
From roni.even@mail01.huawei.com  Thu Nov  1 07:48:46 2012
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From: Roni Even <roni.even@mail01.huawei.com>
To: "avt@ietf.org" <avt@ietf.org>, "payload@ietf.org" <payload@ietf.org>
Thread-Topic: CORRECTION: Help the NomCom
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Date: Thu, 1 Nov 2012 14:48:31 +0000
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Subject: [AVTCORE] FW: CORRECTION: Help the NomCom
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Hi,
Note that NomCopm will receive input till 11/11 and not till the 4th as was=
 stated in previous email
Roni Even

________________________________________
From: wgchairs-bounces@ietf.org [wgchairs-bounces@ietf.org] on behalf of No=
mCom Chair [nomcom-chair@ietf.org]
Sent: Thursday, November 01, 2012 2:17 PM
To: Working Group Chairs
Subject: CORRECTION: Help the NomCom

I sent a message several hours ago requesting that you help and make
your working group aware that the NomCom is looking for input from the
community. This message had a minor error. The NomCom needs to
receive community input by November 11, 2012.

The full text of the corrected message is below
---------------------------------------------------

The IETF Nominations Committee (NomCom) continues to seek input from
the IETF Community. The NomCom would greatly appreciate any help you
could provide in making members of your working group aware of ways in
which they can provide valuable feedback to the NomCom.

In order to ensure that your input is received in time to be useful, the
NomCom needs to receive community feedback on or before Sunday, November 11=
.

The final list of candidates (as per RFC 5680) that the NomCom is
considering for open positions can be found at:
https://www.ietf.org/group/nomcom/2012/input/

The NomCom will be holding office hours during IETF 85, Monday-
Thursday from 1:00pm to 3:00pm in Room 305. The NomCom welcomes
comments on specific individuals, as well as general feedback related to
any of the positions that NomCom is considering.

Note: A list of leadership positions that the NomCom is considering can be
found at: https://www.ietf.org/group/nomcom/2012/

If the NomCom office hours are inconvenient for you or if you cannot
attend IETF 85, the NomCom is happy to take community input via email
to nomcom12 at ietf.org. Additionally, the NomCom is happy to arrange a
meeting outside of office hours, just send us email and we can set
something up.

Comments on specific candidates can also be provided to the NomCom
via the web feedback tool:
https://www.ietf.org/group/nomcom/2012/input/

Thank you for your help,
- Matt Lepinski
  nomcom-chair at ietf.org=

From kevin.gross@avanw.com  Thu Nov  1 10:22:01 2012
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Subject: Re: [AVTCORE] I-D Action: draft-ietf-avtcore-clksrc-01.txt
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--047d7b621ea0fdba8304cd723f0c
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We have revised draft-ietf-avtcore-clksrc based on discussion at IETF 84

   1. We've made minor improvements to language, terminology and formatting.
   2. Fixed a bug in ABNF specifying a traceable PTP clock - no grandmaster
   ID or domain indication is required in this case.
   3. Improved readability for specifying direct-referenced media clock -
   now indicated with "direct" instead of "offset". Made offset specification
   optional and described how to determine it if not supplied.
   4. We have removed use of the term "confidence" in clock discussion and
   replaced it with "quality" to avoid giving the impression that a confidence
   interval is under discussion here. We've also clarified the purpose and
   caveats in communicating clock quality. This should address Johnathan's
   comment that RTCP may be a better means than SDP to convey synchronization
   status.
   5. Jonathan also made a suggestion that instead of allowing indication
   of traceability of various clock sources that we define a generic traceable
   clock and identify it by its stratum value. We researched doing this and
   discovered that stratum designations are used in NTP clock distribution and
   in legacy telecommunications systems. In the NTP context, stratum describes
   how time servers are interconnected but does not describe the quality of
   the clock produced. The usefulness stratum system in the telecommunications
   began eroding with the breakup of the bell system. Telecomm synchronization
   can now be characterized as plesiochronous meaning there is is more of a
   peering than hierarchical relationship between clocks. We believe knowing
   the source of a traceable clock is the most reliable and available
   information for assessing a clock.
   6. Added some background material on Stream-Referenced media clock
   (section 5.3). Added separate signaling of master and slave
   stream-referenced media clocks. Removed address and port specifications
   from clock descriptions.


Kevin Gross
+1-303-447-0517
Media Network Consultant
AVA Networks - www.AVAnw.com <http://www.avanw.com/>, www.X192.org



On Mon, Oct 22, 2012 at 2:29 PM, <internet-drafts@ietf.org> wrote:

>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>  This draft is a work item of the Audio/Video Transport Core Maintenance
> Working Group of the IETF.
>
>         Title           : RTP Clock Source Signalling
>         Author(s)       : Aidan Williams
>                           Kevin Gross
>                           Ray van Brandenburg
>                           Hans Stokking
>         Filename        : draft-ietf-avtcore-clksrc-01.txt
>         Pages           : 21
>         Date            : 2012-10-22
>
> Abstract:
>    NTP timestamps are used by several RTP protocols for synchronisation
>    and statistical measurement.  This memo specifies SDP signalling
>    identifying NTP timestamp clock sources and SDP signalling
>    identifying the media clock sources in a multimedia session.
>
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-avtcore-clksrc
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-avtcore-clksrc-01
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-avtcore-clksrc-01
>
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
>

--047d7b621ea0fdba8304cd723f0c
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<span style=3D"font-family:arial,sans-serif;font-size:13.333333969116211px"=
>We have revised draft-ietf-avtcore-clksrc based on discussion at IETF 84</=
span><div style=3D"font-family:arial,sans-serif;font-size:13.33333396911621=
1px">
<ol><li style=3D"margin-left:15px">We&#39;ve made minor improvements to lan=
guage, terminology and formatting.</li><li style=3D"margin-left:15px">Fixed=
 a bug in ABNF specifying a traceable PTP clock - no grandmaster ID or doma=
in indication is required in this case.</li>
<li style=3D"margin-left:15px">Improved readability for specifying direct-r=
eferenced media clock - now=A0indicated=A0with &quot;direct&quot; instead o=
f &quot;offset&quot;. Made offset specification optional and described how =
to determine it if not supplied.</li>
<li style=3D"margin-left:15px">We have removed use of the term &quot;confid=
ence&quot; in clock discussion and replaced it with &quot;quality&quot; to =
avoid giving the impression that a confidence interval is under discussion =
here. We&#39;ve also clarified the purpose and caveats in communicating clo=
ck quality. This should address Johnathan&#39;s comment that RTCP may be a =
better means than SDP to convey synchronization status.</li>
<li style=3D"margin-left:15px">Jonathan also made a suggestion that instead=
 of allowing indication of traceability of various clock sources that we de=
fine a generic traceable clock and identify it by its stratum value. We res=
earched doing this and discovered that stratum designations are used in NTP=
 clock distribution and in legacy telecommunications systems. In the NTP co=
ntext, stratum describes how time servers are interconnected but does not d=
escribe the quality of the clock produced. The=A0usefulness=A0stratum syste=
m in the telecommunications began eroding with the breakup of the bell syst=
em. Telecomm synchronization can now be characterized as plesiochronous mea=
ning there is is more of a peering than=A0hierarchical=A0relationship betwe=
en clocks. We believe knowing the source of a traceable clock is the most r=
eliable and available information for assessing a clock.</li>
<li style=3D"margin-left:15px">Added some background material on Stream-Ref=
erenced media clock (section 5.3). Added separate signaling of master and s=
lave stream-referenced media clocks. Removed address and port specification=
s from clock descriptions.</li>
</ol></div><div class=3D"gmail_extra"><br clear=3D"all">Kevin Gross<br><div=
>+1-303-447-0517</div><div>Media Network Consultant<br><div>AVA Networks -=
=A0<a href=3D"http://www.avanw.com/" target=3D"_blank">www.AVAnw.com</a>,=
=A0<a href=3D"http://www.X192.org" target=3D"_blank">www.X192.org</a></div>
</div><br>
<br><br><div class=3D"gmail_quote">On Mon, Oct 22, 2012 at 2:29 PM,  <span =
dir=3D"ltr">&lt;<a href=3D"mailto:internet-drafts@ietf.org" target=3D"_blan=
k">internet-drafts@ietf.org</a>&gt;</span> wrote:<br><blockquote class=3D"g=
mail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-l=
eft:1ex">
<br>
A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.<br>
=A0This draft is a work item of the Audio/Video Transport Core Maintenance =
Working Group of the IETF.<br>
<br>
=A0 =A0 =A0 =A0 Title =A0 =A0 =A0 =A0 =A0 : RTP Clock Source Signalling<br>
=A0 =A0 =A0 =A0 Author(s) =A0 =A0 =A0 : Aidan Williams<br>
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Kevin Gross<br>
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Ray van Brandenburg<br>
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Hans Stokking<br>
=A0 =A0 =A0 =A0 Filename =A0 =A0 =A0 =A0: draft-ietf-avtcore-clksrc-01.txt<=
br>
=A0 =A0 =A0 =A0 Pages =A0 =A0 =A0 =A0 =A0 : 21<br>
=A0 =A0 =A0 =A0 Date =A0 =A0 =A0 =A0 =A0 =A0: 2012-10-22<br>
<br>
Abstract:<br>
=A0 =A0NTP timestamps are used by several RTP protocols for synchronisation=
<br>
=A0 =A0and statistical measurement. =A0This memo specifies SDP signalling<b=
r>
=A0 =A0identifying NTP timestamp clock sources and SDP signalling<br>
=A0 =A0identifying the media clock sources in a multimedia session.<br>
<br>
<br>
<br>
The IETF datatracker status page for this draft is:<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-avtcore-clksrc" targ=
et=3D"_blank">https://datatracker.ietf.org/doc/draft-ietf-avtcore-clksrc</a=
><br>
<br>
There&#39;s also a htmlized version available at:<br>
<a href=3D"http://tools.ietf.org/html/draft-ietf-avtcore-clksrc-01" target=
=3D"_blank">http://tools.ietf.org/html/draft-ietf-avtcore-clksrc-01</a><br>
<br>
A diff from the previous version is available at:<br>
<a href=3D"http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-avtcore-clksrc-01"=
 target=3D"_blank">http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-avtcore-cl=
ksrc-01</a><br>
<br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=3D"_blank">ftp://ftp=
.ietf.org/internet-drafts/</a><br>
<br>
_______________________________________________<br>
Audio/Video Transport Core Maintenance<br>
<a href=3D"mailto:avt@ietf.org">avt@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/avt" target=3D"_blank">htt=
ps://www.ietf.org/mailman/listinfo/avt</a><br>
</blockquote></div><br></div>

--047d7b621ea0fdba8304cd723f0c--

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Subject: Re: [AVTCORE] [Technical Errata Reported] RFC5761 (3380)
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Hi,

The errata is correct. I would approve it. This is an important
correction that should be available until the day, if ever this document
is updated.

Cheers

Magnus

On 2012-10-16 18:51, RFC Errata System wrote:
> 
> The following errata report has been submitted for RFC5761,
> "Multiplexing RTP Data and Control Packets on a Single Port".
> 
> --------------------------------------
> You may review the report below and at:
> http://www.rfc-editor.org/errata_search.php?rfc=5761&eid=3380
> 
> --------------------------------------
> Type: Technical
> Reported by: Martin Storsjö <martin@martin.st>
> 
> Section: 4
> 
> Original Text
> -------------
>    o  RTP payload type 80 conflicts with Receiver Summary Information
>       (RSI) packets defined in "RTCP Extensions for Single-Source
>       Multicast Sessions with Unicast Feedback" [6].
> 
> Corrected Text
> --------------
>    o  RTP payload type 81 conflicts with Receiver Summary Information
>       (RSI) packets defined in "RTCP Extensions for Single-Source
>       Multicast Sessions with Unicast Feedback" [6].
> 
> Notes
> -----
> Reference [6], RFC 5760 (IANA likewise), specifies that RTCP RSI has the RTCP packet type number 209, which means that it would conflict with RTP payload type 81, not 80 as stated.
> 
> Instructions:
> -------------
> This errata is currently posted as "Reported". If necessary, please
> use "Reply All" to discuss whether it should be verified or
> rejected. When a decision is reached, the verifying party (IESG)
> can log in to change the status and edit the report, if necessary. 
> 
> --------------------------------------
> RFC5761 (draft-ietf-avt-rtp-and-rtcp-mux-07)
> --------------------------------------
> Title               : Multiplexing RTP Data and Control Packets on a Single Port
> Publication Date    : April 2010
> Author(s)           : C. Perkins, M. Westerlund
> Category            : PROPOSED STANDARD
> Source              : Audio/Video Transport
> Area                : Real-time Applications and Infrastructure
> Stream              : IETF
> Verifying Party     : IESG
> 
> 


-- 

Magnus Westerlund

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


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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>, IETF AVTCore WG <avt@ietf.org>
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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>, "avt@ietf.org" <avt@ietf.org>
Thread-Topic: [AVTCORE] I-D Action: draft-ietf-avtcore-multi-media-rtp-session-01.txt
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Subject: Re: [AVTCORE] I-D Action:	draft-ietf-avtcore-multi-media-rtp-session-01.txt
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I have some comment on this draft.

1) Section 3.3, Architectural Equality, reads as follows, " Provided the me=
dia flows have similar bandwidth requirements, so that the RTCP timing rule=
s work, using the same RTP session for several types of media at once appea=
rs a reasonable choice." This is confusing because the condition stated abo=
ut similar bandwidth is not met by the use case or which this functionality=
 is being proposed (e.g. sessions with audio and video and perhaps data).

2) Section 4, reads, "The goal of the solution is to enable having one or m=
ore RTP sessions, where each RTP session may contain two or more media type=
s." I think the goal is for each RTP session to be able to contain one or m=
ore media types, right?

3) Section 5.1, reads, " The main rule is that if one expects to have equal=
 treatment of all media packets, then this specification might be suitable.=
  The equal treatment [may] include anything from network level up to RTCP =
reporting and feedback." However, some other mechanism not based on the 5-t=
uple for the flow may be used to provide differential treatment, in which c=
ase this multiplexing mechanism may still be suitable despite dissimilar ba=
ndwidths.

Then later in the same section it reads, "The second important consideratio=
n is that all media flows to be sent within a single RTP session need to ha=
ve similar bandwidth ... If an RTP session contains flows with very differe=
nt bandwidths, for example low-rate audio coupled with high-quality video .=
..".
Here is well, as this is one of the intended use cases, I think it is worth=
while to clearly state that this draft provides a [proposed] solution; othe=
rwise, it seems the mechanism defined in this draft is not suitable for the=
 very use case that prompted its creation.

4) Section 5.5, same issue as with section 5.1.=20

5) Section 6.1
s/motivating the previously/motivated previously

And where is reads, "each SSRC will commonly a native media type", did you =
mean to say, "each SSRC will be associated with a specific media type"?

Cheers,
Charles




> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
> Magnus Westerlund
> Sent: Monday, October 22, 2012 11:14 AM
> To: avt@ietf.org
> Subject: Re: [AVTCORE] I-D Action: draft-ietf-avtcore-multi-media-rtp-
> session-01.txt
>=20
> WG,
>=20
> (As author)
>=20
> This updates makes three significant changes:
>=20
> - Updates also RFC 3551
> - Expandes on the RTCP bandwidth issues
>    * Note solution is still an open issue
> - Add a proposal for XOR based FEC handling in this context.
>=20
> Cheers
>=20
> Magnus
>=20
>=20
> On 2012-10-22 17:08, internet-drafts@ietf.org wrote:
> >
> > A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> >  This draft is a work item of the Audio/Video Transport Core Maintenanc=
e
> Working Group of the IETF.
> >
> > 	Title           : Multiple Media Types in an RTP Session
> > 	Author(s)       : Magnus Westerlund
> >                           Colin Perkins
> >                           Jonathan Lennox
> > 	Filename        : draft-ietf-avtcore-multi-media-rtp-session-01.txt
> > 	Pages           : 19
> > 	Date            : 2012-10-22
> >
> > Abstract:
> >    This document specifies how an RTP session can contain media streams
> >    with media from multiple media types such as audio, video, and text.
> >    This has been restricted by the RTP Specification, and thus this
> >    document updates RFC 3550 and RFC 3551 to enable this behavior for
> >    applications that satisfy the applicability for using multiple media
> >    types in a single RTP session.
> >
> >
> > The IETF datatracker status page for this draft is:
> > https://datatracker.ietf.org/doc/draft-ietf-avtcore-multi-media-rtp-
> session
> >
> > There's also a htmlized version available at:
> > http://tools.ietf.org/html/draft-ietf-avtcore-multi-media-rtp-session-0=
1
> >
> > A diff from the previous version is available at:
> > http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-avtcore-multi-media-rtp-
> session-01
> >
> >
> > Internet-Drafts are also available by anonymous FTP at:
> > ftp://ftp.ietf.org/internet-drafts/
> >
> > _______________________________________________
> > Audio/Video Transport Core Maintenance
> > avt@ietf.org
> > https://www.ietf.org/mailman/listinfo/avt
> >
> >
>=20
>=20
> --
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt

From magnus.westerlund@ericsson.com  Sun Nov  4 10:36:13 2012
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Subject: Re: [AVTCORE] [Technical Errata Reported] RFC4585 (3313)
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WG,

I have reviewed this errata and it appears correct. I would recommend
that it is approved. Additional views on this? Please provide any
feedback within a week, after that I will instruct the ADs to approve
this errata.

Cheers

Magnus

On 2012-08-11 16:57, RFC Errata System wrote:
> 
> The following errata report has been submitted for RFC4585,
> "Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)".
> 
> --------------------------------------
> You may review the report below and at:
> http://www.rfc-editor.org/errata_search.php?rfc=4585&eid=3313
> 
> --------------------------------------
> Type: Technical
> Reported by: Mo Zanaty <mzanaty@cisco.com>
> 
> Section: 4.2
> 
> Original Text
> -------------
>       rtcp-fb-ack-param  = SP "rpsi"
>                          / SP "app" [SP byte-string]
>                          / SP token [SP byte-string]
>                          / ; empty
> 
> 
> Corrected Text
> --------------
>       rtcp-fb-ack-param  = SP "rpsi"
>                          / SP "app" [SP byte-string]
>                          / SP token [SP byte-string]
> 
> 
> Notes
> -----
> RFC 4585 defines and allows generic NACK (with no further parameters) but not generic ACK (which always requires further parameters). Section 4.2 says:
> ...
>       Parameters MUST be provided to further distinguish different types
>       of positive acknowledgement feedback.
> ...
>       The feedback type "nack", without parameters, indicates use of the
>       Generic NACK feedback format as defined in Section 6.2.1.
> ...
> The ABNF incorrectly allows nothing after "ack", implying that generic ACK is valid. Even the approved errata, which replaces the invalid empty alternate with optional [], still allows nothing after "ack". Note that recent interest in RTP congestion control (e.g. RMCAT BOF) may lead to defining a standard ACK.
> 
> Instructions:
> -------------
> This errata is currently posted as "Reported". If necessary, please
> use "Reply All" to discuss whether it should be verified or
> rejected. When a decision is reached, the verifying party (IESG)
> can log in to change the status and edit the report, if necessary. 
> 
> --------------------------------------
> RFC4585 (draft-ietf-avt-rtcp-feedback-11)
> --------------------------------------
> Title               : Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)
> Publication Date    : July 2006
> Author(s)           : J. Ott, S. Wenger, N. Sato, C. Burmeister, J. Rey
> Category            : PROPOSED STANDARD
> Source              : Audio/Video Transport
> Area                : Real-time Applications and Infrastructure
> Stream              : IETF
> Verifying Party     : IESG
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
> 


-- 

Magnus Westerlund

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


From alan.d.clark@telchemy.com  Sun Nov  4 10:39:42 2012
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Subject: Re: [AVTCORE] [Technical Errata Reported] RFC4585 (3313)
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Magnus

I also recommend that the proposed change is approved

Regards

Alan


On 11/4/12 1:36 PM, "Magnus Westerlund" <magnus.westerlund@ericsson.com>
wrote:

> WG,
> 
> I have reviewed this errata and it appears correct. I would recommend
> that it is approved. Additional views on this? Please provide any
> feedback within a week, after that I will instruct the ADs to approve
> this errata.
> 
> Cheers
> 
> Magnus
> 
> On 2012-08-11 16:57, RFC Errata System wrote:
>> 
>> The following errata report has been submitted for RFC4585,
>> "Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based
>> Feedback (RTP/AVPF)".
>> 
>> --------------------------------------
>> You may review the report below and at:
>> http://www.rfc-editor.org/errata_search.php?rfc=4585&eid=3313
>> 
>> --------------------------------------
>> Type: Technical
>> Reported by: Mo Zanaty <mzanaty@cisco.com>
>> 
>> Section: 4.2
>> 
>> Original Text
>> -------------
>>       rtcp-fb-ack-param  = SP "rpsi"
>>                          / SP "app" [SP byte-string]
>>                          / SP token [SP byte-string]
>>                          / ; empty
>> 
>> 
>> Corrected Text
>> --------------
>>       rtcp-fb-ack-param  = SP "rpsi"
>>                          / SP "app" [SP byte-string]
>>                          / SP token [SP byte-string]
>> 
>> 
>> Notes
>> -----
>> RFC 4585 defines and allows generic NACK (with no further parameters) but not
>> generic ACK (which always requires further parameters). Section 4.2 says:
>> ...
>>       Parameters MUST be provided to further distinguish different types
>>       of positive acknowledgement feedback.
>> ...
>>       The feedback type "nack", without parameters, indicates use of the
>>       Generic NACK feedback format as defined in Section 6.2.1.
>> ...
>> The ABNF incorrectly allows nothing after "ack", implying that generic ACK is
>> valid. Even the approved errata, which replaces the invalid empty alternate
>> with optional [], still allows nothing after "ack". Note that recent interest
>> in RTP congestion control (e.g. RMCAT BOF) may lead to defining a standard
>> ACK.
>> 
>> Instructions:
>> -------------
>> This errata is currently posted as "Reported". If necessary, please
>> use "Reply All" to discuss whether it should be verified or
>> rejected. When a decision is reached, the verifying party (IESG)
>> can log in to change the status and edit the report, if necessary.
>> 
>> --------------------------------------
>> RFC4585 (draft-ietf-avt-rtcp-feedback-11)
>> --------------------------------------
>> Title               : Extended RTP Profile for Real-time Transport Control
>> Protocol (RTCP)-Based Feedback (RTP/AVPF)
>> Publication Date    : July 2006
>> Author(s)           : J. Ott, S. Wenger, N. Sato, C. Burmeister, J. Rey
>> Category            : PROPOSED STANDARD
>> Source              : Audio/Video Transport
>> Area                : Real-time Applications and Infrastructure
>> Stream              : IETF
>> Verifying Party     : IESG
>> _______________________________________________
>> Audio/Video Transport Core Maintenance
>> avt@ietf.org
>> https://www.ietf.org/mailman/listinfo/avt
>> 
> 




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Cc: HU Zhongji <Zhongji.Hu@alcatel-sbell.com.cn>
Subject: [AVTCORE] FW: New Version Notification for draft-hua-avt-rtp-svcmvc-00.txt
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From ekr@rtfm.com  Mon Nov  5 10:17:43 2012
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From: Eric Rescorla <ekr@rtfm.com>
Date: Mon, 5 Nov 2012 13:17:02 -0500
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Subject: [AVTCORE] Review of draft-ietf-avtcore-aria-srtp-00
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I have reviewed draft-ietf-avtcore-aria-srtp-00.

I'm leaving aside the question of whether it's useful to define
another symmetric encryption algorithm for SRTP (my view is generally
no, especially since we have already specified SEED)
but the WG seems to have decided that.

With that said I don't really understand why you are defining three
key sizes and three counter modes. I would recommend defining only 128
and 256-bit GCM unless there is some actual reason not to.

Other than that, this seems like a reasonably plausible draft.

From mamontor@posgrado.upv.es  Mon Nov  5 04:20:06 2012
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Message-ID: <20121105132002.50214lz37taa88he@webmail.upv.es>
Date: Mon, 05 Nov 2012 13:20:02 +0100
From: Mario Montagud Climent <mamontor@posgrado.upv.es>
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Subject: [AVTCORE] Question about the minimum RTCP report interval
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Hi all,

I would appreciate it very much if you could clarify me a doubt about  
the minimum RTCP reporting interval.

According to RFC 3550, the RTCP report interval is dynamically  
computed based on the allocated session bandwidth, the average size of  
all (received and sent) RTCP packets, the number of participants in  
the session and their role (senders or receivers), or their  
proportion, and the unicast or multicast nature of the session  
(Section 6.2 of RFC 3550).

Nevertheless, it is also specified that this deterministically  
computed RTCP report interval should have a lower bound to avoid  
having bursts (cumpling) of RTCP packets when the number of  
participants is small and the law of large numbers is not helping to  
smooth out the traffic overhead. This would also help to avoid  
excessive frequent reports during transient outages like a network  
partition. The recommended value in RFC 3550 for that fixed RTCP  
minimum interval is 5 seconds.

The RFC 3550 also specifies that in some cases (e.g., if the data rate  
is high and the application demands more timely feedback reports) an  
implementation may scale the minimum RTCP interval to a smaller value  
given by 360 divided by the session bandwidth (in kbps). Consequently,  
this yields to an interval smaller than 5 seconds for bandwidths  
greater than 72 kbps. In multicast sessions, only active senders may  
use that reduced minimum interval, whilst in unicast sessions it also  
may be used by receivers. In the above cases, however, the RTCP  
minimum interval must be still taken into account during the  
membership accounting procedure in order to not prematurely time out  
participants (who can indeed be using it) because of inactivity.

However, RFC 4585 defines an extension to the RTP Audio-Visual Profile  
(AVP) that enables receivers to provide, statistically, more immediate  
feedback messages to the senders and thus allows for short-term  
adaptation and efficient feedback-based repair mechanisms to be  
implemented, while maintaining the bandwidth constraints for RTCP and  
preserving scalability to large groups. In RFC 4585, the RTCP  
transmission interval specified in RFC 3550 is denoted as Regular RTCP  
interval. In addition, RFC 4585 specifies that feedback messages can  
be reported earlier than the next scheduled Regular RTCP transmission  
time if a receiver detects the need to inform about events of interest  
in the media stream (e.g., picture or slice loss) close to their  
occurrence. Otherwise, the Regular RTCP transmission mode is employed,  
but the minimum RTCP report interval of 5 seconds recommended in RFC  
3550 is dropped.

So, my questions about this are:

- When using the Regular RTCP reporting rules (RFC 3550) in multicast  
scenarios, which option should we select for the minimum RTCP  
reporting interval (for a media sender)? Can we use any of the two  
options for selection the minimum thresholds (i.e., 5 sec or  
360/bw_session_kbps)?

- Likewise, should the sentence "the minimum RTCP report interval of 5  
seconds recommended in RFC 3550 is dropped" from RFC 4585 be applied  
in all cases (i.e. not only when using the early RTCP reporting rules)?

Thank you very much in advance.

Best Regards,

Mario Montagud



From hoene@uni-tuebingen.de  Mon Nov  5 11:49:40 2012
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Date: Mon, 05 Nov 2012 20:49:25 +0100
From: "Dr. Christian Hoene" <hoene@uni-tuebingen.de>
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Subject: [AVTCORE] RTCP Circuit breaker
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Hi Colin,

the following paper might be interesting to your work.
http://dl.acm.org/citation.cfm?id=507673

Best regards,

  Christian


From jonathan@vidyo.com  Mon Nov  5 13:13:48 2012
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From: Jonathan Lennox <jonathan@vidyo.com>
To: IETF AVTCore WG <avt@ietf.org>
Date: Mon, 5 Nov 2012 16:13:44 -0500
Thread-Topic: RTP Circuit breakers and report blocks
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Subject: [AVTCORE] RTP Circuit breakers and report blocks
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Hi -- Colin asked me to repeat on the list a comment I made at the Mic in t=
he session today, so the idea doesn't get lost.

There was the discussion about when the circuit breakers should trigger for=
 AVPF early feedback, and the comment was it should be done for compound pa=
ckets containing SR or RR packets, but not for those that don't.

If something like the RGRP proposal is adopted, and in the presence of RFC =
3550 report block round-robining, this will need to be more complex.  I thi=
nk the correct formulation might be that you should respond to SR or RR pac=
kets that contain a report block for one of your sources, but not those tha=
t don't.

--
Jonathan Lennox
jonathan@vidyo.com



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Subject: Re: [AVTCORE] I-D Action: draft-ietf-avtcore-rtp-circuit-breakers-01.txt
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Comments on this draft. Some I mentioned in the session today. Smaller ones
I did not.

Abrstract:

Contains the phrase "is expected to stop transmitting." We probably want to
make a stronger statement here. Normative text in the draft uses "MUST".

Suggest untangling and clarifying this sentence:

Old: "It is expected that all RTP

   applications running on best-effort networks will be able to run

   without triggering these circuit breakers in normal operation."


New: "It is expected that, in the absence of severe congestion, all RTP

   applications running on best-effort networks will be able to run

   without triggering these circuit breakers."


Section 3:

Explain why multicast sessions are out of scope.


Last bullet point on page 4 mentions that RTCP report interval is dependent
on number of participants in the session. Number of participants in a
unicast session is always 2 so this is not a contribution if multicast is
our of scope.

Add editorial clarification to rapid feedback bullet on page 5:

Old: "The use of the RTP/AVPF profile is dependent
      on signalling, but is otherwise generally backwards compatible, as
      it keeps the same average RTCP reporting interval as the base RTP
      specification."

New: "The use of the RTP/AVPF profile is dependent
      on signalling, but is otherwise generally backwards compatible with
RTP/AVP, as
      it keeps the same average RTCP reporting interval as the base RTP
      specification."


Section 4 point 2: Multiple packets per frame is a problem. Multiple frames
per packet is not a problem:

Old: "...the jitter is only meaningful for RTP

       flows that send a single data packet for each RTP timestamp value

       (i.e., audio flows, or video flows where each frame comprises one

       RTP packet)."


New: "...the jitter is only meaningful for RTP

       flows that send a single data packet for each RTP timestamp value

       (i.e., audio flows, or video flows where each packet comprises
one or more video frames)."


p.7 The following text may need work:


"The remaining congestion signals are the packet loss fraction and the

   cumulative number of packets lost.  These are robust indicators of

   congestion in a network where packet loss is primarily due to queue

   overflows, although less accurate in networks where losses can be

   caused by non-congestive packet corruption.  TCP uses packet loss as

   a congestion signal."


TCP intentionally causes congestion to determine available link bandwidth.
It is using packet loss to measure capacity. In the presence of
TCP characterizing loss fraction as a roust indicator is probably an
overstatement. The interaction probably deserves some
some further investigation.


Section 4.1: Delete second paragraph. As disused in the session today, if
things are messed up enough to trigger circuit breaker #1, reducing offered
load is not likely a useful response. Section 4.3 contains a copy of this
recommendation and is probably an appropriate recommendation for circuit
breaker #3 described there.

"What it means to cease transmission..." text appears three times (4.1, 4.2
and 4.3). It probably should just appear once and be reference three times.

Section 4.1: Specify what a "long time" is:

Old: "...but would lead to problematic flows running for a

   long time before being cut off."


New: "...but would lead to problematic flows running for at least 15
seconds before being cut off."


Section 4.2: As discussed in the session today, this section needs to be
improved to discuss devices that don't implement RTCP. One obvious
improvement is to condition the receiver behavior described at the bottom
of page 9. A receiver should not trip this circuit breaker if it is
receiving RTP packets from the sender.

p.10: The equation has a divide-by-zero hazard. You may want to explicitly
state that the circuit breaker does not trip if the denominator is 0.

Section 4.3: Simulation is required to gain confidence that there are no
false-trigger conditions for this circuit breaker. Test in the presence of
competing TCP flows. Test different RTCP RR reporting delay and packet loss
scenarios. Solicit more ideas from others.

Kevin Gross
+1-303-447-0517
Media Network Consultant
AVA Networks - www.AVAnw.com <http://www.avanw.com/>, www.X192.org



On Tue, Oct 30, 2012 at 11:05 AM, Colin Perkins <csp@csperkins.org> wrote:

> A summary of the changes in this version is as follows:
>
> - Clarify that multicast congestion control is outside the scope of this
> memo.
>
> - Clarify what it means for an implementation to cease transmission when
> the circuit breaker fires. Specifically, the intention is that the
> application will stop sending RTP data packets until the user makes an
> explicit attempt to restart the call. The draft notes that RTP flows halted
> by the circuit breaker should not be restarted automatically unless the
> sender has received information that the congestion has dissipated.
>
> - Add an explicit RTCP Timeout circuit breaker in Section 4.2 (this is a
> revised and extended version of Section 8 of the -00 draft). Unlike the
> media timeout in Section 4.1, which fires when returning RTCP RR packets
> show that media isn't reaching the receiver, this circuit breaker is
> triggered if RTCP packets are no longer returning to the sender.
>
> - In section 4.3, clarify when high-rate senders can back-off to a lower
> rate when the circuit breaker is triggered, and when they need to cease
> transmission.
>
> - Expand and clarify Section 5 about RTP circuit breakers for systems
> using the RTP/AVPF profile. A sender needs to process early RTCP reports
> that contain an SR/RR packet as-if they were regular RTCP reports, but
> ignore early non-compound RTCP packets without an SR/RR. This allows the
> use of low-overhead early RTP/AVPF feedback without triggering the RTP
> circuit breaker, and so is suitable for RTP congestion control algorithms
> that need to quickly report loss events in between regular RTCP reports.
>
> - Expand and clarify Section 6 about the impact of RTCP XR, to note that
> RTCP XR packets are ignored by the circuit breaker algorithm.
>
> - Expand and clarify Section 7 about the impact of ECN, to note that
> reports of ECN-CE marks are treated as packet loss for the congestion based
> circuit breaker, but ignored for the media timeout and RTCP timeout circuit
> breakers.
>
> - Expand and clarify Section 8 on the security considerations to discuss
> authentication requirements.
>
> I believe this draft is on the agenda for the AVTCORE meeting in Atlanta,
> but feedback via the list is also welcome.
>
> Colin
>
>
>
>
> On 22 Oct 2012, at 23:16, Internet-Drafts@ietf.org wrote:
> > A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> > This draft is a work item of the Audio/Video Transport Core Maintenance
> Working Group of the IETF.
> >
> >       Title           : RTP Congestion Control: Circuit Breakers for
> Unicast Sessions
> >       Author(s)       : Colin Perkins
> >                          Varun Singh
> >       Filename        : draft-ietf-avtcore-rtp-circuit-breakers-01.txt
> >       Pages           : 16
> >       Date            : 2012-10-22
> >
> > Abstract:
> >   The Real-time Transport Protocol (RTP) is widely used in telephony,
> >   video conferencing, and telepresence applications.  Such applications
> >   are often run on best-effort UDP/IP networks.  If congestion control
> >   is not implemented in the applications, then network congestion will
> >   deteriorate the user's multimedia experience.  This document does not
> >   propose a congestion control algorithm; rather, it defines a minimal
> >   set of "circuit-breakers".  Circuit-breakers are conditions under
> >   which an RTP flow is expected to stop transmitting media to protect
> >   the network from excessive congestion.  It is expected that all RTP
> >   applications running on best-effort networks will be able to run
> >   without triggering these circuit breakers in normal operation.  Any
> >   future RTP congestion control specification is expected to operate
> >   within the envelope defined by these circuit breakers.
> >
> >
> > The IETF datatracker status page for this draft is:
> > https://datatracker.ietf.org/doc/draft-ietf-avtcore-rtp-circuit-breakers
> >
> > There's also a htmlized version available at:
> > http://tools.ietf.org/html/draft-ietf-avtcore-rtp-circuit-breakers-01
> >
> > A diff from the previous version is available at:
> >
> http://www.ietf.org/rfcdiff?url2=draft-ietf-avtcore-rtp-circuit-breakers-01
> >
> >
> > Internet-Drafts are also available by anonymous FTP at:
> > ftp://ftp.ietf.org/internet-drafts/
> >
> > _______________________________________________
> > Audio/Video Transport Core Maintenance
> > avt@ietf.org
> > https://www.ietf.org/mailman/listinfo/avt
>
>
>
> --
> Colin Perkins
> http://csperkins.org/
>
>
>
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
>

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Comments on this draft. Some I mentioned in the session today.=A0Smaller=A0=
ones I did not.<div><br></div><div>Abrstract:</div><div><br></div><blockquo=
te style=3D"margin:0px 0px 0px 40px;border:none;padding:0px"><div>Contains =
the phrase &quot;is expected to stop transmitting.&quot; We probably want t=
o make a stronger statement here. Normative text in the draft uses &quot;MU=
ST&quot;.</div>
<div><br></div><div>Suggest untangling and clarifying this sentence:=A0</di=
v><div><br></div><div>Old: &quot;<span style=3D"color:rgb(0,0,0);font-size:=
1em">It is expected that all RTP</span></div><pre style=3D"font-size:1em;ma=
rgin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">
   applications running on best-effort networks will be able to run</pre><p=
re style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)=
">   without triggering these circuit breakers in normal operation.&quot;</=
pre>
<pre style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,=
0)"><br></pre><pre style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;=
color:rgb(0,0,0)">New: &quot;<span style=3D"font-size:1em">It is expected t=
hat, in the absence of severe congestion, all RTP</span></pre>
<pre style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,=
0)">   applications running on best-effort networks will be able to run</pr=
e><pre style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,=
0,0)">
   without triggering these circuit breakers.&quot;</pre></blockquote><div>=
<br></div><div>Section 3:</div><div><br></div><blockquote style=3D"margin:0=
px 0px 0px 40px;border:none;padding:0px"><div>Explain why multicast session=
s are out of scope.</div>
</blockquote><div><br></div><div>Last=A0bullet=A0point on page 4 mentions t=
hat RTCP report interval is dependent on number of participants in the sess=
ion. Number of participants in a unicast session is always 2 so this is not=
 a=A0contribution if multicast is our of scope.</div>
<div><br></div><div>Add editorial=A0clarification=A0to rapid feedback bulle=
t on page 5:</div><div><br></div><blockquote style=3D"margin:0 0 0 40px;bor=
der:none;padding:0px">Old: &quot;The use of the RTP/AVPF profile is depende=
nt<br>
=A0 =A0 =A0 on signalling, but is otherwise generally backwards compatible,=
 as<br>=A0 =A0 =A0 it keeps the same average RTCP reporting interval as the=
 base RTP<br>=A0 =A0 =A0 specification.&quot;<br><br>New: &quot;The use of =
the RTP/AVPF profile is dependent<br>
=A0 =A0 =A0 on signalling, but is otherwise generally backwards compatible =
with RTP/AVP, as<br>=A0 =A0 =A0 it keeps the same average RTCP reporting in=
terval as the base RTP<br>=A0 =A0 =A0 specification.&quot;</blockquote><div=
><br></div><div>
Section 4 point 2: Multiple packets per frame is a problem. Multiple frames=
 per packet is not a problem:</div><div><br></div><blockquote style=3D"marg=
in:0 0 0 40px;border:none;padding:0px"><div>Old: &quot;...<span style=3D"co=
lor:rgb(0,0,0);font-size:1em">the jitter is only meaningful for RTP</span><=
/div>
<pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;col=
or:rgb(0,0,0)">       flows that send a single data packet for each RTP tim=
estamp value</pre><pre class=3D"" style=3D"font-size:1em;margin-top:0px;mar=
gin-bottom:0px;color:rgb(0,0,0)">
       (i.e., audio flows, or video flows where each frame comprises one</p=
re><pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;=
color:rgb(0,0,0)">       RTP packet).&quot;</pre><pre class=3D"" style=3D"f=
ont-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">
<br></pre><pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-bott=
om:0px;color:rgb(0,0,0)">New: &quot;...<span style=3D"font-size:1em">the ji=
tter is only meaningful for RTP</span></pre><pre class=3D"" style=3D"font-s=
ize:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">
       flows that send a single data packet for each RTP timestamp value</p=
re><pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;=
color:rgb(0,0,0)">       (i.e., audio flows, or video flows where each pack=
et comprises one or more video frames).&quot;</pre>
</blockquote><pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-b=
ottom:0px;color:rgb(0,0,0)"><br></pre><pre class=3D"" style=3D"font-size:1e=
m;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">p.7 The following text=
 may need work:</pre>
<pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;col=
or:rgb(0,0,0)"><span style=3D"font-size:1em"><br></span></pre><blockquote s=
tyle=3D"margin:0 0 0 40px;border:none;padding:0px"><pre class=3D"" style=3D=
"font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">
&quot;The remaining congestion signals are the packet loss fraction and the=
</pre><pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-bottom:0=
px;color:rgb(0,0,0)">   cumulative number of packets lost.  These are robus=
t indicators of</pre>
<pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;col=
or:rgb(0,0,0)">   congestion in a network where packet loss is primarily du=
e to queue</pre><pre class=3D"" style=3D"font-size:1em;margin-top:0px;margi=
n-bottom:0px;color:rgb(0,0,0)">
   overflows, although less accurate in networks where losses can be</pre><=
pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;colo=
r:rgb(0,0,0)">   caused by non-congestive packet corruption.  TCP uses pack=
et loss as</pre>
<pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;col=
or:rgb(0,0,0)"><span style=3D"font-size:1em">   a congestion signal.&quot;<=
/span><span style=3D"font-size:1em"> </span></pre><pre class=3D"" style=3D"=
font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">
<span style=3D"font-size:1em"><br></span></pre>TCP intentionally causes con=
gestion to determine available link bandwidth. It is using packet loss to m=
easure capacity. In the presence of TCP=A0characterizing=A0loss fraction as=
 a roust indicator is probably an overstatement. The interaction probably=
=A0deserves=A0some some=A0further=A0investigation.</blockquote>
<div><br></div><div>Section 4.1: Delete=A0second paragraph. As=A0disused in=
 the session=A0today, if things are messed up enough to trigger circuit bre=
aker #1, reducing offered load is not likely a useful response. Section 4.3=
 contains a copy of this recommendation and is probably an appropriate reco=
mmendation for circuit breaker #3 described there.</div>
<div><br></div><div>&quot;What it means to cease transmission...&quot; text=
 appears three times (4.1, 4.2 and 4.3). It probably should just appear onc=
e and be reference three times.</div><div><br></div><div>Section 4.1: Speci=
fy what a &quot;long time&quot; is:</div>
<div><br></div><blockquote style=3D"margin:0 0 0 40px;border:none;padding:0=
px"><div>Old: &quot;...<span style=3D"color:rgb(0,0,0);font-size:1em">but w=
ould lead to problematic flows running for a</span></div><pre class=3D"" st=
yle=3D"font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">
   long time before being cut off.&quot;</pre><pre class=3D"" style=3D"font=
-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><br></pre><pre=
 class=3D"" style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;color:r=
gb(0,0,0)">
New: &quot;...<span style=3D"font-size:1em">but would lead to problematic f=
lows running for at least 15 seconds</span><span style=3D"font-size:1em"> b=
efore being cut off.&quot;</span></pre></blockquote><pre class=3D"" style=
=3D"font-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">
<span style=3D"font-size:1em"><br></span></pre>Section 4.2: As discussed in=
 the session today, this section needs to be improved to discuss devices th=
at don&#39;t implement RTCP. One obvious improvement is to condition the re=
ceiver behavior described at the bottom of page 9. A receiver should not tr=
ip this circuit breaker if it is receiving RTP packets from the sender.<div=
>
<br></div><div>p.10: The equation has a divide-by-zero hazard. You may want=
 to explicitly state that the=A0circuit=A0breaker does not trip if the deno=
minator is 0.</div><div><br></div><div>Section 4.3: Simulation is required =
to gain=A0confidence=A0that there are no false-trigger conditions for this =
circuit breaker. Test in the presence of competing TCP flows. Test differen=
t RTCP RR reporting delay and packet loss scenarios. Solicit more ideas fro=
m others.</div>
<div class=3D"gmail_extra"><br clear=3D"all">Kevin Gross<br><div>+1-303-447=
-0517</div><div>Media Network Consultant<br><div>AVA Networks -=A0<a href=
=3D"http://www.avanw.com/" target=3D"_blank">www.AVAnw.com</a>,=A0<a href=
=3D"http://www.X192.org" target=3D"_blank">www.X192.org</a></div>
</div><br>
<br><br><div class=3D"gmail_quote">On Tue, Oct 30, 2012 at 11:05 AM, Colin =
Perkins <span dir=3D"ltr">&lt;<a href=3D"mailto:csp@csperkins.org" target=
=3D"_blank">csp@csperkins.org</a>&gt;</span> wrote:<br><blockquote class=3D=
"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding=
-left:1ex">
A summary of the changes in this version is as follows:<br>
<br>
- Clarify that multicast congestion control is outside the scope of this me=
mo.<br>
<br>
- Clarify what it means for an implementation to cease transmission when th=
e circuit breaker fires. Specifically, the intention is that the applicatio=
n will stop sending RTP data packets until the user makes an explicit attem=
pt to restart the call. The draft notes that RTP flows halted by the circui=
t breaker should not be restarted automatically unless the sender has recei=
ved information that the congestion has dissipated.<br>

<br>
- Add an explicit RTCP Timeout circuit breaker in Section 4.2 (this is a re=
vised and extended version of Section 8 of the -00 draft). Unlike the media=
 timeout in Section 4.1, which fires when returning RTCP RR packets show th=
at media isn&#39;t reaching the receiver, this circuit breaker is triggered=
 if RTCP packets are no longer returning to the sender.<br>

<br>
- In section 4.3, clarify when high-rate senders can back-off to a lower ra=
te when the circuit breaker is triggered, and when they need to cease trans=
mission.<br>
<br>
- Expand and clarify Section 5 about RTP circuit breakers for systems using=
 the RTP/AVPF profile. A sender needs to process early RTCP reports that co=
ntain an SR/RR packet as-if they were regular RTCP reports, but ignore earl=
y non-compound RTCP packets without an SR/RR. This allows the use of low-ov=
erhead early RTP/AVPF feedback without triggering the RTP circuit breaker, =
and so is suitable for RTP congestion control algorithms that need to quick=
ly report loss events in between regular RTCP reports.<br>

<br>
- Expand and clarify Section 6 about the impact of RTCP XR, to note that RT=
CP XR packets are ignored by the circuit breaker algorithm.<br>
<br>
- Expand and clarify Section 7 about the impact of ECN, to note that report=
s of ECN-CE marks are treated as packet loss for the congestion based circu=
it breaker, but ignored for the media timeout and RTCP timeout circuit brea=
kers.<br>

<br>
- Expand and clarify Section 8 on the security considerations to discuss au=
thentication requirements.<br>
<br>
I believe this draft is on the agenda for the AVTCORE meeting in Atlanta, b=
ut feedback via the list is also welcome.<br>
<br>
Colin<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
<br>
<br>
On 22 Oct 2012, at 23:16, <a href=3D"mailto:Internet-Drafts@ietf.org">Inter=
net-Drafts@ietf.org</a> wrote:<br>
&gt; A New Internet-Draft is available from the on-line Internet-Drafts dir=
ectories.<br>
&gt; This draft is a work item of the Audio/Video Transport Core Maintenanc=
e Working Group of the IETF.<br>
&gt;<br>
&gt; =A0 =A0 =A0 Title =A0 =A0 =A0 =A0 =A0 : RTP Congestion Control: Circui=
t Breakers for Unicast Sessions<br>
&gt; =A0 =A0 =A0 Author(s) =A0 =A0 =A0 : Colin Perkins<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Varun Singh<br>
&gt; =A0 =A0 =A0 Filename =A0 =A0 =A0 =A0: draft-ietf-avtcore-rtp-circuit-b=
reakers-01.txt<br>
&gt; =A0 =A0 =A0 Pages =A0 =A0 =A0 =A0 =A0 : 16<br>
&gt; =A0 =A0 =A0 Date =A0 =A0 =A0 =A0 =A0 =A0: 2012-10-22<br>
&gt;<br>
&gt; Abstract:<br>
&gt; =A0 The Real-time Transport Protocol (RTP) is widely used in telephony=
,<br>
&gt; =A0 video conferencing, and telepresence applications. =A0Such applica=
tions<br>
&gt; =A0 are often run on best-effort UDP/IP networks. =A0If congestion con=
trol<br>
&gt; =A0 is not implemented in the applications, then network congestion wi=
ll<br>
&gt; =A0 deteriorate the user&#39;s multimedia experience. =A0This document=
 does not<br>
&gt; =A0 propose a congestion control algorithm; rather, it defines a minim=
al<br>
&gt; =A0 set of &quot;circuit-breakers&quot;. =A0Circuit-breakers are condi=
tions under<br>
&gt; =A0 which an RTP flow is expected to stop transmitting media to protec=
t<br>
&gt; =A0 the network from excessive congestion. =A0It is expected that all =
RTP<br>
&gt; =A0 applications running on best-effort networks will be able to run<b=
r>
&gt; =A0 without triggering these circuit breakers in normal operation. =A0=
Any<br>
&gt; =A0 future RTP congestion control specification is expected to operate=
<br>
&gt; =A0 within the envelope defined by these circuit breakers.<br>
&gt;<br>
&gt;<br>
&gt; The IETF datatracker status page for this draft is:<br>
&gt; <a href=3D"https://datatracker.ietf.org/doc/draft-ietf-avtcore-rtp-cir=
cuit-breakers" target=3D"_blank">https://datatracker.ietf.org/doc/draft-iet=
f-avtcore-rtp-circuit-breakers</a><br>
&gt;<br>
&gt; There&#39;s also a htmlized version available at:<br>
&gt; <a href=3D"http://tools.ietf.org/html/draft-ietf-avtcore-rtp-circuit-b=
reakers-01" target=3D"_blank">http://tools.ietf.org/html/draft-ietf-avtcore=
-rtp-circuit-breakers-01</a><br>
&gt;<br>
&gt; A diff from the previous version is available at:<br>
&gt; <a href=3D"http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-avtcore-rtp-c=
ircuit-breakers-01" target=3D"_blank">http://www.ietf.org/rfcdiff?url2=3Ddr=
aft-ietf-avtcore-rtp-circuit-breakers-01</a><br>
&gt;<br>
&gt;<br>
&gt; Internet-Drafts are also available by anonymous FTP at:<br>
&gt; <a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=3D"_blank">ftp:=
//ftp.ietf.org/internet-drafts/</a><br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; Audio/Video Transport Core Maintenance<br>
&gt; <a href=3D"mailto:avt@ietf.org">avt@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/avt" target=3D"_blank=
">https://www.ietf.org/mailman/listinfo/avt</a><br>
<br>
<br>
<br>
</div></div><span class=3D"HOEnZb"><font color=3D"#888888">--<br>
Colin Perkins<br>
<a href=3D"http://csperkins.org/" target=3D"_blank">http://csperkins.org/</=
a><br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
<br>
_______________________________________________<br>
Audio/Video Transport Core Maintenance<br>
<a href=3D"mailto:avt@ietf.org">avt@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/avt" target=3D"_blank">htt=
ps://www.ietf.org/mailman/listinfo/avt</a><br>
</div></div></blockquote></div><br></div>

--047d7b621ede0d0ad204cdc68f40--

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From: Kevin Gross <kevin.gross@avanw.com>
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Subject: Re: [AVTCORE] FW: New Version Notification for draft-rescorla-avtcore-6222bis-00.txt
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--047d7b62243cff4a5f04cdc6dc43
Content-Type: text/plain; charset=ISO-8859-1

Since it looks like this will be accepted as a workgroup item, here are a
few comments:

I'm not clear on how a bis works but I assume this draft needs to be able
to stand on it's own as it proposes to obsolete RFC 6222. Some or all
references to RFC 6222 probably need to be removed. Particularly confusing
are statements in the abstract that seem to indicate that only parts of RFC
6222 are being replaced.

Second bullet point in section 4.2 Item (a): I don't understand how a CNAME
based on MAC address would be considered short-term. Seems like this would
have same or better persistence properties as the UUID in the first bullet
point.

Second bullet point in section 4.2 Item (b): Proposes truncating our nice
96-bit random CNAME to 48 bits. I think we have an unacceptable opportunity
for duplication with this approach. This CNAME should probably use RFC 4648
in which case these CNAMEs take the same form as the per-session CNAMES but
differ in the requirement to create once at software initialization. Is it
necessary for the different types of CNAMEs to have different appearance?

p.6: Phrase, "(to compromise between packet size and uniqueness)" is scary.
Exude more confidence here and/or insert a forward reference to section 6.1.

Section 6.1: Please provide a collision probability for a worst-case
population. 2^48 is an unreasonable population for this application (and
50% is a scary probability). Is 2^40 a worst-case population. Seems large.

Kevin Gross
+1-303-447-0517
Media Network Consultant
AVA Networks - www.AVAnw.com <http://www.avanw.com/>, www.X192.org



On Sat, Oct 13, 2012 at 10:32 PM, Ali C. Begen (abegen) <abegen@cisco.com>wrote:

> We have submitted a draft to replace 6222. This document is a new version
> of 6222 and has the fixes proposed by Eric in the last meeting.
>
> Comments are welcome.
> -acbegen
>
> -----Original Message-----
> From: "internet-drafts@ietf.org" <internet-drafts@ietf.org>
> Date: Saturday, October 13, 2012 10:30 PM
> To: "Ali C. Begen" <abegen@cisco.com>
> Cc: "Eric com>" <ekr@rtfm.com>
> Subject: New Version Notification for draft-rescorla-avtcore-6222bis-00.txt
>
> >
> >A new version of I-D, draft-rescorla-avtcore-6222bis-00.txt
> >has been successfully submitted by Ali Begen and posted to the
> >IETF repository.
> >
> >Filename:       draft-rescorla-avtcore-6222bis
> >Revision:       00
> >Title:          Guidelines for Choosing RTP Control Protocol (RTCP)
> Canonical
> >Names (CNAMEs)
> >Creation date:  2012-10-14
> >WG ID:          Individual Submission
> >Number of pages: 10
> >URL:
> >http://www.ietf.org/internet-drafts/draft-rescorla-avtcore-6222bis-00.txt
> >Status:
> >http://datatracker.ietf.org/doc/draft-rescorla-avtcore-6222bis
> >Htmlized:
> >http://tools.ietf.org/html/draft-rescorla-avtcore-6222bis-00
> >
> >
> >Abstract:
> >   The RTP Control Protocol (RTCP) Canonical Name (CNAME) is a
> >   persistent transport-level identifier for an RTP endpoint.  While the
> >   Synchronization Source (SSRC) identifier of an RTP endpoint may
> >   change if a collision is detected or when the RTP application is
> >   restarted, its RTCP CNAME is meant to stay unchanged, so that RTP
> >   endpoints can be uniquely identified and associated with their RTP
> >   media streams.
> >
> >   For proper functionality, RTCP CNAMEs should be unique within the
> >   participants of an RTP session.  However, the existing guidelines for
> >   choosing the RTCP CNAME provided in the RTP standard are insufficient
> >   to achieve this uniqueness.  RFC 6222 was published to update those
> >   guidelines to allow endpoints to choose unique RTCP CNAMEs.
> >   Unfortunately, later investigations showed that some parts of the new
> >   algorithms were unnecessarily complicated and/or ineffective.  This
> >   document specifies a replacement for those parts.
> >
> >
> >
> >
> >
> >The IETF Secretariat
> >
>
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
>

--047d7b62243cff4a5f04cdc6dc43
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Since it looks like this will be accepted as a workgroup item, here are a f=
ew comments:<div><br></div><div>I&#39;m not clear on how a bis works but I =
assume this draft needs to be able to stand on it&#39;s own as it proposes =
to obsolete RFC 6222. Some or all references to RFC 6222 probably need to b=
e removed. Particularly confusing are statements in the abstract that seem =
to indicate that only parts of RFC 6222 are being replaced.</div>
<div><br></div><div>Second bullet point in section 4.2=A0Item (a): I don&#3=
9;t understand how a CNAME based on MAC address would be considered short-t=
erm. Seems like this would have same or better persistence properties as th=
e UUID in the first bullet point.</div>
<div><br></div><div>Second bullet point in section 4.2=A0Item (b): Proposes=
 truncating our nice 96-bit random CNAME to 48 bits. I think we have an una=
cceptable opportunity for duplication with this approach. This CNAME should=
 probably use RFC 4648 in which case these CNAMEs take the same form as the=
 per-session CNAMES but differ in the requirement to create once at softwar=
e=A0initialization. Is it necessary for the different types of CNAMEs to ha=
ve different appearance?<br>
</div><div><br></div><div>p.6: Phrase, &quot;<font color=3D"#000000"><span =
style=3D"font-size:1em">(to compromise between packet size and uniqueness)&=
quot; is scary. Exude more=A0</span>confidence<span style=3D"font-size:1em"=
>=A0here and/or insert a forward reference to section 6.1.</span></font></d=
iv>
<div><font color=3D"#000000"><span style=3D"font-size:1em"><br></span></fon=
t></div><div><font color=3D"#000000"><span style=3D"font-size:1em">Section =
6.1: Please provide a=A0</span>collision<span style=3D"font-size:1em">=A0</=
span>probability<span style=3D"font-size:1em">=A0for a worst-case populatio=
n. 2^48 is an unreasonable population for this application (and 50% is a sc=
ary=A0</span>probability<span style=3D"font-size:1em">). Is 2^40 a worst-ca=
se population. Seems large.</span></font></div>
<div class=3D"gmail_extra"><br clear=3D"all">Kevin Gross<br><div>+1-303-447=
-0517</div><div>Media Network Consultant<br><div>AVA Networks -=A0<a href=
=3D"http://www.avanw.com/" target=3D"_blank">www.AVAnw.com</a>,=A0<a href=
=3D"http://www.X192.org" target=3D"_blank">www.X192.org</a></div>
</div><br>
<br><br><div class=3D"gmail_quote">On Sat, Oct 13, 2012 at 10:32 PM, Ali C.=
 Begen (abegen) <span dir=3D"ltr">&lt;<a href=3D"mailto:abegen@cisco.com" t=
arget=3D"_blank">abegen@cisco.com</a>&gt;</span> wrote:<br><blockquote clas=
s=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pad=
ding-left:1ex">
We have submitted a draft to replace 6222. This document is a new version<b=
r>
of 6222 and has the fixes proposed by Eric in the last meeting.<br>
<br>
Comments are welcome.<br>
-acbegen<br>
<br>
-----Original Message-----<br>
From: &quot;<a href=3D"mailto:internet-drafts@ietf.org">internet-drafts@iet=
f.org</a>&quot; &lt;<a href=3D"mailto:internet-drafts@ietf.org">internet-dr=
afts@ietf.org</a>&gt;<br>
Date: Saturday, October 13, 2012 10:30 PM<br>
To: &quot;Ali C. Begen&quot; &lt;<a href=3D"mailto:abegen@cisco.com">abegen=
@cisco.com</a>&gt;<br>
Cc: &quot;Eric com&gt;&quot; &lt;<a href=3D"mailto:ekr@rtfm.com">ekr@rtfm.c=
om</a>&gt;<br>
Subject: New Version Notification for draft-rescorla-avtcore-6222bis-00.txt=
<br>
<br>
&gt;<br>
&gt;A new version of I-D, draft-rescorla-avtcore-6222bis-00.txt<br>
&gt;has been successfully submitted by Ali Begen and posted to the<br>
&gt;IETF repository.<br>
&gt;<br>
&gt;Filename: =A0 =A0 =A0 draft-rescorla-avtcore-6222bis<br>
&gt;Revision: =A0 =A0 =A0 00<br>
&gt;Title: =A0 =A0 =A0 =A0 =A0Guidelines for Choosing RTP Control Protocol =
(RTCP) Canonical<br>
&gt;Names (CNAMEs)<br>
&gt;Creation date: =A02012-10-14<br>
&gt;WG ID: =A0 =A0 =A0 =A0 =A0Individual Submission<br>
&gt;Number of pages: 10<br>
&gt;URL:<br>
&gt;<a href=3D"http://www.ietf.org/internet-drafts/draft-rescorla-avtcore-6=
222bis-00.txt" target=3D"_blank">http://www.ietf.org/internet-drafts/draft-=
rescorla-avtcore-6222bis-00.txt</a><br>
&gt;Status:<br>
&gt;<a href=3D"http://datatracker.ietf.org/doc/draft-rescorla-avtcore-6222b=
is" target=3D"_blank">http://datatracker.ietf.org/doc/draft-rescorla-avtcor=
e-6222bis</a><br>
&gt;Htmlized:<br>
&gt;<a href=3D"http://tools.ietf.org/html/draft-rescorla-avtcore-6222bis-00=
" target=3D"_blank">http://tools.ietf.org/html/draft-rescorla-avtcore-6222b=
is-00</a><br>
&gt;<br>
&gt;<br>
&gt;Abstract:<br>
&gt; =A0 The RTP Control Protocol (RTCP) Canonical Name (CNAME) is a<br>
&gt; =A0 persistent transport-level identifier for an RTP endpoint. =A0Whil=
e the<br>
&gt; =A0 Synchronization Source (SSRC) identifier of an RTP endpoint may<br=
>
&gt; =A0 change if a collision is detected or when the RTP application is<b=
r>
&gt; =A0 restarted, its RTCP CNAME is meant to stay unchanged, so that RTP<=
br>
&gt; =A0 endpoints can be uniquely identified and associated with their RTP=
<br>
&gt; =A0 media streams.<br>
&gt;<br>
&gt; =A0 For proper functionality, RTCP CNAMEs should be unique within the<=
br>
&gt; =A0 participants of an RTP session. =A0However, the existing guideline=
s for<br>
&gt; =A0 choosing the RTCP CNAME provided in the RTP standard are insuffici=
ent<br>
&gt; =A0 to achieve this uniqueness. =A0RFC 6222 was published to update th=
ose<br>
&gt; =A0 guidelines to allow endpoints to choose unique RTCP CNAMEs.<br>
&gt; =A0 Unfortunately, later investigations showed that some parts of the =
new<br>
&gt; =A0 algorithms were unnecessarily complicated and/or ineffective. =A0T=
his<br>
&gt; =A0 document specifies a replacement for those parts.<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;The IETF Secretariat<br>
&gt;<br>
<br>
_______________________________________________<br>
Audio/Video Transport Core Maintenance<br>
<a href=3D"mailto:avt@ietf.org">avt@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/avt" target=3D"_blank">htt=
ps://www.ietf.org/mailman/listinfo/avt</a><br>
</blockquote></div><br></div>

--047d7b62243cff4a5f04cdc6dc43--

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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: Kevin Gross <kevin.gross@avanw.com>
Thread-Topic: [AVTCORE] FW: New Version Notification for draft-rescorla-avtcore-6222bis-00.txt
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From:  Kevin Gross <kevin.gross@avanw.com>
Date:  Monday, November 5, 2012 6:18 PM
To:  "Ali C. Begen" <abegen@cisco.com>
Cc:  "avt@ietf.org" <avt@ietf.org>
Subject:  Re: [AVTCORE] FW: New Version Notification for
draft-rescorla-avtcore-6222bis-00.txt


>
>Since it looks like this will be accepted as a workgroup item, here are a
>few comments:
>
>I'm not clear on how a bis works but I assume this draft needs to be able
>to stand on it's own as it proposes to obsolete RFC 6222. Some or all
>references to RFC 6222 probably need to be removed. Particularly
>confusing are statements in the abstract that
> seem to indicate that only parts of RFC 6222 are being replaced.
>

I think it is ok to refer to 6222 to explain what things caused a concern
and are being fixed in this document.Now reading the last sentence in the
abstract, yeah, it sounds a bit funny. It should rather say "This document
addresses these concerns and replaces RFC 6222."

>
>Second bullet point in section 4.2 Item (a): I don't understand how a
>CNAME based on MAC address would be considered short-term. Seems like
>this would have same or better persistence properties as the UUID in the
>first bullet point.

Well, there could be multiple MACs associated with an endpoint and they
could be changed via software, etc. So, I don't think it is as persistent
as UUID.

>
>Second bullet point in section 4.2 Item (b): Proposes truncating our nice
>96-bit random CNAME to 48 bits. I think we have an unacceptable
>opportunity for duplication with this approach. This CNAME should
>probably use RFC 4648 in which case these CNAMEs
> take the same form as the per-session CNAMES but differ in the
>requirement to create once at software initialization. Is it necessary
>for the different types of CNAMEs to have different appearance?

I don=B9t think there is such a requirement and your suggestion makes sense=
.
BTW, there is a sentence that needs to be deleted here (artifact from
6222): "hence, only the initial SSRC value chosen by the endpoint is used"

>
>p.6: Phrase, "(to compromise between packet size and uniqueness)" is
>scary. Exude more confidence here and/or insert a forward reference to
>section 6.1.

OK.

>
>Section 6.1: Please provide a collision probability for a worst-case
>population. 2^48 is an unreasonable population for this application
> (and 50% is a scary probability). Is 2^40 a worst-case population. Seems
>large.

Eric, do you have any numbers here?

-acbegen

>
>Kevin Gross
>+1-303-447-0517
>Media Network Consultant
>AVA Networks - www.AVAnw.com <http://www.avanw.com/>, www.X192.org
><http://www.X192.org>
>
>
>
>
>On Sat, Oct 13, 2012 at 10:32 PM, Ali C. Begen (abegen)
><abegen@cisco.com> wrote:
>
>We have submitted a draft to replace 6222. This document is a new version
>of 6222 and has the fixes proposed by Eric in the last meeting.
>
>Comments are welcome.
>-acbegen
>
>-----Original Message-----
>From: "internet-drafts@ietf.org" <internet-drafts@ietf.org>
>Date: Saturday, October 13, 2012 10:30 PM
>To: "Ali C. Begen" <abegen@cisco.com>
>Cc: "Eric com>" <ekr@rtfm.com>
>Subject: New Version Notification for
>draft-rescorla-avtcore-6222bis-00.txt
>
>>
>>A new version of I-D, draft-rescorla-avtcore-6222bis-00.txt
>>has been successfully submitted by Ali Begen and posted to the
>>IETF repository.
>>
>>Filename:       draft-rescorla-avtcore-6222bis
>>Revision:       00
>>Title:          Guidelines for Choosing RTP Control Protocol (RTCP)
>>Canonical
>>Names (CNAMEs)
>>Creation date:  2012-10-14
>>WG ID:          Individual Submission
>>Number of pages: 10
>>URL:
>>http://www.ietf.org/internet-drafts/draft-rescorla-avtcore-6222bis-00.txt
>>Status:
>>http://datatracker.ietf.org/doc/draft-rescorla-avtcore-6222bis
>>Htmlized:
>>http://tools.ietf.org/html/draft-rescorla-avtcore-6222bis-00
>>
>>
>>Abstract:
>>   The RTP Control Protocol (RTCP) Canonical Name (CNAME) is a
>>   persistent transport-level identifier for an RTP endpoint.  While the
>>   Synchronization Source (SSRC) identifier of an RTP endpoint may
>>   change if a collision is detected or when the RTP application is
>>   restarted, its RTCP CNAME is meant to stay unchanged, so that RTP
>>   endpoints can be uniquely identified and associated with their RTP
>>   media streams.
>>
>>   For proper functionality, RTCP CNAMEs should be unique within the
>>   participants of an RTP session.  However, the existing guidelines for
>>   choosing the RTCP CNAME provided in the RTP standard are insufficient
>>   to achieve this uniqueness.  RFC 6222 was published to update those
>>   guidelines to allow endpoints to choose unique RTCP CNAMEs.
>>   Unfortunately, later investigations showed that some parts of the new
>>   algorithms were unnecessarily complicated and/or ineffective.  This
>>   document specifies a replacement for those parts.
>>
>>
>>
>>
>>
>>The IETF Secretariat
>>
>
>_______________________________________________
>Audio/Video Transport Core Maintenance
>avt@ietf.org
>https://www.ietf.org/mailman/listinfo/avt
>
>
>
>
>
>
>



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Thread-Topic: Do RFC 4585 and RFC 5104 mandate the usage of the a=rtcp-fb SDP attribute?
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From: "Belling, Thomas (NSN - DE/Munich)" <thomas.belling@nsn.com>
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Subject: [AVTCORE] Do RFC 4585 and RFC 5104 mandate the usage of the a=rtcp-fb SDP attribute?
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Dear RTP/AVPF experts,

Triggered by some recent discussion in 3GPP SA4, I would welcome your =
opinion on the following question:


Do RFC 4585 and RFC 5104 allow sending the RTCP feedback messages =
defined in those RFCs, if AVPF has been negotiated using SDP =
offer-answer, but no  a=3Drtcp-fb SDP attribute was included in the SDP =
negotiation?



A more detailed discussion is below:


>From RFC 4585:

4.2.  RTCP Feedback Capability Attribute

...
   The "rtcp-fb" attribute SHOULD be used to indicate which RTCP FB
   messages MAY be used in this media session for the indicated payload  =
type.
...
   If no "rtcp-fb" attribute is specified, the RTP receivers MAY send
   feedback using other suitable RTCP feedback packets as defined for
   the respective media type.  The RTP receivers MUST NOT rely on the
   RTP senders reacting to any of the FB messages.  The RTP sender MAY
   choose to ignore some feedback messages.

   If one or more "rtcp-fb" attributes are present in a media session
   description, the RTCP receivers for the media session(s) containing
   the "rtcp-fb"

   o  MUST ignore all "rtcp-fb" attributes of which they do not fully
      understand the semantics (i.e., where they do not understand the
      meaning of all values in the "a=3Drtcp-fb" line);

   o  SHOULD provide feedback information as specified in this document
      using any of the RTCP feedback packets as specified in one of the
      "rtcp-fb" attributes for this media session; and

   o  MUST NOT use other FB messages than those listed in one of the
      "rtcp-fb" attribute lines.

   When used in conjunction with the offer/answer model [8], the offerer
   MAY present a set of these AVPF attributes to its peer.  The answerer
   MUST remove all attributes it does not understand as well as those it
   does not support in general or does not wish to use in this
   particular media session.  The answerer MUST NOT add feedback
   parameters to the media description and MUST NOT alter values of such
   parameters.  The answer is binding for the media session, and both
   offerer and answerer MUST only use feedback mechanisms negotiated in
   this way.  Both offerer and answerer MAY independently decide to send
   RTCP FB messages of only a subset of the negotiated feedback
   mechanisms, but they SHOULD react properly to all types of the
   negotiated FB messages when received.


>From RFC 5104

7.2.  Offer-Answer

   The Offer/Answer [RFC3264] implications for codec control protocol
   feedback messages are similar to those described in [RFC4585].  The
   offerer MAY indicate the capability to support selected codec
   commands and indications.  The answerer MUST remove all CCM
   parameters corresponding to the CCMs that it does not wish to support
   in this particular media session (for example, because it does not
   implement the message in question, or because its application logic
   suggests that support of the message adds no value).  The answerer
   MUST NOT add new ccm parameters in addition to what has been offered.
   The answer is binding for the media session and both offerer and
   answerer MUST NOT use any feedback messages other than what both
   sides have explicitly indicated as being supported.  In other words,
   only the joint subset of CCM parameters from the offer and answer may
   be used.

   Note that including a CCM parameter in an offer or answer indicates
   that the party (offerer or answerer) is at least capable of receiving
   the corresponding CCM(s) and act upon them.  In cases when the
   reception of a negotiated CCM mandates the party to respond with
   another CCM, it must also have that capability.  Although it is not
   mandated to initiate CCMs of any negotiated type, it is generally
   expected that a party will initiate CCMs when appropriate.

   The session maximum packet rate parameter part of the TMMBR
   indication is declarative, and the highest value from offer and
   answer SHALL be used.  If the session maximum packet rate parameter
   is not present in an offer, it SHALL NOT be included by the answerer.




A suggested Interpretation for RFC 4585 in 3GPP SA4 was :

The SDP offer may or may not include one or more a=3Drtcp-fb attributes =
for negotiating various AVPF feedback messages. (this is of course =
assuming that AVPF is negotiated on the m=3D line).

IF attribute lines with a=3Drtcp-fb are NOT included THEN
      Both end-points may send any AVPF FB messages (if AVPF is agreed)
      It is not guaranteed that any AVPF FB messages will work (in =
either direction)

IF attribute lines with a=3Drtcp-fb are included for some or all of the =
AVPF FB messages THEN
      The answerer must remove the attribute lines for the AVPF FB =
messages it does not want to use
      Only those AVPF FB messages that are in the SDP answer are allowed =
to be used, i.e. it is forbidden to send the AVPF FB messages that are =
not in the SDP answer

Is that correct?




Does the same interpretation apply for RFC 5104?

Based on:  "The answer is binding for the media session and both offerer =
and answerer MUST NOT use any feedback messages other than what both  =
sides have explicitly indicated as being supported. " , the following =
alternative interpretation has been suggested:
*     Every CCM FB message that one intend to use must be negotiated in =
SDP offer-answer.
*     Any CCM FB message that has not been negotiated (and therefore is =
not agreed) in SDP offer-answer cannot be used in the session.



Many thanks for your help,

Thomas



----------------------------------=20
Dr. Thomas Belling=20
3GPP Standardisation
Nokia Siemens Networks=20



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<!-- Converted from text/rtf format -->

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Calibri">Dear RTP/AVPF =
experts,</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Calibri">T</FONT><FONT =
FACE=3D"Calibri">riggered</FONT> <FONT FACE=3D"Calibri">by some recent =
discussion in 3GPP SA4, I would welcome your opinion on the following =
question</FONT></SPAN><SPAN LANG=3D"en-us"><FONT =
FACE=3D"Calibri">:</FONT></SPAN></P>
<BR>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Calibri">Do RFC 4585 and =
RFC 5104</FONT></SPAN><SPAN LANG=3D"en-us"> <FONT =
FACE=3D"Calibri">allow</FONT></SPAN><SPAN LANG=3D"en-us"> <FONT =
FACE=3D"Calibri">sending</FONT></SPAN><SPAN LANG=3D"en-us"><FONT =
FACE=3D"Calibri"></FONT></SPAN><SPAN LANG=3D"en-us"> <FONT =
FACE=3D"Calibri">the</FONT></SPAN><SPAN LANG=3D"en-us"> <FONT =
FACE=3D"Calibri">RTCP feedba</FONT></SPAN><SPAN LANG=3D"en-us"><FONT =
FACE=3D"Calibri">ck m</FONT><FONT FACE=3D"Calibri">essages</FONT> <FONT =
FACE=3D"Calibri">defined in those RFCs,</FONT> <FONT FACE=3D"Calibri">if =
AVPF has been</FONT></SPAN><SPAN LANG=3D"en-us"> <FONT =
FACE=3D"Calibri">negotiated</FONT></SPAN><SPAN LANG=3D"en-us"><FONT =
FACE=3D"Calibri"></FONT></SPAN><SPAN LANG=3D"en-us"> <FONT =
FACE=3D"Calibri">using SDP offer-answer, but no</FONT></SPAN><SPAN =
LANG=3D"en-us">&nbsp;<FONT FACE=3D"Calibri"> a=3Drtcp-fb SDP =
attribute</FONT></SPAN><SPAN LANG=3D"en-us"><FONT FACE=3D"Calibri"> =
w</FONT><FONT FACE=3D"Calibri">as included</FONT></SPAN><SPAN =
LANG=3D"en-us"><FONT FACE=3D"Calibri"> in the SDP =
negotiation</FONT></SPAN><SPAN LANG=3D"en-us"><FONT =
FACE=3D"Calibri">?</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN></P>
<BR>
<BR>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Calibri">A more detailed =
discussion is below:</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Calibri">From RFC =
4585:</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT =
SIZE=3D2 FACE=3D"Courier New">4.2.&nbsp; RTCP Feedback Capability =
Attribute</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT =
FACE=3D"Calibri">&#8230;</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT =
SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; The &quot;rtcp-fb&quot; =
attribute SHOULD be used to indicate which RTCP FB</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; messages MAY be used in this media session for the =
indicated payload</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New"></FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us">&nbsp;<FONT FACE=3D"Times New =
Roman"> type.</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT =
FACE=3D"Calibri">&#8230;</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN LANG=3D"de"></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; If no =
&quot;rtcp-fb&quot; attribute is specified, the RTP receivers MAY =
send</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; feedback using other suitable RTCP feedback packets as =
defined for</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; the respective media type.&nbsp; The RTP receivers =
MUST NOT rely on the</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; RTP senders reacting to any of the FB messages.&nbsp; =
The RTP sender MAY</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; choose to ignore some feedback =
messages.</FONT></SPAN><SPAN LANG=3D"de"></SPAN><SPAN =
LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"de"></SPAN><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"de"></SPAN><SPAN LANG=3D"en-us"><FONT =
SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; If one or more =
&quot;rtcp-fb&quot; attributes are present in a media =
session</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; description, the RTCP receivers for the media =
session(s) containing</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; the &quot;rtcp-fb&quot;</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; o&nbsp; MUST ignore all &quot;rtcp-fb&quot; attributes =
of which they do not fully</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; understand the semantics (i.e., =
where they do not understand the</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; meaning of all values in the =
&quot;a=3Drtcp-fb&quot; line);</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; o&nbsp; SHOULD provide feedback information as =
specified in this document</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; using any of the RTCP feedback =
packets as specified in one of the</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &quot;rtcp-fb&quot; attributes for =
this media session; and</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; o&nbsp; MUST NOT use other FB messages than those =
listed in one of the</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &quot;rtcp-fb&quot; attribute =
lines.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; When used in conjunction with the offer/answer model =
[8], the offerer</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; MAY present a set of these AVPF attributes to its =
peer.&nbsp; The answerer</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; MUST remove all attributes it does not understand as =
well as those it</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; does not support in general or does not wish to use in =
this</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; particular media session.&nbsp; The answerer MUST NOT =
add feedback</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; parameters to the media description and MUST NOT alter =
values of such</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; parameters.&nbsp; The answer is binding for the media =
session, and both</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; offerer and answerer MUST only use feedback mechanisms =
negotiated in</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; this way.&nbsp; Both offerer and answerer MAY =
independently decide to send</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; RTCP FB messages of only a subset of the negotiated =
feedback</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; mechanisms, but they SHOULD react properly to all =
types of the</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; negotiated FB messages when =
received.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"de"></SPAN><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"de"></SPAN><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT =
FACE=3D"Calibri">From</FONT></SPAN><SPAN LANG=3D"en-us"> <FONT =
FACE=3D"Calibri">RFC 5104</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN LANG=3D"de"></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New">7.2.&nbsp; =
Offer-Answer</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; The Offer/Answer [RFC3264] implications for codec =
control protocol</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; feedback messages are similar to those described in =
[RFC4585].&nbsp; The</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; offerer MAY indicate the capability to support =
selected codec</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; commands and indications.&nbsp; The answerer MUST =
remove all CCM</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; parameters corresponding to the CCMs that it does not =
wish to support</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; in this particular media session (for example, because =
it does not</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; implement the message in question, or because its =
application logic</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; suggests that support of the message adds no =
value).&nbsp; The answerer</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; MUST NOT add new ccm parameters in addition to what =
has been offered.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; The answer is binding for the media session and both =
offerer and</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; answerer MUST NOT use any feedback messages other than =
what both</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; sides have explicitly indicated as being =
supported.&nbsp; In other words,</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; only the joint subset of CCM parameters from the offer =
and answer may</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; be used</FONT></SPAN><SPAN LANG=3D"de"></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New">.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"de"></SPAN><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"de"></SPAN><SPAN LANG=3D"en-us"><FONT =
SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; Note that including a CCM =
parameter in an offer or answer indicates</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; that the party (offerer or answerer) is at least =
capable of receiving</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; the corresponding CCM(s) and act upon them.&nbsp; In =
cases when the</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; reception of a negotiated CCM mandates the party to =
respond with</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; another CCM, it must also have that capability.&nbsp; =
Although it is not</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; mandated to initiate CCMs of any negotiated type, it =
is generally</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; expected that a party will initiate CCMs when =
appropriate.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; The session maximum packet rate parameter part of the =
TMMBR</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; indication is declarative, and the highest value from =
offer and</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; answer SHALL be used.&nbsp; If the session maximum =
packet rate parameter</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; is not present in an offer, it SHALL NOT be included =
by the answerer.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"de"></SPAN><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Calibri">A =
suggested</FONT><FONT FACE=3D"Calibri"> =
Interpretation</FONT></SPAN><SPAN LANG=3D"en-us"><FONT FACE=3D"Calibri"> =
for</FONT></SPAN><SPAN LANG=3D"en-us"> <FONT FACE=3D"Calibri">RFC =
4585</FONT></SPAN><SPAN LANG=3D"en-us"><FONT FACE=3D"Calibri"></FONT> =
<FONT FACE=3D"Calibri">in</FONT></SPAN><SPAN LANG=3D"en-us"> <FONT =
FACE=3D"Calibri">3GPP</FONT></SPAN><SPAN LANG=3D"en-us"> <FONT =
FACE=3D"Calibri">SA</FONT><FONT FACE=3D"Calibri">4</FONT></SPAN><SPAN =
LANG=3D"en-us"><FONT FACE=3D"Calibri"></FONT></SPAN><SPAN =
LANG=3D"en-us"> <FONT FACE=3D"Calibri">was</FONT> <FONT =
FACE=3D"Calibri">:</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Calibri">The SDP offer =
may or may not include one or more a=3Drtcp-fb attributes for =
negotiating various AVPF feedback messages. (this is of course assuming =
that AVPF is negotiated on the m=3D line).</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Calibri">IF attribute =
lines with a=3Drtcp-fb are NOT included THEN</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Calibri">&nbsp;</FONT> =
<FONT FACE=3D"Calibri">&nbsp;&nbsp;&nbsp; Both end-points may send any =
AVPF FB messages (if AVPF is agreed)</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT =
FACE=3D"Calibri">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; It is not guaranteed =
that any AVPF FB messages will work (in either =
direction)</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Calibri">IF attribute =
lines with a=3Drtcp-fb are included for some or all of the AVPF FB =
messages THEN</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT =
FACE=3D"Calibri">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; The answerer must remove =
the attribute lines for the AVPF FB messages it does not want to =
use</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT =
FACE=3D"Calibri">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Only those AVPF FB =
messages that are in the SDP answer are allowed to be used, i.e. it is =
forbidden to send the AVPF FB messages that are not in the S</FONT><FONT =
FACE=3D"Calibri">DP answer</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Calibri">Is that =
correct?</FONT></SPAN></P>
<BR>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Calibri">Does the same =
interpretation apply for RFC 5104?</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Calibri">Based =
on</FONT></SPAN><SPAN LANG=3D"en-us"><FONT =
FACE=3D"Calibri">:</FONT></SPAN><SPAN LANG=3D"en-us">&nbsp;<FONT =
FACE=3D"Calibri"></FONT></SPAN><SPAN LANG=3D"en-us"> <FONT =
FACE=3D"Calibri">&#8220;</FONT><FONT FACE=3D"Calibri">The answer is =
binding for the media session and both offerer and</FONT></SPAN><SPAN =
LANG=3D"en-us"><FONT FACE=3D"Calibri"></FONT></SPAN><SPAN =
LANG=3D"en-us"> <FONT FACE=3D"Calibri">answerer MUST NOT use any =
feedback messages other than what both</FONT><FONT =
FACE=3D"Calibri">&nbsp; sides have explicitly indicated as being =
supported.</FONT> <FONT FACE=3D"Calibri">&#8220;</FONT><FONT =
FACE=3D"Calibri"></FONT></SPAN><SPAN LANG=3D"en-us"> <FONT =
FACE=3D"Calibri">, the following alternative interpretation has been =
suggested:</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT =
FACE=3D"Calibri">*&nbsp;&nbsp;&nbsp;&nbsp;</FONT> <FONT =
FACE=3D"Calibri">Every CCM FB message that one intend to use must be =
negot</FONT><FONT FACE=3D"Calibri">iated in SDP =
offer-answer.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT =
FACE=3D"Calibri">*&nbsp;&nbsp;&nbsp;&nbsp;</FONT> <FONT =
FACE=3D"Calibri">Any CCM FB message that has not been negotiated (and =
therefore is not agreed) in SDP offer-answer cannot be used in the =
session.</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN></P>
<BR>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Calibri">Many thanks for =
your help,</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT =
FACE=3D"Calibri">Thomas</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"de-de"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"de-de"><FONT SIZE=3D2 =
FACE=3D"Arial">----------------------------------</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"de-de"></SPAN><SPAN LANG=3D"de-de"><FONT FACE=3D"Arial"><BR>
</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"de-de"></SPAN><SPAN =
LANG=3D"de-de"><FONT SIZE=3D2 FACE=3D"Arial">Dr. Thomas =
Belling</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"de-de"></SPAN><SPAN =
LANG=3D"de-de"><FONT FACE=3D"Arial"> </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"de-de"></SPAN><SPAN =
LANG=3D"de-de"><FONT SIZE=3D2 FACE=3D"Arial">3GPP Standardisation<BR>
Nokia Siemens Networks=A0</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"de-de"><BR>
</SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"de-de"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN></P>

</BODY>
</HTML>
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Date: Thu, 8 Nov 2012 14:11:57 -0500
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To: "Belling, Thomas (NSN - DE/Munich)" <thomas.belling@nsn.com>
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Subject: Re: [AVTCORE] Do RFC 4585 and RFC 5104 mandate the usage of the a=rtcp-fb SDP attribute?
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Hi Thomas,
>
> 
> Do RFC 4585 and RFC 5104allowsendingtheRTCP feedback messages defined in
> those RFCs, if AVPF has beennegotiatedusing SDP offer-answer, but
> no a=rtcp-fb SDP attributewas includedin the SDP negotiation?
> 

Yes, I think it is allowed based on Section 4.2 of RFC 4585:

   The "rtcp-fb" attribute SHOULD be used to indicate which RTCP FB
   messages MAY be used in this media session for the indicated payload
   type.

The above SHOULD allows for exception to explicitly indicate the FB
messages that is to be used. However, I would remember that this SHOULD
is an IETF SHOULD. See RFC 2119:

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

So, if one is not going to do this one better have a good reason for not
using a=rtcp-fb attribute to indicate which feedback messages that are
supported. I would guess that the intention for the SHOULD is to allow
cases where SDP is used, but it is not really possible for the entity
providing the SDP to indicate the actual set of feedback messages that
are supported on the end-point(s).

>From my perspective, I would use the signaling if it is available. It
makes it explicit which of the messages types that are supported by the
end-points.

Cheers

Magnus

> 
> 
> A more detailed discussion is below:
> 
> From RFC 4585:
> 
> 4.2.  RTCP Feedback Capability Attribute
> 
> …
> 
>    The "rtcp-fb" attribute SHOULD be used to indicate which RTCP FB
> 
>    messages MAY be used in this media session for the indicated
> payload type.
> 
> …
> 
>    If no "rtcp-fb" attribute is specified, the RTP receivers MAY send
> 
>    feedback using other suitable RTCP feedback packets as defined for
> 
>    the respective media type.  The RTP receivers MUST NOT rely on the
> 
>    RTP senders reacting to any of the FB messages.  The RTP sender MAY
> 
>    choose to ignore some feedback messages.
> 
>    If one or more "rtcp-fb" attributes are present in a media session
> 
>    description, the RTCP receivers for the media session(s) containing
> 
>    the "rtcp-fb"
> 
>    o  MUST ignore all "rtcp-fb" attributes of which they do not fully
> 
>       understand the semantics (i.e., where they do not understand the
> 
>       meaning of all values in the "a=rtcp-fb" line);
> 
>    o  SHOULD provide feedback information as specified in this document
> 
>       using any of the RTCP feedback packets as specified in one of the
> 
>       "rtcp-fb" attributes for this media session; and
> 
>    o  MUST NOT use other FB messages than those listed in one of the
> 
>       "rtcp-fb" attribute lines.
> 
>    When used in conjunction with the offer/answer model [8], the offerer
> 
>    MAY present a set of these AVPF attributes to its peer.  The answerer
> 
>    MUST remove all attributes it does not understand as well as those it
> 
>    does not support in general or does not wish to use in this
> 
>    particular media session.  The answerer MUST NOT add feedback
> 
>    parameters to the media description and MUST NOT alter values of such
> 
>    parameters.  The answer is binding for the media session, and both
> 
>    offerer and answerer MUST only use feedback mechanisms negotiated in
> 
>    this way.  Both offerer and answerer MAY independently decide to send
> 
>    RTCP FB messages of only a subset of the negotiated feedback
> 
>    mechanisms, but they SHOULD react properly to all types of the
> 
>    negotiated FB messages when received.
> 
> FromRFC 5104
> 
> 7.2.  Offer-Answer
> 
>    The Offer/Answer [RFC3264] implications for codec control protocol
> 
>    feedback messages are similar to those described in [RFC4585].  The
> 
>    offerer MAY indicate the capability to support selected codec
> 
>    commands and indications.  The answerer MUST remove all CCM
> 
>    parameters corresponding to the CCMs that it does not wish to support
> 
>    in this particular media session (for example, because it does not
> 
>    implement the message in question, or because its application logic
> 
>    suggests that support of the message adds no value).  The answerer
> 
>    MUST NOT add new ccm parameters in addition to what has been offered.
> 
>    The answer is binding for the media session and both offerer and
> 
>    answerer MUST NOT use any feedback messages other than what both
> 
>    sides have explicitly indicated as being supported.  In other words,
> 
>    only the joint subset of CCM parameters from the offer and answer may
> 
>    be used.
> 
>    Note that including a CCM parameter in an offer or answer indicates
> 
>    that the party (offerer or answerer) is at least capable of receiving
> 
>    the corresponding CCM(s) and act upon them.  In cases when the
> 
>    reception of a negotiated CCM mandates the party to respond with
> 
>    another CCM, it must also have that capability.  Although it is not
> 
>    mandated to initiate CCMs of any negotiated type, it is generally
> 
>    expected that a party will initiate CCMs when appropriate.
> 
>    The session maximum packet rate parameter part of the TMMBR
> 
>    indication is declarative, and the highest value from offer and
> 
>    answer SHALL be used.  If the session maximum packet rate parameter
> 
>    is not present in an offer, it SHALL NOT be included by the answerer.
> 
> A suggestedInterpretationforRFC 4585in3GPPSA4was :
> 
> The SDP offer may or may not include one or more a=rtcp-fb attributes
> for negotiating various AVPF feedback messages. (this is of course
> assuming that AVPF is negotiated on the m= line).
> 
> IF attribute lines with a=rtcp-fb are NOT included THEN
> 
>       Both end-points may send any AVPF FB messages (if AVPF is agreed)
> 
>       It is not guaranteed that any AVPF FB messages will work (in
> either direction)
> 
> IF attribute lines with a=rtcp-fb are included for some or all of the
> AVPF FB messages THEN
> 
>       The answerer must remove the attribute lines for the AVPF FB
> messages it does not want to use
> 
>       Only those AVPF FB messages that are in the SDP answer are allowed
> to be used, i.e. it is forbidden to send the AVPF FB messages that are
> not in the SDP answer
> 
> Is that correct?
> 
> 
> Does the same interpretation apply for RFC 5104?
> 
> Based on: “The answer is binding for the media session and both offerer
> andanswerer MUST NOT use any feedback messages other than what both 
> sides have explicitly indicated as being supported. “, the following
> alternative interpretation has been suggested:
> 
> *     Every CCM FB message that one intend to use must be negotiated in
> SDP offer-answer.
> 
> *     Any CCM FB message that has not been negotiated (and therefore is
> not agreed) in SDP offer-answer cannot be used in the session.
> 
> 
> Many thanks for your help,
> 
> Thomas
> 
> ----------------------------------
> Dr. Thomas Belling
> 
> 3GPP Standardisation
> Nokia Siemens Networks 
> 
> 
> 
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
> 


-- 

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
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Date: Thu, 8 Nov 2012 16:40:18 -0500
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: [AVTCORE] Comments on draft-ietf-avtcore-leap-second-01
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Hi,

I have reviewed the leap second document and have the following comments.

1. Section 1. It needs to say also in the introduction that it updates
RFC 3550, not only in the abstract.

2. Section 3.4:
   +-------+--------------+--------------+--------------+--------------+
   |  RTP  |      TAI     |      UTC     |     POSIX    |      NTP     |
   +-------+--------------+--------------+--------------+--------------+
   |  8000 | 00:00:32.500 | 23:59:58.500 | 23:59:58.500 | 23:59:58.500 |
   | 12000 | 00:00:33.000 | 23:59:59.000 | 23:59:59.000 | 23:59:59.000 |
   | 16000 | 00:00:33.500 | 23:59:59.500 | 23:59:59.500 | 23:59:59.500 |
   | 20000 | 00:00:34.000 | 23:59:60.000 | 23:59:59.000 | 00:00:00.000 |
   | 24000 | 00:00:34.500 | 23:59:60.500 | 23:59:59.500 | 00:00:00.000 |
   | 28000 | 00:00:35.000 | 00:00:00.000 | 00:00:00.000 | 00:00:00.000 |
   | 32000 | 00:00:35.500 | 00:00:00.500 | 00:00:00.500 | 00:00:00.500 |
   +-------+--------------+--------------+--------------+--------------+

                                  Table 1

I am wondering if the RTP 20000 and 24000 is really correctly
representing what you write in section 3.2 for the NTP column. Should't
the 00:00:00.000 actually  00:00:00.000-, i.e. something that is prior
to 00:00:00.000 but just?

3. Section 4.1:

   RTP Senders working to a leap-second-bearing reference SHOULD NOT
   generate sender reports containing an originating NTP timestamp in
   the vicinity of a leap second.  Receivers SHOULD ignore timestamps in
   any such reports inadvertently generated.

I wonder how you in practice are going to fulfill the sender
recommendation. That only works if your regular reporting interval is on
the order of a second or longer. Otherwise not sending a SR packet for a
RTP packet sender within 1.5*Td (where Td is your deterministic
reporting interval) will be counted towards source time-out or other
methods detecting missing packets. So for a reporting interval of 100 or
200 ms you can't actually follow the SHOULD.

4. Section 5. Security Consideration.

I wonder if this section really are covering some potential issues.
First are there security replay mechanisms that uses the clock? SRTP is
not effected as it uses packet index, i.e. roll over counter + seq_nr.
But there might be other things that could be confused for example by
the posix clock.

Are there attack vectors that are created when by possibly fool an
end-host to believe a leap-second event is happening and thus thinks the
clock are freezing or repeating that otherwise would be possible?

5. Reference section:

I don't believe that all references are normative. I think for example
the note in section 3.0 uses [3] in an informative one. Additional ones
may be similar. Please check the usage of all the references if they
really are normative.

Cheers

Magnus Westerlund

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


From kevin.gross@avanw.com  Thu Nov  8 14:06:18 2012
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Cc: IETF AVTCore WG <avt@ietf.org>, draft-ietf-avtcore-leap-second@tools.ietf.org
Subject: Re: [AVTCORE] Comments on draft-ietf-avtcore-leap-second-01
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--047d7b603a8ad5063b04ce030992
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Thanks for the review. Responses below.

On Thu, Nov 8, 2012 at 4:40 PM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> Hi,
>
> I have reviewed the leap second document and have the following comments.
>
> 1. Section 1. It needs to say also in the introduction that it updates
> RFC 3550, not only in the abstract.
>

Ok.

>
> 2. Section 3.4:
>    +-------+--------------+--------------+--------------+--------------+
>    |  RTP  |      TAI     |      UTC     |     POSIX    |      NTP     |
>    +-------+--------------+--------------+--------------+--------------+
>    |  8000 | 00:00:32.500 | 23:59:58.500 | 23:59:58.500 | 23:59:58.500 |
>    | 12000 | 00:00:33.000 | 23:59:59.000 | 23:59:59.000 | 23:59:59.000 |
>    | 16000 | 00:00:33.500 | 23:59:59.500 | 23:59:59.500 | 23:59:59.500 |
>    | 20000 | 00:00:34.000 | 23:59:60.000 | 23:59:59.000 | 00:00:00.000 |
>    | 24000 | 00:00:34.500 | 23:59:60.500 | 23:59:59.500 | 00:00:00.000 |
>    | 28000 | 00:00:35.000 | 00:00:00.000 | 00:00:00.000 | 00:00:00.000 |
>    | 32000 | 00:00:35.500 | 00:00:00.500 | 00:00:00.500 | 00:00:00.500 |
>    +-------+--------------+--------------+--------------+--------------+
>
>                                   Table 1
>
> I am wondering if the RTP 20000 and 24000 is really correctly
> representing what you write in section 3.2 for the NTP column. Should't
> the 00:00:00.000 actually  00:00:00.000-, i.e. something that is prior
> to 00:00:00.000 but just?
>

A common system behavior is that when the NTP clock is
frozen, successive calls to retrieve system return incrementally larger
time values. This keeps applications from faulting from assumptions about
time being continuous and monotonic. This behavior occurs in the LS bits of
the time value and so is not generally visible in the human-readable format
shown in the table. I can add a discussion of this if you think it is
useful.

>
> 3. Section 4.1:
>
>    RTP Senders working to a leap-second-bearing reference SHOULD NOT
>    generate sender reports containing an originating NTP timestamp in
>    the vicinity of a leap second.  Receivers SHOULD ignore timestamps in
>    any such reports inadvertently generated.
>
> I wonder how you in practice are going to fulfill the sender
> recommendation. That only works if your regular reporting interval is on
> the order of a second or longer. Otherwise not sending a SR packet for a
> RTP packet sender within 1.5*Td (where Td is your deterministic
> reporting interval) will be counted towards source time-out or other
> methods detecting missing packets. So for a reporting interval of 100 or
> 200 ms you can't actually follow the SHOULD.
>

That's a good point. If we need to keep sending SRs during the 2second leap
second interval to avoid timeouts, they can be sent with a 0 NTP timestamp.
This might be a better approach for all RTCP rates. I have to research what
special receiver behavior may be required to accommodate this.

>
> 4. Section 5. Security Consideration.
>
> I wonder if this section really are covering some potential issues.
> First are there security replay mechanisms that uses the clock? SRTP is
> not effected as it uses packet index, i.e. roll over counter + seq_nr.
> But there might be other things that could be confused for example by
> the posix clock.
>
> Are there attack vectors that are created when by possibly fool an
> end-host to believe a leap-second event is happening and thus thinks the
> clock are freezing or repeating that otherwise would be possible?
>

Today I volunteered to review draft-ietf-tictoc-security-requirements. I
will take this and your comments into consideration and try to improve the
security considerations section.
>
>
> 5. Reference section:
>
> I don't believe that all references are normative. I think for example
> the note in section 3.0 uses [3] in an informative one. Additional ones
> may be similar. Please check the usage of all the references if they
> really are normative.
>

Sounds like I need to look at dividing informative vs. normative
references.

>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
>

--047d7b603a8ad5063b04ce030992
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Thanks for the review. Responses below.<br><div class=3D"gmail_extra"><br c=
lear=3D"all">On Thu, Nov 8, 2012 at 4:40 PM, Magnus Westerlund <span dir=3D=
"ltr">&lt;<a href=3D"mailto:magnus.westerlund@ericsson.com" target=3D"_blan=
k">magnus.westerlund@ericsson.com</a>&gt;</span> wrote:<br>
<div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margi=
n:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204=
);border-left-style:solid;padding-left:1ex">Hi,<br>
<br>
I have reviewed the leap second document and have the following comments.<b=
r>
<br>
1. Section 1. It needs to say also in the introduction that it updates<br>
RFC 3550, not only in the abstract.<br></blockquote><div><br></div><div>Ok.=
=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8e=
x;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-styl=
e:solid;padding-left:1ex">

<br>
2. Section 3.4:<br>
=A0 =A0+-------+--------------+--------------+--------------+--------------=
+<br>
=A0 =A0| =A0RTP =A0| =A0 =A0 =A0TAI =A0 =A0 | =A0 =A0 =A0UTC =A0 =A0 | =A0 =
=A0 POSIX =A0 =A0| =A0 =A0 =A0NTP =A0 =A0 |<br>
=A0 =A0+-------+--------------+--------------+--------------+--------------=
+<br>
=A0 =A0| =A08000 | 00:00:32.500 | 23:59:58.500 | 23:59:58.500 | 23:59:58.50=
0 |<br>
=A0 =A0| 12000 | 00:00:33.000 | 23:59:59.000 | 23:59:59.000 | 23:59:59.000 =
|<br>
=A0 =A0| 16000 | 00:00:33.500 | 23:59:59.500 | 23:59:59.500 | 23:59:59.500 =
|<br>
=A0 =A0| 20000 | 00:00:34.000 | 23:59:60.000 | 23:59:59.000 | 00:00:00.000 =
|<br>
=A0 =A0| 24000 | 00:00:34.500 | 23:59:60.500 | 23:59:59.500 | 00:00:00.000 =
|<br>
=A0 =A0| 28000 | 00:00:35.000 | 00:00:00.000 | 00:00:00.000 | 00:00:00.000 =
|<br>
=A0 =A0| 32000 | 00:00:35.500 | 00:00:00.500 | 00:00:00.500 | 00:00:00.500 =
|<br>
=A0 =A0+-------+--------------+--------------+--------------+--------------=
+<br>
<br>
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Table 1=
<br>
<br>
I am wondering if the RTP 20000 and 24000 is really correctly<br>
representing what you write in section 3.2 for the NTP column. Should&#39;t=
<br>
the 00:00:00.000 actually =A000:00:00.000-, i.e. something that is prior<br=
>
to 00:00:00.000 but just?<br></blockquote><div><br></div><div>A common syst=
em behavior is that when the NTP clock is frozen,=A0successive=A0calls to=
=A0retrieve=A0system return incrementally larger time values. This keeps ap=
plications from faulting from assumptions about time being continuous and=
=A0monotonic. This behavior=A0occurs=A0in the LS bits of the time value and=
 so is not generally visible in the human-readable format shown in the tabl=
e. I can add a discussion of this if you think it is useful.</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<br>
3. Section 4.1:<br>
<br>
=A0 =A0RTP Senders working to a leap-second-bearing reference SHOULD NOT<br=
>
=A0 =A0generate sender reports containing an originating NTP timestamp in<b=
r>
=A0 =A0the vicinity of a leap second. =A0Receivers SHOULD ignore timestamps=
 in<br>
=A0 =A0any such reports inadvertently generated.<br>
<br>
I wonder how you in practice are going to fulfill the sender<br>
recommendation. That only works if your regular reporting interval is on<br=
>
the order of a second or longer. Otherwise not sending a SR packet for a<br=
>
RTP packet sender within 1.5*Td (where Td is your deterministic<br>
reporting interval) will be counted towards source time-out or other<br>
methods detecting missing packets. So for a reporting interval of 100 or<br=
>
200 ms you can&#39;t actually follow the SHOULD.<br></blockquote><div><br><=
/div><div>That&#39;s a good point. If we need to keep sending SRs during th=
e 2second leap second interval to avoid timeouts, they can be sent with a 0=
 NTP timestamp. This=A0might=A0be a better approach for all RTCP rates. I h=
ave to research what special receiver behavior may be required to=A0accommo=
date=A0this.</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<br>
4. Section 5. Security Consideration.<br>
<br>
I wonder if this section really are covering some potential issues.<br>
First are there security replay mechanisms that uses the clock? SRTP is<br>
not effected as it uses packet index, i.e. roll over counter + seq_nr.<br>
But there might be other things that could be confused for example by<br>
the posix clock.<br>
<br>
Are there attack vectors that are created when by possibly fool an<br>
end-host to believe a leap-second event is happening and thus thinks the<br=
>
clock are freezing or repeating that otherwise would be possible?<br></bloc=
kquote><div><br></div>Today I volunteered to review draft-ietf-tictoc-secur=
ity-requirements. I will take this and your comments into consideration and=
 try to improve the security considerations section.<blockquote class=3D"gm=
ail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-l=
eft-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">

<br>
5. Reference section:<br>
<br>
I don&#39;t believe that all references are normative. I think for example<=
br>
the note in section 3.0 uses [3] in an informative one. Additional ones<br>
may be similar. Please check the usage of all the references if they<br>
really are normative.<br></blockquote><div><br></div><div>Sounds like I nee=
d to look at dividing informative vs. normative references.=A0</div><blockq=
uote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-wi=
dth:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-=
left:1ex">

<br>
Cheers<br>
<br>
Magnus Westerlund<br>
<br>
----------------------------------------------------------------------<br>
Multimedia Technologies, Ericsson Research EAB/TVM<br>
----------------------------------------------------------------------<br>
Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0+46 10 7148287<br>
F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile +46 73 0949079<br>
SE-164 80 Stockholm, Sweden| mailto: <a href=3D"mailto:magnus.westerlund@er=
icsson.com">magnus.westerlund@ericsson.com</a><br>
----------------------------------------------------------------------<br>
<br>
_______________________________________________<br>
Audio/Video Transport Core Maintenance<br>
<a href=3D"mailto:avt@ietf.org">avt@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/avt" target=3D"_blank">htt=
ps://www.ietf.org/mailman/listinfo/avt</a><br>
</blockquote></div><br></div>

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Date: Thu, 8 Nov 2012 18:08:41 -0500
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Subject: [AVTCORE] Key diversification with DTLS-SRTP with Shim
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Section 6.3.2 of
http://tools.ietf.org/id/draft-westerlund-avtcore-transport-multiplexing-04.txt
states:

 Instead we propose that an DTLS-SRTP key-derivation change is
   introduced.  By including the Session ID value in the derivation of
   the keying material a single DTLS-SRTP key-management operation could
   apply keys and parameters for all the RTP sessions in the same
   transport flow.  Thus the keying cost is significantly reduced,
   especially in regards to network communication and delay impact and
   vunerability to packet loss.

This seems totally reasonable.

DTLS-SRTP uses RFC 5705 TLS Exporters. These take an optional "context" value
which is currently not used. I propose we set the "context" to be the shim ID
(represented as a 16-bit big-endian unsigned integer).

-Ekr

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Date: Fri, 9 Nov 2012 06:40:31 -0500
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: [AVTCORE] Comments on draft-ietf-avtcore-clksrc-01
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Hi,

I have review the Clock Source document draft-ietf-avtcore-clksrc-01 and
have the following comments:

1. General comment on all the use of hanging lists. Please put in some
separator characters like ":" between the label and the bullet text.

2. Section 3. media stream

Yes, it is made clear above that this is SDP terminology. But as "media
stream" is also used for the sequnce of packets comming from an SSRC in
RTP I think it is prudent to call this type of media stream "SDP media
stream" to avoid confusion in the reader.

3. Section 4.6
This is really a question of how traceable clocks works. I know in
Sweden we have "Swedish Time" which is produced by a governmental body
using a number of time production centers and time sources. Several of
the time sources are also used to help produce UTC. Commonly the
difference between Swedish time and UTC will be minimal, i.e on the nano
second level. However, in the case Sweden would be isolated from UTC
users of Swedish time will be able to use a common time-base. This is
clearly tracable, but how do you deal with the fact that there are
different sources of traceable time.

For example, my home computer's clocks runs of the Swedish strata one
NTP servers, but I think it is traceable also.

4. Section 4.8:

   clksrc = ntp / ptp / gps / gal / local / private

I think this should ad a syntax element that make clear the syntax of a
future extension shall have. An example:

   clksrc = ntp / ptp / gps / gal / local / private / clksrc-ext

   clksrc-ext = token ; token as in SDP

The similar extensibility I think should exist in the following rules:

ptp-version
ptp-server

5. Section 4.8:

hostname      =  *( domainlabel "." ) toplabel [ "." ]

Where is this rule taken from. I try to remember why you need a hostname
to end on a ".".

6. Section 4.8
    port = 1*DIGIT

I would recommend that we restrict the port number to 5 digits maximum
length, i.e.     port = 1*5DIGIT

Or do you see any need for totally unlimited number of digits?

7. I wonder if it should be more explicit that the a=clksrc attribute
actually are defined to usage at the source (a=SSRC) level in the
definition text by including a mention of source-specific and a ref to
RFC 5576?

8. Section 5.3:

   In a given system, master clock identifiers must be unique.  Such
   identifiers MAY be manually configured, however 17 octet string
   identifiers SHOULD be generated according to the "short-term
   persistent RTCP CNAME" algorithm as described in RFC6222 [7].

I can see that a short term persistent is sufficient, but aren't it
desired to actually use a long term persistent identifier for clock
sources so that one can track their usage over different RTP sessions?

9. Section 5.4:

               mediaclock = refclk / streamid / sender

I think the above ABNF syntax rules shall include a syntax definition
for future extensions of the attribute.

10. Section 4 and 5:

The SDP attributes are lacking clear rules for interpretation in a SDP
Offer/Answer context and thus also the declarative usage should be
contrasted and clear.


But, I think this document is shaping up nicely and the authors are
doing an excellent work with it. So lets get these last things settled
so that we can last call it.

Cheers

Magnus Westerlund

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


From magnus.westerlund@ericsson.com  Fri Nov  9 03:53:53 2012
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Date: Fri, 9 Nov 2012 06:53:47 -0500
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Cc: IETF AVTCore WG <avt@ietf.org>, draft-ietf-avtcore-leap-second@tools.ietf.org
Subject: Re: [AVTCORE] Comments on draft-ietf-avtcore-leap-second-01
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Hi,

Responding to the ones with question or feedback desired.

On 2012-11-08 17:05, Kevin Gross wrote:
> Thanks for the review. Responses below.
> 
> On Thu, Nov 8, 2012 at 4:40 PM, Magnus Westerlund
> <magnus.westerlund@ericsson.com <mailto:magnus.westerlund@ericsson.com>>
> wrote:
> 
> 
>     2. Section 3.4:
>        +-------+--------------+--------------+--------------+--------------+
>        |  RTP  |      TAI     |      UTC     |     POSIX    |      NTP     |
>        +-------+--------------+--------------+--------------+--------------+
>        |  8000 | 00:00:32.500 | 23:59:58.500 | 23:59:58.500 | 23:59:58.500 |
>        | 12000 | 00:00:33.000 | 23:59:59.000 | 23:59:59.000 | 23:59:59.000 |
>        | 16000 | 00:00:33.500 | 23:59:59.500 | 23:59:59.500 | 23:59:59.500 |
>        | 20000 | 00:00:34.000 | 23:59:60.000 | 23:59:59.000 | 00:00:00.000 |
>        | 24000 | 00:00:34.500 | 23:59:60.500 | 23:59:59.500 | 00:00:00.000 |
>        | 28000 | 00:00:35.000 | 00:00:00.000 | 00:00:00.000 | 00:00:00.000 |
>        | 32000 | 00:00:35.500 | 00:00:00.500 | 00:00:00.500 | 00:00:00.500 |
>        +-------+--------------+--------------+--------------+--------------+
> 
>                                       Table 1
> 
>     I am wondering if the RTP 20000 and 24000 is really correctly
>     representing what you write in section 3.2 for the NTP column. Should't
>     the 00:00:00.000 actually  00:00:00.000-, i.e. something that is prior
>     to 00:00:00.000 but just?
> 
> 
> A common system behavior is that when the NTP clock is
> frozen, successive calls to retrieve system return incrementally larger
> time values. This keeps applications from faulting from assumptions
> about time being continuous and monotonic. This behavior occurs in the
> LS bits of the time value and so is not generally visible in the
> human-readable format shown in the table. I can add a discussion of this
> if you think it is useful.

I think a short comment / note why this doesn't show up in the table
might be beneficial to avoid confusion in the reader.

> 
> 
>     3. Section 4.1:
> 
>        RTP Senders working to a leap-second-bearing reference SHOULD NOT
>        generate sender reports containing an originating NTP timestamp in
>        the vicinity of a leap second.  Receivers SHOULD ignore timestamps in
>        any such reports inadvertently generated.
> 
>     I wonder how you in practice are going to fulfill the sender
>     recommendation. That only works if your regular reporting interval is on
>     the order of a second or longer. Otherwise not sending a SR packet for a
>     RTP packet sender within 1.5*Td (where Td is your deterministic
>     reporting interval) will be counted towards source time-out or other
>     methods detecting missing packets. So for a reporting interval of 100 or
>     200 ms you can't actually follow the SHOULD.
> 
> 
> That's a good point. If we need to keep sending SRs during the 2second
> leap second interval to avoid timeouts, they can be sent with a 0 NTP
> timestamp. This might be a better approach for all RTCP rates. I have to
> research what special receiver behavior may be required to accommodate this.
> 

I think using 0 makes it very clear that the NTP field is not to be
trusted for these packets. That is very clear for the updated and aware
implementation. It is quite bad for an old implementation. Using the
system clock across a leap second event can clearly result in a
desynchronization of audio and video with a second. If you use the 0
value a naive implementation can suddenly believe that the media is
desynchronized with 112+ years. I have seen this occur in at least one
implementation when feeding it bad time-synch data. The implementation
started buffering until it run out of memory.

> 
>     4. Section 5. Security Consideration.
> 
>     I wonder if this section really are covering some potential issues.
>     First are there security replay mechanisms that uses the clock? SRTP is
>     not effected as it uses packet index, i.e. roll over counter + seq_nr.
>     But there might be other things that could be confused for example by
>     the posix clock.
> 
>     Are there attack vectors that are created when by possibly fool an
>     end-host to believe a leap-second event is happening and thus thinks the
>     clock are freezing or repeating that otherwise would be possible?
> 
> 
> Today I volunteered to review draft-ietf-tictoc-security-requirements. I
> will take this and your comments into consideration and try to improve
> the security considerations section.
> 

Good, I encourage people to think creatively if there are ways of using
the rules provided for attacks. For example I think the insertion of 0
in the NTP field might have some implications.

Cheers

Magnus Westerlund

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


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Subject: Re: [AVTCORE] Key diversification with DTLS-SRTP with Shim
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Thanks,

On 2012-11-08 18:08, Eric Rescorla wrote:
> Section 6.3.2 of
> http://tools.ietf.org/id/draft-westerlund-avtcore-transport-multiplexing-04.txt
> states:
> 
>  Instead we propose that an DTLS-SRTP key-derivation change is
>    introduced.  By including the Session ID value in the derivation of
>    the keying material a single DTLS-SRTP key-management operation could
>    apply keys and parameters for all the RTP sessions in the same
>    transport flow.  Thus the keying cost is significantly reduced,
>    especially in regards to network communication and delay impact and
>    vunerability to packet loss.
> 
> This seems totally reasonable.
> 
> DTLS-SRTP uses RFC 5705 TLS Exporters. These take an optional "context" value
> which is currently not used. I propose we set the "context" to be the shim ID
> (represented as a 16-bit big-endian unsigned integer).

Do one want this value to be prefixed with something like "SHIM" or
other so that others can feed in their values without causing risk of
extracting the same material?

Cheers

Magnus Westerlund

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Subject: Re: [AVTCORE] Fwd: I-D Action:	draft-westerlund-avtcore-rtp-topologies-update-01.txt
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Hi,

Sorry for the delay in responding.

On 2012-10-25 15:00, Roni Even wrote:
> Hi Magnus,
> I read the draft so to answer the questions
> 
> 1. I think the document need also to discuss the SDES behavior (CNAME must
> be forwarded to support the CSRC) specified in 7.2 and 7.3 of RFC 3550 for
> the different topologies (we had recently a discussion about using SDES
> messages). 

Yes, that is a good idea.

> 2. I did not fully understand the SSRC behavior for source projecting
> middlebox and how it is different from the topologies in 3.5.

The difference between a RTP mixer and a source projecting mixer and its
usage of SSRCs are as follow.

Source projection: SSRC has a one to one relation to a CSRC
RTP Mixer: Has one SSRC that carries content from potentially multiple
CSRCs.

The big difference is that an RTP mixer's SSRC contains either a
selection or a media level mix/composition of the contributed media
streams. A source projection mixer really is a way of projecting the
other media sources in the conference into the RTP session between the
Mixer and this particular conference participant.

Does this make it clearer?

Cheers

Magnus Westerlund

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Hi,

Thanks for the comments.

On 2012-11-03 18:41, Charles Eckel (eckelcu) wrote:
> I think this is a useful document; However, I do have a few comments
> and concerns regarding its focus and relationship with other drafts.
> 
> 1) Section 3, Motivations. It currently reads as follows:
> 
>    This section looks at the motivations why an additional solution is
>    needed assuming that you can do both the classical method of having
>    one RTP session per transport flow as defined by the RTP
>    specification [RFC3550] and when you have multiple media types within
>    one RTP session [I-D.ietf-avtcore-multi-media-rtp-session].
> 
> How about the following instead:
> 
>    This section looks at the motivations for standardizing two mechanisms 
>    for multiplexing multiple media types within a single transport flow:
>   1) having one RTP session per transport flow, as defined by the RTP
>    specification [RFC3550], and multiplexing multiple RTP session per transport
>    flow, as defined in this document ( or elsewhere if preferred to split it out).
>   2) having  multiple media types within one RTP session, as defined in
>   [I-D.ietf-avtcore-multi-media-rtp-session].

Yes, that is clearer.

> 

> 2) Sections 3.1 and 3.2 are motivations for (2) and are duplicated
> in
[I-D.ietf-avtcore-multi-media-rtp-session] , whereas sections 3.3 and
3.4 are motivations for (1). Perhaps it is best to remove sections 3.1
and 3.2 and let this draft focus on the reasons why (1) is needed, as
that is real focus of this draft.
> 

I will need to think about this. It might be that some restructuring are
needed in general to consider both when a single transport is motivated
and what the motivation is for having multiple RTP sessions over one
transport flow. Thus maybe further tweaking of the introduction section
and the structure.

> 3) Section 3.5 is more a statement of the requirement for a way to
negotiate (1) vs (2), rather than a motivation for either.

Then I don't think the message comes across that this session tries to
describe. To deploy a mix of legacy multiple RTP sessions clients and
single RTP session with multiple media types clients are not possible in
a joint multi-party conference unless one have a full RTP level
translation capable middlebox.

> 
> 4) Section 6.2, Signaling, contains the following:>
>    The 'session-mux-id' attribute is included for a media description,
>    in order to indicate the Session ID for that particular media
>    description.  Every media description that shares a common attribute
>    value is assumed to be part of a single RTP session.  
> 
> Is that true? I think you meant to say these may exist in one or
> more RTP sessions, with the RTP sessions being multiplexed on a single
> transport flow.
>
> And the later it reads as follows:
> 
>    An SDP Offerer
>    MUST include the 'session-mux-id' attribute for every media
>    description associated with a 'SHIM' group.  If the SDP Answer does
>    not contain the SHIM group, the SDP Offerer MUST NOT use SHIM based
>    layering.  However, if that is separate RTP sessions or BUNDLE is
>    determined on what was present in the offer and answer.  This will
>    depend on what the offering party likes to happen. 
> 
> You lost me here. Hopefully some  rewording would help.

I wonder if what you are getting confused over is the statement that
multiple m= lines where each m= lines contains the same SID would form a
joint RTP session. That was a notion that is included, but based on the
BUNDLE discussions I think should be avoided as the majority of the
difficult issues with BUNDLE is due to having multiple m= lines be part
of describing one RTP session.

Cheers

Magnus Westerlund

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To: Magnus Westerlund <magnus.westerlund@ericsson.com>
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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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To: Mario Montagud Climent <mamontor@posgrado.upv.es>
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Subject: Re: [AVTCORE] Question about the minimum RTCP report interval
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On 2012-11-05 13:20, Mario Montagud Climent wrote:
> 
> Hi all,
> 
> I would appreciate it very much if you could clarify me a doubt about
> the minimum RTCP reporting interval.
> 
> According to RFC 3550, the RTCP report interval is dynamically computed
> based on the allocated session bandwidth, the average size of all
> (received and sent) RTCP packets, the number of participants in the
> session and their role (senders or receivers), or their proportion, and
> the unicast or multicast nature of the session (Section 6.2 of RFC 3550).
> 
> Nevertheless, it is also specified that this deterministically computed
> RTCP report interval should have a lower bound to avoid having bursts
> (cumpling) of RTCP packets when the number of participants is small and
> the law of large numbers is not helping to smooth out the traffic
> overhead. This would also help to avoid excessive frequent reports
> during transient outages like a network partition. The recommended value
> in RFC 3550 for that fixed RTCP minimum interval is 5 seconds.
> 
> The RFC 3550 also specifies that in some cases (e.g., if the data rate
> is high and the application demands more timely feedback reports) an
> implementation may scale the minimum RTCP interval to a smaller value
> given by 360 divided by the session bandwidth (in kbps). Consequently,
> this yields to an interval smaller than 5 seconds for bandwidths greater
> than 72 kbps. In multicast sessions, only active senders may use that
> reduced minimum interval, whilst in unicast sessions it also may be used
> by receivers. In the above cases, however, the RTCP minimum interval
> must be still taken into account during the membership accounting
> procedure in order to not prematurely time out participants (who can
> indeed be using it) because of inactivity.
> 
> However, RFC 4585 defines an extension to the RTP Audio-Visual Profile
> (AVP) that enables receivers to provide, statistically, more immediate
> feedback messages to the senders and thus allows for short-term
> adaptation and efficient feedback-based repair mechanisms to be
> implemented, while maintaining the bandwidth constraints for RTCP and
> preserving scalability to large groups. In RFC 4585, the RTCP
> transmission interval specified in RFC 3550 is denoted as Regular RTCP
> interval. In addition, RFC 4585 specifies that feedback messages can be
> reported earlier than the next scheduled Regular RTCP transmission time
> if a receiver detects the need to inform about events of interest in the
> media stream (e.g., picture or slice loss) close to their occurrence.
> Otherwise, the Regular RTCP transmission mode is employed, but the
> minimum RTCP report interval of 5 seconds recommended in RFC 3550 is
> dropped.
> 
> So, my questions about this are:
> 
> - When using the Regular RTCP reporting rules (RFC 3550) in multicast
> scenarios, which option should we select for the minimum RTCP reporting
> interval (for a media sender)? Can we use any of the two options for
> selection the minimum thresholds (i.e., 5 sec or 360/bw_session_kbps)?
> 

To my understanding you can use any of them. This is actually a
potential interoperability issue.

> - Likewise, should the sentence "the minimum RTCP report interval of 5
> seconds recommended in RFC 3550 is dropped" from RFC 4585 be applied in
> all cases (i.e. not only when using the early RTCP reporting rules)?

It applies also for regular reporting and instead the trr-int mechanism
is introduced.

I would however strongly recommend that one uses 5 seconds in the
formula when timing out sources to avoid spurious timing out of sources
that uses the 5 seconds if oneself are uses the scaled value.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
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From internet-drafts@ietf.org  Mon Nov 19 14:51:41 2012
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Subject: [AVTCORE] I-D Action: draft-ietf-avt-srtp-not-mandatory-11.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Audio/Video Transport Core Maintenance Wo=
rking Group of the IETF.

	Title           : Securing the RTP Protocol Framework: Why RTP Does Not Ma=
ndate a Single Media Security Solution
	Author(s)       : Colin Perkins
                          Magnus Westerlund
	Filename        : draft-ietf-avt-srtp-not-mandatory-11.txt
	Pages           : 10
	Date            : 2012-11-19

Abstract:
   This memo discusses the problem of securing real-time multimedia
   sessions, and explains why the Real-time Transport Protocol (RTP),
   and the associated RTP control protocol (RTCP), do not mandate a
   single media security mechanism.  Guidelines for designers and
   reviewers of future RTP extensions are provided, to ensure that
   appropriate security mechanisms are mandated, and that any such
   mechanisms are specified in a manner that conforms with the RTP
   architecture.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-avt-srtp-not-mandatory

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-avt-srtp-not-mandatory-11

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-avt-srtp-not-mandatory-11


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


From csp@csperkins.org  Mon Nov 19 14:59:17 2012
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From: Colin Perkins <csp@csperkins.org>
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Subject: Re: [AVTCORE] I-D Action: draft-ietf-avt-srtp-not-mandatory-11.txt
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This version of the draft updates the recommendations in Section 6 based =
on discussion with the Security Area Directors. There are no changes in =
the rest of the draft.

Colin


On 19 Nov 2012, at 22:51, Internet-Drafts@ietf.org wrote:
> A New Internet-Draft is available from the on-line Internet-Drafts =
directories.
> This draft is a work item of the Audio/Video Transport Core =
Maintenance Working Group of the IETF.
>=20
> 	Title           : Securing the RTP Protocol Framework: Why RTP =
Does Not Mandate a Single Media Security Solution
> 	Author(s)       : Colin Perkins
>                          Magnus Westerlund
> 	Filename        : draft-ietf-avt-srtp-not-mandatory-11.txt
> 	Pages           : 10
> 	Date            : 2012-11-19
>=20
> Abstract:
>   This memo discusses the problem of securing real-time multimedia
>   sessions, and explains why the Real-time Transport Protocol (RTP),
>   and the associated RTP control protocol (RTCP), do not mandate a
>   single media security mechanism.  Guidelines for designers and
>   reviewers of future RTP extensions are provided, to ensure that
>   appropriate security mechanisms are mandated, and that any such
>   mechanisms are specified in a manner that conforms with the RTP
>   architecture.
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-avt-srtp-not-mandatory
>=20
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-avt-srtp-not-mandatory-11
>=20
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-avt-srtp-not-mandatory-11


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




From cris.pascual.gonzalez@gmail.com  Tue Nov 20 07:10:47 2012
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Subject: [AVTCORE] Deadline extension| MMEDIA 2013 || April 22 - 26, 2013 - Venice, Italy
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INVITATION:

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

Please consider to contribute to and/or forward to the appropriate groups the 
following opportunity to submit and publish original scientific results to MMEDIA 
2013.

The submission deadline is December 8, 2012.

Authors of selected papers will be invited to submit extended article versions to 
one of the IARIA Journals: http://www.iariajournals.org

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


============== MMEDIA 2013 | Call for Papers ===============

CALL FOR PAPERS, TUTORIALS, PANELS

MMEDIA 2013, The Fifth International Conferences on Advances in Multimedia

April 22 - 26, 2013 - Venice, Italy


General page: http://www.iaria.org/conferences2013/MMEDIA13.html

Call for Papers: http://www.iaria.org/conferences2013/CfPMMEDIA13.html

- regular papers

- short papers (work in progress)

- posters

Submission page: http://www.iaria.org/conferences2013/SubmitMMEDIA13.html


Submission deadline: December 8, 2012

Sponsored by IARIA, www.iaria.org


Extended versions of selected papers will be published in IARIA Journals:  
http://www.iariajournals.org

Print proceedings will be available via Curran Associates, Inc.: 
http://www.proceedings.com/9769.html

Articles will be archived in the free access ThinkMind Digital Library: 
http://www.thinkmind.org


Please note the Poster and Work in Progress options.

The topics suggested by the conference can be discussed in term of concepts, state 
of the art, research, standards, implementations, running experiments, 
applications, and industrial case studies. Authors are invited to submit complete 
unpublished papers, which are not under review in any other conference or journal 
in the following, but not limited to, topic areas.
All tracks are open to both research and industry contributions, in terms of 
Regular papers, Posters, Work in progress, Technical/marketing/business 
presentations, Demos, Tutorials, and Panels.

Before submission, please check and conform with the Editorial rules: 
http://www.iaria.org/editorialrules.html


MMEDIA 2013 Topics (topics and submission details: see CfP on the site)


Fundamentals in multimedia

   Multimedia systems, architecture, and applications; New multimedia platforms; 
Multimedia architectural specification languages; Theoretical aspects and 
algorithms for multimedia; Multimedia content delivery networks; Network support 
for multimedia data; Multimedia data storage; Multimedia meta-modeling techniques 
and operating systems; Multimedia signal coding and processing (audio, video, 
image); Multimedia applications (telepresence, triple-play, quadruple-play, ?); 
Multimedia tools (authoring, analyzing, editing, browsing, ?); Computational 
multimedia intelligence (fuzzy logic, neural networks, genetic algorithms, ?); 
Intelligent agents for multimedia content creation, distribution, and analysis; 
Multimedia networking; Wired and wireless multimedia systems; Distributed 
multimedia systems; Multisensor data integration and fusion; Multimedia and P2P; 
Multimedia standards

Multimedia content and modeling

   Interfaces for multimedia creation; Multimedia streaming and services; Image 
modeling and editing; Audio modeling and transformation; Video modeling and 
transformation; Image recognition; Multimedia databases; Multimedia coding and 
encryption; Multimedia modeling for learning content; Multimedia description 
languages; Image clustering; Media fusion for communication and presentation

Self-organizing multimedia architectures

   Self-organization in multimedia systems; Self-organization in multimedia 
communities; Self-organized multimedia networks; Multimedia content distribution 
and consumption; Adaptive multimedia interfaces; Multimedia retrieval

Multimedia content-based retrieval and analysis

   Multimodal data analysis; Multimedia databases; Semi-automatic and automatic 
methods for multimedia annotation; Image/video/audio databases; Content-based image 
retrieval; Semantics-based search and integration of multimedia and digital 
content; Multimedia data modeling, indexing, and mining; Statistical modeling of 
multimedia data; Multimedia extraction and annotation; Content 
search/browsing/retrieval; Internet imaging and multimedia; Multimodal content 
analysis; Multimedia abstraction and summarization; Semantic analysis of multimedia 
data; Media assimilation and fusion

Perception and cognition for multimedia users

   Quality of experience; Relevance feedback; Human-computer interaction; 
Multimodal interaction; Multimodal user interfaces; Mobile user-centered 
interfaces; Peer-to-peer multimedia systems and streaming; Pervasive and 
interactive multimedia systems (digital TV, mobile systems, gaming,?); Multimedia 
in personal, sensor and ad-hoc networks; Visualization and virtual reality; 
Intelligent browsing and visualization; Perception and cognition; Perception and 
modeling of the environment; Multimedia collaboration; Social networking
Multimedia ontology

   Multimedia semantics; Emergent semantics; Media ontology learning; Ontology for 
media web mining; Multimedia ontologies; Multimedia information management; 
Approaches using metadata standards; Conceptual clustering; Modeling and 
recognition of visual objects and actions
Mobile and ubiquitous multimedia

   Architectures, protocols, and algorithms for multimedia mobility; Middleware and 
distributed computing support for mobile and ubiquitous multimedia; Mobile and 
ubiquitous multimedia in intelligent transportation systems; Enabling platforms for 
mobile multimedia; Roaming and limited bandwidth; Intermittent connectivity; 
Streaming mobile multimedia; Mobile multimedia software architectures; Mobile 
multimedia applications and services; Communication and cooperation via mobile 
multimedia; Business models for mobile multimedia; Provisioning of mobile 
multimedia services; Context-aware mobile and ubiquitous multimedia; Mobile 
computer graphics, games and entertainment; Mobile and ubiquitous multimedia in ad 
hoc networks; Personalization, privacy and security in mobile multimedia; Social 
and regulatory aspects of mobile multimedia; Multimedia in the Extended Home; 
Ubiquitous/Seamless content sharing

Multimedia services

   Reliability, availability, serviceability of multimedia services; Multimedia 
content distribution services; Real-time multimedia services; Audio-visual 
multimedia services; Multimedia signal processing and communications; Media 
representation and algorithms; Audio, image, video processing, coding and 
compression; Multimedia database, content delivery and transport; Multimedia 
service protocols; Mobility of multimedia services; Internet telephony and 
hypermedia technologies and systems; Media enabled eCommerce service; Case studies, 
field trials and evaluation of new multimedia services

Multimedia applications

   Real-time interactive multimedia applications Adaptive and context-aware 
multimedia applications; Ambiance multimedia applications; Media applications on 
mobile devices; Multi-modal interaction; Virtual environments; Personalization; 
Collaboration, contextual metadata, collaborative tagging; Web applications; 
Multimedia authoring; Multimedia-enabled new applications (eLearning, 
entertainment,?..); Cooperative networks and applications; Mobile multimedia 
applications & services; Semantic metadata for mobile applications; Semantics 
enabled multimedia applications; Semantics enabled networks and middleware for 
multimedia applications; Wireless ad-hoc and sensor networks/RFID applications 
Industrial use-cases and applications
Multimedia security and content protection

   Multimedia security (watermark, encryption,? ); Mobile multimedia systems and 
services; Security, privacy, and cryptographic protocols; Network security issues 
and protocols; Key management and authentication; Authentication and access 
control; Intrusion detection and prevention; Content protection and digital rights 
management; Trusted computing; Information hiding; Protection of user-generated 
content

Multimedia control and management

   Wireless and mobile multimedia network management; Multimedia measurement, 
control, and management; Content management and delivery; IP multimedia system 
operations and management; Managing the quality of experience and quality of 
service; Measuring the quality of performance in multimedia systems; Mobile 
multimedia network traffic engineering and optimization; Monitoring and managing 
mobile multimedia; Resource reservation for multimedia services; Multicast and 
broadcast multimedia service management; Management of service oriented 
architectures; Pricing, accounting and billing for multimedia services

---------------------------
Committee: http://www.iaria.org/conferences2013/ComMMEDIA13.html
================================================  

From wwwrun@rfc-editor.org  Fri Nov 23 09:50:12 2012
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Subject: [AVTCORE] RFC 6792 on Guidelines for Use of the RTP Monitoring Framework
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A new Request for Comments is now available in online RFC libraries.

        
        RFC 6792

        Title:      Guidelines for Use of the 
                    RTP Monitoring Framework 
        Author:     Q. Wu, Ed.,
                    G. Hunt, P. Arden
        Status:     Informational
        Stream:     IETF
        Date:       November 2012
        Mailbox:    sunseawq@huawei.com, 
                    r.geoff.hunt@gmail.com, 
                    philip.arden@bt.com
        Pages:      17
        Characters: 42473
        Updates/Obsoletes/SeeAlso:   None

        I-D Tag:    draft-ietf-avtcore-monarch-22.txt

        URL:        http://www.rfc-editor.org/rfc/rfc6792.txt

This memo proposes an extensible Real-time Transport Protocol (RTP)
monitoring framework for extending the RTP Control Protocol (RTCP)
with a new RTCP Extended Reports (XR) block type to report new
metrics regarding media transmission or reception quality.  In this
framework, a new XR block should contain a single metric or a small
number of metrics relevant to a single parameter of interest or
concern, rather than containing a number of metrics that attempt to
provide full coverage of all those parameters of concern to a
specific application.  Applications may then "mix and match" to
create a set of blocks that cover their set of concerns.  Where
possible, a specific block should be designed to be reusable across
more than one application, for example, for all of voice, streaming
audio, and video.  This document is not an Internet Standards Track 
specification; it is published for informational purposes.

This document is a product of the Audio/Video Transport Core Maintenance Working Group of the IETF.


INFORMATIONAL: This memo provides information for the Internet community.
It does not specify an Internet standard of any kind. Distribution of
this memo is unlimited.

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Subject: Re: [AVTCORE] I-D Action: draft-ietf-avt-srtp-not-mandatory-11.txt
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On 11/19/2012 11:59 PM, Colin Perkins wrote:
> This version of the draft updates the recommendations in Section 6 based on discussion with the Security Area Directors. There are no changes in the rest of the draft.

Mostly looks OK to me.

One thing I did not understand about the new section 6 ....
It describes RTP/AVPF as a profile that is an example of where a single 
security mechanism is not reasonable to mandate, because it's used in 
many other contexts.
I think that's OK - but isn't it true that RTP/AVPF *disallows* the use 
of SRTP, since it would then be RTP/SAVPF?

This can be confusing to the reader - it may be clearer if one mentions 
explicitly that the RTP/AVPF, which is a set of building blocks that 
don't need a security mandate, is used to build RTP/SAVPF, which *is* a 
security mandate.

Or am I the one confused?

                 Harald

>
> Colin
>
>
> On 19 Nov 2012, at 22:51, Internet-Drafts@ietf.org wrote:
>> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>> This draft is a work item of the Audio/Video Transport Core Maintenance Working Group of the IETF.
>>
>> 	Title           : Securing the RTP Protocol Framework: Why RTP Does Not Mandate a Single Media Security Solution
>> 	Author(s)       : Colin Perkins
>>                           Magnus Westerlund
>> 	Filename        : draft-ietf-avt-srtp-not-mandatory-11.txt
>> 	Pages           : 10
>> 	Date            : 2012-11-19
>>
>> Abstract:
>>    This memo discusses the problem of securing real-time multimedia
>>    sessions, and explains why the Real-time Transport Protocol (RTP),
>>    and the associated RTP control protocol (RTCP), do not mandate a
>>    single media security mechanism.  Guidelines for designers and
>>    reviewers of future RTP extensions are provided, to ensure that
>>    appropriate security mechanisms are mandated, and that any such
>>    mechanisms are specified in a manner that conforms with the RTP
>>    architecture.
>>
>>
>> The IETF datatracker status page for this draft is:
>> https://datatracker.ietf.org/doc/draft-ietf-avt-srtp-not-mandatory
>>
>> There's also a htmlized version available at:
>> http://tools.ietf.org/html/draft-ietf-avt-srtp-not-mandatory-11
>>
>> A diff from the previous version is available at:
>> http://www.ietf.org/rfcdiff?url2=draft-ietf-avt-srtp-not-mandatory-11
>


From magnus.westerlund@ericsson.com  Mon Nov 26 00:08:06 2012
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Subject: Re: [AVTCORE] I-D Action: draft-ietf-avt-srtp-not-mandatory-11.txt
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Hi Harald,


On 2012-11-25 00:46, Harald Alvestrand wrote:
> On 11/19/2012 11:59 PM, Colin Perkins wrote:

> One thing I did not understand about the new section 6 ....
> It describes RTP/AVPF as a profile that is an example of where a single
> security mechanism is not reasonable to mandate, because it's used in
> many other contexts.
> I think that's OK - but isn't it true that RTP/AVPF *disallows* the use
> of SRTP, since it would then be RTP/SAVPF?
> 
> This can be confusing to the reader - it may be clearer if one mentions
> explicitly that the RTP/AVPF, which is a set of building blocks that
> don't need a security mandate, is used to build RTP/SAVPF, which *is* a
> security mandate.
> 
> Or am I the one confused?

I guess you are getting tripped up on this sentence:

   In other cases, though,
   an RTP profile is applicable to such a wide range of applications
   that it would not make sense for that profile to mandate particular
   security building blocks be used (the RTP/AVPF profile [RFC4585] is
   an example of this type of RTP profile, since it provides building
   blocks that can be used in different styles of application).

This is an example why RTP profiles, independent of which profile it may
be is not the appropriate level of putting in the mandatory to implement
security solution alternative. And frankly SAVPF and SAVP is only half a
security mandate, they lack key-management and without a mandatory to
implement key-management solution they do not specify a MTI security
solution.

And if I count correctly in draft-ietf-avtcore-rtp-security-options
there are at least 11 options for keying SRTP.

Cheers

Magnus Westerlund

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


From harald@alvestrand.no  Mon Nov 26 00:49:31 2012
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On 11/26/2012 09:08 AM, Magnus Westerlund wrote:
> Hi Harald,
>
>
> On 2012-11-25 00:46, Harald Alvestrand wrote:
>> On 11/19/2012 11:59 PM, Colin Perkins wrote:
>> One thing I did not understand about the new section 6 ....
>> It describes RTP/AVPF as a profile that is an example of where a single
>> security mechanism is not reasonable to mandate, because it's used in
>> many other contexts.
>> I think that's OK - but isn't it true that RTP/AVPF *disallows* the use
>> of SRTP, since it would then be RTP/SAVPF?
>>
>> This can be confusing to the reader - it may be clearer if one mentions
>> explicitly that the RTP/AVPF, which is a set of building blocks that
>> don't need a security mandate, is used to build RTP/SAVPF, which *is* a
>> security mandate.
>>
>> Or am I the one confused?
> I guess you are getting tripped up on this sentence:
>
>     In other cases, though,
>     an RTP profile is applicable to such a wide range of applications
>     that it would not make sense for that profile to mandate particular
>     security building blocks be used (the RTP/AVPF profile [RFC4585] is
>     an example of this type of RTP profile, since it provides building
>     blocks that can be used in different styles of application).
>
> This is an example why RTP profiles, independent of which profile it may
> be is not the appropriate level of putting in the mandatory to implement
> security solution alternative. And frankly SAVPF and SAVP is only half a
> security mandate, they lack key-management and without a mandatory to
> implement key-management solution they do not specify a MTI security
> solution.
Yes, my opinion of the usefulness of the RTP "profile" field is 
sometimes acerbic.
SAVP(F) mandates use of SRTP, but that gets you only some of the way.
>
> And if I count correctly in draft-ietf-avtcore-rtp-security-options
> there are at least 11 options for keying SRTP.
Sigh.
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> Färögatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>


From mcgrew@cisco.com  Mon Nov 26 08:18:33 2012
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From: "David McGrew (mcgrew)" <mcgrew@cisco.com>
To: Harald Alvestrand <harald@alvestrand.no>, Magnus Westerlund <magnus.westerlund@ericsson.com>
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On 11/26/12 3:49 AM, "Harald Alvestrand" <harald@alvestrand.no> wrote:

>On 11/26/2012 09:08 AM, Magnus Westerlund wrote:
>> Hi Harald,
>>
>>
>> On 2012-11-25 00:46, Harald Alvestrand wrote:
>>> On 11/19/2012 11:59 PM, Colin Perkins wrote:
>>> One thing I did not understand about the new section 6 ....
>>> It describes RTP/AVPF as a profile that is an example of where a single
>>> security mechanism is not reasonable to mandate, because it's used in
>>> many other contexts.
>>> I think that's OK - but isn't it true that RTP/AVPF *disallows* the use
>>> of SRTP, since it would then be RTP/SAVPF?
>>>
>>> This can be confusing to the reader - it may be clearer if one mentions
>>> explicitly that the RTP/AVPF, which is a set of building blocks that
>>> don't need a security mandate, is used to build RTP/SAVPF, which *is* a
>>> security mandate.
>>>
>>> Or am I the one confused?
>> I guess you are getting tripped up on this sentence:
>>
>>     In other cases, though,
>>     an RTP profile is applicable to such a wide range of applications
>>     that it would not make sense for that profile to mandate particular
>>     security building blocks be used (the RTP/AVPF profile [RFC4585] is
>>     an example of this type of RTP profile, since it provides building
>>     blocks that can be used in different styles of application).
>>
>> This is an example why RTP profiles, independent of which profile it may
>> be is not the appropriate level of putting in the mandatory to implement
>> security solution alternative. And frankly SAVPF and SAVP is only half a
>> security mandate, they lack key-management and without a mandatory to
>> implement key-management solution they do not specify a MTI security
>> solution.
>Yes, my opinion of the usefulness of the RTP "profile" field is
>sometimes acerbic.
>SAVP(F) mandates use of SRTP, but that gets you only some of the way.
>>
>> And if I count correctly in draft-ietf-avtcore-rtp-security-options
>> there are at least 11 options for keying SRTP.
>Sigh.

Eleven keying method sounds right; that would be DTLS-SRTP, SDP Security
Descriptions, EKT, plus the eight different MIKEY methods.  Too many
options, but not all of them are standards track.

David

>>
>> Cheers
>>
>> Magnus Westerlund
>>
>> ----------------------------------------------------------------------
>> Multimedia Technologies, Ericsson Research EAB/TVM
>> ----------------------------------------------------------------------
>> Ericsson AB                | Phone  +46 10 7148287
>> F=E4r=F6gatan 6                | Mobile +46 73 0949079
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From hannes.tschofenig@gmx.net  Mon Nov 26 09:22:59 2012
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Subject: Re: [AVTCORE] I-D Action: draft-ietf-avt-srtp-not-mandatory-11.txt
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On Nov 26, 2012, at 6:18 PM, David McGrew (mcgrew) wrote:

> Eleven keying method sounds right; that would be DTLS-SRTP, SDP =
Security
> Descriptions, EKT, plus the eight different MIKEY methods.  Too many
> options, but not all of them are standards track.

Given that everyone was (and still is) allowed to publish their favorite =
VoIP media key exchange protocol it is a surprise that we only have so =
few.
Some of the mechanisms have been published in the IETF only because of a =
too restrictive registration policy established by the MIKEY RFC.

On the other hand, if you look at the deployment then the situation is a =
bit different..=

From woohwankim@gmail.com  Mon Nov 26 18:45:09 2012
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From: Woo-Hwan Kim <whkim5@ensec.re.kr>
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Cc: Magnus Westerlund <magnus.westerlund@ericsson.com>, Daesung Kwon <ds_kwon@ensec.re.kr>, Je Hong Park <jhpark@ensec.re.kr>
Subject: Re: [AVTCORE] Review of draft-ietf-avtcore-aria-srtp-00
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--f46d04428c6eb0118804cf7109cb
Content-Type: text/plain; charset=ISO-8859-1

 Thanks for the review.

 Both SEED and ARIA were established as KS(Korean Standard) by the Ministry
of Knowledge Economy of Korea.
 But SEED and ARIA have different application areas each other.
 While SEED is mainly used for for electronic commerce and financial
service, ARIA is for government use and public purpose.
 As the governmental area is growing recently,
 we need to standardize SRTP-ARIA even though SRTP-SEED is already defined
in RFC 5669.

 As for the ciphersuites, we intended to provide the same funtionality as
SRTP-AES.
 While preparing TLS-ARIA(RFC 6209), we added all the ciphersuites which
are defined for TLS-AES in one document
 following the opinion of TLS ADs.
 So, we referred to RFC 6188 which defines CM-AES-192/256 and the
I.D.(draft-ietf-avtcore-srtp-aes-gcm) on the GCM-AES-128/256 and
CCM-AES-128/256.

Thanks.

Woo-Hwan Kim


> Date: Mon, 5 Nov 2012 13:17:02 -0500
> From: Eric Rescorla <ekr@rtfm.com>
> To: avt@ietf.org
> Subject: [AVTCORE] Review of draft-ietf-avtcore-aria-srtp-00
> Message-ID:
> <CABcZeBNWmUYaMZTo5BTN9_YJATNQERrsNpn_CF5Q4FegOwcYPw@mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> I have reviewed draft-ietf-avtcore-aria-srtp-00.
>
> I'm leaving aside the question of whether it's useful to define
> another symmetric encryption algorithm for SRTP (my view is generally
> no, especially since we have already specified SEED)
> but the WG seems to have decided that.
>
> With that said I don't really understand why you are defining three
> key sizes and three counter modes. I would recommend defining only 128
> and 256-bit GCM unless there is some actual reason not to.
>
> Other than that, this seems like a reasonably plausible draft.
>
>

--f46d04428c6eb0118804cf7109cb
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<div>=A0Thanks for the review.<br>=A0<br>=A0Both SEED and ARIA were establi=
shed=20
as KS(Korean Standard) by the Ministry of Knowledge Economy of Korea.<br>=
=A0But=20
SEED and ARIA have different application areas each other.<br>=A0While SEED=
 is=20
mainly used for for electronic commerce and financial service, ARIA is for=
=20
government use and public purpose.<br>=A0As the governmental area is growin=
g=20
recently, <br>=A0we need to standardize SRTP-ARIA even though SRTP-SEED is =
already=20
defined in RFC 5669.<br>=A0<br>=A0As for the ciphersuites, we intended to p=
rovide=20
the same funtionality as SRTP-AES.<br>=A0While preparing TLS-ARIA(RFC 6209)=
, we=20
added all the ciphersuites which are defined for TLS-AES in one document=20
<br>=A0following the opinion of TLS ADs.<br>=A0So, we referred to RFC 6188 =
which=20
defines CM-AES-192/256 and the I.D.(draft-ietf-avtcore-srtp-aes-gcm) on the=
=20
GCM-AES-128/256 and CCM-AES-128/256.<br></div><div><br></div><div>Thanks.</=
div><div><br></div><div>Woo-Hwan Kim</div><span style=3D"color:rgb(34,34,34=
);font-family:arial,sans-serif;font-size:14px;background-color:rgb(255,255,=
255)"><div>
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
4px;background-color:rgb(255,255,255)"><br></span></div><div><span style=3D=
"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:14px;background=
-color:rgb(255,255,255)"><br>
</span></div>&gt; Date: Mon, 5 Nov 2012 13:17:02 -0500</span><br style=3D"c=
olor:rgb(34,34,34);font-family:arial,sans-serif;font-size:14px;background-c=
olor:rgb(255,255,255)"><span style=3D"color:rgb(34,34,34);font-family:arial=
,sans-serif;font-size:14px;background-color:rgb(255,255,255)">&gt; From: Er=
ic Rescorla &lt;</span><a href=3D"mailto:ekr@rtfm.com" style=3D"color:rgb(1=
7,85,204);font-family:arial,sans-serif;font-size:14px;background-color:rgb(=
255,255,255)">ekr@rtfm.com</a><span style=3D"color:rgb(34,34,34);font-famil=
y:arial,sans-serif;font-size:14px;background-color:rgb(255,255,255)">&gt;</=
span><br style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-siz=
e:14px;background-color:rgb(255,255,255)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
4px;background-color:rgb(255,255,255)">&gt; To:=A0</span><a href=3D"mailto:=
avt@ietf.org" style=3D"color:rgb(17,85,204);font-family:arial,sans-serif;fo=
nt-size:14px;background-color:rgb(255,255,255)">avt@ietf.org</a><br style=
=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:14px;backgro=
und-color:rgb(255,255,255)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
4px;background-color:rgb(255,255,255)">&gt; Subject: [AVTCORE] Review of dr=
aft-ietf-avtcore-aria-srtp-</span><span style=3D"color:rgb(34,34,34);font-f=
amily:arial,sans-serif;font-size:14px;background-color:rgb(255,255,255)">00=
</span><br style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-s=
ize:14px;background-color:rgb(255,255,255)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
4px;background-color:rgb(255,255,255)">&gt; Message-ID:</span><br style=3D"=
color:rgb(34,34,34);font-family:arial,sans-serif;font-size:14px;background-=
color:rgb(255,255,255)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
4px;background-color:rgb(255,255,255)">&gt; &lt;</span><a href=3D"mailto:CA=
BcZeBNWmUYaMZTo5BTN9_YJATNQERrsNpn_CF5Q4FegOwcYPw@mail.gmail.com" style=3D"=
color:rgb(17,85,204);font-family:arial,sans-serif;font-size:14px;background=
-color:rgb(255,255,255)">CABcZeBNWmUYaMZTo5BTN9_YJATNQERrsNpn_CF5Q4FegOwcYP=
w@mail.gmail.com</a><span style=3D"color:rgb(34,34,34);font-family:arial,sa=
ns-serif;font-size:14px;background-color:rgb(255,255,255)">&gt;</span><br s=
tyle=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:14px;bac=
kground-color:rgb(255,255,255)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
4px;background-color:rgb(255,255,255)">&gt; Content-Type: text/plain; chars=
et=3DISO-8859-1</span><br style=3D"color:rgb(34,34,34);font-family:arial,sa=
ns-serif;font-size:14px;background-color:rgb(255,255,255)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
4px;background-color:rgb(255,255,255)">&gt;</span><br style=3D"color:rgb(34=
,34,34);font-family:arial,sans-serif;font-size:14px;background-color:rgb(25=
5,255,255)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
4px;background-color:rgb(255,255,255)">&gt; I have reviewed draft-ietf-avtc=
ore-aria-srtp-</span><span style=3D"color:rgb(34,34,34);font-family:arial,s=
ans-serif;font-size:14px;background-color:rgb(255,255,255)">00.</span><br s=
tyle=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:14px;bac=
kground-color:rgb(255,255,255)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
4px;background-color:rgb(255,255,255)">&gt;</span><br style=3D"color:rgb(34=
,34,34);font-family:arial,sans-serif;font-size:14px;background-color:rgb(25=
5,255,255)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
4px;background-color:rgb(255,255,255)">&gt; I&#39;m leaving aside the quest=
ion of whether it&#39;s useful to define</span><br style=3D"color:rgb(34,34=
,34);font-family:arial,sans-serif;font-size:14px;background-color:rgb(255,2=
55,255)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
4px;background-color:rgb(255,255,255)">&gt; another symmetric encryption al=
gorithm for SRTP (my view is generally</span><br style=3D"color:rgb(34,34,3=
4);font-family:arial,sans-serif;font-size:14px;background-color:rgb(255,255=
,255)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
4px;background-color:rgb(255,255,255)">&gt; no, especially since we have al=
ready specified SEED)</span><br style=3D"color:rgb(34,34,34);font-family:ar=
ial,sans-serif;font-size:14px;background-color:rgb(255,255,255)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
4px;background-color:rgb(255,255,255)">&gt; but the WG seems to have decide=
d that.</span><br style=3D"color:rgb(34,34,34);font-family:arial,sans-serif=
;font-size:14px;background-color:rgb(255,255,255)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
4px;background-color:rgb(255,255,255)">&gt;</span><br style=3D"color:rgb(34=
,34,34);font-family:arial,sans-serif;font-size:14px;background-color:rgb(25=
5,255,255)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
4px;background-color:rgb(255,255,255)">&gt; With that said I don&#39;t real=
ly understand why you are defining three</span><br style=3D"color:rgb(34,34=
,34);font-family:arial,sans-serif;font-size:14px;background-color:rgb(255,2=
55,255)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
4px;background-color:rgb(255,255,255)">&gt; key sizes and three counter mod=
es. I would recommend defining only 128</span><br style=3D"color:rgb(34,34,=
34);font-family:arial,sans-serif;font-size:14px;background-color:rgb(255,25=
5,255)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
4px;background-color:rgb(255,255,255)">&gt; and 256-bit GCM unless there is=
 some actual reason not to.</span><br style=3D"color:rgb(34,34,34);font-fam=
ily:arial,sans-serif;font-size:14px;background-color:rgb(255,255,255)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
4px;background-color:rgb(255,255,255)">&gt;</span><br style=3D"color:rgb(34=
,34,34);font-family:arial,sans-serif;font-size:14px;background-color:rgb(25=
5,255,255)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
4px;background-color:rgb(255,255,255)">&gt; Other than that, this seems lik=
e a reasonably plausible draft.</span><br style=3D"color:rgb(34,34,34);font=
-family:arial,sans-serif;font-size:14px;background-color:rgb(255,255,255)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
4px;background-color:rgb(255,255,255)">&gt;</span><br style=3D"color:rgb(34=
,34,34);font-family:arial,sans-serif;font-size:14px;background-color:rgb(25=
5,255,255)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
4px;background-color:rgb(255,255,255)">&gt;</span>

--f46d04428c6eb0118804cf7109cb--

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Subject: [AVTCORE] Working group last call on draft-ietf-avtcore-srtp-aes-gcm-03
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WG,

This announces the WG last call on AES-GCM and AES-CCM Authenticated
Encryption in Secure RTP (SRTP)
https://datatracker.ietf.org/doc/draft-ietf-avtcore-srtp-aes-gcm/

Please provide any comments no later than the 12th of December. Also
comments of the nature of "I have read it and have no comments and think
it should be published" are highly valuable.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
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----------------------------------------------------------------------
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From magnus.westerlund@ericsson.com  Tue Nov 27 07:07:55 2012
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Subject: [AVTCORE] Confirmation of adoption of draft-rescorla-avtcore-6222bis-00
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WG,

At the IETF 85th meeting's first AVTCORE session we took a hum for
adopting Guidelines for Choosing RTP Control Protocol (RTCP) Canonical
Names (CNAMEs) (draft-rescorla-avtcore-6222bis-00) as a WG item with the
intention to have it replace RFC 6222.

I would like to confirm this call for adoption also on the WG mailing
list. So people that didn't provided input into this in the meeting
please provide your support or objections no later than the 6th of December.

Cheers

Magnus Westerlund

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


From magnus.westerlund@ericsson.com  Tue Nov 27 07:32:52 2012
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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: [AVTCORE] Minutes for AVTCORE at IETF85
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WG,

Here is the initial version of the minutes.

http://www.ietf.org/proceedings/85/minutes/minutes-85-avtcore

please review and provide any comments for things you like us to address.

Cheers

Magnus Westerlund

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----------------------------------------------------------------------
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Färögatan 6                | Mobile +46 73 0949079
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From adam@nostrum.com  Tue Nov 27 08:54:19 2012
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Cc: IETF AVTCore WG <avt@ietf.org>
Subject: Re: [AVTCORE] Confirmation of adoption of draft-rescorla-avtcore-6222bis-00
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I support adoption of this document as a working group item.

/a

On 11/27/12 07:07, Magnus Westerlund wrote:
> WG,
>
> At the IETF 85th meeting's first AVTCORE session we took a hum for
> adopting Guidelines for Choosing RTP Control Protocol (RTCP) Canonical
> Names (CNAMEs) (draft-rescorla-avtcore-6222bis-00) as a WG item with the
> intention to have it replace RFC 6222.
>
> I would like to confirm this call for adoption also on the WG mailing
> list. So people that didn't provided input into this in the meeting
> please provide your support or objections no later than the 6th of December.
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> Färögatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
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From jonathan@vidyo.com  Tue Nov 27 09:31:33 2012
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From: Jonathan Lennox <jonathan@vidyo.com>
To: IETF AVTCore WG <avt@ietf.org>
Date: Tue, 27 Nov 2012 12:31:05 -0500
Thread-Topic: [AVTCORE] Working group last call on draft-ietf-avtcore-srtp-aes-gcm-03
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Subject: Re: [AVTCORE] Working group last call on draft-ietf-avtcore-srtp-aes-gcm-03
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I have read the draft.

It would be useful to have test vectors for the various algorithms, for imp=
lementers to validate their implementations.

The draft needs to specify how header extension encryption (as specified in=
 draft-ietf-avtcore-srtp-encrypted-header-ext) is to be done when using the=
 AES-GCM and AES-CCM algorithms.

I believe the consensus was that it should be done using the equivalent CTR=
-mode encryption transform, so I'd suggest adding something like the follow=
ing text to this draft:

[Section Number]. Header Extension Encryption.

When RTP Header Extension Encryption [I-D.ietf-avtcore-srtp-encrypted-heade=
r-ext] is in use, a separate keystream is generated to encrypt selected RTP=
 header extension elements.  For the AEAD_AES_128_GCM and the AEAD_AES_128_=
CCM algorithms, this keystream MUST be generated in the manner defined in [=
I-D.ietf-avtcore-srtp-encrypted-header-ext] for the AES_128_CM transform.  =
For the AEAD_AES_256_GCM and the AEAD_AES_256_CCM algorithms, the keystream=
 MUST be generated in the manner defined for the AES_256_CM transform.

-----Original Message-----
From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]=20
Sent: Tuesday, November 27, 2012 9:21 AM
To: IETF AVTCore WG
Subject: [AVTCORE] Working group last call on draft-ietf-avtcore-srtp-aes-g=
cm-03

WG,

This announces the WG last call on AES-GCM and AES-CCM Authenticated Encryp=
tion in Secure RTP (SRTP) https://datatracker.ietf.org/doc/draft-ietf-avtco=
re-srtp-aes-gcm/

Please provide any comments no later than the 12th of December. Also commen=
ts of the nature of "I have read it and have no comments and think it shoul=
d be published" are highly valuable.

Cheers

Magnus Westerlund

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

_______________________________________________
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From: "David McGrew (mcgrew)" <mcgrew@cisco.com>
To: Jonathan Lennox <jonathan@vidyo.com>, IETF AVTCore WG <avt@ietf.org>
Thread-Topic: [AVTCORE] Working group last call on draft-ietf-avtcore-srtp-aes-gcm-03
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Subject: Re: [AVTCORE] Working group last call on draft-ietf-avtcore-srtp-aes-gcm-03
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Thanks Jonathan for the review and suggested text.   I agree about the
header extension encryption.

David

On 11/27/12 12:31 PM, "Jonathan Lennox" <jonathan@vidyo.com> wrote:

>I have read the draft.
>
>It would be useful to have test vectors for the various algorithms, for
>implementers to validate their implementations.
>
>The draft needs to specify how header extension encryption (as specified
>in draft-ietf-avtcore-srtp-encrypted-header-ext) is to be done when using
>the AES-GCM and AES-CCM algorithms.
>
>I believe the consensus was that it should be done using the equivalent
>CTR-mode encryption transform, so I'd suggest adding something like the
>following text to this draft:
>
>[Section Number]. Header Extension Encryption.
>
>When RTP Header Extension Encryption
>[I-D.ietf-avtcore-srtp-encrypted-header-ext] is in use, a separate
>keystream is generated to encrypt selected RTP header extension elements.
> For the AEAD_AES_128_GCM and the AEAD_AES_128_CCM algorithms, this
>keystream MUST be generated in the manner defined in
>[I-D.ietf-avtcore-srtp-encrypted-header-ext] for the AES_128_CM
>transform.  For the AEAD_AES_256_GCM and the AEAD_AES_256_CCM algorithms,
>the keystream MUST be generated in the manner defined for the AES_256_CM
>transform.
>
>-----Original Message-----
>From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
>Sent: Tuesday, November 27, 2012 9:21 AM
>To: IETF AVTCore WG
>Subject: [AVTCORE] Working group last call on
>draft-ietf-avtcore-srtp-aes-gcm-03
>
>WG,
>
>This announces the WG last call on AES-GCM and AES-CCM Authenticated
>Encryption in Secure RTP (SRTP)
>https://datatracker.ietf.org/doc/draft-ietf-avtcore-srtp-aes-gcm/
>
>Please provide any comments no later than the 12th of December. Also
>comments of the nature of "I have read it and have no comments and think
>it should be published" are highly valuable.
>
>Cheers
>
>Magnus Westerlund
>
>----------------------------------------------------------------------
>Multimedia Technologies, Ericsson Research EAB/TVM
>----------------------------------------------------------------------
>Ericsson AB                | Phone  +46 10 7148287
>F=E4r=F6gatan 6                | Mobile +46 73 0949079
>SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>----------------------------------------------------------------------
>
>_______________________________________________
>Audio/Video Transport Core Maintenance
>avt@ietf.org
>https://www.ietf.org/mailman/listinfo/avt
>
>_______________________________________________
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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: Adam Roach <adam@nostrum.com>, Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: [AVTCORE] Confirmation of adoption of draft-rescorla-avtcore-6222bis-00
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+1

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
> Adam Roach
> Sent: Tuesday, November 27, 2012 8:54 AM
> To: Magnus Westerlund
> Cc: IETF AVTCore WG
> Subject: Re: [AVTCORE] Confirmation of adoption of draft-rescorla-avtcore=
-
> 6222bis-00
>=20
> I support adoption of this document as a working group item.
>=20
> /a
>=20
> On 11/27/12 07:07, Magnus Westerlund wrote:
> > WG,
> >
> > At the IETF 85th meeting's first AVTCORE session we took a hum for
> > adopting Guidelines for Choosing RTP Control Protocol (RTCP) Canonical
> > Names (CNAMEs) (draft-rescorla-avtcore-6222bis-00) as a WG item with
> the
> > intention to have it replace RFC 6222.
> >
> > I would like to confirm this call for adoption also on the WG mailing
> > list. So people that didn't provided input into this in the meeting
> > please provide your support or objections no later than the 6th of
> December.
> >
> > Cheers
> >
> > Magnus Westerlund
> >
> > ----------------------------------------------------------------------
> > Multimedia Technologies, Ericsson Research EAB/TVM
> > ----------------------------------------------------------------------
> > Ericsson AB                | Phone  +46 10 7148287
> > F=E4r=F6gatan 6                | Mobile +46 73 0949079
> > SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> > ----------------------------------------------------------------------
> >
> > _______________________________________________
> > Audio/Video Transport Core Maintenance
> > avt@ietf.org
> > https://www.ietf.org/mailman/listinfo/avt
>=20
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt

From csp@csperkins.org  Tue Nov 27 14:22:06 2012
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On 27 Nov 2012, at 15:07, Magnus Westerlund wrote:
> At the IETF 85th meeting's first AVTCORE session we took a hum for
> adopting Guidelines for Choosing RTP Control Protocol (RTCP) Canonical
> Names (CNAMEs) (draft-rescorla-avtcore-6222bis-00) as a WG item with =
the intention to have it replace RFC 6222.
>=20
> I would like to confirm this call for adoption also on the WG mailing
> list. So people that didn't provided input into this in the meeting
> please provide your support or objections no later than the 6th of =
December.


I have read the draft, and support it being adopted as a working draft =
item.

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




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From: "Roni Even" <ron.even.tlv@gmail.com>
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WG,

 

At the IETF 85th meeting's second AVTCORE session we took a hum for adopting
Update to Recommended Codecs for the AVP RTP Profile

(draft-terriberry-avp-codecs-00) as a WG item with the intention to update
RFC3551.

 

I would like to confirm this call for adoption also on the WG mailing list.
So people that didn't provided input into this in the meeting please provide
your support or objections no later than December 10th.

 

Thanks

Roni


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	mso-style-priority:99;
	mso-style-link:"Plain Text";
	font-family:"Calibri","sans-serif";}
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	{mso-style-name:"HTML Preformatted Char";
	mso-style-priority:99;
	mso-style-link:"HTML Preformatted";
	font-family:"Courier New";}
span.h11
	{mso-style-name:h11;
	font-family:"Courier New";
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.MsoChpDefault
	{mso-style-type:export-only;
	font-family:"Calibri","sans-serif";}
@page WordSection1
	{size:8.5in 11.0in;
	margin:1.0in 1.25in 1.0in 1.25in;}
div.WordSection1
	{page:WordSection1;}
--></style><!--[if gte mso 9]><xml>
<o:shapedefaults v:ext=3D"edit" spidmax=3D"1026" />
</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext=3D"edit">
<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]--></head><body lang=3DEN-US link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p =
class=3DMsoPlainText>WG,<o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><pre><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'>At the =
IETF 85th meeting's second AVTCORE session we took a hum for adopting =
</span><span lang=3DEN =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'>Update to =
Recommended Codecs for the AVP RTP Profile<o:p></o:p></span></pre><p =
class=3DMsoPlainText> (draft-terriberry-avp-codecs-00) as a WG item with =
the intention to update &nbsp;RFC3551.<o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p class=3DMsoPlainText>I =
would like to confirm this call for adoption also on the WG mailing =
list. So people that didn't provided input into this in the meeting =
please provide your support or objections no later than December =
10<sup>th</sup>.<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>Thanks<o:p></o:p></p><p =
class=3DMsoNormal>Roni<o:p></o:p></p></div></body></html>
------=_NextPart_000_04B8_01CDCD43.A5156C80--


From wwwrun@rfc-editor.org  Wed Nov 28 06:39:40 2012
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Subject: [AVTCORE] [Technical Errata Reported] RFC3711 (3420)
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The following errata report has been submitted for RFC3711,
"The Secure Real-time Transport Protocol (SRTP)".

--------------------------------------
You may review the report below and at:
http://www.rfc-editor.org/errata_search.php?rfc=3711&eid=3420

--------------------------------------
Type: Technical
Reported by: Matthias Schertler <matthias.schertler@gmx.net>

Section: 3.1.

Original Text
-------------
   The "Encrypted Portion" of an SRTP packet consists of the encryption
   of the RTP payload (including RTP padding when present) of the
   equivalent RTP packet.

Corrected Text
--------------
   The "Encrypted Portion" of an SRTP packet consists of the encryption
   of the RTP payload (including RTP padding and RTP pad count when present)
   of the equivalent RTP packet.  

Notes
-----
In Figure 1 "RTP padding" and "RTP pad count" are different things. The text should use the same terminology in order to make clear that the padding count is encrypted.

Instructions:
-------------
This errata is currently posted as "Reported". If necessary, please
use "Reply All" to discuss whether it should be verified or
rejected. When a decision is reached, the verifying party (IESG)
can log in to change the status and edit the report, if necessary. 

--------------------------------------
RFC3711 (draft-ietf-avt-srtp-09)
--------------------------------------
Title               : The Secure Real-time Transport Protocol (SRTP)
Publication Date    : March 2004
Author(s)           : M. Baugher, D. McGrew, M. Naslund, E. Carrara, K. Norrman
Category            : PROPOSED STANDARD
Source              : Audio/Video Transport
Area                : Real-time Applications and Infrastructure
Stream              : IETF
Verifying Party     : IESG

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