up to ILL=14 and 
only two frames per packet. It is assumed that the sender will use the 
combination most suitable for the current network performance and 
congestion state.

There is no need for the payload format to put in more limitations. The 
RTP payload format has great flexibility and for any receiver 
implementing the interleaving mode can decode these payloads independent 
  of which ratio between ILL and N that are used. The I restriction 
value is there to put a limit on buffering and interleaving induced 
delay. However using many AMR frames per packet is generally not a good 
idea. If one would use the 130 frames per packet that you calculate, I 
hope you are aware that this represent 2.6 seconds. That can be a whole 
sentence lost if this single packet is lost. So filling voice packet to 
the limit is not suitable in environments where losses occur. However if 
one has retransmission or uses a reliable channel more frames per 
packets may give satisfactory performance. My opinion is that more than 
5-10 frames per packet is to much. And if one is going as high as 10 
frames per packet one should definitely consider using interleaving with 
fairly decent length.


[Chairs hat on]

RFC 3267 is a payload format NOT a profile. A RTP profile is something 
else and must not be confused with payload formats. I have seen this 
mistake being done also by others. The current available profiles are 
RFC 1890 (AVP) or rather draft-ietf-avt-profile-new-13. Two new profiles 
are with the IESG, SRTP (SAVP) draft-ietf-avt-srtp-09 and Extended RTCP 
feedback (AVPF) draft-ietf-avt-rtcp-feedback-07.

[chairs hat off]

Best Regards


Magnus


B Prashanth wrote:

> Dear All
> The frame interleaving in the AMR profile (RFC 3267) states that
> 
> " The sender of the payload MUST only apply interleaving if the
>    receiver has signalled its use through out-of-band means.  Since
>    interleaving will increase buffering requirements at the receiver,
>    the receiver uses MIME parameter "interleaving=I" to set the maximum
>    number of frame-blocks allowed in an interleaving group to I.
> 
>    When performing interleaving the sender MUST use a proper number of
>    frame-blocks per payload (N) and ILL so that the resulting size of an
>    interleave group is less or equal to I, i.e., N*(L+1)<=I. "
> 
> ILL is the interleaving Length i.e. number of RTP packets across which 
> interleaving happens.
> NOTE: The value of ILL starts from 0. Hence, one needs to do ILL +1 in 
> the above formula.
> 
> Nmax, the maximum value N (number of frame-blocks per payload) can take, 
> is calculated by the formula ->
> Nmax = (RTP MTU SIZE)/(The frame size of a frame block)
> 
> Assuming,
> 
> RTP MTU SIZE = 1500 bytes, and
> The frame size of a AMR frame block = 13 octets for AMR frame type=0
> 
> The above assumption gives:
> Nmax = 1500/13 = 115 packets.
> 
> The information obtained from out-of-band means, say, SDP is 
> "interleaving=30", which means that an interleaving group can have a 
> maximum of 30 frames per group such that:
> (ILL+1) * N less than or equal to I  
>         where    N varies from 0<N<Nmax.
>                       ILL is the interleaving Length i.e. number of RTP 
> packets across which interleaving happens.
> 
> Now, in the case taken by me above, Nmax > I   (i.e. 115>30). So, the 
> condition (ILL+1)*N less than or equal to I with N=Nmax doesn't hold good.
> 
> So, how does one select the value of N and ILL such that it is less than 
> I, in the above case?
> 
> Right now, I can think of only one option of handling this:
> My option:  Assume the minimum value of ILL, say 1. A value of 2 makes 
> sense because one needs at least 2 RTP packets to do interleaving. Based 
> on this value calculate N
> N = (30/2)-1
> N = 14
> With this value of N we satisfy both the conditions, namely,
> (ILL+1) * N less than or equal to I  -> (2+1)
> and
> N varies from 0<N<Nmax.
> 
> I wanted to ask the others on this list whether this solution is the 
> only solution? Or there are other ways of handling this?
> Shouldn't the values of N and ILL be very apparent rather than AMR 
> profile designer making such assumptions about it?
> 
> -- 
> Prashanth B.P.
> Sr. Design Engr.
> NeoMagic Design Center.
> Polyplex Building, First Floor
> B-37 Sector 1
> Noida 
> INDIA 201301
> E-Mail:  bprashanth@neomagic.com
> Tel:     +91-120-2536208
> Fax:     +91-120-2536207
> 

-- 

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Thu Jul 24 06:02:12 2003
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Subject: Re: [AVT] Frame Interleaving in AMR profile->RFC 3267
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Hi Magnus!

    Thanks for replying in detail. It has cleared lot of my doubts.

Regards,
Prashanth

Magnus Westerlund wrote:

> Hi,
>
> When it comes to meeting the limitations the sender side has the 
> freedom to select any combination of ILL and number of frames that 
> satisfies the restrictions. Another restriction then I that may be 
> used is "maxptime" which puts a limit on the number of frames per packet.
>
> In your example with I=30 the sender has the choice to go from ILL=0 
> (No interleaving, with 30 frames in a single RTP packet, up to ILL=14 
> and only two frames per packet. It is assumed that the sender will use 
> the combination most suitable for the current network performance and 
> congestion state.
>
> There is no need for the payload format to put in more limitations. 
> The RTP payload format has great flexibility and for any receiver 
> implementing the interleaving mode can decode these payloads 
> independent  of which ratio between ILL and N that are used. The I 
> restriction value is there to put a limit on buffering and 
> interleaving induced delay. However using many AMR frames per packet 
> is generally not a good idea. If one would use the 130 frames per 
> packet that you calculate, I hope you are aware that this represent 
> 2.6 seconds. That can be a whole sentence lost if this single packet 
> is lost. So filling voice packet to the limit is not suitable in 
> environments where losses occur. However if one has retransmission or 
> uses a reliable channel more frames per packets may give satisfactory 
> performance. My opinion is that more than 5-10 frames per packet is to 
> much. And if one is going as high as 10 frames per packet one should 
> definitely consider using interleaving with fairly decent length.
>
>
> [Chairs hat on]
>
> RFC 3267 is a payload format NOT a profile. A RTP profile is something 
> else and must not be confused with payload formats. I have seen this 
> mistake being done also by others. The current available profiles are 
> RFC 1890 (AVP) or rather draft-ietf-avt-profile-new-13. Two new 
> profiles are with the IESG, SRTP (SAVP) draft-ietf-avt-srtp-09 and 
> Extended RTCP feedback (AVPF) draft-ietf-avt-rtcp-feedback-07.
>
> [chairs hat off]
>
> Best Regards
>
>
> Magnus
>
>
> B Prashanth wrote:
>
>> Dear All
>> The frame interleaving in the AMR profile (RFC 3267) states that
>>
>> " The sender of the payload MUST only apply interleaving if the
>>    receiver has signalled its use through out-of-band means.  Since
>>    interleaving will increase buffering requirements at the receiver,
>>    the receiver uses MIME parameter "interleaving=I" to set the maximum
>>    number of frame-blocks allowed in an interleaving group to I.
>>
>>    When performing interleaving the sender MUST use a proper number of
>>    frame-blocks per payload (N) and ILL so that the resulting size of an
>>    interleave group is less or equal to I, i.e., N*(L+1)<=I. "
>>
>> ILL is the interleaving Length i.e. number of RTP packets across 
>> which interleaving happens.
>> NOTE: The value of ILL starts from 0. Hence, one needs to do ILL +1 
>> in the above formula.
>>
>> Nmax, the maximum value N (number of frame-blocks per payload) can 
>> take, is calculated by the formula ->
>> Nmax = (RTP MTU SIZE)/(The frame size of a frame block)
>>
>> Assuming,
>>
>> RTP MTU SIZE = 1500 bytes, and
>> The frame size of a AMR frame block = 13 octets for AMR frame type=0
>>
>> The above assumption gives:
>> Nmax = 1500/13 = 115 packets.
>>
>> The information obtained from out-of-band means, say, SDP is 
>> "interleaving=30", which means that an interleaving group can have a 
>> maximum of 30 frames per group such that:
>> (ILL+1) * N less than or equal to I          where    N varies from 
>> 0<N<Nmax.
>>                       ILL is the interleaving Length i.e. number of 
>> RTP packets across which interleaving happens.
>>
>> Now, in the case taken by me above, Nmax > I   (i.e. 115>30). So, the 
>> condition (ILL+1)*N less than or equal to I with N=Nmax doesn't hold 
>> good.
>>
>> So, how does one select the value of N and ILL such that it is less 
>> than I, in the above case?
>>
>> Right now, I can think of only one option of handling this:
>> My option:  Assume the minimum value of ILL, say 1. A value of 2 
>> makes sense because one needs at least 2 RTP packets to do 
>> interleaving. Based on this value calculate N
>> N = (30/2)-1
>> N = 14
>> With this value of N we satisfy both the conditions, namely,
>> (ILL+1) * N less than or equal to I  -> (2+1)
>> and
>> N varies from 0<N<Nmax.
>>
>> I wanted to ask the others on this list whether this solution is the 
>> only solution? Or there are other ways of handling this?
>> Shouldn't the values of N and ILL be very apparent rather than AMR 
>> profile designer making such assumptions about it?
>>
>> -- 
>> Prashanth B.P.
>> Sr. Design Engr.
>> NeoMagic Design Center.
>> Polyplex Building, First Floor
>> B-37 Sector 1
>> Noida INDIA 201301
>> E-Mail:  bprashanth@neomagic.com
>> Tel:     +91-120-2536208
>> Fax:     +91-120-2536207
>>
>




_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.

	Title		: The Secure Real-time Transport Protocol
	Author(s)	: M. Baugher et al.
	Filename	: draft-ietf-avt-srtp-09.txt
	Pages		: 56
	Date		: 2003-7-23
	
This document describes the Secure Real-time Transport Protocol
(SRTP), a profile of the Real-time Transport Protocol (RTP), which
can provide confidentiality, message authentication, and replay
protection to the RTP traffic and to the control traffic for RTP,
the Real-time Transport Control Protocol (RTCP).

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avt-srtp-09.txt

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Internet-Drafts can also be obtained by e-mail.

Send a message to:
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NOTE:	The mail server at ietf.org can return the document in
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	feature, insert the command "ENCODING mime" before the "FILE"
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	a MIME-compliant mail reader.  Different MIME-compliant mail readers
	exhibit different behavior, especially when dealing with
	"multipart" MIME messages (i.e. documents which have been split
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implementation to automatically retrieve the ASCII version of the
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_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Thu Jul 24 11:38:16 2003
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Subject: [AVT] Retransmission draft
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Hi all,

for your information:

Matsushita has sent an IPR disclosure to the IETF IPR website regarding
patent rights applicable to the draft-ietf-rtp-retransmission-08.txt.

More information can be found under http://www.ietf.org/ipr


cheers,

Jos=E9 Rey____ mailto:rey@panasonic.de
Mobile Multimedia Team -  MCOM Group
Panasonic European Laboratories GmbH
Tel: +49(0)6103-766-134
Fax: +49(0)6103-766-166
http://www.pel.panasonic.de
____________________________________


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Thu Jul 24 14:13:15 2003
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From: The IESG <iesg-secretary@ietf.org>
Date: Thu, 24 Jul 2003 14:11:21 -0400
Subject: [AVT] Protocol Action: RTP Control Protocol Extended Reports (RTCP XR) to
 Proposed Standard
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The IESG has approved the Internet-Draft 'RTP Control Protocol
Extended Reports (RTCP XR)' <draft-ietf-avt-rtcp-report-extns-06.txt>
as a Proposed Standard. This document is the product of the
Audio/Video Transport Working Group. 

The IESG contact persons are A. Mankin and J. Peterson

Technical Summary

      This document defines the Extended Report (XR) packet type for the
      RTP Control Protocol (RTCP), and defines how the use of XR packets
      can be signaled by an application if it employs the Session
      Description Protocol (SDP). XR packets are composed of report
      blocks, and seven block types are defined here. The purpose of the
      extended reporting format is to convey information that supplements
      the six statistics that are contained in the report blocks used by
      RTCP's Sender Report (SR) and Receiver Report (RR) packets. Some
      applications, such as multicast inference of network characteristics
      (MINC) or voice over IP (VoIP) monitoring, require other and more
      detailed statistics. In addition to the block types defined here,
      additional block types may be defined in the future by adhering to
      the framework that this document provides. Security considerations
      including the use of Secure RTCP to overcome the heightened amount of
      information provided by the XR packet type are described.

Working Group Summary

      The working group divided for a time on whether to include more complex
      types of monitoring in the initial seven block types, but the resolution
      on the framework achieved a very good consensus.

Protocol Quality

      Prototype implementations exist of the XR packet type and have been
      discussed in reviewing the specification.

      The review of the specification was very active due to strong interest
      by a variety of communities.

      The review for the IESG was done by Allison Mankin.


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Thu Jul 24 17:23:09 2003
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Subject: [AVT] timestamp
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Hello,
can u help me out on how to actually generate and implement the RTP 
timestamp for mpeg1.
Thank you
sitesh

_________________________________________________________________
Attention NRIs. Send money home. 
http://server1.msn.co.in/msnleads/citibankrca/citibankrca2.asp?type=txt Do 
it in a jiffy!


_______________________________________________
Audio/Video Transport Working Group
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https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Fri Jul 25 10:30:17 2003
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Date: Fri, 25 Jul 2003 16:27:54 +0200
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Subject: [AVT] Comments on draft-chen-rtp-bv-00.txt
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Hi,

Here are some comments on this draft. Before this I would like to say 
thank you to the authors for providing a very good initial version of 
the draft. I have very few comments.

1. There is need to tighten up the use of aggregation of consecutive 
frames. I would propose that in section 3.2 and 4.2 add a sentence 
stating that any frames in payload MUST be consecutive. If that is not 
intended there is need for having some mechanism for recovering the 
timestamp for each individual frame.

2. Section 3 and 4 are both missing specification for how to use the 
marker bit. I would suggest the normal definition that for packets with 
there first frame being an speech onset the marker bit is set to 1, 
otherwise 0. Please see draft-ietf-avt-profile-new-13.txt

3. I would also like to see some more text on how one recovers the RTP 
timestamp for an aggregated frame. I now that as long as you only intend 
to aggregate consecutive frames it will be RTP TS + TS_rate*0.005*(n-1) 
however this should be written out.

4. I would also suggest that you tighten the language in Section 5 
regarding that BroadVoice 16 and 32 frames shall not be mixed in a file. 
Please use the suitable RFC 2119 language for this.

5. Section 6.1, the usage of maxptime. Are there any need to define a 
default value for maxptime and especially of selecting 200 ms?

6. Section 6.1, Public Specification. This needs to be a complete 
reference.

7. Section 7. I would suggest that you improve your text for how to map 
the parameters to SDP. I would suggest borrowing text from section 8.3 
of RFC 3267.

8. The reference list needs to be split into normative and informative 
references.


Best Regards


Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Hi everybody,

I will be on vacation for three weeks from now. I will be back the 18th 
of July. Any questions will have to be answered by Colin and Steve. But 
please cc me anyway so I can know what has happened.

Cheers

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Sat, 26 Jul 2003 16:37:16 +0100
From: Colin Perkins <csp@csperkins.org>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [AVT] Comments on draft-chen-rtp-bv-00.txt
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Hi,

--> Magnus Westerlund <magnus.westerlund@ericsson.com> writes:
> Here are some comments on this draft. Before this I would like to say 
> thank you to the authors for providing a very good initial version of 
> the draft. I have very few comments.
...[snip several good comments]...
> 5. Section 6.1, the usage of maxptime. Are there any need to define a 
> default value for maxptime and especially of selecting 200 ms?

The maxptime attribute can probably be defined by reference to draft-ietf-
mmusic-sdp-new-13.txt, rather than by copying the definition. 

> 6. Section 6.1, Public Specification. This needs to be a complete 
> reference.
> 
> 7. Section 7. I would suggest that you improve your text for how to map 
> the parameters to SDP. I would suggest borrowing text from section 8.3 
> of RFC 3267.
> 
> 8. The reference list needs to be split into normative and informative 
> references.

9. The mention, in section 2, that BroadVoice16 might be adopted in a
   future PacketCable standard is not appropriate, and will need to be
   removed or replaced by a reference to the published PacketCable
   standard before this draft can become an RFC (I made a similar comment
   to the iLBC authors during at meeting).

Cheers,
-- 
Colin Perkins
csp@csperkins.org

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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This is a multi-part message in MIME format.

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Folks,

Attached are draft minutes for the AVT meeting at the 57th IETF in Vienna.
Please send any comments or corrections to the chairs by 1st August.

-- 
Colin Perkins
csp@csperkins.org

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Audio/Video Transport Working Group Minutes

Reported by Magnus Westerlund and Colin Perkins=20

   The AVT working group met once at the 57th IETF in Vienna. During this
   session the working group discussed the updated charter, current status,
   VoIP congestion control concerns raised by the IAB, the SSM RTCP Extensi=
on
   and numerous RTP payload formats.=20


Introduction and document status

   The meeting started by Stephen Casner announcing that this will be his
   last meeting as AVT co-chair. Allison Mankin presented Stephen with a
   plaque enumerating the RFCs that Stephen has been shepherding during his
   time as chair for AVT since its start in 1992. The working group showed
   its appreciation for his long service with thundering applause.

   Stephen Casner outlined the current working group status. No RFCs
   relating to AVT have been published since the last meeting, however
   there are a currently 7 in the RFC editor's queue. The RTP specification
   is currently in author's 48 hours and has gone through one substantial
   change. The usage of the SDP RR and RS bandwidth modifier results in
   change to the RTCP interval calculation mechanism. An RTP active sender
   is allowed to take its part of the RR part when the number of senders
   are larger than RS/(RS+RR).  This is a generalisation of the previous
   25% rule that was present.  The senders=3D0 clause in the example
   algorithm (A.7) has also been removed to prevent senders bandwidth
   bleeding to receivers when RR=3D0. =20
  =20
   The other AVT drafts in the RFC editor's queue are the AVP profile, the
   RTP profile MIME registration, the SDP bandwidth modifier for RTCP, the
   RTP payload format for EVRC/SMV and the RTP payload format for ETSI ES
   201 108 DSR.

   The specification for Enhanced IP/UDP/RTP header compression is with the
   RFC Editor, but the authors would like to fix one technical shortcoming
   before publication. This can be done if the Area Director and Working
   Group approve of the change. The issue is to include the CSRC list in
   the calculation of the Headers Checksum (HDRCKSUM), to enable header
   validation and prevent error propagation in the CSRC list. The working
   group was given until 18th July (the end of the IETF week) to comment on
   this change. Positive comments were made during the meeting, and there
   were no objections on the mailing list.

   The working group also has a number of drafts in IESG review: Tunnelling
   multiplexed compressed RTP (TCRTP), Secure RTP (which has had some recen=
t=20
   modifications in response to AD review), the Payload format for MPEG-4,
   RTCP extended reports, RTCP feedback (draft-ietf-avt-rtcp-feedback-07)
   and associated simulations (draft-burmeister-avt-rtcp-feedback-sim-02),
   and RTP retransmission (draft-ietf-avt-rtp-retransmission-08).

   The drafts of Uneven Level Protection (draft-ietf-avt-ulp-07.txt) and
   Unequal erasure protection (draft-ietf-avt-uxp-05.txt) were not updated
   for this meeting.=20

   Colin Perkins noted that ITU-T SG16 Q.14 is working on standards for FAX
   over RTP. He expressed interest in increasing coordination between ITU-T
   and AVT to ensure that solutions developed do not conflict. He also
   noted that a MIME type registration must be done to ensure that the FAX
   over RTP work can be referenced by IETF protocols - such a registration
   should be coordinated with AVT.


Charter Update

   A preliminary updated WG charter with milestones was presented and has
   also be posted to the mailing list. Please review and send any comments
   to the list. The updated charter contains the following items:=20

    - Advance RTP and the A/V profile to Full standard RFC
    - Review and revise RTP payload formats for Draft Standard RFC, as
      appropriate
    - Develop new RTP payload formats as needed.=20
    - Complete the generic FEC formats (ULP & UXP)
    - Complete the RTCP extensions for SSM sessions=20
    - Develop a framing method for use of RTP/RTCP over TCP & TLS=20
    - Review applicability of RTP header compression for MPLS=20
    - Develop a new RTP profile combining AVPF and SAVP=20
    - Coordinate with the DCCP working group to ensure efficient transport
      of RTP over DCCP, and to ensure an appropriate API is developed

   Longer term goals of the working group are to move to Draft Standard=20
   the SRTP Profile, the Extended RTP Profile for RTCP-based Feedback,=20
   the Compressed RTP (CRTP) framework, including Enhanced CRTP, and the
   RTP MIB. Further development of media codecs is not expected to be
   chartered.=20


DCCP
http://www.ietf.org/html.charters/dccp-charter.html

   Aaron Falk made a short informational speech about DCCP and its upcoming
   design review. He encouraged people to read the specification, and come
   to the design review. He especially desired people in AVT to comment on
   any API related issues, as this relates to the possible usage of DCCP.=20


RTP Payload Format for MIDI
draft-ietf-avt-mwpp-midi-rtp-08.txt
draft-lazzaro-avt-mwpp-coding-guidelines-03.txt

   Colin Perkins outlined some slides from John Lazarro regarding the RTP
   payload format for MIDI. This format has been reviewed by the MMA (the
   MIDI Manufacturers Association) and their comments have been addressed.
   An implementation exists and will be updated in regards to the changes
   during the summer. It is expected that the draft will be ready for
   working group last call before the next meeting.=20

   The companion implementation guide for RTP MIDI has been updated to more
   concern the MIDI parts, like recovery journal, than to deal with general
   RTP things. John Lazzaro has requested that this document becomes a AVT
   working group document. A hum was performed with some indication for
   approval and none against. =20


RTP framing for TCP and TLS
draft-lazzaro-avt-rtp-framing-contrans-01.txt

   Colin Perkins updated the group on the RTP framing for TCP and TLS. This
   draft has been awaiting resolution of the MMUSIC WG comedia work and the
   rechartering. It can be expected to make progress once the new charter
   is accepted.=20


IAB Concerns Regarding VoIP congestion control=20
draft-iab-congestion-00.txt

   Sally Floyd made a short presentation on an initial draft of IAB
   recommendations regarding congestion control for Voice over IP flows.
   The basic recommendation is that any flow that is facing persistent high
   rates of packet loss should stop sending. Having congestion control of
   some type will ensure that congestion collapse does not happen, user
   quality can be maintained, and that fairness against other VoIP flows
   and TCP flows can be maintained in a best effort network.=20

   The draft does not recommend any particular deployment path for VoIP,
   like best effort, QoS reservations, etc. It also not specifying any
   behaviour, it is only a recommendation that there is a need to address
   these issues. It also tries to address some of the framing problems and
   what assumptions that can be made, like what is fairness, and is the
   network, bit-rate or processor limited.=20

   Several questions and comments where raised. One was the need to clarify
   what it means to terminate a connection: in this case stop sending media
   (not necessarily to terminate any signalling relationship). It was asked
   if RTCP information is sufficient for deciding when to terminate the data
   flow, which will need further investigation.=20

   Is was also asked why the draft is restricted to VoIP and doesn't addres=
s=20
   other media types? This is not because other media don't need congestion
   control for other media, rather it's a reaction to the special treatment
   often proposed for voice (e.g. because flows are low bandwidth). This
   special treatment is not considered appropriate.  It was also clarified
   that the document does not address congestion control for multicast.=20


RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals=20
draft-ietf-avt-rfc2833bis-03.txt

   Henning Schulzrinne presented the current status of the draft. No major
   technical changes have occurred, and it is now believed that all events
   have been tracked down (however, review from someone working in this
   area would be appreciated). The examples in the draft have also been
   improved. Henning believes that the document is ready for WG last call,
   however some more dedicated review would be beneficial. Tom Taylor and
   Flemming Andreasen volunteered to perform a review of the document. The
   goal is to have the draft ready for last call before or around the next
   IETF meeting.=20

   This draft was originally intended to become a Draft Standard RFC,
   however so far the interop testing has not progressed. Henning
   questioned the level of interoperability testing that will be needed: if
   every event needs to be tested that will be a futile effort as they are
   over a 100 in total, and from several different systems. Alternatives
   are:=20

    - interop testing only needs to show that the basic timing and redundan=
cy=20
      mechanisms works
    - to strip all events that can't be tested, which probably will result
      in only DTMF tones will be left
    - the last alternative is to leave this as a proposed standard for all
      future.=20

   The chairs will ask the area directors if basic timing and redundancy
   interoperability is sufficient for testing for draft standard. It was
   proposed that the draft be continued to be progressed to WG last call
   and be published as Proposed Standard RFC, while work continues on
   interoperability testing. The only consideration is if any of the
   interoperability will result in technical changes to the document,=20
   in that case it might be better to wait. It would be advantageous if
   someone actually doing implementation work using this format could be
   participate in the interoperability testing.=20


Payload format for Speex Codec=20
draft-herlein-speex-rtp-profile-01.txt

   Phil Kerr presented an update to the RTP payload format for Speex.  The
   format has received extensive comments on the mailing list that need to
   be addressed in next version. Comments in the meeting were regarding the
   usage of the RTP timestamp rate. There has been some consideration that
   a codec parameter for rate may be redundant, since the codec has some
   functionality for recovering sampling rate. Another alternative would be
   to use different RTP payload types and timestamp rates depending on
   codec rate, this is however not without problem as Stephen Casner
   pointed out. The issue needs further investigation. The recovery of the
   timestamp for different frames when performing aggregation also needs to
   be clarified.=20


Payload format for Vorbis=20
draft-kerr-avt-vorbis-rtp-02.txt

   Phil Kerr presented an update, where the main issue is related to the
   codebooks that the codec uses. One thing introduced in the last version
   is caching of the codebooks. This needs to be clarified, especially in
   cases where the code book needs to transfered. A codebook is not limited
   in size but they are commonly in the size 10-14 kbytes. So far suggested
   mechanism are: 1) Transfer in HTTP, 2) send it RTCP APP packets, 3) Use
   RTP over TCP, 4) try to define a limited set of codebooks that has to be
   cached and are simply referenced. Steve Casner commented that in RTP
   version 1 there was a mechanism to transfer parameters in parallel.
   However this was removed due to synchronisation problems. Having any
   such mechanism is a slippery slope and needs to be threaded carefully.=20

   What is the gain from performing application level fragmentation, rather
   than using IP level fragmentation? Need to be further clarified and
   evaluated if enough gain is present to warrant the complexity of the
   mechanism. There has also been a number of other general comments on the
   draft on the mailing list regarding usage of SDP parameters, Path MTU
   discovery, RTP timestamp recovery, and frame aggregation that needs to
   be resolved.=20


Internet Low Bit Rate Codec and RTP payload format
draft-ietf-avt-ilbc-codec-02.txt
draft-ietf-avt-rtp-ilbc-02.txt

   Alan Duric presented an update of the codec and payload format. The
   codec has been reviewed by speech coding experts and their comments=20
   will be distributed when the last call is announced. As a result of=20
   the review the codec specification has been clarified and corrected.
   The new version should be possible to implement based only on the
   description rather then the source code.

   The payload format has an open issue with mode flag if only one side=20
   is capable of supporting 20 ms frame length in an SDP Offer/Answer
   negotiation. The proposal to fix this is to include an offer/answer
   section in the draft, specifying bi-directional behaviour for the mode
   flag. A related issue is that ptime can't double as mode flag (this
   should be explained in the text, since it is a common question).=20

   As there has been no demand for comfort noise (CN) generation it will
   not be included in the Codec. Henry Sinnreich commented that Voice
   Activity Detection (VAD) and CN generation is important. The counter
   comment was that an external VAD and the generic CN can be used anyway.
   There might also be future work to provide iLBC with VAD and CN.=20

   Colin Perkins asked if there is any need for test vectors? The answer
   from Alan is that the codec specification is desired to remain being a
   floating-point defined one and therefore exact test vectors can't be
   used.  However there is a possibility to provide test vector and then
   use PESQ measurements to check how far from the nominal output ones
   implementation produces. If any vector will be provided they will be in
   a separate draft.

   Jon Peterson asked if the mode parameter is a more generic parameter and
   therefore should be defined as such. Alan said that he had seen it in
   some other codecs. Magnus Westerlund commented that any mode parameter
   and their values are codec specific and should therefore be defined as
   such.=20


RTP Payload Format for Uncompressed Video=20
draft-ietf-avt-uncomp-video-03.txt

   The current status and remaining open issues were presented by Ladan
   Gharai. There have been few technical changes since the last version,
   mostly editorial and related to understanding. The two payload
   parameters sub-sampling and color-mode has been combined into one
   parameter (sampling) to avoid the usage of "impossible" combinations.=20

   The remaining open issue is if the currently defined colorimetry values
   are sufficient or if there are more that should be defined? David Singer
   also had an comment that YUV-4:4:2 sampling can have their chroma
   samples in different orders. This needs to be addressed.=20

   After the next update the draft is expected to be mature enough for
   working group last call.=20


RTP Payload for H.264/JVT Video=20
draft-ietf-avt-rtp-h264-02.txt

   Stephan Wenger presented the current status. The video codec itself has
   now been approved by ITU as H.264 and will soon also be approved by
   ISO/IEC. Work will continue in the JVT for professional extensions,
   however the NALU interface will not be changed, so this work is not
   believed to effect the payload format. The updated draft has been
   significantly improved regarding understanding, especially of the
   decoding order number (DON) concept.=20

   There are a few remaining open issues. Should we do base64 encoding of
   the parameters sets instead of base16? There does not seem to be any
   formal problem of using this, and it will provide better efficiency.=20
   The security consideration needs to be improved and carefully reviewed.
   There is concern over Timing SEI messages, since broken implementations
   may disregard the RTP timestamp. There is need for strong clarifications
   on the problem of doing that. Steve Casner commented that at least the
   blame and cause are on the same side, namely with the decoder.=20


RTP Payload format for 3GPP Timed Text=20
draft-rey-avt-3gpp-timed-text-00.txt

   Jose Rey presented for the first time a proposal for an RTP payload
   format for the 3GPP timed text format. The timed text format provides
   decorated and timed text, that can be used for karaoke and subtitling.
   There is a need to complete this work fairly quickly as the 3GPP Release
   6 deadline is officially march 2004, and may move to June 2004.=20

   Steve Casner commented that the current proposal seems to provide much
   flexibility that does not provide any useful functionality, and only
   complicates the format. Colin Perkins also questioned how this work
   relates to RFC 2793, RTP Payload for Text Conversation. The answer is
   that text conversation does not provide modifiers for style and fonts
   and other decorations as 3GPP Timed Text provides.=20

   Joerg Ott raised the most substantial question: should this be a more
   generic work. The applications are not limited to 3GPP mobile phones
   with streaming, and a generic format seams appropriate. David Singer
   commented that he had hoped that the W3C and MPEG work on authoring and
   transport of decorated and timed text would be further progressed at
   this time, and could provide input into this work. However Dave also
   pointed out that there is a strong need to meet the 3GPP deadline.
   Magnus Westerlund further commented that he hoped that people were
   talking about a generic payload format rather than specifying a generic
   set of operations on text (i.e. a text codec) in duplication of work
   ongoing in other standards bodies. David singer further commented that
   there are usually no comments when people standardise multiple different
   codecs, that one generic is enough. The conclusion is that there is need
   to further explore the scope of the work here. The subject should be
   clarified through a mail discussion or internet draft.=20

   The request for working group status has been delayed until the open
   issues regarding how generic the work should be can be resolved.=20


RTCP Extensions for SSM Sessions with Unicast Feedback=20
draft-ietf-avt-rtcpssm-04.txt

   Joerg Ott presented the update of the draft. The format has reverted
   back to using its own RSI RTCP format instead of defining XR extension
   formats. Some editorial issues have not been fixed yet. The authors
   believe there are only a few remaining technical issues for this draft.

   One issue is if the distribution source should be allowed to use only
   the senders RTCP bandwidth part or also the receivers? Steve commented
   that the appropriate thing would be to allow the distribution source to
   use all RTCP bandwidth as it summaries all RTCP. Magnus Westerlund also
   commented that care should be take to define the RTCP behaviour for SSM
   in the draft as the default, however future profiles may redefine this
   behaviour.=20

   The second issue is the possible need to have values indicating for
   which sources, or number of sources, the gathered information in RSI
   mode applies to. This may be an unnecessary complication of the format,
   and Magnus Westerlund commented that it only seems to be applicable to
   some of the formats. Joerg asked if there is any need for this feature=20
   otherwise the authors plan to go for simplicity here (meaning that the
   receivers have to trust the sender to do the reasonable thing when
   summarising).=20

   The third issue is that RTCP BYE packets are reported five times (as is
   done for SSRC collisions). However there seems to be little necessity of
   doing this for normal RTCP BYE packets. Steve Casner commented that it
   seems to be an advantage of having this to work in the same way as
   standard RTCP. Colin commented that there is probably no need to report
   a collision more than one every time you detect it: if the receiver that
   causes the collision does not receive the report packet, the collision
   will continue and a new collision report will be sent after the next
   observation of it by the source.=20

   There is significant interest in this work and the authors should do
   their best to meet the December milestone in the preliminary revised
   charter. Colin Perkins pointed out that the definition of the "a=3Drtcp"=
=20
   attribute contradicts the definition in draft-ietf-mmusic-sdp-nat-05.txt=
.=20
   The intention from the authors was to use it as defined in the SDP NAT
   draft.=20

   Colin Perkins noted that while the Security Considerations section has a
   thorough declaration of the problems present with this RTCP extension,
   there is a need to have normative declaration on how these security
   considerations shall be addressed in different applications. If not,
   there will not be possible to have an interoperable implementations.=20


Requirements for performing RTP header compression over MPLS networks
draft-ash-e2e-voip-hdr-comp-rqmts-00.txt
draft-ash-e2e-vompls-hdr-compress-01.txt
draft-ash-e2e-crtp-hdr-compress-01.txt

   Jerry Ash presented an introduction and motivation for the work that has
   previously been discussed in the MPLS working group. There are several
   ideas that has been proposed in this context. The main idea is to
   provide RTP/UDP/IP header compression over several hops in backbones
   that have expensive bit-rate costs. Ideas include that one can perform
   header compression from an ingress point in an MPLS network to the
   egress point without doing compression/decompression for each link.
   Instead the MPLS router can forward the compressed packets based on
   either MPLS labels, or the compression context IDs. However the
   scalability of each solution needs to be further investigated. The
   context switch idea could also be applied outside an MPLS network for
   other links. Another part of the complete framework is to ensure that
   the header compression mechanism can handle the long delay that may
   result between the compression and decompression points. So far the
   assumption is that ECRTP will be enough for this.=20

   There were several voices raised questioning the need of this work.
   There seems to be interest to migrate legacy voice channels done over
   circuit switched connections to instead use VoIP. Further there was a
   lot of confusion in the public in regards to what End-to-End in the
   presentation meant. However this is either MPLS ingress to egress, or in
   some optimised cases may be through end-to-end if all nodes from end-
   point to end-point support CCID switching and header compression. This
   mechanism will not be media dependent as it will be able to compress the
   header for any RTP transported media. The actual work to be done is not
   clear and some people address concern that it seems to be a very strange
   idea. It is clear that the work must first start with writing an
   requirements document that people agree on and to get the people
   understand what the goals of this work is.=20

   Co-coordination with other working groups will be needed, in particular
   the MPLS working group, and it is hoped that the people from other WG
   will participate in also in the initial phases defining requirements.  =
=20

				   - * -


--Multipart_Sun__27_Jul_2003_16:17:04_-0700_0817bc00--

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: Colin Perkins <csp@csperkins.org>
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This is to announce a working group last call on the RTP Payload for DTMF
Digits, Telephony Tones and Signals <draft-ietf-avt-rfc2833bis-03.txt>.
Please send any final comments on this draft to the mailing list by 11th
August 2003. If no substantive technical issues are raised, we intend to
submit this draft for publication as a proposed standard RFC to obsolete
RFC 2833.

Note that this draft will not be advanced until we receive at least one
valid review from someone not directly involved with production of the
draft (i.e., not an author or contributor). 

-- 
Colin Perkins
csp@csperkins.org

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: [AVT] 3GPP timed text draft: more generic?
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Hi all,

during the presentation of the payload format for 3GPP timed text some
concerns arose on whether this work should be more generic.  On the
other hand I also emphasized the fact that there's a deadline to meet
here for 3GPP Release 6: March '04.

It is our opinion that, in this case, a more specific payload format for
the 3GPP timed text format is the right way to go. Does anybody think
otherwise? We would like to hear comments/critic on this, in order to
progress to working group work item.  How could it be made more generic?

thanks in advance,


Jos=E9 Rey____ mailto:rey@panasonic.de
Mobile Multimedia Team -  MCOM Group
Panasonic European Laboratories GmbH
Tel: +49(0)6103-766-134
Fax: +49(0)6103-766-166
http://www.pel.panasonic.de
____________________________________


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Mon Jul 28 09:20:30 2003
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Date: Mon, 28 Jul 2003 06:17:58 -0700
From: Colin Perkins <csp@csperkins.org>
To: "Jose Rey" <rey@panasonic.de>
Cc: avt@ietf.org
Subject: Re: [AVT] 3GPP timed text draft: more generic?
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--> "Jose Rey" <rey@panasonic.de> writes:
> 
> Hi all,
> 
> during the presentation of the payload format for 3GPP timed text some
> concerns arose on whether this work should be more generic.  On the
> other hand I also emphasized the fact that there's a deadline to meet
> here for 3GPP Release 6: March '04.
> 
> It is our opinion that, in this case, a more specific payload format for
> the 3GPP timed text format is the right way to go. Does anybody think
> otherwise? We would like to hear comments/critic on this, in order to
> progress to working group work item.  How could it be made more generic?

A specific payload format for 3GPP timed text might be appropriate, but I
would like to understand the options and alternatives before we make that
choice. Is 3GPP the only body doing timed and/or decorated text? If not,
what are the alternatives, and how do they relate?

-- 
Colin Perkins
csp@csperkins.org

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Mon, 28 Jul 2003 06:24:32 -0700
From: Colin Perkins <csp@csperkins.org>
To: "Jose Rey" <rey@panasonic.de>
Cc: avt@ietf.org
Subject: Re: [AVT] Retransmission draft
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Jose,

--> "Jose Rey" <rey@panasonic.de> writes:
> for your information:
> 
> Matsushita has sent an IPR disclosure to the IETF IPR website regarding
> patent rights applicable to the draft-ietf-rtp-retransmission-08.txt.
> 
> More information can be found under http://www.ietf.org/ipr

Is it possible for you to elaborate on which parts of the retransmission
format are covered by the Matsushita IPR?

-- 
Colin Perkins
csp@csperkins.org

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Neo Anderson" <neo@wiseone.zzn.com>
Message-Id: <4F2B0A162724F4A478553D3C4FC99A12@neo.wiseone.zzn.com>
Date: Tue, 29 Jul 2003 15:21:17 +0530
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To: Premendra.Kumar@infineon.com, avt@ietf.org
Subject: Re: [AVT] (no subject)
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<html><head><meta Name=3D'keywords' Content=3D'commtouch, pronto, mail, =
free email, free, branded, web based, free web based email, communicatio=
ns, internet, software, advertising banners, e-mail, free software'></he=
ad><body   ><div align=3D'left'><font   ><blockquote><blockquote><TT>---=
- Begin Original Message ----<BR>
<BR>
 From: &quot;Kumar Premendra (IFIN DC COM)&quot; &lt;Premendra.Kumar@inf=
ineon.com&gt;<BR>
Sent: Tue, 22 Jul 2003 13:50:26 +0530<BR>
To: avt@ietf.org<BR>
Subject: [AVT] (no subject)<BR>
<BR>
<BR>
<BR>
Hello,<BR>
<BR>
I have some doubt while implementing RTP/RTCP....please clarify my <BR>=

doubt if<BR>
possible.<BR>
<BR>
Q. =A0We must send RTCP packet =A0in compound packet format ,Always ?<B=
R>
<BR>
Yes.<BR>
<BR>
&gt; If not, how do we identify whether received packet is =A0normal RT=
CP <BR>
packet or compound packet.<BR>
<BR>
RFC 1889, 6.1 Says:<BR>
 &quot;There is no explicit count of individual RTCP packets in the <BR=
>
compound packet since the lower layer protocols are expected to <BR>
provide an overall length to determine the end of the compound <BR>
packet.&quot;<BR>
<BR>
Using udp for example, recvfrom () will give the length of the packet <=
BR>
which is received. This length has to be crosschecked with the length <=
BR>
field in the fixed header of RTCP packet(s).<BR>
<BR>
--Neo<BR>
<BR>
<BR>
<BR>
<BR>
</TT><br><br><font><p align=3Dleft><br>Get your own Free E-mail at Witc=
h Wisdom http://www.witchwisdom.com<br>_________________________________=
__________________________<br>Get your own Web-based E-mail Service at h=
ttp://www.zzn.com</blockquote></blockquote></div></font></body></html>

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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A new Request for Comments is now available in online RFC libraries.


        RFC 3550

        Title:      RTP: A Transport Protocol for Real-Time
                    Applications
        Author(s):  H. Schulzrinne, S. Casner, R. Frederick,
                    V. Jacobson 
        Status:     Standards Track
        Date:       July 2003
        Mailbox:    schulzrinne@cs.columbia.edu, casner@acm.org,
                    ronf@bluecoat.com, van@packetdesign.com
        Pages:      104
        Characters: 259985
        Obsoletes:  1889

        I-D Tag:    draft-ietf-avt-rtp-new-12.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3550.txt


This memorandum describes RTP, the real-time transport protocol.  RTP
provides end-to-end network transport functions suitable for
applications transmitting real-time data, such as audio, video or
simulation data, over multicast or unicast network services.  RTP
does not address resource reservation and does not guarantee
quality-of-service for real-time services.  The data transport is
augmented by a control protocol (RTCP) to allow monitoring of the
data delivery in a manner scalable to large multicast networks, and
to provide minimal control and identification functionality.  RTP and
RTCP are designed to be independent of the underlying transport and
network layers.  The protocol supports the use of RTP-level
translators and mixers.

Most of the text in this memorandum is identical to RFC 1889 which it
obsoletes.  There are no changes in the packet formats on the wire,
only changes to the rules and algorithms governing how the protocol
is used.  The biggest change is an enhancement to the scalable timer
algorithm for calculating when to send RTCP packets in order to
minimize transmission in excess of the intended rate when many
participants join a session simultaneously.

This document is a product of the Audio/Video Tranposrt Working Group
of the IETF.

This is now a Draft Standard Protocol.

This document specifies an Internet standards track protocol for
the Internet community, and requests discussion and suggestions
for improvements.  Please refer to the current edition of the
"Internet Official Protocol Standards" (STD 1) for the
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_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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--NextPart


A new Request for Comments is now available in online RFC libraries.


        RFC 3551

        Title:      RTP Profile for Audio and Video Conferences
                    with Minimal Control
        Author(s):  H. Schulzrinne, S. Casner
        Status:     Standards Track
        Date:       July 2003
        Mailbox:    schulzrinne@cs.columbia.edu, casner@acm.org
        Pages:      44
        Characters: 106621
        Obsoletes:  1890

        I-D Tag:    draft-ietf-avt-profile-new-13.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3551.txt


This document describes a profile called "RTP/AVP" for the use of the
real-time transport protocol (RTP), version 2, and the associated
control protocol, RTCP, within audio and video multiparticipant
conferences with minimal control.  It provides interpretations of
generic fields within the RTP specification suitable for audio and
video conferences.  In particular, this document defines a set of
default mappings from payload type numbers to encodings.

This document also describes how audio and video data may be carried
within RTP.  It defines a set of standard encodings and their names
when used within RTP.  The descriptions provide pointers to reference
implementations and the detailed standards.  This document is meant
as an aid for implementors of audio, video and other real-time
multimedia applications.

This memorandum obsoletes RFC 1890.  It is mostly backwards-compatible
except for functions removed because two interoperable implementations
were not found.  The additions to RFC 1890 codify existing practice in
the use of payload formats under this profile and include new payload
formats defined since RFC 1890 was published.

This document is a product of the Audio/Video Transport Working Group
of the IETF.

This is now a Draft Standard Protocol.

This document specifies an Internet standards track protocol for
the Internet community, and requests discussion and suggestions
for improvements.  Please refer to the current edition of the
"Internet Official Protocol Standards" (STD 1) for the
standardization state and status of this protocol.  Distribution
of this memo is unlimited.

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Joyce K. Reynolds and Sandy Ginoza
USC/Information Sciences Institute

...

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_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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--NextPart


A new Request for Comments is now available in online RFC libraries.


        RFC 3555

        Title:      MIME Type Registration of RTP Payload Formats
        Author(s):  S. Casner, P. Hoschka
        Status:     Standards Track
        Date:       July 2003
        Mailbox:    casner@acm.org, ph@w3.org
        Pages:      45
        Characters: 56618
        Updates/Obsoletes/SeeAlso:  None

        I-D Tag:    draft-ietf-avt-rtp-mime-06.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3555.txt


This document defines the procedure to register RTP Payload Formats
as audio, video or other MIME subtype names.  This is useful in a
text-based format or control protocol to identify the type of an
RTP transmission.  This document also registers all the RTP payload
formats defined in the RTP Profile for Audio and Video Conferences
as MIME subtypes.  Some of these may also be used for transfer
modes other than RTP.

This document is a product of the Audio/Video Transport Working Group
of the IETF.

This is now a Proposed Standard Protocol.

This document specifies an Internet standards track protocol for
the Internet community, and requests discussion and suggestions
for improvements.  Please refer to the current edition of the
"Internet Official Protocol Standards" (STD 1) for the
standardization state and status of this protocol.  Distribution
of this memo is unlimited.

This announcement is sent to the IETF list and the RFC-DIST list.
Requests to be added to or deleted from the IETF distribution list
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Details on obtaining RFCs via FTP or EMAIL may be obtained by sending
an EMAIL message to rfc-info@RFC-EDITOR.ORG with the message body 
help: ways_to_get_rfcs.  For example:

        To: rfc-info@RFC-EDITOR.ORG
        Subject: getting rfcs

        help: ways_to_get_rfcs

Requests for special distribution should be addressed to either the
author of the RFC in question, or to RFC-Manager@RFC-EDITOR.ORG.  Unless
specifically noted otherwise on the RFC itself, all RFCs are for
unlimited distribution.echo 
Submissions for Requests for Comments should be sent to
RFC-EDITOR@RFC-EDITOR.ORG.  Please consult RFC 2223, Instructions to RFC
Authors, for further information.


Joyce K. Reynolds and Sandy Ginoza
USC/Information Sciences Institute

...

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_______________________________________________
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avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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        RFC 3556

        Title:      Session Description Protocol (SDP) Bandwidth
                    Modifiers for RTP Control Protocol (RTCP)
                    Bandwidth
        Author(s):  S. Casner
        Status:     Standards Track
        Date:       July 2003
        Mailbox:    casner@acm.org
        Pages:      8
        Characters: 15310
        Updates/Obsoletes/SeeAlso:  None

        I-D Tag:    draft-ietf-avt-rtcp-bw-05.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3556.txt


This document defines an extension to the Session Description Protocol
(SDP) to specify two additional modifiers for the bandwidth attribute.
These modifiers may be used to specify the bandwidth allowed for RTP
Control Protocol (RTCP) packets in a Real-time Transport Protocol
(RTP) session.

This document is a product of the Audio/Video Transport Working Group
of the IETF.

This is now a Proposed Standard Protocol.

This document specifies an Internet standards track protocol for
the Internet community, and requests discussion and suggestions
for improvements.  Please refer to the current edition of the
"Internet Official Protocol Standards" (STD 1) for the
standardization state and status of this protocol.  Distribution
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USC/Information Sciences Institute

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_______________________________________________
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        RFC 3557

        Title:      RTP Payload Format for European Telecommunications
                    Standards Institute (ETSI) European Standard
                    ES 201 108 Distributed Speech Recognition Encoding
        Author(s):  Q. Xie, Ed.
        Status:     Standards Track
        Date:       July 2003
        Mailbox:    Qiaobing.Xie@motorola.com
        Pages:      15
        Characters: 29692
        Updates/Obsoletes/SeeAlso:  None

        I-D Tag:    draft-ietf-avt-dsr-05.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3557.txt


This document specifies an RTP payload format for encapsulating
European Telecommunications Standards Institute (ETSI) European
Standard (ES) 201 108 front-end signal processing feature streams
for distributed speech recognition (DSR) systems.

This document is a product of the Audio/Video Transport Working Group
of the IETF.

This is now a Proposed Standard Protocol.

This document specifies an Internet standards track protocol for
the Internet community, and requests discussion and suggestions
for improvements.  Please refer to the current edition of the
"Internet Official Protocol Standards" (STD 1) for the
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USC/Information Sciences Institute

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_______________________________________________
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https://www1.ietf.org/mailman/listinfo/avt



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A new Request for Comments is now available in online RFC libraries.


        RFC 3558

        Title:      RTP Payload Format for Enhanced Variable Rate
                    Codecs (EVRC) and Selectable Mode Vocoders (SMV)
        Author(s):  A. Li
        Status:     Standards Track
        Date:       July 2003
        Mailbox:    adamli@icsl.ucla.edu
        Pages:      23
        Characters: 49787
        Updates/Obsoletes/SeeAlso:  None

        I-D Tag:    draft-ietf-avt-evrc-smv-03.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3558.txt


This document describes the RTP payload format for Enhanced Variable
Rate Codec (EVRC) Speech and Selectable Mode Vocoder (SMV)
Speech.  Two sub-formats are specified for different application
scenarios.  A bundled/interleaved format is included to reduce the
effect of packet loss on speech quality and amortize the overhead of
the RTP header over more than one speech frame.  A non-bundled format
is also supported for conversational applications.

This document is a product of the Audio/Video Transport Working Group
of the IETF.

This is now a Proposed Standard Protocol.

This document specifies an Internet standards track protocol for
the Internet community, and requests discussion and suggestions
for improvements.  Please refer to the current edition of the
"Internet Official Protocol Standards" (STD 1) for the
standardization state and status of this protocol.  Distribution
of this memo is unlimited.

This announcement is sent to the IETF list and the RFC-DIST list.
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Joyce K. Reynolds and Sandy Ginoza
USC/Information Sciences Institute

...

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_______________________________________________
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avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Wed Jul 30 09:15:50 2003
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Content-Class: urn:content-classes:message
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Subject: RE: [AVT] Retransmission draft
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Hi Colin, all,

> -----Original Message-----
> From: Colin Perkins [mailto:csp@csperkins.org]
> Sent: Monday, July 28, 2003 3:25 PM
> To: Jose Rey
> Cc: avt@ietf.org
> Subject: Re: [AVT] Retransmission draft
>
>
> Jose,
>
> --> "Jose Rey" <rey@panasonic.de> writes:
> > for your information:
> >
> > Matsushita has sent an IPR disclosure to the IETF IPR
> website regarding
> > patent rights applicable to the
> draft-ietf-rtp-retransmission-08.txt.
> >
> > More information can be found under http://www.ietf.org/ipr
>
> Is it possible for you to elaborate on which parts of the
> retransmission
> format are covered by the Matsushita IPR?

The disclosed IPR covers the fact that a packet
has two sequence numbers, one of each is used to determine the
transmission
order and the other one is for the reproduction order.  Applying this to
the draft these two sequence numbers map to the "standard" RTP SN and to
the
Original Sequence Number (OSN) fields, respectively.

cheers,

Jose
>
> --
> Colin Perkins
> csp@csperkins.org


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Subject: [AVT] RFC 3545 on Enhanced Compressed RTP (CRTP) for Links with High Delay, Packet Loss and Reordering
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A new Request for Comments is now available in online RFC libraries.


        RFC 3545

        Title:      Enhanced Compressed RTP (CRTP) for Links with High
                    Delay, Packet Loss and Reordering
        Author(s):  T. Koren, S. Casner, J. Geevarghese, B. Thompson,
                    P. Ruddy
        Status:     Standards Track
        Date:       July 2003
        Mailbox:    tmima@cisco.com, casner@acm.org,
                    geevjohn@hotmail.com, brucet@cisco.com,
                    pruddy@cisco.com
        Pages:      22
        Characters: 48278
        Updates/Obsoletes/SeeAlso:  None

        I-D Tag:    draft-ietf-avt-crtp-enhance-07.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3545.txt


This document describes a header compression scheme for point to
point links with packet loss and long delays.  It is based on
Compressed Real-time Transport Protocol (CRTP), the IP/UDP/RTP
header compression described in RFC 2508.  CRTP does not perform well
on such links: packet loss results in context corruption and due to
the long delay, many more packets are discarded before the context
is repaired.  To correct the behavior of CRTP over such links, a few
extensions to the protocol are specified here.  The extensions aim to
reduce context corruption by changing the way the compressor updates
the context at the decompressor: updates are repeated and include
updates to full and differential context parameters.  With these
extensions, CRTP performs well over links with packet loss, packet
reordering and long delays.

This document is a product of the Audio Video Transport Working Group
of the IETF.

This is now a Proposed Standard Protocol.

This document specifies an Internet standards track protocol for
the Internet community, and requests discussion and suggestions
for improvements.  Please refer to the current edition of the
"Internet Official Protocol Standards" (STD 1) for the
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Joyce K. Reynolds and Sandy Ginoza
USC/Information Sciences Institute

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