
From magnus.westerlund@ericsson.com  Mon Apr  2 05:43:09 2012
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Date: Mon, 2 Apr 2012 14:43:05 +0200
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Subject: Re: [AVTCORE] WG last call on Monitoring Architecture for RTP (draft-ietf-avtcore-monarch-11)
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WG,

The date set as end for this WG last call has passed. However, I will
keep it open until the end of the week to let some more people comment
on it. I do hope that the XR block people will provide some feedback on it.

Thanks,

Magnus

On 2012-03-05 08:44, Magnus Westerlund wrote:
> WG,
> 
> This initiates the Working group last call on Monitoring Architecture
> for RTP (draft-ietf-avtcore-monarch-11) to be published as Informational
> RFC. The WG last call will run for three weeks and end Monday the 26th.
> 
> The draft can be retrieved from here:
> https://datatracker.ietf.org/doc/draft-ietf-avtcore-monarch/
> 
> Please provide feedback if you have read it, even if you have no
> comments or think it is good enough, this is important information in
> judging the support and readiness for publication.
> 
> Thanks
> 
> 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
> 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 ray.vanbrandenburg@tno.nl  Mon Apr  2 05:47:39 2012
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From: "Brandenburg, R. (Ray) van" <ray.vanbrandenburg@tno.nl>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: [xrblock] [AVTCORE] WG last call on Monitoring Architecture for	RTP (draft-ietf-avtcore-monarch-11)
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Subject: Re: [AVTCORE] [xrblock] WG last call on Monitoring Architecture for	RTP (draft-ietf-avtcore-monarch-11)
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Hi,

I thought I had already send a mail before, but that mail seems to have nev=
er arrived.  =


Anyway, I have read the document and I think it is ready. =


Ray

On 2 apr. 2012, at 14:43, "Magnus Westerlund" <magnus.westerlund@ericsson.c=
om> wrote:

> WG,
> =

> The date set as end for this WG last call has passed. However, I will
> keep it open until the end of the week to let some more people comment
> on it. I do hope that the XR block people will provide some feedback on i=
t.
> =

> Thanks,
> =

> Magnus
> =

> On 2012-03-05 08:44, Magnus Westerlund wrote:
>> WG,
>> =

>> This initiates the Working group last call on Monitoring Architecture
>> for RTP (draft-ietf-avtcore-monarch-11) to be published as Informational
>> RFC. The WG last call will run for three weeks and end Monday the 26th.
>> =

>> The draft can be retrieved from here:
>> https://datatracker.ietf.org/doc/draft-ietf-avtcore-monarch/
>> =

>> Please provide feedback if you have read it, even if you have no
>> comments or think it is good enough, this is important information in
>> judging the support and readiness for publication.
>> =

>> Thanks
>> =

>> 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
>> =

> =

> =

> -- =

> =

> 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
> ----------------------------------------------------------------------
> =

> _______________________________________________
> xrblock mailing list
> xrblock@ietf.org
> https://www.ietf.org/mailman/listinfo/xrblock
This e-mail and its contents are subject to the DISCLAIMER at http://www.tn=
o.nl/emaildisclaimer


From csp@csperkins.org  Mon Apr  2 08:09:25 2012
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From: Colin Perkins <csp@csperkins.org>
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Subject: Re: [AVTCORE] [xrblock] WG last call on Monitoring Architecture for RTP (draft-ietf-avtcore-monarch-11)
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Magnus,

I think this draft is in good shape, but there are two issues that came =
up at IETF83 that should be addressed:

- In the discussion about QoE metrics, it became clear that there are =
some classes of metric that can only be interpreted with knowledge of =
the media codec that is being used (MOS score was the triggering =
example, but there may be others). It might be worth adding a short =
section between the current Sections 5.1 and 5.2 on "Correlating RTCP XR =
with RTP Data", to mention that in some cases this correlation is =
needed, and note that it's necessary to include both the payload type of =
the reported media in such reports, and signal details of the payload =
format and parameters to which that payload type is bound.

- Section 5.1 talks about the need to use single metrics blocks. This is =
fine as far as it goes, but it would be appropriate to highlight that it =
is appropriate to not-only define metrics separately, but also to do so =
in separate drafts where possible, since this makes it easier to evolve =
the metrics (update each block separately), and also makes it easier to =
require compliance with particular reports.

Colin



On 2 Apr 2012, at 13:43, Magnus Westerlund wrote:
> WG,
>=20
> The date set as end for this WG last call has passed. However, I will
> keep it open until the end of the week to let some more people comment
> on it. I do hope that the XR block people will provide some feedback =
on it.
>=20
> Thanks,
>=20
> Magnus
>=20
> On 2012-03-05 08:44, Magnus Westerlund wrote:
>> WG,
>>=20
>> This initiates the Working group last call on Monitoring Architecture
>> for RTP (draft-ietf-avtcore-monarch-11) to be published as =
Informational
>> RFC. The WG last call will run for three weeks and end Monday the =
26th.
>>=20
>> The draft can be retrieved from here:
>> https://datatracker.ietf.org/doc/draft-ietf-avtcore-monarch/
>>=20
>> Please provide feedback if you have read it, even if you have no
>> comments or think it is good enough, this is important information in
>> judging the support and readiness for publication.
>>=20
>> Thanks
>>=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
>>=20
>=20
>=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
> _______________________________________________
> xrblock mailing list
> xrblock@ietf.org
> https://www.ietf.org/mailman/listinfo/xrblock


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




From p.ohanlon@gmail.com  Tue Apr  3 05:15:07 2012
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From: Piers O'Hanlon <p.ohanlon@gmail.com>
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Subject: Re: [AVTCORE] Adopting new work on RTP Congestion Circuit Breakers
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I support adoption by the WG - it would be useful to have such circuit =
breakers available sooner than later.

Piers O'Hanlon

On 29 Mar 2012, at 13:25, Magnus Westerlund wrote:

> WG,
>=20
> At the IETF 83 there was a presentation in the AVTCORE and also in the
> ICCRG of "RTP Congestion Control: Circuit Breakers for Unicast =
Sessions"
>=20
> =
https://datatracker.ietf.org/doc/draft-perkins-avtcore-rtp-circuit-breaker=
s/
>=20
> This was well received and there was support for adopting this as WG
> item. Thus I would like to confirm on the mailing list that we should
> adopt a WG item that works on providing RTP congestion control circuit
> breakers for unicast transport flows.
>=20
> WG, please comment if you support or disagree of such adaptation. The =
WG
> chairs will call a consensus after Friday the 13th.
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt


From magnus.westerlund@ericsson.com  Tue Apr  3 07:00:26 2012
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On 2012-03-30 02:53, James M. Polk wrote:
> While I generally think this draft is 
> interesting, even the author admits it's awfully 
> drafty at this point (both in AVTCORE and 
> TSV-Dir). And I think perhaps too much so in this 
> version. In other words, I'd like to see more meat on them bones.

James, do you think the draft is so drafty that you don't understand the
intent on what is to be defined? I did ask for adoption of the WG item,
i.e. the task to provide a specific functionality rather than a specific
document. I do find it most likely that we adopt the referenced
document, but step one is to create a milestone.

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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At 09:00 AM 4/3/2012, Magnus Westerlund wrote:
>On 2012-03-30 02:53, James M. Polk wrote:
> > While I generally think this draft is
> > interesting, even the author admits it's awfully
> > drafty at this point (both in AVTCORE and
> > TSV-Dir). And I think perhaps too much so in this
> > version. In other words, I'd like to see more meat on them bones.
>
>James, do you think the draft is so drafty that you don't understand the
>intent on what is to be defined? I did ask for adoption of the WG item,
>i.e. the task to provide a specific functionality rather than a specific
>document. I do find it most likely that we adopt the referenced
>document, but step one is to create a milestone.

I support the milestone.

I think the draft is going in the right=20
direction, but needs a bit more meat on them=20
bones before I support adoption of this draft to=20
fulfill the milestone. In other words, I don't=20
believe the authors will take a wrong turn when=20
filling in the rest of the draft, but want to=20
have some more there so we know one or more of=20
the existing thin areas doesn't do just that. I=20
believe Colin will do a good job.

Does this explain where I'm coming from?

James


>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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Cc: IETF AVTCore WG <avt@ietf.org>
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On 2012-04-04 02:52, James M. Polk wrote:
> At 09:00 AM 4/3/2012, Magnus Westerlund wrote:
>> On 2012-03-30 02:53, James M. Polk wrote:
>>> While I generally think this draft is
>>> interesting, even the author admits it's awfully
>>> drafty at this point (both in AVTCORE and
>>> TSV-Dir). And I think perhaps too much so in this
>>> version. In other words, I'd like to see more meat on them bones.
>>
>> James, do you think the draft is so drafty that you don't understand the
>> intent on what is to be defined? I did ask for adoption of the WG item,
>> i.e. the task to provide a specific functionality rather than a specific
>> document. I do find it most likely that we adopt the referenced
>> document, but step one is to create a milestone.
> 
> I support the milestone.
> 
> I think the draft is going in the right 
> direction, but needs a bit more meat on them 
> bones before I support adoption of this draft to 
> fulfill the milestone. In other words, I don't 
> believe the authors will take a wrong turn when 
> filling in the rest of the draft, but want to 
> have some more there so we know one or more of 
> the existing thin areas doesn't do just that. I 
> believe Colin will do a good job.
> 
> Does this explain where I'm coming from?
> 

Yes, and I think it is a very valid opinion. My plan is if there is
sufficient support for the milestone to go ahead and create it. Then we
can do the adoption of the document at a somewhat later stage. However,
I do intended to be aggressive time wise on this item to ensure that we
have something sufficiently along the publication process that RTCWEB
will not hesitate to normatively reference it in their first set of RTP
usage specification. I do hope the authors will be diligent and
responsive on this draft to make fast progress.

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 csp@csperkins.org  Wed Apr  4 00:36:07 2012
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Mark,

So far, we haven't really considered RTP/AVPF rapid feedback (Section 5 =
of the draft is essentially a placeholder, and the mechanisms in Section =
4 are specifically for baseline RTCP). As you've noticed, it changes the =
dynamics significantly, and we need to think carefully on how the =
various rapid feedback modes interact with the circuit breaker =
conditions before we define anything.

Colin



On 4 Apr 2012, at 02:55, Mark Handley wrote:
> Colin, Varun,
>=20
> Can you clarify how the circuit breaker mechanism would work with RFC
> 4585 extended feedback.  To make things concrete, I'm thinking about
> the following scenario:
>=20
> - Sender is sending 2Mb/s video into a 4Mb/s DSL line.  RTT is 100ms.
> - Receiver is allowed to send up to 2.5% of this as RTCP feedback in
> ACK mode, so that's 6.5 packets per RTT if I got the maths right.
> - A TCP flow slowstarts, doubles the window for the last time and
> collides with the RTP flow in the wonderful way slowstart sometimes
> does.  If you're unlucky, you'll get 33% packet loss for one full RTT
> before TCP backs off.
> - The RTCP receiver sends 6 successive RTCP reports over one RTT
> indicating 33% loss.  If doesn't really matter if ACKs or NACKs are
> sent in this scenario.
>=20
> The calculated TCP rate would be ~18Kb/s for those parameters, so the
> circuit breaker would kick in if I understood the draft correctly
> because that's more than two reporting intervals with excessively high
> loss.
>=20
> This is the main reason for loss event rate in the TFRC specification
> - you avoid responding to such transients too strongly.
>=20
> Cheers,
> Mark
>=20
>=20
>=20
> On 5 March 2012 23:32, Colin Perkins <csp@csperkins.org> wrote:
>> Here's our initial attempt at a "circuit breakers" draft for RTCWeb. =
Comments welcome - this is very much a straw-man for discussion, rather =
than a final solution.
>>=20
>> Colin
>>=20
>>=20
>>=20
>> Begin forwarded message:
>>> From: internet-drafts@ietf.org
>>> Subject: I-D Action: =
draft-perkins-avtcore-rtp-circuit-breakers-00.txt
>>> Date: 5 March 2012 20:17:59 GMT
>>> To: i-d-announce@ietf.org
>>> Reply-To: internet-drafts@ietf.org
>>>=20
>>>=20
>>> A New Internet-Draft is available from the on-line Internet-Drafts =
directories.
>>>=20
>>>       Title           : RTP Congestion Control: Circuit Breakers for =
Unicast Sessions
>>>       Author(s)       : Colin Perkins
>>>                          Varun Singh
>>>       Filename        : =
draft-perkins-avtcore-rtp-circuit-breakers-00.txt
>>>       Pages           : 14
>>>       Date            : 2012-03-05
>>>=20
>>>   The Real-time Transport Protocol (RTP) is widely used for =
telephony,
>>>   video conferencing, and telepresence applications.  These
>>>   applications are often used over best-effort UDP/IP networks.  If
>>>   congestion control is not implemented then network congestion will
>>>   deteriorate the user's multimedia experience.  This document does =
not
>>>   propose a congestion control algorithm.  Instead, it specifies a
>>>   minimal set of "circuit-breakers".  Circuit-breakers are =
conditions
>>>   under which an RTP flow should cease to transmit 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.
>>>=20
>>>=20
>>> A URL for this Internet-Draft is:
>>> =
http://www.ietf.org/internet-drafts/draft-perkins-avtcore-rtp-circuit-brea=
kers-00.txt
>>=20
>> --
>> Colin Perkins
>> http://csperkins.org/
>>=20
>>=20
>>=20
>> _______________________________________________
>> Audio/Video Transport Core Maintenance
>> avt@ietf.org
>> https://www.ietf.org/mailman/listinfo/avt


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




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To: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Cc: "James M. Polk" <jmpolk@cisco.com>, IETF AVTCore WG <avt@ietf.org>
Subject: Re: [AVTCORE] Adopting new work on RTP Congestion Circuit Breakers
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On 4 Apr 2012, at 07:27, Magnus Westerlund wrote:
> On 2012-04-04 02:52, James M. Polk wrote:
>> At 09:00 AM 4/3/2012, Magnus Westerlund wrote:
>>> On 2012-03-30 02:53, James M. Polk wrote:
>>>> While I generally think this draft is
>>>> interesting, even the author admits it's awfully
>>>> drafty at this point (both in AVTCORE and
>>>> TSV-Dir). And I think perhaps too much so in this
>>>> version. In other words, I'd like to see more meat on them bones.
>>> 
>>> James, do you think the draft is so drafty that you don't understand the
>>> intent on what is to be defined? I did ask for adoption of the WG item,
>>> i.e. the task to provide a specific functionality rather than a specific
>>> document. I do find it most likely that we adopt the referenced
>>> document, but step one is to create a milestone.
>> 
>> I support the milestone.
>> 
>> I think the draft is going in the right 
>> direction, but needs a bit more meat on them 
>> bones before I support adoption of this draft to 
>> fulfill the milestone. In other words, I don't 
>> believe the authors will take a wrong turn when 
>> filling in the rest of the draft, but want to 
>> have some more there so we know one or more of 
>> the existing thin areas doesn't do just that. I 
>> believe Colin will do a good job.
>> 
>> Does this explain where I'm coming from?
>> 
> 
> Yes, and I think it is a very valid opinion. My plan is if there is
> sufficient support for the milestone to go ahead and create it. Then we
> can do the adoption of the document at a somewhat later stage. However,
> I do intended to be aggressive time wise on this item to ensure that we
> have something sufficiently along the publication process that RTCWEB
> will not hesitate to normatively reference it in their first set of RTP
> usage specification. I do hope the authors will be diligent and
> responsive on this draft to make fast progress.


I'm hoping to...

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




From vsingh.ietf@gmail.com  Wed Apr  4 06:08:37 2012
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From: Varun Singh <vsingh.ietf@gmail.com>
Date: Wed, 4 Apr 2012 16:08:15 +0300
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Cc: rtp-congestion@alvestrand.no, Mark Handley <M.Handley@cs.ucl.ac.uk>, "avt@ietf.org WG" <avt@ietf.org>
Subject: Re: [AVTCORE] [R-C] I-D Action: draft-perkins-avtcore-rtp-circuit-breakers-00.txt
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Hi Mark,

As Colin pointed out, currently the circuit breakers are defined for
baseline RTCP. Just to clarify: the circuit breaker should give
sufficient time for a rate-control to operate in. For example, the
REMB message should be sent (as a non-compound RTCP) much before the
circuit breaker kicks in and at the same time the circuit breaker
should not make the receiver send the REMB message prematurely.

Varun

On Wed, Apr 4, 2012 at 10:36, Colin Perkins <csp@csperkins.org> wrote:
> Mark,
>
> So far, we haven't really considered RTP/AVPF rapid feedback (Section 5 o=
f the draft is essentially a placeholder, and the mechanisms in Section 4 a=
re specifically for baseline RTCP). As you've noticed, it changes the dynam=
ics significantly, and we need to think carefully on how the various rapid =
feedback modes interact with the circuit breaker conditions before we defin=
e anything.
>
> Colin
>
>
>
> On 4 Apr 2012, at 02:55, Mark Handley wrote:
>> Colin, Varun,
>>
>> Can you clarify how the circuit breaker mechanism would work with RFC
>> 4585 extended feedback. =A0To make things concrete, I'm thinking about
>> the following scenario:
>>
>> - Sender is sending 2Mb/s video into a 4Mb/s DSL line. =A0RTT is 100ms.
>> - Receiver is allowed to send up to 2.5% of this as RTCP feedback in
>> ACK mode, so that's 6.5 packets per RTT if I got the maths right.
>> - A TCP flow slowstarts, doubles the window for the last time and
>> collides with the RTP flow in the wonderful way slowstart sometimes
>> does. =A0If you're unlucky, you'll get 33% packet loss for one full RTT
>> before TCP backs off.
>> - The RTCP receiver sends 6 successive RTCP reports over one RTT
>> indicating 33% loss. =A0If doesn't really matter if ACKs or NACKs are
>> sent in this scenario.
>>
>> The calculated TCP rate would be ~18Kb/s for those parameters, so the
>> circuit breaker would kick in if I understood the draft correctly
>> because that's more than two reporting intervals with excessively high
>> loss.
>>
>> This is the main reason for loss event rate in the TFRC specification
>> - you avoid responding to such transients too strongly.
>>
>> Cheers,
>> Mark
>>
>>
>>
>> On 5 March 2012 23:32, Colin Perkins <csp@csperkins.org> wrote:
>>> Here's our initial attempt at a "circuit breakers" draft for RTCWeb. Co=
mments welcome - this is very much a straw-man for discussion, rather than =
a final solution.
>>>
>>> Colin
>>>
>>>
>>>
>>> Begin forwarded message:
>>>> From: internet-drafts@ietf.org
>>>> Subject: I-D Action: draft-perkins-avtcore-rtp-circuit-breakers-00.txt
>>>> Date: 5 March 2012 20:17:59 GMT
>>>> To: i-d-announce@ietf.org
>>>> Reply-To: internet-drafts@ietf.org
>>>>
>>>>
>>>> A New Internet-Draft is available from the on-line Internet-Drafts dir=
ectories.
>>>>
>>>> =A0 =A0 =A0 Title =A0 =A0 =A0 =A0 =A0 : RTP Congestion Control: Circui=
t Breakers for Unicast Sessions
>>>> =A0 =A0 =A0 Author(s) =A0 =A0 =A0 : Colin Perkins
>>>> =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Varun Singh
>>>> =A0 =A0 =A0 Filename =A0 =A0 =A0 =A0: draft-perkins-avtcore-rtp-circui=
t-breakers-00.txt
>>>> =A0 =A0 =A0 Pages =A0 =A0 =A0 =A0 =A0 : 14
>>>> =A0 =A0 =A0 Date =A0 =A0 =A0 =A0 =A0 =A0: 2012-03-05
>>>>
>>>> =A0 The Real-time Transport Protocol (RTP) is widely used for telephon=
y,
>>>> =A0 video conferencing, and telepresence applications. =A0These
>>>> =A0 applications are often used over best-effort UDP/IP networks. =A0I=
f
>>>> =A0 congestion control is not implemented then network congestion will
>>>> =A0 deteriorate the user's multimedia experience. =A0This document doe=
s not
>>>> =A0 propose a congestion control algorithm. =A0Instead, it specifies a
>>>> =A0 minimal set of "circuit-breakers". =A0Circuit-breakers are conditi=
ons
>>>> =A0 under which an RTP flow should cease to transmit media to protect =
the
>>>> =A0 network from excessive congestion. =A0It is expected that all RTP
>>>> =A0 applications running on best-effort networks will be able to run
>>>> =A0 without triggering these circuit breakers in normal operation.
>>>>
>>>>
>>>> A URL for this Internet-Draft is:
>>>> http://www.ietf.org/internet-drafts/draft-perkins-avtcore-rtp-circuit-=
breakers-00.txt
>>>
>>> --
>>> Colin Perkins
>>> http://csperkins.org/
>>>
>>>
>>>
>>> _______________________________________________
>>> Audio/Video Transport Core Maintenance
>>> avt@ietf.org
>>> https://www.ietf.org/mailman/listinfo/avt
>
>
> --
> Colin Perkins
> http://csperkins.org/
>
>
>
> _______________________________________________
> Rtp-congestion mailing list
> Rtp-congestion@alvestrand.no
> http://www.alvestrand.no/mailman/listinfo/rtp-congestion



--=20
http://www.netlab.tkk.fi/~varun/

From mark.j.handley@gmail.com  Tue Apr  3 18:55:54 2012
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Subject: Re: [AVTCORE] Fwd: I-D Action: draft-perkins-avtcore-rtp-circuit-breakers-00.txt
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Colin, Varun,

Can you clarify how the circuit breaker mechanism would work with RFC
4585 extended feedback.  To make things concrete, I'm thinking about
the following scenario:

 - Sender is sending 2Mb/s video into a 4Mb/s DSL line.  RTT is 100ms.
 - Receiver is allowed to send up to 2.5% of this as RTCP feedback in
ACK mode, so that's 6.5 packets per RTT if I got the maths right.
 - A TCP flow slowstarts, doubles the window for the last time and
collides with the RTP flow in the wonderful way slowstart sometimes
does.  If you're unlucky, you'll get 33% packet loss for one full RTT
before TCP backs off.
 - The RTCP receiver sends 6 successive RTCP reports over one RTT
indicating 33% loss.  If doesn't really matter if ACKs or NACKs are
sent in this scenario.

The calculated TCP rate would be ~18Kb/s for those parameters, so the
circuit breaker would kick in if I understood the draft correctly
because that's more than two reporting intervals with excessively high
loss.

This is the main reason for loss event rate in the TFRC specification
- you avoid responding to such transients too strongly.

Cheers,
Mark



On 5 March 2012 23:32, Colin Perkins <csp@csperkins.org> wrote:
> Here's our initial attempt at a "circuit breakers" draft for RTCWeb. Comm=
ents welcome - this is very much a straw-man for discussion, rather than a =
final solution.
>
> Colin
>
>
>
> Begin forwarded message:
>> From: internet-drafts@ietf.org
>> Subject: I-D Action: draft-perkins-avtcore-rtp-circuit-breakers-00.txt
>> Date: 5 March 2012 20:17:59 GMT
>> To: i-d-announce@ietf.org
>> Reply-To: internet-drafts@ietf.org
>>
>>
>> A New Internet-Draft is available from the on-line Internet-Drafts direc=
tories.
>>
>> =A0 =A0 =A0 Title =A0 =A0 =A0 =A0 =A0 : RTP Congestion Control: Circuit =
Breakers for Unicast Sessions
>> =A0 =A0 =A0 Author(s) =A0 =A0 =A0 : Colin Perkins
>> =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Varun Singh
>> =A0 =A0 =A0 Filename =A0 =A0 =A0 =A0: draft-perkins-avtcore-rtp-circuit-=
breakers-00.txt
>> =A0 =A0 =A0 Pages =A0 =A0 =A0 =A0 =A0 : 14
>> =A0 =A0 =A0 Date =A0 =A0 =A0 =A0 =A0 =A0: 2012-03-05
>>
>> =A0 The Real-time Transport Protocol (RTP) is widely used for telephony,
>> =A0 video conferencing, and telepresence applications. =A0These
>> =A0 applications are often used over best-effort UDP/IP networks. =A0If
>> =A0 congestion control is not implemented then network congestion will
>> =A0 deteriorate the user's multimedia experience. =A0This document does =
not
>> =A0 propose a congestion control algorithm. =A0Instead, it specifies a
>> =A0 minimal set of "circuit-breakers". =A0Circuit-breakers are condition=
s
>> =A0 under which an RTP flow should cease to transmit media to protect th=
e
>> =A0 network from excessive congestion. =A0It is expected that all RTP
>> =A0 applications running on best-effort networks will be able to run
>> =A0 without triggering these circuit breakers in normal operation.
>>
>>
>> A URL for this Internet-Draft is:
>> http://www.ietf.org/internet-drafts/draft-perkins-avtcore-rtp-circuit-br=
eakers-00.txt
>
> --
> Colin Perkins
> http://csperkins.org/
>
>
>
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt

From magnus.westerlund@ericsson.com  Tue Apr 10 23:23:51 2012
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Subject: [AVTCORE] Have you or plan to implement draft-ietf-avtcore-feedback-supression-rtp
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WG,

IESG is asking about implementations or at least plans to implement RTCP
Extension for Third-party Loss Report
(draft-ietf-avtcore-feedback-supression-rtp-16). I had in the write up
simple indicated that I didn't know about any. But I didn't ask the list.

So if you are implementing or plan to implement can you please send a
statement to the list or at least me privately so that I can more
accurately represent the situation. That will hopefully reduce the
concerns from the IESG.

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: [AVTCORE] IESG comment regarding draft-ietf-avtcore-feedback-supression-rtp-16
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WG,

A comment from the IESG (Brian Haberman) said the following regarding
the second paragraph of section 3:

1. The Protocol Overview section states : "Intermediaries in the network
that receive a RTCP TPLR SHOULD NOT send their own additional
Third-Party Loss Report messages for the same packet sequence numbers."
 Why is this not a MUST?  Is it simply to handle intermediate devices
that don't support this function?  If there is another scenario where a
device may send a TPLR that overlaps, it would
be good to spell that out.

The document editor has agreed with the IESG that it appears suitable to
change this SHOULD NOT to a MUST NOT.

Before that goes into the specification that will be approved I want to
give the WG a chance to comment on this change.

Please provide any feedback on this change no later than April the 20th.

Regards

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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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 Maintenanc=
e Working Group of the IETF.

	Title           : RTCP Extension for Third-party Loss Report
	Author(s)       : Qin Wu
                          Frank Xia
                          Roni Even
	Filename        : draft-ietf-avtcore-feedback-supression-rtp-17.txt
	Pages           : 19
	Date            : 2012-04-13

   In a large RTP session using the RTP Control Protocol (RTCP) feedback
   mechanism defined in RFC 4585, a feedback target may experience
   transient overload if some event causes a large number of receivers
   to send feedback at once.  This overload is usually avoided by
   ensuring that feedback reports are forwarded to all receivers,
   allowing them to avoid sending duplicate feedback reports.  However,
   there are cases where it is not recommended to forward feedback
   reports, and this may allow feedback implosion.  This memo discusses
   these cases and defines a new RTCP third-party loss report that can
   be used to inform receivers that the feedback target is aware of some
   loss event, allowing them to suppress feedback.  Associated SDP
   signaling is also defined.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avtcore-feedback-supression-=
rtp-17.txt

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

This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-avtcore-feedback-supression-r=
tp-17.txt


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From magnus.westerlund@ericsson.com  Fri Apr 13 07:20:14 2012
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Subject: Re: [AVTCORE] I-D Action:draft-ietf-avtcore-feedback-supression-rtp-17.txt
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WG,

This version will not be approved for publication by the AD until after
the previous stated dead line of next Friday (20th of April). This way
you have the opportunity to review all the proposed changes resulting
from the IESG review.

Cheers

Magnus Westerlund


On 2012-04-13 11:53, Qin Wu wrote:
> On 13 April , 2012 5:29 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           : RTCP Extension for Third-party Loss Report
>> Author(s)       : Qin Wu
>>                          Frank Xia
>>                          Roni Even
>> Filename        : draft-ietf-avtcore-feedback-supression-rtp-17.txt
>> Pages           : 19
>> Date            : 2012-04-13
>>
>>   In a large RTP session using the RTP Control Protocol (RTCP) feedback
>>   mechanism defined in RFC 4585, a feedback target may experience
>>   transient overload if some event causes a large number of receivers
>>   to send feedback at once.  This overload is usually avoided by
>>   ensuring that feedback reports are forwarded to all receivers,
>>   allowing them to avoid sending duplicate feedback reports.  However,
>>   there are cases where it is not recommended to forward feedback
>>   reports, and this may allow feedback implosion.  This memo discusses
>>   these cases and defines a new RTCP third-party loss report that can
>>   be used to inform receivers that the feedback target is aware of some
>>   loss event, allowing them to suppress feedback.  Associated SDP
>>   signaling is also defined.
>>
>>
>> A URL for this Internet-Draft is:
>> http://www.ietf.org/internet-drafts/draft-ietf-avtcore-feedback-supression-rtp-17.txt
> 
> Here is one minor update to feedback suppression draft based on comments during IETF LC and from Gen-Art Review:
>     o  Remove the discussion of improper dithering in the introduction section.
>     o  Change of the normative language from SHOULD NOT to MUST NOT in
>         the protocol overview section.
>     o  Editorial changes based on Gen-ART review.
>     o  Move Example use case section before Protocol overview section.
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
> 


-- 

Magnus Westerlund

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From magnus.westerlund@ericsson.com  Mon Apr 16 07:45:13 2012
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Subject: Re: [AVTCORE] WG last call on Monitoring Architecture for RTP (draft-ietf-avtcore-monarch-11)
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WGs, Authors

This WG last call has ended. I had wished for even more reviews but at
least four has stated that they have reviewed the document. There has
also been a number of comments on the drafts that should be addressed
prior to publication.

Authors, please start addressing the comments and submit a new version
when ready.

Cheers

Magnus Westerlund

On 2012-03-05 08:44, Magnus Westerlund wrote:
> WG,
> 
> This initiates the Working group last call on Monitoring Architecture
> for RTP (draft-ietf-avtcore-monarch-11) to be published as Informational
> RFC. The WG last call will run for three weeks and end Monday the 26th.
> 
> The draft can be retrieved from here:
> https://datatracker.ietf.org/doc/draft-ietf-avtcore-monarch/
> 
> Please provide feedback if you have read it, even if you have no
> comments or think it is good enough, this is important information in
> judging the support and readiness for publication.
> 
> Thanks
> 
> 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
> 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
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From: "Bill Ver Steeg (versteb)" <versteb@cisco.com>
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Sorry for the tardy input, but I support the topic as a WG item. I do =
agree that the draft is a bit thin in some crucial areas, but it is a =
good start. I will try to find time to contribute to the process on this =
one, as it is quite important work.

In fact, circuit breakers are just the first requirement and we need to =
finish that up and address congestion control in all of its glory, IMHO. =
First things first, though.

Bill VerSteeg

-----Original Message-----
From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of =
Magnus Westerlund
Sent: Thursday, March 29, 2012 7:26 AM
To: IETF AVTCore WG
Subject: [AVTCORE] Adopting new work on RTP Congestion Circuit Breakers

WG,

At the IETF 83 there was a presentation in the AVTCORE and also in the
ICCRG of "RTP Congestion Control: Circuit Breakers for Unicast Sessions"

https://datatracker.ietf.org/doc/draft-perkins-avtcore-rtp-circuit-breake=
rs/

This was well received and there was support for adopting this as WG
item. Thus I would like to confirm on the mailing list that we should
adopt a WG item that works on providing RTP congestion control circuit
breakers for unicast transport flows.

WG, please comment if you support or disagree of such adaptation. The WG
chairs will call a consensus after Friday the 13th.

Cheers

Magnus Westerlund

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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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Date: Tue, 17 Apr 2012 11:21:37 -0600
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From: Kevin Gross <kevin.gross@avanw.com>
To: "Bill Ver Steeg (versteb)" <versteb@cisco.com>
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Cc: Magnus Westerlund <magnus.westerlund@ericsson.com>, IETF AVTCore WG <avt@ietf.org>
Subject: Re: [AVTCORE] Adopting new work on RTP Congestion Circuit Breakers
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Congestion is a topic of interest for me. I will find time to review and
contribute.

Kevin Gross

On Tue, Apr 17, 2012 at 11:11 AM, Bill Ver Steeg (versteb) <
versteb@cisco.com> wrote:

> Sorry for the tardy input, but I support the topic as a WG item. I do
> agree that the draft is a bit thin in some crucial areas, but it is a goo=
d
> start. I will try to find time to contribute to the process on this one, =
as
> it is quite important work.
>
> In fact, circuit breakers are just the first requirement and we need to
> finish that up and address congestion control in all of its glory, IMHO.
> First things first, though.
>
> Bill VerSteeg
>
> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
> Magnus Westerlund
> Sent: Thursday, March 29, 2012 7:26 AM
> To: IETF AVTCore WG
> Subject: [AVTCORE] Adopting new work on RTP Congestion Circuit Breakers
>
> WG,
>
> At the IETF 83 there was a presentation in the AVTCORE and also in the
> ICCRG of "RTP Congestion Control: Circuit Breakers for Unicast Sessions"
>
>
> https://datatracker.ietf.org/doc/draft-perkins-avtcore-rtp-circuit-breake=
rs/
>
> This was well received and there was support for adopting this as WG
> item. Thus I would like to confirm on the mailing list that we should
> adopt a WG item that works on providing RTP congestion control circuit
> breakers for unicast transport flows.
>
> WG, please comment if you support or disagree of such adaptation. The WG
> chairs will call a consensus after Friday the 13th.
>
> 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
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
>

--f46d044268e6d381bb04bde32b9e
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Congestion is a topic of interest for me. I will find time to review and co=
ntribute.<div><br clear=3D"all">Kevin Gross<br><div><br></div><div class=3D=
"gmail_quote">On Tue, Apr 17, 2012 at 11:11 AM, Bill Ver Steeg (versteb) <s=
pan dir=3D"ltr">&lt;<a href=3D"mailto:versteb@cisco.com">versteb@cisco.com<=
/a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Sorry for the tardy input, but I support the=
 topic as a WG item. I do agree that the draft is a bit thin in some crucia=
l areas, but it is a good start. I will try to find time to contribute to t=
he process on this one, as it is quite important work.<br>

<br>
In fact, circuit breakers are just the first requirement and we need to fin=
ish that up and address congestion control in all of its glory, IMHO. First=
 things first, though.<br>
<br>
Bill VerSteeg<br>
<div class=3D"im HOEnZb"><br>
-----Original Message-----<br>
From: <a href=3D"mailto:avt-bounces@ietf.org">avt-bounces@ietf.org</a> [mai=
lto:<a href=3D"mailto:avt-bounces@ietf.org">avt-bounces@ietf.org</a>] On Be=
half Of Magnus Westerlund<br>
Sent: Thursday, March 29, 2012 7:26 AM<br>
To: IETF AVTCore WG<br>
</div><div class=3D"HOEnZb"><div class=3D"h5">Subject: [AVTCORE] Adopting n=
ew work on RTP Congestion Circuit Breakers<br>
<br>
WG,<br>
<br>
At the IETF 83 there was a presentation in the AVTCORE and also in the<br>
ICCRG of &quot;RTP Congestion Control: Circuit Breakers for Unicast Session=
s&quot;<br>
<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-perkins-avtcore-rtp-circu=
it-breakers/" target=3D"_blank">https://datatracker.ietf.org/doc/draft-perk=
ins-avtcore-rtp-circuit-breakers/</a><br>
<br>
This was well received and there was support for adopting this as WG<br>
item. Thus I would like to confirm on the mailing list that we should<br>
adopt a WG item that works on providing RTP congestion control circuit<br>
breakers for unicast transport flows.<br>
<br>
WG, please comment if you support or disagree of such adaptation. The WG<br=
>
chairs will call a consensus after Friday the 13th.<br>
<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<a href=3D"tel:%2B46%=
2010%207148287" value=3D"+46107148287">+46 10 7148287</a><br>
F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile <a href=3D"tel:%2B4=
6%2073%200949079" value=3D"+46730949079">+46 73 0949079</a><br>
SE-164 80 Stockholm, Sweden| mailto: <a href=3D"mailto:magnus.westerlund@er=
icsson.com">magnus.westerlund@ericsson.com</a><br>
----------------------------------------------------------------------<br>
<br>
_______________________________________________<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>
_______________________________________________<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>

--f46d044268e6d381bb04bde32b9e--

From internet-drafts@ietf.org  Tue Apr 17 20:58:35 2012
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Subject: [AVTCORE] I-D Action: draft-ietf-avtcore-monarch-12.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 Maintenanc=
e Working Group of the IETF.

	Title           : Monitoring Architecture for RTP
	Author(s)       : Qin Wu
                          Geoff Hunt
                          Philip Arden
	Filename        : draft-ietf-avtcore-monarch-12.txt
	Pages           : 29
	Date            : 2012-04-17

   This memo proposes an architecture for extending RTP Control Protocol
   (RTCP) with a new RTCP Extended Reports (XR) (RFC3611) block type to
   report new metrics regarding media transmission or reception quality,
   following RTCP guideline established in RFC5968.  This memo suggests
   that a new 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 which 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 which covers their set of concerns.  Where possible, a
   specific block should be designed to be re-usable across more than
   one application, for example, for all of voice, streaming audio and
   video.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avtcore-monarch-12.txt

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

This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-avtcore-monarch-12.txt


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From bill.wu@huawei.com  Tue Apr 17 21:42:39 2012
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From magnus.westerlund@ericsson.com  Sun Apr 22 23:37:03 2012
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Subject: Re: [AVTCORE] I-D Action:draft-ietf-avtcore-feedback-supression-rtp-17.txt
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WG,

I have seen no objections against the changes. Therefore I state that
the WG is satisfied with the editors changes in response to the IESG's
comments.

Robert, from the WG's perspective this is ready for publication.

Cheers

Magnus

On 2012-04-13 16:20, Magnus Westerlund wrote:
> WG,
> 
> This version will not be approved for publication by the AD until after
> the previous stated dead line of next Friday (20th of April). This way
> you have the opportunity to review all the proposed changes resulting
> from the IESG review.
> 
> Cheers
> 
> Magnus Westerlund
> 
> 
> On 2012-04-13 11:53, Qin Wu wrote:
>> On 13 April , 2012 5:29 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           : RTCP Extension for Third-party Loss Report
>>> Author(s)       : Qin Wu
>>>                          Frank Xia
>>>                          Roni Even
>>> Filename        : draft-ietf-avtcore-feedback-supression-rtp-17.txt
>>> Pages           : 19
>>> Date            : 2012-04-13
>>>
>>>   In a large RTP session using the RTP Control Protocol (RTCP) feedback
>>>   mechanism defined in RFC 4585, a feedback target may experience
>>>   transient overload if some event causes a large number of receivers
>>>   to send feedback at once.  This overload is usually avoided by
>>>   ensuring that feedback reports are forwarded to all receivers,
>>>   allowing them to avoid sending duplicate feedback reports.  However,
>>>   there are cases where it is not recommended to forward feedback
>>>   reports, and this may allow feedback implosion.  This memo discusses
>>>   these cases and defines a new RTCP third-party loss report that can
>>>   be used to inform receivers that the feedback target is aware of some
>>>   loss event, allowing them to suppress feedback.  Associated SDP
>>>   signaling is also defined.
>>>
>>>
>>> A URL for this Internet-Draft is:
>>> http://www.ietf.org/internet-drafts/draft-ietf-avtcore-feedback-supression-rtp-17.txt
>>
>> Here is one minor update to feedback suppression draft based on comments during IETF LC and from Gen-Art Review:
>>     o  Remove the discussion of improper dithering in the introduction section.
>>     o  Change of the normative language from SHOULD NOT to MUST NOT in
>>         the protocol overview section.
>>     o  Editorial changes based on Gen-ART review.
>>     o  Move Example use case section before Protocol overview section.
>> _______________________________________________
>> 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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Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
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WG,

Draft minutes for Paris are now available. Please review and provide
feedback.

http://www.ietf.org/proceedings/83/minutes/minutes-83-avtcore.htm

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 iesg-secretary@ietf.org  Mon Apr 23 12:05:00 2012
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Cc: avtcore chair <avtcore-chairs@tools.ietf.org>, avtcore mailing list <avt@ietf.org>, RFC Editor <rfc-editor@rfc-editor.org>
Subject: [AVTCORE] Protocol Action: 'RTCP Extension for Third-party Loss Report' to	Proposed Standard (draft-ietf-avtcore-feedback-supression-rtp-17.txt)
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The IESG has approved the following document:
- 'RTCP Extension for Third-party Loss Report'
  (draft-ietf-avtcore-feedback-supression-rtp-17.txt) as a Proposed
Standard

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

The IESG contact persons are Robert Sparks and Gonzalo Camarillo.

A URL of this Internet Draft is:
http://datatracker.ietf.org/doc/draft-ietf-avtcore-feedback-supression-rtp/




  Technical Summary
        In a large RTP session using the RTCP feedback mechanism defined in RFC
        4585, a feedback target may experience transient overload if some event
        causes a large number of receivers to send feedback at once. This
        overload is usually avoided by ensuring that feedback reports are
        forwarded to all receivers, allowing them to avoid sending duplicate
        feedback reports.  However, there are cases where it is not recommended
        to forward feedback reports, and this may allow feedback implosion. 
        This memo discusses these cases and defines a new RTCP third-party loss
        report that can be used to inform receivers that the feedback target is
        aware of some loss event, allowing them to suppress feedback. 
        Associated SDP signalling is also defined.

    Working Group Summary
        There is strong consensus among an adequate number of WG
        participants on this solution.
       
    Document Quality
    There are not yet any reported implementations. The document
    has had reasonable review. 

Personnel

  Magnus Westerlund is the document shepherd.
  Robert Sparks is the responsible AD


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WG,


I have received a correction from Jonathan Lennox. When trying to upload
the new version I found a bug in the meeting material manager. In trying
to work around it I deleted the online version of minutes. Until a new
version is available online please use the attached one.

Cheers

Magnus

On 2012-04-23 12:57, Magnus Westerlund wrote:
> WG,
> 
> Draft minutes for Paris are now available. Please review and provide
> feedback.
> 
> http://www.ietf.org/proceedings/83/minutes/minutes-83-avtcore.htm
> 
> 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
> 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
----------------------------------------------------------------------

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<p class=MsoTitle><span lang=EN-US style='mso-fareast-font-family:"Times New Roman";
mso-ansi-language:EN-US;mso-fareast-language:SV'>AVTCORE IETF83 Minutes<o:p></o:p></span></p>

</div>

<p class=MsoNormal><span class=GramE><span lang=EN-US style='mso-ansi-language:
EN-US;mso-fareast-language:SV'>Monday, 26<sup>th</sup> of March at 13.00-16.10
2012.</span></span><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'><o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Note Takers: Brian Rosen, Charles Eckel<o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Chairs: Roni Even, Magnus Westerlund<o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Audio recordings:<o:p></o:p></span></p>

<p class=MsoListParagraphCxSpFirst style='text-indent:-18.0pt;mso-list:l3 level1 lfo1'><![if !supportLists]><span
lang=EN-US style='font-family:Symbol;mso-fareast-font-family:Symbol;mso-bidi-font-family:
Symbol;mso-ansi-language:EN-US;mso-fareast-language:SV'><span style='mso-list:
Ignore'>·<span style='font:7.0pt "Times New Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><a
href="http://www.ietf.org/audio/ietf83/ietf83-252b-20120326-1256-pm.mp3"><span
lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:SV'>http://www.ietf.org/audio/ietf83/ietf83-252b-20120326-1256-pm.mp3</span></a><span
lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:SV'><o:p></o:p></span></p>

<p class=MsoListParagraphCxSpLast style='text-indent:-18.0pt;mso-list:l3 level1 lfo1'><![if !supportLists]><span
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</span></span></span><![endif]><a
href="http://www.ietf.org/audio/ietf83/ietf83-252b-20120326-1506-pm.mp3"><span
lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:SV'>http://www.ietf.org/audio/ietf83/ietf83-252b-20120326-1506-pm.mp3</span></a><span
lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:SV'><o:p></o:p></span></p>

<h1><span lang=EN-US style='mso-fareast-font-family:"Times New Roman";
mso-ansi-language:EN-US;mso-fareast-language:SV'>1. AVTCore Status Update
Chairs</span><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'><o:p></o:p></span></h1>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>The WG chairs provided a short status update after having covered the Note
Well and the Agenda. The WG has had one RFC being published, three documents in
IESG processing. <span class=GramE>draft-ietf-avt-srtp-ekt-03</span> have
recently had some discussion on the list in relation to RTCWEB. <span
class=GramE>draft-ietf-avt-srtp-aes-gcm-02</span> will progress after EKT. <span
class=GramE>draft-ietf-avtcore-srtp-encrypted-header-ext-01</span> has finished
WGLC. It will need to address comments before publication. <span class=GramE>draft-ietf-avtcore-monarch-11</span>
is in WGLC. <o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>The WG chairs made a milestone review. There were a number of milestone
dates that was missed. Update proposed with new dates. There have also been
proposals for new milestones, Overview of RTP security solutions, RTP clock
source signaling, and SRTP crypto transform and key management based on ARIA
algorithm.<o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>The chairs also informed the WG about the change in Publication Request
write-up (PROTO). This includes asking the authors about confirmation about IPR
disclosures related to the document. <o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Magnus Westerlund also reminded the WG about the importance of filling in
the blue sheets. One reason is so that we get the right sized room. <o:p></o:p></span></p>

<h1><span lang=EN-US style='mso-fareast-font-family:"Times New Roman";
mso-ansi-language:EN-US;mso-fareast-language:SV'>2. Why RTP Sessions Should Be
Content Neutral <o:p></o:p></span></h1>

<p class=MsoNormal><span class=GramE><span lang=EN-US style='mso-ansi-language:
EN-US;mso-fareast-language:SV'>draft-alvestrand-rtp-sess-neutral-00</span></span><span
lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:SV'><o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Harald Alvestrand presented his personal view around media flows, transport
pairs and their relation to RTP session. Since RTP and SDP were published the <span
class=GramE>amount of bandwidth capacity has gone up significantly and RSVP</span>
and Multicast hasn’t deployed. The network can provide prioritization. But that
is only needed when you actually want prioritization. Harald claims that RFC
3550 is making a leap of faith. <span class=GramE>One being that media types
align with desired prioritization.</span> Section 5.2, bullet 4 (Slide 6) blind
mixing only works for audio conferencing. If one does video one would not like
to do blind mixing. <o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Cullen Jennings asked about the leap of logic, he can’t quite understand
the argument. Harald answered that if you believe the statement, then a mixer
with sufficient <span class=GramE>meta</span> data could mix something nice out
of 3 audio and 2 video streams. So if you don’t do it blindly the statement is
false. Doing it blindly the statement is irrelevant. Colin Perkins followed up
this questions saying that of course one can’t mix blindly. The argument in the
RFC is that at the time of writing the appropriate method of providing the <span
class=GramE>meta</span> data needed was using transport streams. Harald
responded that he thinks that argument is wrong.<o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Harald moved on to Section 5.2 bullet 5 (Slide 7). The prioritization
between media streams are not function of top level media types it is of
desired functionality. Dave Oran asked for a clarification about <span
class=SpellE>Harald’s</span> assumption. Do you think “multiple media” in the
sentence being multiple media types (audio and video) or multiple media
streams? Harald couldn’t make head and tails of that and has assumed multiple
media types. Dave Oran then commented that he and likely many others don’t by
the crux of the argumentation because we are not agreeing with that
interpretation. Harald replied that interpreting it as multiple media streams
would preclude multiple SSRCs in one RTP session which no one has argued
against. Roni Even commented that this list in the RFC 3550 is in the context
of a discussion of multiple media types, not streams. <o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Harald moved on to Corrective Action (Slide 8). Thinks there is no need to
change RTP and also the RTCP isn’t connected to the media type, rather media
stream asymmetry. What <span class=GramE>is</span> needed is SDP corrections.
Those are underway in </span><a
href="https://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-bundle-negotiation/"><span
lang=EN-US style='mso-ansi-language:EN-US'>draft-ietf-mmusic-sdp-bundle-negotiation-00</span></a><span
lang=EN-US style='mso-ansi-language:EN-US'>. <span class=GramE>Now time for
discussion.</span><o:p></o:p></span></p>

<p class=MsoNormal><span class=SpellE><span lang=EN-US style='mso-ansi-language:
EN-US'>Tholis</span></span><span lang=EN-US style='mso-ansi-language:EN-US'>
Eckert? <span class=GramE>did</span> you establish and arguments why life would
be better with RTP with what you want? Harald responded that the reason this
was started was that in Google when they needed to establish four transport
flows there was a high failure rate. This got much better by deploying RTP and
RTCP multiplexing when only two flows where needed. Now we know what the cost
is and therefore we want the possibility to go to one flow when it is
appropriate. <o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Colin Perkins stated that have we had this conversation 15 years ago I
might have actually agreed with everything you said. Unfortunately we have a
fair amount of history with RTP and SDP. RFC 3550 is not the only thing
impacted by this. This can work in some cases including the one you are
interested in. There exist serious problems in other use cases. You appear to ignore
these other cases. Need to document when it works and when it doesn't not only
change the SHOULD in RFC 3550.<span style='mso-spacerun:yes'>  </span>There are
drafts that say where it's a problem. Harald responded that one level I do
agree with what Colin said. The reason I went ahead with this is that there are
cases which are argued against by this quote. This change of permission can’t
render any existing application unusable. Colin asked, who is arguing against
it, only seen recent arguments that it doesn’t work in all cases. Harald, then
we can finish bashing in this open door and document this. <o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Cullen Jennings wondered what you are asking for. According to the slide we
are on track to do things with all the issues that need work. Hadriel Kaplan sarcastically
commented that if that one use case is sufficient for a solution, while all
others need to do things generic. But stepping back, </span><a
href="https://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-bundle-negotiation/"><span
lang=EN-US style='mso-ansi-language:EN-US'>draft-ietf-mmusic-sdp-bundle-negotiation-00</span></a><span
lang=EN-US style='mso-ansi-language:EN-US'> is far from done. Also, this is a
proposal of changing RTP which is what you want. Magnus Westerlund has another
proposal of how to do multiple RTP sessions over a single transport (Solution
B) which would be an alternative and we can do that. And Cullen commented that
I don’t see the problem. Hadriel responded that the advantage of multiple RTP
sessions over one <span class=GramE>transport <span
style='mso-spacerun:yes'> </span>is</span> that it support other use cases.
Which also don’t want to create multiple transport flows but have multiple RTP <span
class=GramE>sessions.</span> <span class=GramE>For example, to simplify
transcoding of the audio and not the video.</span> Just add a sixth tuple so
that one can separate them inside of the five tuple used today. Cullen
responded that he wants to send the same RTP packets I send today on a single
transport. Hadriel responded that this is actually not the same packet you sent
yesterday. For example if using SRTP with a single RTP session there will be a
single key for all media streams. Cullen responded that this is not in his
slides. Cullen stated that Harald is arguing that all is fine, nothing needs to
be done. Cullen doesn’t think that is what Harald really wants.</span><span
lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:SV'><o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Roni Even (as WG chair) commented that one need to understand when it works
and the limitations are. In the end what is important is that one understands
when to use a single RTP session and when there is a need for multiple
sessions. Hadriel asked if Roni thinks both solutions are needed. Roni
responded that even if only one is needed it needs to be documented with its
limitations. Colin Perkins commented that his proposal is that we do A and B
and write yet another document to explain the trade-offs. Hadriel asked why
would we do both if B solves all the cases and A has <span class=GramE>limitations?</span>
Colin agreed with that, but many don’t see these limitations as serious. Roni
commented that B doesn’t use the same RTP as today. Colin concluded that we
have been arguing about this for months and we are not agreeing. Thus let’s do
both and document the trade-offs. <o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Hadriel asked the proponents for A if they really cared if it is A or B,
doesn’t you just want the hammer? Is the two or one byte header that big a
deal? Harald commented that on an architectural level we should not continue to
enforce an unsound architectural choice. Desire that we do get the consensus of
the AVTCORE WG so that we don’t need to have this discussion anymore. As a <span
class=SpellE>deployer</span> I want to put audio and video in the same session
without getting grief from AVTCORE. Hadriel asked as if you really care about
one or multiple RTP sessions. Harald answered one due to lines of code. Hadriel
responded that you anyway need to support multiple RTP sessions and thus
already have the code. To add a session multiplexing layer would mean more
lines of code. Hadriel responded that he believes that in the end it would be <span
class=GramE>less lines</span> of codes. Today you should have no code and
supporting a multiplexing layer is; if supported, then read two bytes and send
here. Harald responded that don’t you think we have implemented single session
solution after such a long argument? Cullen Jennings commented that he doesn’t
think this is a valid argument. We are arguing over 5 lines of code, when we
are dumping many thousands of lines of code into the browsers.<o:p></o:p></span></p>

<p class=MsoNormal><span class=GramE><span lang=EN-US style='mso-ansi-language:
EN-US;mso-fareast-language:SV'>Moving on to the next topic.</span></span><span
lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:SV'> <o:p></o:p></span></p>

<h1><span lang=EN-US style='mso-fareast-font-family:"Times New Roman";
mso-ansi-language:EN-US;mso-fareast-language:SV'>3. RTP Multiplexing
Architecture - Magnus Westerlund<o:p></o:p></span></h1>

<p class=MsoNormal><span class=GramE><span lang=EN-US style='mso-ansi-language:
EN-US;mso-fareast-language:SV'>draft-westerlund-avtcore-multiplex-architecture-01</span></span><span
lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:SV'><o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Was skipped, to allow for continued discussion around single transport
multiplexing, i.e. next topic. <o:p></o:p></span></p>

<h1><span lang=EN-US style='mso-fareast-font-family:"Times New Roman";
mso-ansi-language:EN-US;mso-fareast-language:SV'>4. Multiple RTP Session on a
Single Lower-Layer Transport - Magnus Westerlund<o:p></o:p></span></h1>

<p class=MsoNormal><span class=GramE><span lang=EN-US style='mso-ansi-language:
EN-US;mso-fareast-language:SV'>draft-westerlund-avtcore-transport-multiplexing-02</span></span><span
lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:SV'><o:p></o:p></span></p>

<p class=MsoNormal><span class=GramE><span lang=EN-US style='mso-ansi-language:
EN-US;mso-fareast-language:SV'>Starting time 49 minutes into first audio
recording.</span></span><span lang=EN-US style='mso-ansi-language:EN-US;
mso-fareast-language:SV'><o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Magnus Westerlund (as individual) went into a presentation intended to
drive discussion and achieve the goal of knowing which work should be done.
Therefore there is quick check pretty soon and unless clear consensus exists we
will go into the details. On slide 4 the options are presented; 1) Define a
mechanism that enables multiple RTP sessions over a single lower layer
transport. 2) Clarify the RTP specification and operation of a single RTP session
with multiple media types in it. 3) Do both 1) and 2). 4) Do nothing.<o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Cullen Jennings commented that he didn't get the slide (4). Is option 1
include the option of adding a mux header, and the second using something like
payload type to differentiate the media types. Magnus responded yes, the main
goal he has is to have full RTP sessions with full RTP functionality. Cullen
responded that he doesn’t know what an RTP session is. Magnus clarified that
one really important function for option 1) is to have multiple RTP sessions,
each having their full 32-bit SSRC spaces available for usage, including having
the same SSRC value being used for different media streams in both session,
which is not possible in option 2. Roni commented that this is part of the
discussion in the multiplexing architecture document (draft-westerlund-avtcore-multiplex-architecture-01).
Colin Perkins commented that the proposal in draft-westerlund-avtcore-transport-multiplexing-02
is to add a <span class=SpellE>demultiplexing</span> shim to realize option 1.
The second option is basically to do what Harald proposes. Cullen asked<span
class=GramE>,</span> can you clarify what Harald is proposing? Magnus suggested
that we should go through the details of the proposed work on slide 5.<o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Magnus presented his and Colin Perkins proposal to do option 3), i.e. doing
both option 1) and 2). The details of what needs to be worked on are described
on the slide. Cullen Jennings wonder what happened to the bundle proposal, i.e.
take a bunch of stuff that are on different m= lines in the SDP and allow them
to run on a single 5-tuple? Magnus answered that this is <span class=SpellE>Harald’s</span>,
i.e. option 2. Cullen disagreed that there is a difference. Hadriel Kaplan
supported Magnus view. Cullen followed up asking if there are two video cameras
are they two m= lines? Magnus responded that it is possible but not necessarily
and that has not been the main discussion point. The main discussion has been
around having both audio and video in an RTP session. Cullen wants to get
clarification on an easy thing, what about two video cameras. Magnus responded
that this is not part of the discussion at all. Cullen responded that this what
he got was different in the neutral discussion compared to how bundle currently
stands. Magnus responded that he thinks the issue Cullen is after is how you
represent in SDP when one end-point <span class=GramE>have</span> multiple
media sources. This issue hasn’t been much discussed at all. Cullen commented
that he thought this is what was discussed when started down the bundle path.
Magnus responded that he at least wasn’t discussing this issue. Magnus tried to
conclude that the question of how you represent multiple media sources in an
RTP session in SDP will still be there independent of the choices and proposals
in this slide deck. Colin Perkins commented that it is important to separate
the signaling issue from the RTP level issues. People have been putting
multiple cameras in the same RTP session for 15 years. We might argue how you
signal this but people have been doing this for a long time. <o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Colin Perkins reiterated the options. 1) We have the concept of RTP
sessions and find them useful and like to keep them when transporting them over
a single transport flow (5-tuple). There is also use cases for putting things
together in the same RTP session that previously <span class=GramE>were<span
style='mso-spacerun:yes'>  </span>not</span>, i.e. option 2), which some find
useful and can make it work, like WebRTC. We also need a document that says
what the issues and considerations are. <o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Harald stated that when he has been discussing bundle he hasn’t been
discussing if an m= line should represent one, two or fifty sources of the same
type. Magnus commented that it is good that at least <span class=SpellE>Harald</span>
and he have been discussing the same thing.<span style='mso-spacerun:yes'> 
</span><o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Richard <span class=SpellE>Azack</span> asked why can’t we have multiple
unique sets of payload type and use these to determine which RTP session the
packet belongs to when sending multiple RTP sessions over the same 5-tuple.
Magnus answered that in most implementations you start at port level, the
payload type indicates encoding, which would force one to use Port + PT as
determination of RTP session. Colin Perkins commented that Richard should read draft-westerlund-avtcore-multiplex-architecture-01
for extensive discussion of why it wouldn’t work.<o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Magnus proposed that we do a consensus check to see if the discussion so
far has accomplished anything. Roni Even is the one that judges the consensus,
if any, and he shouldn’t call it unless it is very clear. Magnus iterated the
choices. Cullen Jennings commented he doesn’t know what any of these questions
mean. That we want to work on something around multiplexing is the only one
that is clear and <span class=GramE>have</span> been for several meetings. Roni
Even clarified that all Magnus attempts to determine is how to go forward.
Cullen Jennings stated a proposal for the WG. Let’s take the requirements
people have expressed and analyze the drafts against these. Stop talking about
sessions and transport and answer simple questions such as; can I send multiple
video cameras over one UDP port? Can I offer multiple codecs for the same
camera and switch between them without additional signaling? Can I run all my
stuff on one port, two ports, <span class=GramE>N</span> ports. Cullen couldn’t
answer these questions for any of the proposals. Those should be clear to go
forward. Colin Perkins answered that the answer to the first two questions is
yes and have been for the last 15 years. The answer to the third one is what we
try to decide. Roni added that both of the solution options can resolve
question three, but we are trying to determine if we should do either or both. <o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Hadriel Kaplan commented that he hasn’t seen the advantages disadvantages
between the options. From his perspective option 1 has advantages; don't know what
they are for 2). Are we going have any slides on the advantages disadvantages?
Are we voting without knowing the pro and cons? Magnus stated that he can
present what the limitations are one the single RTP session, which is part of
the slide deck. Hadriel commented that the only downside he knows of for 1) is
the overhead (one or two bytes). That is what you are seeing on the wire
commented Roni. From the point of the receiver it is two different streams.
Which Hadriel commented is exactly what he has today. They are just <span
class=SpellE>demultiplexing</span> <span class=GramE>this</span> two bytes in,
which is why it is simple, following the KISS principle. Roni responded that we
today have both the single (multiplex inside) and the multi-session (multiplex
on transport) <span class=GramE>cases,</span> the only limitation that exist is
the SHOULD NOT in RFC 3550 about audio and video in the same session. Hadriel
responded that this is the big rub isn’t it? There are reasons, and what Harald
is saying is that these may not be applicable to his use case. And this is only
<span class=GramE>a should</span>. So people for which <span class=GramE>a
single RTP session with multiple media types are</span> applicable can do it.
Thus we would not need to develop what Magnus is proposing. However, there are still
people thinking that it is needed… Hadriel commented that there are clear
advantages to Magnus’ proposal. Hadriel is flabbergasted that people doesn’t
understand how easy it is to add two bytes. The attribute issues we discussed
in MMUSIC this morning. With Magnus proposal these all goes away as each m=
line is still a separate RTP session. Roni responded that this is a question
for MMUSIC. There is the question Cullen raised if one describe each SSRC as an
individual m= line or have multiple SSRCs in a given m= line. Hadriel responded
there aspects that matter in this group. For example by having separate RTP sessions
there is separation of the RTCP reports also. Thus the reports are orthogonal
and independent between the RTP sessions. In the single RTP session case the
reports needs to be about different media types and one issue in fact the
payload type relation on what one reports on. You are mixing oil and water and <span
class=GramE>tries</span> to get mayonnaise.<o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Jonathan Lennox commented first that a lot of <span class=GramE>the
disconnect</span> between Colin and Cullen is two models. The first one is the
way RTP works in the head of the ones that designed RTP, meaning that you can
have up to 2^32 SSRCs in an RTP session. Then it is the second model in which
RTP works, the one being used by SIP which is one SSRC in each direction, end
of the story. The one understanding the second gets confused over the first
one. The ones understanding the first one gets confused over when people make
statements following the second one, as the response from the first one is that
of course they can do that. Jonathan prefers the first model, it has many
benefits, unfortunately there is lot of SDP is designed according to the second
model which is broken. This is MMUSIC’s problem. A second comment was to
Hadriel, that these two bytes are only two bytes and having to rewrite anything
that needs to look at the packet on the wire, like Wireshark and Deep Packet
Inspection. It is mostly a question of who does the work; the people doing the
media plane, or the ones doing the signaling plane. Hadriel clear do not want
it to be him. Jonathan clearly <span class=GramE>don’t</span> want it to be him
either.<o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Paul <span class=SpellE>Kyzivat</span> commented that he doesn’t know
enough to choose between the two options. <span class=GramE>But is concerned
over doing both.</span> If you do it both you will have lots more
interoperability. One side <span class=GramE>do</span> one and signal that and
the other side do the other, we will have a mess. Magnus responded that <span
class=GramE>a class of applications actually have</span> to agree on which
method they are going to use to avoid that issue. An application or class of
applications must determine if they can accept the limitations of a single RTP
session or actually need multiple RTP sessions. RTP is used in a wide variety
of applications. Unfortunately a lot of them aren’t represented in this room.
Paul commented that you assume that there are applications that are firewalled
from each other? Magnus responded <span class=SpellE>thatyes</span> there are
classes, like sensor networks or sensor data transport. <o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Cullen Jennings commented he would like to try to up level the discussion.
Colin appear frustrated by the “this is signaling problem take it over there”
and Cullen himself are frustrated that when he takes it to MMUSIC he is told to
take it AVTCORE. And in AVTCORE he is told to take it to MMUSIC. And we are
both the same people. RTP clearly is used in different context, and Cullen is
interested in usage inside the RFC 3264 SDP Offer/Answer usage of RTP. <span
style='mso-spacerun:yes'> </span>In this discussion we come up on cases that
are non-compatible. It might be SDP O/A that is wrong, but independently that
is irrelevant. We must consider what is built and deployed and what we need to
interoperate with. That is the hard part of SDP and how to make this work
together. Cullen wants to see clear descriptions of how we are going to
interoperate with existing implementations. Magnus asked if what you are
suggesting is to continue working on both tracks to get the real details of
both proposals. Roni commented that the choices and what is need to be
addressed in MMUSIC will be different depending on if you do only Multiple RTP
sessions over a single transport or if you do multiple media types in a single
RTP session or both. If we can come to a decision it would make it easier in
MMUSIC. Magnus commented that if we follow the line Hadriel is arguing doing
only Multiple RTP sessions over single transport then life in MMUSIC is likely
easier. If we do both solutions, the choice between the solutions will be out
of our hands, it will be in other places where the choice is made between the
solutions. When it comes to “Multiple Media types in a single session” one way
of looking at it is that it has really been SDP that enforced the SHOULD NOT,
and as bundle is happening the limitation has already been released and it will
happen. Personally I think it is doing both or only Multiple RTP sessions over
a single transport. Doing nothing would in fact mean that we need to go kill
bundle in MMUSIC so that there is no way of signaling multiple media types for
a single RTP session.<o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Hadriel Kaplan followed up on Jonathan regarding Wireshark. Wireshark will
have to be changed independent on which we do. Jonathan responded that this is
not true as multiple SSRCs are already supported. Hadriel is that really true,
can it really play back both audio and video in that single transport flow? <span
class=GramE>[Confused partially off the <span class=SpellE>mic</span>
comments].</span> Roni commented he thinks what we trying to accomplish is to
have everything working that is currently within RFC 3550 and in addition allow
multiple RTP sessions being transported on the same flow. Hadriel responded
that isn’t true you are trying to have multiple media types in a single RTP
session. Roni disagreed. Magnus supported <span class=SpellE>Hadriels</span>
view and went on to comment. There are two different things but they are
brought together by the commonality in enabling multiple media types over a
single transport flow. Hadriel followed up saying that there are two choices
for solving that problem. We <span class=GramE>either have a single RTP session
with multiple media types and</span> get all the issues in MMUSIC or we take
what we have today and get an extra <span class=SpellE>muxing</span> a few
bytes in, a 6-tuple instead of 5-tuple everything else stays the same. Of
course it is going to be simpler for me, why do you think I am up at the <span
class=SpellE>mic.</span> It will also be useful in other contexts, like SIP.
Jonathan Lennox commented that this could be used in SIP. Hadriel responded,
not realistically assuming SBCs. Hadriel went on to comments Magnus previous
comments as being confusing. Of course we should always try to do only one
thing? Roni commented that the previous decision in RTCWEB was to start with
the single RTP session and then develop the multi session. <span class=GramE>Hadriel,</span>
responded that this is not their decision to make, they have to follow the
decision made in AVTCORE and MMUSIC. If some vendors want to do it, then it is
a proprietary solution. If I am held to the fire and have to come to this group
and get things through then they will have to do the same. Cullen Jennings
commented that it wasn’t some random group that decided what Roni said, it was
this WG. Magnus stated that <span class=GramE>minutes for Quebec IETF says</span>
that we are to do both. It was Magnus that brought up the question of doing
only one solution in Taipei, otherwise we would be further along the track to
do both. Hadriel clarified that my statement that both will be in the
marketplace anyway doesn’t matter, we IETF can decide which are standardized
solutions and we can decided to do only one.<span style='mso-spacerun:yes'> 
</span><span style='mso-spacerun:yes'> </span><o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Roni commented that in terms of RTP nothing prevents you from having
multiple media types in a single RTP session. It is an SDP issue. What we can
do in this WG is to capture what it means to do that. The text in RFC 3550 is
that yes you can do it but be careful due to the following reasons. <span
class=SpellE>Roni’s</span> interpretation of what Magnus says is that AVTCORE
should document this. For the issues of doing it in SDP we need to discuss that
and see if it works. There might be other signaling protocols where this works.<o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Harald commented with his process <span class=GramE>lawyers</span> hat that
WGs don’t get to override other WGs decisions. IESG is the ones that get to
override. If RTCWEB makes a decision that is incompatible with AVTCOREs then we
have some ADs with a headache. <span class=SpellE>Harald’s</span> personal
architect hat comments that only doing Multiple RTP sessions over a single
transport does solves <span class=SpellE>Hadriel’s</span> transport
multiplexing, not <span class=SpellE>Harald’s</span> architectural issues. The
decision of top level media type is still stupid and should be fixed no matter
the cost. The <span class=SpellE>implementor</span> hat is that we will get
something that works. Colin responded, surely you are not proposing that we fix
every stupid IETF decision<o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Jonathan Lennox commented that we either can make RTP a bit uglier to work
around SDP or we can make SDP uglier to work with RTP. Following Alice's
Restaurant principal that one big pile of garbage is always better than two
smaller piles of garbage, Jonathan thinks we should put the garbage on top of
the existing garbage and make SDP uglier. <o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Jonathan’s second comment was that he <span class=GramE>don’t</span> get
the cases where multiple media types in one single RTP session doesn’t work.
Either they haven’t been well explained or they don’t apply to real use cases.
I am happy to do both as long as the use cases are good enough and someone else
is happy to do the work. Magnus commented that he can present the two biggest
limitations.<o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Magnus explained slide 10 pressing on that one want to establish a
communication session with multiple participants or have a point to point
session that goes through a media gateway. Assume that one client C and the
central point are supporting and enables using multiple media types in a single
RTP session. On the far side of the central point you have one or more clients
not capable of using multiple media types in the same RTP session, instead they
fall back to using different RTP sessions over independent transports. Thus you
have SSRCs that comes from different RTP sessions that suddenly need to
function in the context of a single RTP session without collisions. If they are
colliding they force the need to do SSRC translation. SSRC translation is
costly as it forces re-encryption of the content and forces the translator to
be aware of all RTCP extensions being used in the RTP session to guarantee
correct translation. Roni commented that it is <span class=GramE>an</span>
Mixer so it shouldn’t be a problem. Magnus answered that this is primarily for
nodes that don’t do full processing of the media, such as transport
translators, media gateways or RTP mixers that switches between sources.
Jonathan commented that if these are offer/answers you can’t do that anyway
(see later presentation). Roni also commented that he agrees with Jonathan if
you are talking about RTP mixers they by definition present a different SSRC
space to each leg. But Magnus are correct for a gateway. Magnus responded that
in RTCWEB it appears that the limitation Jonathan talks about isn’t going to
exist due to JSEP. Cullen Jennings asked why you can’t rely on the SSRC
collision <span class=GramE>mechanism?</span> Magnus responded that it might
not work if client <span class=GramE>A</span> uses the same SSRC on both RTP
sessions, which is allowed as they are different sessions, then it will not see
a collision. It is the merger of the RTP sessions that causes the collision.
Colin Perkins also commented that there exists signaling that overrides the
collision detection rules in RTP. Cullen responded that as Jonathan said if
people are doing offer/answer this is not a problem. Colin responded that it is
a problem if one does a=<span class=SpellE>ssrc</span> and the other isn’t, <span
class=GramE>then</span> you have a problem.<o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US'>Richard <span
class=SpellE>Ejzak</span></span><span lang=EN-US style='mso-ansi-language:EN-US;
mso-fareast-language:SV'> asked if there is negotiation between the client and
the central node then it is clear if shall be used or not. Magnus responded
that then it becomes a deployment problem. If one want to use a single RTP
session you have to forklift upgrade your clients. The reason is that you
negotiate the transport individually between the client and the central node. The
goal is to agree locally if you can use it or not. Otherwise you will need to
know that all clients that may join this particular session will support using
a single transport.<o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Roni Even commented that we are running out of time and should try to give
some guidance on how to proceed. Magnus brought up the Options slide and asked
if anyone is a supporting the option to do nothing. No one supported that.
Cullen commented that his understanding of the previous agreement is to do
multiple media types in a single RTP sessions and then continue working on
enabling multiple RTP sessions over a single transport. There is too little
clarity to make a decision.<o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Justin <span class=SpellE>Uberti</span> commented that he thought that we
had agreed that we have one proposal “Multiple Media Types in a single session”
that works for a set of interesting use case. We can clear work on another
solution that works for all use cases. But I thought we had agreed on doing “Multiple
Media Types in a single session”. Magnus commented that he thinks that both
Justin and Cullen in fact are arguing for the Proposal on slide 5. Justin
responded that the agreement was for multiple SSRCs and a single port and
potentially multiple media types. Colin Perkins commented that what he
remembers we go forward with the proposal for multiple media types in a single
RTP session. We also bring forward a proposal for multiple RTP sessions over a
single transport. Which is what we are doing and we are saying that we should
go forward with both of them. <o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Roni Even stated that we can’t guarantee that we will in the end be able to
support all of them. After <span class=SpellE>AVTcore</span> WG makes a
decision, and described the RTP aspects we will have to specify how to use them
in SDP. And that will not happen here. We can only describe how to do both
using RTP, not the signaling.<o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Jonathan Lennox commented that with the exception of Hadriel we appear to
have some consensus for doing Option 3 (both) module the understanding of what
three is, including continuing with individual submission for Multiple RTP
sessions over a single transport. <o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Roni Even stated that for the single session that you will need to expand
the architecture document to describe when it can be used without bias for
Multiple RTP sessions. The Multiple RTP sessions needs a separate draft. Lennox
commented that it clearly needs normative work. Single RTP session needs to
remove <span class=GramE>the should</span> and some descriptive work. Roni
followed up that means it is an update to RFC 3550 in the end. Magnus stated
that his understanding is that the people are saying doing both independently.
There was proposed that the main proponents of different views would meet and
figure out the document structure and who does the work? Roni asked if anyone
was <span class=GramE>objecting?</span> Documents will be not adopted at this <span
class=GramE>stage,</span> we first need to figure out the right set of
documents. We are to progress with the assumption that we will update RFC 3550
in regards to the SHOULD and also continue with draft-westerlund-avtcoretransport-multiplexing-02.
That is the direction we are going to take. Brian Rosen asked what about the
part that describes single RTP session with multiple media types needs to
discuss its limitations. Roni responded certainly. Roni concluded that we
appear to have a consensus of how to progress. We also need discuss with MMUSIC
how to deal with the SDP aspects. <o:p></o:p></span></p>

<h1><span lang=EN-US style='mso-fareast-font-family:"Times New Roman";
mso-ansi-language:EN-US;mso-fareast-language:SV'>5. RTP Congestion Control:
Circuit Breakers for Unicast Sessions - Colin Perkins<o:p></o:p></span></h1>

<p class=MsoNormal><span class=GramE><span lang=EN-US style='mso-ansi-language:
EN-US;mso-fareast-language:SV'>draft-perkins-avtcore-rtp-circuit-breakers-00</span></span><span
lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:SV'><o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Starts at 1:45:00 in the first audio recording<o:p></o:p></span></p>

<p class=MsoNormal><span class=GramE><span lang=EN-US style='mso-ansi-language:
EN-US;mso-fareast-language:SV'>Colin Perkins presenting.</span></span><span
lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:SV'> The
motivation is that RTCWEB is talking about congestion control, there <span
class=GramE>are some proposal</span> but it is a difficult problem. This works
is about conditions where if encountered everyone in this room would agree is
problematic and therefore should stop. Anyone using congestion control would
always be within these limits. <o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Colin will also be talking about this in ICCRG’s second session on Tuesday.<o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>The base assumption is RFC 3550 RTCP information. Future work will look
into refining <span class=GramE>this using RTP extensions</span>. Using
RTT/Jitter every 5 seconds is too seldom. Packet loss statistics every 5
seconds is not good for congestion control but for overload detection is
probably good enough. Two circuit breakers are proposed. The first one is
timeout, no increased RTCP extended sequence number for two reporting intervals.
The second breaker is if one gets 10 times the TCP throughput equation says an
equivalent TCP connection would get. <o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>There are open issues that need to be discussed. Include the assumption for
the circuit breakers and if they in themselves are appropriate. The second
major thing is around using any available RTP extensions. <o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Cullen Jennings commented that both look great. They are easy,
implementable exactly what we need for circuit breakers. Think we (Cisco) tried
this in our Telepresence equipment. Think that a single packet loss could
result in that HD rates were not possible any more. Colin responded that it
might be correct as the model is standard TCP and does not do high rates very
well. But it is something we need to look into. <o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Flemming Andreasen asked if this assumes RTP packet losses are <span
class=GramE>congestion?</span> Colin responded yes, and clearly this is not
always the case. If one can’t do the assumption how do we do a circuit breaker
at all? Flemming responded that might mean that this isn’t applicable in all
cases. <span class=GramE>Needs to be clear on this.</span> <o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Harald Alvestrand asked if Colin have looked at some example numbers <span
class=GramE>yet?</span> Colin responded no, not yet. Harald had a second
question regarding the packet size, appears odd that we would get lower
throughput if we use smaller packets. Colin responded that there was the
reduced size work for TFRC and we should consider taking that into account.
Harald proposed that we could use the MTU for that. Colin responded that this
is a slightly bigger issue and there was discussion for DCCP around this. <span
class=GramE>Will look into this.</span> Harald commented that it probably
doesn’t matter much if we are aiming at hitting within a factor of 10 for this.
Magnus commented that the question comes down to if one is byte or packet
congested. Colin followed up that the things that use small packets are
commonly <span class=GramE>audio</span> which is lower bit-rate than video and
thus less likely to cause congestion. <o:p></o:p></span></p>

<p class=MsoNormal><span class=SpellE><span lang=EN-US style='mso-ansi-language:
EN-US;mso-fareast-language:SV'>Tholios</span></span><span lang=EN-US
style='mso-ansi-language:EN-US;mso-fareast-language:SV'> Eckert? <span
class=GramE>asked</span> what is the magic with 10 times the throughput? Colin
responded that it is number picked out of thin air. <span class=SpellE>Tholios</span>
asked if it the fairness argument, that RTP can do 10 times what TCP could over
the same path. Colin responded that the proposal is not a congestion control
algorithm it is a stop you are breaking things algorithm. The factor 10 is a
number that makes it clear that it is significant more than what TCP gets and
thus a clear sign that something is wrong. <span class=SpellE>Tholios</span>
commented that 10 times is so unfair that you should give up. Colin stated that
10 <span class=GramE>is</span> not set in stone it is a starting point for the
discussion.<o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>James Polk stated that he had sent an email to the RTCWEB <span
class=SpellE>Congstion</span> list (</span><a
href="mailto:rtp-congestion@alvestrand.no"><span lang=EN-US style='mso-ansi-language:
EN-US;mso-fareast-language:SV'>rtp-congestion@alvestrand.no</span></a><span
lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:SV'>) and <span
class=GramE>wonders if that was</span> the right list to provide feedback.
Colin <span class=GramE>think</span> that list is the right one. He will reply
to the question which had to do if the solution is for best effort only, and
what if <span class=GramE>is</span> not best effort, and multiple DSCP being
used?<o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Magnus Westerlund (as chair) stated that this works appear to be very
relevant for this WG. It will likely end up as part of congestion control
discussions in a number of RTP extensions etc. This will be the outer limit on
what can be done, and hopefully in the future we will see more fair, better etc
algorithms. Colin responded that what is important is that we agree on that
this type of draft is what we need to do. Then we can argue what the right
content is. James Polk commented that if the transport area is spinning up <span
class=GramE>a RTP</span> congestion WG then this work will belong there. Magnus
commented that until that happens AVTCORE appear a good home. The goal with
this was to produce something quickly that RTCWEB could reference. We hope that
the future WG will produce something that we more generally can reference in
these cases. James asked Harald if he is motivated to get the WG spun up by <span
class=GramE>Vancouver?</span> Harald did acknowledge that.<o:p></o:p></span></p>

<h1><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:SV'>6.
RTP Topology Considerations for Offer/Answer-Initiated Sessions - Jonathan
Lennox<o:p></o:p></span></h1>

<p class=MsoNormal><span class=GramE><span lang=EN-US style='mso-ansi-language:
EN-US;mso-fareast-language:SV'>draft-lennox-avtcore-rtp-topo-offer-answer-00</span></span><span
lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:SV'><o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Presentation starts 1:59:00 into the first audio recording.<o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Jonathan Lennox presenting RTP Topology considerations for Offer/Answer
initiated sessions. Offer/Answer (O/A) has two modes, one for multicast addresses
and another for unicast (the more common). Jonathan has analyzed the topologies
in RFC 5117 and all except one can be assembled from Offer/Answer unicast, the
exception is <span class=SpellE>Topo</span>-Transport-Translator. The <span
class=SpellE>Topo</span>-Transport-Translator is basically a relay of packets
from one participant to all the others. This basically behaves like a multicast
session would. For example all participants need to have a single view of the
session. The rest of the topologies do some level of modification to the media,
RTP packets or RTCP reporting. In essence the <span class=SpellE>middlebox</span>
ensures that a given participant can understand what it sends from the other
participants.<o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>So why doesn’t <span class=SpellE>Topo</span>-Transport-Translator work
with O/A? Where the topology requires a common session configuration, unicast
O/A allows each end-point to establish its own view of the session. Some
examples from the core of SDP O/A are 1) Bandwidth. When you send an Offer you
express the bandwidth you want to receive. When the answerer sends an answer it
expresses what it want to receive. These values are used to determine what the
RTCP bandwidth is using the 5% rule. If the two end-points express
significantly different values you end up with different RTCP reporting
intervals and timeout values. In a session where a common view needs to be
established, no-one can tell what the common value is to be as there <span
class=GramE>exist</span> no obvious answer. 2) Media Types. Participants remove
media types (Payload Types) from an offer in the answer or updated Offer. An
answer can even renumber the payload type numbers used for a given
configuration. 3) There exist additional parameters such as <span class=SpellE>ptime</span>
and many SDP extensions that will have issues. <o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>The above is a good thing, as you know that you in fact are only talking to
one other party. That may be forwarding or rewriting other participant’s media,
but you can optimize for this case. Thus you can make optimizations which
aren’t possible if you blindly are talking to a large group of participants.
Colin Perkins commented that everything except transport translators and
multicast. Jonathan agreed as multicast do have special rules in SDP O/A and
can detect that. It doesn’t know about transport translators. <o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>When you have asymmetric bandwidth the optimization you can do is to have
the peer use 5% of my receiver bandwidth towards me, and I will use 5% of your
receiver bandwidth towards you. Timeout is calculated accordingly. We also
resolve the issue that if A uses one SSRC and B 5 SSRCs, then B adds another
SSRC<span class=GramE>,</span> then A’s bandwidth is reduced, despite it not
being shared with any other SSRC in this direction. They are not rivals for the
same bitrate capacity.<o:p></o:p></span></p>

<p class=MsoNormal><span class=GramE><span lang=EN-US style='mso-ansi-language:
EN-US;mso-fareast-language:SV'>RTCP receiver reports.</span></span><span
lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:SV'> In a RTP
session all receivers sends reports on all senders. In a many to many session
this goes towards number of reception reports being quadratic to the number of
session members. If you are actually talking to the same person they are
basically useless. Colin Perkins asked for clarification that they will be
stacked up in the same packet. Jonathan agreed, but when you are on the order
of 150 participants, they can’t be stacked up into the same packet due to MTU.
Colin commented that in that case you are round-robin between the SSRC on which
you report. Jonathan responded that starts to slow things down. The point is
that the <span class=SpellE>middlebox</span> is consuming most of the RTCP
reports itself. So the current rule uses up RTCP bandwidth that could otherwise
be used for more useful and timely things. If you have multiple sources coming
from the same end-point, having them report on each other is completely
useless. <o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Normative recommendations for unicast SDP offer/answer<o:p></o:p></span></p>

<p class=MsoListParagraphCxSpFirst style='text-indent:-18.0pt;mso-list:l0 level1 lfo2'><![if !supportLists]><span
lang=EN-US style='font-family:Symbol;mso-fareast-font-family:Symbol;mso-bidi-font-family:
Symbol;mso-ansi-language:EN-US;mso-fareast-language:SV'><span style='mso-list:
Ignore'>·<span style='font:7.0pt "Times New Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span lang=EN-US style='mso-ansi-language:EN-US;
mso-fareast-language:SV'>RTCP bandwidth and timeouts MUST be calculated
independently in each direction<o:p></o:p></span></p>

<p class=MsoListParagraphCxSpMiddle style='text-indent:-18.0pt;mso-list:l0 level1 lfo2'><![if !supportLists]><span
lang=EN-US style='font-family:Symbol;mso-fareast-font-family:Symbol;mso-bidi-font-family:
Symbol;mso-ansi-language:EN-US;mso-fareast-language:SV'><span style='mso-list:
Ignore'>·<span style='font:7.0pt "Times New Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span lang=EN-US style='mso-ansi-language:EN-US;
mso-fareast-language:SV'>endpoints SHOULD NOT send reports from their own
sources<o:p></o:p></span></p>

<p class=MsoListParagraphCxSpLast style='text-indent:-18.0pt;mso-list:l0 level1 lfo2'><![if !supportLists]><span
lang=EN-US style='font-family:Symbol;mso-fareast-font-family:Symbol;mso-bidi-font-family:
Symbol;mso-ansi-language:EN-US;mso-fareast-language:SV'><span style='mso-list:
Ignore'>·<span style='font:7.0pt "Times New Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span lang=EN-US style='mso-ansi-language:EN-US;
mso-fareast-language:SV'>endpoints SHOULD pick single reporting source to send
reception reports<o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Way forward:<o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Colin Perkins agrees that there is an issue with RTCP bandwidth and
asymmetric paths. We have known about this a long time but no one has been
annoyed sufficiently to do something about it. For the other two points, he
believes that the end-points must send reports. The whole point of the
transport translator is that it behaves as a multicast group, so don't try to
change it; if you do not want this, use another topology (e.g. terminate RTP
session on the mixer). Jonathan responded that the point is that I know I am
using one of the other models. Explicitly then use one of the other topologies
so that you don’t see all the other SSRCs and don’t have to report on them.
Don’t change the definition of RTP so it breaks. Jonathan responded that even
if you are terminating RTP sessions on the mixer but are sending a lot of
sources in each terminated session you still have the same problem. If I have
150 SSRCs in a session and you’re the peer with 150 SSRCs we do have 150^2
reporting blocks. Colin, but then use an RTP mixer so that you don’t have as
many SSRCs. Jonathan responded, we are not using <span class=GramE>an</span>
mixer, I am exaggerating a bit, but we are seeing this problem. Colin Perkins
responded that having built such a system 15 years ago without any issue what
is the problem? Bandwidth is cheap. Jonathan responded that he don’t like
complexities that are <span class=GramE>O(</span>N^2). But it ends up being <span
class=GramE>wasted/stealing</span> from more important things. Colin responded,
<span class=GramE>then</span> configure your bandwidth fraction sensibly.
Jonathan responded that he can’t send his feedback in a timely and frequent
fashion. Colin responded<span class=GramE>,</span> you have 150 reporting
intervals, send them on any of them. Use AVPF, don’t see the problem.<o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Joerg Ott commented that you might want to engineer this off-line. This
looks interesting but don’t know if it is the right suggestions. Joerg has
looked at more complex overlay structures using RTCP point to point in
segments. Jonathan responded that he wants to know more about <span
class=SpellE>Joerg’s</span> use case. <o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Magnus Westerlund (as individual) thinks the problem should be split up. We
should at least clarify how entities with multiple sources should behave. The
second two bullets (Slide 15) apply to all the topologies. Jonathan responded
that there is a difference between transport translator and multicast and the
rest. Magnus followed up that one of his main concerns is that we shouldn’t
have different behaviors depending on topology. If one clarifies how any
end-point with multiple SSRCs <span class=GramE>are</span> to behave it will
apply to all topologies. Magnus also supported looking into the asymmetric
bandwidth, but the WG should target generic solutions that aren’t specific to
offer/answer. Please remember that we have other usages of RTP that aren’t
established O/A signaling. Jonathan asked if they apply asymmetric bandwidth.
Magnus responded yes. Jonathan commented that he think in that case you can’t
do a transport translator. Magnus thinks dividing the proposals up into
independent pieces which can each be discussed on their own merits is the way
forward.<o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Harald Alvestrand asked if the last two bullets on slide 15 are
clarifications or changes. Jonathan responded that it is probably a question in
debate. Jonathan has a way of constructing it so that it is within the spec if
you assume a weird internal topology. Harald responded that he has worked with
an engineer trying to interpret RFC 3550 rules and several times thought they
can’t mean that. So please clarify. Jonathan followed up that the bullets can
be interpreted so that the local SSRCs can’t see each other and therefore can’t
report on each other. The issue is that if the other side looks at this, it can
think the network is broken as 2/3 of the sources can’t see me. If they know
you are doing this it is not a problem. Colin commented that is the issue.
There are devices that will think this is broken. Jonathan asked<span
class=GramE>,</span> are they devices that can use offer/answer. Colin
responded that he don’t think O/A is the right thing to think about. Joerg Ott
commented that you need to clarify what you think your co-located sources are.
There is work on SIP end-point decomposition ongoing, so you don’t quite know
how well the thing you suppose is sitting next to you really can talk to you or
not. Please keep this in mind when you go through this.<o:p></o:p></span></p>

<h1><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:SV'>7.
Multipath RTP <br>
Varun Singh, presented by Joerg Ott<o:p></o:p></span></h1>

<p class=MsoNormal><span class=GramE><span lang=EN-US style='mso-ansi-language:
EN-US;mso-fareast-language:SV'>draft-singh-avtcore-mprtp-04</span></span><span
lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:SV'><o:p></o:p></span></p>

<p class=MsoNormal><span class=GramE><span lang=EN-US style='mso-ansi-language:
EN-US;mso-fareast-language:SV'>Starts 19 min into the second audio recording.</span></span><span
lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:SV'><o:p></o:p></span></p>

<p class=MsoNormal><span class=GramE><span lang=EN-US style='mso-ansi-language:
EN-US;mso-fareast-language:SV'>Joerg Ott presenting Multipath RTP proposal.</span></span><span
lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:SV'> The authors
have received quite a lot of comments, mostly about signaling.<span
style='mso-spacerun:yes'>  </span><span class=GramE>Recap and summary of
updates since -02.</span> <o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Two mechanisms to convey interface information; an RTCP based <span
class=SpellE>inband</span> and a SDP based out of band. SDP takes precedence
over RTP. <o:p></o:p></span></p>

<p class=MsoNormal><span class=GramE><span lang=EN-US style='mso-ansi-language:
EN-US;mso-fareast-language:SV'>Implemented multipath to test a bandwidth
distribution algorithm.</span></span><span lang=EN-US style='mso-ansi-language:
EN-US;mso-fareast-language:SV'><span style='mso-spacerun:yes'>  </span>Results
of single path vs. multiple paths, if enough capacity, both <span class=GramE>work</span>
well. Similarly; as loss increases multipath helps.<o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Next steps<o:p></o:p></span></p>

<p class=MsoListParagraphCxSpFirst style='text-indent:-18.0pt;mso-list:l2 level1 lfo4'><![if !supportLists]><span
lang=EN-US style='mso-ascii-font-family:Calibri;mso-fareast-font-family:Calibri;
mso-hansi-font-family:Calibri;mso-bidi-font-family:Calibri;mso-ansi-language:
EN-US;mso-fareast-language:SV'><span style='mso-list:Ignore'>-<span
style='font:7.0pt "Times New Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span lang=EN-US style='mso-ansi-language:EN-US;
mso-fareast-language:SV'>Turn into WG item?<o:p></o:p></span></p>

<p class=MsoListParagraphCxSpMiddle style='text-indent:-18.0pt;mso-list:l2 level1 lfo4'><![if !supportLists]><span
lang=EN-US style='mso-ascii-font-family:Calibri;mso-fareast-font-family:Calibri;
mso-hansi-font-family:Calibri;mso-bidi-font-family:Calibri;mso-ansi-language:
EN-US;mso-fareast-language:SV'><span style='mso-list:Ignore'>-<span
style='font:7.0pt "Times New Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span lang=EN-US style='mso-ansi-language:EN-US;
mso-fareast-language:SV'>Add security considerations.<o:p></o:p></span></p>

<p class=MsoListParagraphCxSpLast style='text-indent:-18.0pt;mso-list:l2 level1 lfo4'><![if !supportLists]><span
lang=EN-US style='mso-ascii-font-family:Calibri;mso-fareast-font-family:Calibri;
mso-hansi-font-family:Calibri;mso-bidi-font-family:Calibri;mso-ansi-language:
EN-US;mso-fareast-language:SV'><span style='mso-list:Ignore'>-<span
style='font:7.0pt "Times New Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span lang=EN-US style='mso-ansi-language:EN-US;
mso-fareast-language:SV'>Double-check with mmusic.<o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Currently the draft is targeted for experimental RFC. <o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Roni as Chair: Is the AVTCORE WG interested in a WG item working on an
experimental multipath solution? Jonathan Lennox commented this is more forward
looking than most care about immediately, but will be useful in near future. <span
class=GramE>Inclined to support this work.</span> Joerg Ott one additional
target deployments are multicast overlays for streaming is one place. Magnus,
as Chair: not strong support, but will check on list as well. As long as
someone drives it is necessary, but we also need some from the community that
review and comment on the work.<o:p></o:p></span></p>

<h1><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:SV'>8.
RTP Clock Source Signaling<span style='mso-tab-count:1'> </span><br>
Aidan Williams<o:p></o:p></span></h1>

<p class=MsoNormal><span class=GramE><span lang=EN-US style='mso-ansi-language:
EN-US;mso-fareast-language:SV'>draft-williams-avtcore-clksrc-00</span></span><span
lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:SV'><o:p></o:p></span></p>

<p class=MsoNormal><span class=GramE><span lang=EN-US style='mso-ansi-language:
EN-US;mso-fareast-language:SV'>Starts 28 minutes into second audio recording.</span></span><span
lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:SV'><o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Aidan Williams presents the new draft on RTP Clock Source signaling. The
draft is a result of discussion at IETF 82. Several people had interest in this
so making it a general building block is a good idea.<span
style='mso-spacerun:yes'>  </span>IDMS and AVB sync both had SDP signaling for
clock source, combined into this document. Need mechanism to provide explicit
SDP signaling of NTP timestamp clock source.<o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Motivations are that a single reference clock used to create NTP timestamps
in RTP protocols in <span class=GramE>a</span> end-point. No indication
currently of which clock is being used. Use cases include social <span
class=SpellE><span class=GramE>tv</span></span>, video walls, networked audio,
and sensor arrays. <o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Using SDP to let end-points describe what clock source they are using. IP
addresses, EUI-<span class=GramE>64, …</span> Definition of equivalent clocks
is also given. Can describe various clock types, NTP, 1588 family, GPS, local.
There is also a concept of Traceable clock. If you can track it back to a
global time source. <o:p></o:p></span></p>

<p class=MsoNormal><span class=GramE><span lang=EN-US style='mso-ansi-language:
EN-US;mso-fareast-language:SV'>Going through the example on Slide 7.</span></span><span
lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:SV'> Ted Hardie
asking <span class=GramE>If</span> you have NTP <span class=SpellE>attr</span>
on one m-line, shouldn't it be on all of them, i.e. session level? NTP is after
all a system level time source. Aidan responded that this is an artificial
example in slide. Commonly you would have one for the whole session. Ted
followed up that he wanted to be certain that a receiver of the SDP would not
make a decision on the m= lines based on time source as the same time source
would be available on to all m= lines. Aidan responded that for a given system
yes. We want to be able to detect systems that aren’t using the same clock
source or master clock. Ted commented that in this emitter case are you
presuming that if I have a video source it may have a different time source
that the general system time source? Aidan confirmed that could occur. Jonathan
Lennox commented that in the case of disaggregated media you might not actually
want different clock source and should actually raise an alarm. Ted asked if
there is a method for interrogating the sources of disaggregated media for
their time source. Aidan responded that he was not aware of any standard
mechanisms. <o:p></o:p></span></p>

<p class=MsoNormal><span class=SpellE><span lang=EN-US style='mso-ansi-language:
EN-US;mso-fareast-language:SV'>Toerless</span></span><span lang=EN-US
style='mso-ansi-language:EN-US;mso-fareast-language:SV'> asked if you expect it
to become reference for any upcoming work. This works appears to allow
end-points to determine if they are synchronized or not in a binary fashion,
possibly even how desynchronized. Is it up to these end-points what they can do
with <span class=GramE>that.</span> Aidan responded yes, that is what it
provides. IDMS is the most immediate use case. Aidan clarified that he himself
is primarily in the synchronized audio playback case. <span class=SpellE>Toerless</span>
asked if the end-points will have well defined behavior depending on if they
are synchronized or not. Aidan responded yes, for a synchronized audio playback
a device would probably mute, but this will be application dependent. <span
class=SpellE>Toerless</span> asked if this actually help if one only has a
local <span class=GramE>clock?</span> Can I calculate the drift and use that
somehow? Aidan responded that what really helps is the knowledge if the sender
and receiver have a synchronized time base or not. <o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Magnus as Chair asked the WG if there is interested in working on a
solution within IETF in this WG. There was some interest indicated through
raised hands. The reason asking the question in very general terms was that the
chairs needs to discuss with Ads and MMUSIC WG chairs if it is appropriate to
take the work on. <o:p></o:p></span></p>

<h1><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:SV'>9.
RTCP for inter-destination media synchronization<br>
Ray van Brandenburg<o:p></o:p></span></h1>

<p class=MsoNormal><span class=GramE><span lang=EN-US style='mso-ansi-language:
EN-US;mso-fareast-language:SV'>draft-ietf-avtcore-idms-03</span></span><span
lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:SV'><o:p></o:p></span></p>

<p class=MsoNormal><span class=GramE><span lang=EN-US style='mso-ansi-language:
EN-US;mso-fareast-language:SV'>Starts 43 min into second audio recording.</span></span><span
lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:SV'><o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Ray van Brandenburg provided a short status update. Using synchronized
clocks with IDMS improves performance but is not necessary. Clock source signaling
has been moved out as an optional but normative reference. Risk that this
delays IDMS. Other changes are clarified terminology, dealing with leap
seconds, and general editorial clean up. Think it is ready for WG last call.<o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Colin Perkins asked if the leap second issue is IDMS specific or a general clarification
to RFC 3550. Ray responded that it is applicable to any use cases requiring
tight synchronization. It could be a clarification on RFC 3550. Currently there
is a general remark in the clock source draft and a more specific one in IDMS. <span
class=GramE>So based on your comment we might need an even more general for RFC
3550.</span> Colin responded<span class=GramE>,</span> that yes if this will
affect others than IDMS then making it a very short draft updating RFC 3550
that IDMS can reference. Ray agreed to look into creating such a draft. <o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Glen Zorn was curious why reference to clock source draft needs to be
normative? Ray responded that his interpretation of the normative vs.
informative guidance is clear that an optimal but performance improving part
should be normative reference. Glen <span class=GramE>commented,</span> that
the clock synchronization is optional. Magnus Westerlund (as chair) stated that
even for optional parts, if you do need to understand the reference to
implement it, it is still a normative reference. Glen was skeptical clearly
normative if you need to understand it to implement it and be compliant. But in
this case it is optional and thus you don’t need it to be compliant. Magnus
responded that it depends on the use case and an implementer may actually have
to include this optional part. <o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Roni Even concluding as chair; First thing is to write leap second as
separate draft applicable to RFC 3550. Secondly, regarding the WG last call we
can go ahead without clock source but the final publication will be held until
the clock source is available. <o:p></o:p></span></p>

<h1><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:SV'>10.
Duplicating RTP Streams<span style='mso-tab-count:1'>   </span><br>
Ali Begen<o:p></o:p></span></h1>

<p class=MsoNormal><span class=GramE><span lang=EN-US style='mso-ansi-language:
EN-US;mso-fareast-language:SV'>draft-begen-avtcore-rtp-duplication-01</span></span><span
lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:SV'><o:p></o:p></span></p>

<p class=MsoNormal><span class=GramE><span lang=EN-US style='mso-ansi-language:
EN-US;mso-fareast-language:SV'>Starts 51 min into the second audio recording.</span></span><span
lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:SV'><o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Ali Begen presented this individual proposal which last was presented in
Taipei. Duplication of RTP stream and RTCP reports is not well specified, this draft
addresses that need. Temporal (time shifted) redundancy use case. <span
class=GramE>The SDP using SSRC grouping to indicate the duplication.</span>
When doing spatial redundancy the SDP uses media stream (m=) grouping, plus use
of <span class=SpellE>cname</span> and <span class=SpellE>srcname</span> (</span><a
href="https://datatracker.ietf.org/doc/draft-westerlund-avtext-rtcp-sdes-srcname/"><span
lang=EN-US style='mso-ansi-language:EN-US'>draft-westerlund-avtext-rtcp-sdes-srcname-00</span></a><span
lang=EN-US style='mso-ansi-language:EN-US'>)</span><span lang=EN-US
style='mso-ansi-language:EN-US;mso-fareast-language:SV'> to indicate which
SSRCs in the different RTP sessions that are duplication of each other. This is
necessary if there is more than one SSRC from a given <span class=SpellE>cname</span>.<o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>One remaining issue, if stream merging happens inside network element, how
to deal with it. Propose treating as mixer. Roni Even commented (as individual)
that it appears to be the same issue as for the AVTEXT WG item </span><a
href="https://datatracker.ietf.org/doc/draft-ietf-avtext-splicing-for-rtp/"><span
lang=EN-US style='mso-ansi-language:EN-US'>draft-ietf-avtext-splicing-for-rtp-07</span></a><span
lang=EN-US style='mso-ansi-language:EN-US'>. Colin Perkins commented that both
a mixer and translator would work, there is slightly different trade-offs. The
Mixer makes it visible and lets you report and the translator makes it
invisible and don’t lets you report. Jonathan Lennox commented that it appears
what is happing is that one takes two streams and mixes them. Unless you
deliberately try to hide this use a mixer. </span><span lang=EN-US
style='mso-ansi-language:EN-US;mso-fareast-language:SV'>Roni (as individual) asked
for clarification that when Ali said merge it resulting stream will include
packets from both streams. A mixer will result in only one outgoing SSRC. <o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:
SV'>Ali asked for adoption of this draft in AVTCORE WG. They are already
working on getting the signaling part adopted in MMUSIC. Magnus Westerlund (as
chair) asked Ali if he believes that the push back <span class=GramE>raised</span>
previously has been resolved about the utility of duplication? Ali responded
that the push back has been around why you wouldn’t use FEC instead and those
have been well explained in the draft. Roni also commented that another issue
was around congested paths. Ali commented that this is not the intended usage
to address congested paths. Magnus called for a hum of who are interested in
this as a WG item. A fair amount of people supporting a WG item, none against, the
chairs will take to list.<o:p></o:p></span></p>

<h1><span lang=EN-US style='mso-ansi-language:EN-US;mso-fareast-language:SV'>End
of meeting.<o:p></o:p></span></h1>

<p class=MsoNormal><span lang=EN-US style='mso-ansi-language:EN-US'><o:p>&nbsp;</o:p></span></p>

</div>

</body>

</html>

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WG, Colin and Charles

This version intends to address the WG last call comments. I would
recommend that people review the changes. I will give people a week to
review this. Please provide any feedback on the changes no later than
1st of May.

Cheers

Magnus

On 2012-04-18 05:58, 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           : Monitoring Architecture for RTP
> 	Author(s)       : Qin Wu
>                           Geoff Hunt
>                           Philip Arden
> 	Filename        : draft-ietf-avtcore-monarch-12.txt
> 	Pages           : 29
> 	Date            : 2012-04-17
> 
>    This memo proposes an architecture for extending RTP Control Protocol
>    (RTCP) with a new RTCP Extended Reports (XR) (RFC3611) block type to
>    report new metrics regarding media transmission or reception quality,
>    following RTCP guideline established in RFC5968.  This memo suggests
>    that a new 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 which 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 which covers their set of concerns.  Where possible, a
>    specific block should be designed to be re-usable across more than
>    one application, for example, for all of voice, streaming audio and
>    video.
> 
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-avtcore-monarch-12.txt
> 
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
> 
> This Internet-Draft can be retrieved at:
> ftp://ftp.ietf.org/internet-drafts/draft-ietf-avtcore-monarch-12.txt
> 
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
> 


-- 

Magnus Westerlund

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Multimedia Technologies, Ericsson Research EAB/TVM
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From magnus.westerlund@ericsson.com  Tue Apr 24 06:18:18 2012
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Date: Tue, 24 Apr 2012 15:17:46 +0200
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [AVTCORE] I-D Action: draft-ietf-avtcore-monarch-12.txt
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Authors,

I have reviewed the latest changes and do wonder if the proposed
formulations really are the correct one.

Full diff:
http://tools.ietf.org/rfcdiff?url2=draft-ietf-avtcore-monarch-12

Section 5.2:
Report blocks that require such correlation need to include the payload
type of the reported media, and must signal details and parameters of
the payload format to which that payload type is bound.

I think my grief with the above sentence is the second half. I don't
find it appropriate that each report block include details and
parameters of the the payload format. I think the approriate level is to
include the payload type and use the functions you anyway have for
associating signalling with media session capture to store the actual
parameter?

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 ron.even.tlv@gmail.com  Tue Apr 24 13:03:21 2012
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>, <avt@ietf.org>
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Date: Tue, 24 Apr 2012 23:01:34 +0300
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Subject: Re: [AVTCORE] I-D Action: draft-ietf-avtcore-monarch-12.txt
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Hi Magnus,
I think that one reason for having the information in the RTCP report =
may be
for RTCP monitors who just see the RTCP messages and not the signaling.
Anyhow it is relevant to codec metrics and not to the transport metrics
Roni

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
> Magnus Westerlund
> Sent: Tuesday, April 24, 2012 4:18 PM
> To: avt@ietf.org
> Subject: Re: [AVTCORE] I-D Action: draft-ietf-avtcore-monarch-12.txt
>=20
> Authors,
>=20
> I have reviewed the latest changes and do wonder if the proposed
> formulations really are the correct one.
>=20
> Full diff:
> http://tools.ietf.org/rfcdiff?url2=3Ddraft-ietf-avtcore-monarch-12
>=20
> Section 5.2:
> Report blocks that require such correlation need to include the =
payload
> type of the reported media, and must signal details and parameters of
> the payload format to which that payload type is bound.
>=20
> I think my grief with the above sentence is the second half. I don't
> find it appropriate that each report block include details and
> parameters of the the payload format. I think the approriate level is
> to include the payload type and use the functions you anyway have for
> associating signalling with media session capture to store the actual
> parameter?
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
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From bill.wu@huawei.com  Tue Apr 24 19:35:02 2012
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Magnus,

On 24 Apr 2012, at 14:17, Magnus Westerlund wrote:
> I have reviewed the latest changes and do wonder if the proposed
> formulations really are the correct one.
>=20
> Full diff:
> http://tools.ietf.org/rfcdiff?url2=3Ddraft-ietf-avtcore-monarch-12
>=20
> Section 5.2:
> Report blocks that require such correlation need to include the =
payload
> type of the reported media, and must signal details and parameters of
> the payload format to which that payload type is bound.
>=20
> I think my grief with the above sentence is the second half. I don't
> find it appropriate that each report block include details and
> parameters of the the payload format. I think the approriate level is =
to
> include the payload type and use the functions you anyway have for
> associating signalling with media session capture to store the actual
> parameter?


I don't interpret that text as requiring the report blocks to include =
details of the parameters of the payload format. Rather, those =
parameters need to be signalled. Perhaps a better phrasing would be:

"Report blocks that require such correlation need to include the payload =
type of the reported media. In addition, it is necessary to signal the =
details and parameters of the payload format to which that payload type =
is bound using some out-of-band means (e.g., as part of an SDP =
offer/answer exchange)."
=20


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




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Magnus,

I'm happy with the changes (although you might want to include a =
clarification to Section 5.2 along the lines that I just proposed).

Colin



On 24 Apr 2012, at 14:13, Magnus Westerlund wrote:
> WG, Colin and Charles
>=20
> This version intends to address the WG last call comments. I would
> recommend that people review the changes. I will give people a week to
> review this. Please provide any feedback on the changes no later than
> 1st of May.
>=20
> Cheers
>=20
> Magnus
>=20
> On 2012-04-18 05:58, internet-drafts@ietf.org wrote:
>>=20
>> 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           : Monitoring Architecture for RTP
>> 	Author(s)       : Qin Wu
>>                          Geoff Hunt
>>                          Philip Arden
>> 	Filename        : draft-ietf-avtcore-monarch-12.txt
>> 	Pages           : 29
>> 	Date            : 2012-04-17
>>=20
>>   This memo proposes an architecture for extending RTP Control =
Protocol
>>   (RTCP) with a new RTCP Extended Reports (XR) (RFC3611) block type =
to
>>   report new metrics regarding media transmission or reception =
quality,
>>   following RTCP guideline established in RFC5968.  This memo =
suggests
>>   that a new 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 which 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 which covers their set of concerns.  Where possible, a
>>   specific block should be designed to be re-usable across more than
>>   one application, for example, for all of voice, streaming audio and
>>   video.
>>=20
>>=20
>> A URL for this Internet-Draft is:
>> http://www.ietf.org/internet-drafts/draft-ietf-avtcore-monarch-12.txt
>>=20
>> Internet-Drafts are also available by anonymous FTP at:
>> ftp://ftp.ietf.org/internet-drafts/
>>=20
>> This Internet-Draft can be retrieved at:
>> ftp://ftp.ietf.org/internet-drafts/draft-ietf-avtcore-monarch-12.txt
>>=20
>> _______________________________________________
>> Audio/Video Transport Core Maintenance
>> avt@ietf.org
>> https://www.ietf.org/mailman/listinfo/avt
>>=20
>=20
>=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


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




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Subject: Re: [AVTCORE] I-D Action: draft-ietf-avtcore-monarch-12.txt
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On 2012-04-25 11:45, Colin Perkins wrote:
> Magnus,
> 
> On 24 Apr 2012, at 14:17, Magnus Westerlund wrote:
>> I have reviewed the latest changes and do wonder if the proposed 
>> formulations really are the correct one.
>> 
>> Full diff: 
>> http://tools.ietf.org/rfcdiff?url2=draft-ietf-avtcore-monarch-12
>> 
>> Section 5.2: Report blocks that require such correlation need to
>> include the payload type of the reported media, and must signal
>> details and parameters of the payload format to which that payload
>> type is bound.
>> 
>> I think my grief with the above sentence is the second half. I
>> don't find it appropriate that each report block include details
>> and parameters of the the payload format. I think the approriate
>> level is to include the payload type and use the functions you
>> anyway have for associating signalling with media session capture
>> to store the actual parameter?
> 
> 
> I don't interpret that text as requiring the report blocks to include
> details of the parameters of the payload format. Rather, those
> parameters need to be signalled. Perhaps a better phrasing would be:
> 
> "Report blocks that require such correlation need to include the
> payload type of the reported media. In addition, it is necessary to
> signal the details and parameters of the payload format to which that
> payload type is bound using some out-of-band means (e.g., as part of
> an SDP offer/answer exchange)."
> 

The proposed text is much clearer on the signaling being out of band in
relation to RTCP and that is how I personally think it should be.

Authors, are you also fine with the text proposed by Colin?

cheers

Magnus Westerlund

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From magnus.westerlund@ericsson.com  Wed Apr 25 07:46:57 2012
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Date: Wed, 25 Apr 2012 16:46:37 +0200
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: [AVTCORE] Adopting a WG item for RTP stream duplication?
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WG,

In Paris we took a hum on adopting a WG item for RTP Stream Duplication.
The problem this tries to address is described in
https://datatracker.ietf.org/doc/draft-begen-avtcore-rtp-duplication/

The result in the meeting room was minuted as: A fair amount of people
supporting a WG item, none against.

See more about this item (Agenda Item 10) in the minutes
http://www.ietf.org/proceedings/83/minutes/minutes-83-avtcore.htm

Thus I would like to ask the rest of the WG if they support or oppose
the creation of a WG item for defining how to best perform RTP stream
Duplication.

Cheers

Magnus Westerlund
(WG Chair)

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WG,

(As individual)

We have updated our individual draft for the maximum SSRC SDP signaling.
The main change is that the fallback policy has been changed from a
single SSRC to being application specific.

Feedback much appreciated.

Cheers

Magnus Westerlund

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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.

	Title           : Multiple Synchronization sources (SSRC) in RTP Session S=
ignaling
	Author(s)       : Magnus Westerlund
                          Bo Burman
                          Fredrik Jansson
	Filename        : draft-westerlund-avtcore-max-ssrc-01.txt
	Pages           : 13
	Date            : 2012-04-26

   RTP has always been a protocol that supports multiple participants
   each sending their own media streams in an RTP session.
   Unfortunately many implementations are designed only for point to
   point voice over IP with a single source in each end-point.  Even
   client implementations aimed at video conferences have often been
   built with the assumption around central mixers that only deliver a
   single media stream per media type.  Thus any application that wants
   to allow for more advance usage where multiple media streams are sent
   and received by an end-point has an issue with legacy
   implementations.  This document describes the problem and proposes a
   signalling solution for how to use multiple SSRCs within one RTP
   session and at the same time handle the legacy issues.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-westerlund-avtcore-max-ssrc-01.tx=
t

Internet-Drafts are also available by anonymous FTP at:
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This Internet-Draft can be retrieved at:
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The IETF datatracker page for this Internet-Draft is:
https://datatracker.ietf.org/doc/draft-westerlund-avtcore-max-ssrc/

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From eckelcu@cisco.com  Fri Apr 27 11:21:51 2012
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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>, <avt@ietf.org>, "Colin Perkins" <csp@csperkins.org>
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Subject: Re: [AVTCORE] I-D Action: draft-ietf-avtcore-monarch-12.txt
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The changes addressed all my comments, with the exception of the =
following typo:

> 13) Section 5.3,
> r/even if though/even though

Note that section 5.3 is not section 5.4, but the same typo exists.

Cheers,
Charles

> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Tuesday, April 24, 2012 6:14 AM
> To: avt@ietf.org; Colin Perkins; Charles Eckel (eckelcu)
> Subject: Re: [AVTCORE] I-D Action: draft-ietf-avtcore-monarch-12.txt
>=20
> WG, Colin and Charles
>=20
> This version intends to address the WG last call comments. I would
> recommend that people review the changes. I will give people a week to
> review this. Please provide any feedback on the changes no later than
> 1st of May.
>=20
> Cheers
>=20
> Magnus
>=20
> On 2012-04-18 05:58, 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           : Monitoring Architecture for RTP
> > 	Author(s)       : Qin Wu
> >                           Geoff Hunt
> >                           Philip Arden
> > 	Filename        : draft-ietf-avtcore-monarch-12.txt
> > 	Pages           : 29
> > 	Date            : 2012-04-17
> >
> >    This memo proposes an architecture for extending RTP Control
> Protocol
> >    (RTCP) with a new RTCP Extended Reports (XR) (RFC3611) block type
> to
> >    report new metrics regarding media transmission or reception
> quality,
> >    following RTCP guideline established in RFC5968.  This memo
> suggests
> >    that a new 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 which 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 which covers their set of concerns.  Where possible, a
> >    specific block should be designed to be re-usable across more =
than
> >    one application, for example, for all of voice, streaming audio
> and
> >    video.
> >
> >
> > A URL for this Internet-Draft is:
> > =
http://www.ietf.org/internet-drafts/draft-ietf-avtcore-monarch-12.txt
> >
> > Internet-Drafts are also available by anonymous FTP at:
> > ftp://ftp.ietf.org/internet-drafts/
> >
> > This Internet-Draft can be retrieved at:
> > ftp://ftp.ietf.org/internet-drafts/draft-ietf-avtcore-monarch-12.txt
> >
> > _______________________________________________
> > 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
> ----------------------------------------------------------------------


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