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From: Colin Perkins <csp@csperkins.org>
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References: <EC3FD58E75D43A4F8807FDE0749175461827F981@FRMRSSXCHMBSB1.dc-m.alcatel-lucent.com> <CDD3E15B-F46F-4033-9DBD-335605AB4C50@csperkins.org> <7C8D947749B24396AA011FB286CC69E1@china.huawei.com> <EC3FD58E75D43A4F8807FDE074917546182EA1B0@FRMRSSXCHMBSB1.dc-m.alcatel-lucent.com> <489F755BBD284D18AE9AD47F17317B6F@china.huawei.com> <EC3FD58E75D43A4F8807FDE074917546182EA93D@FRMRSSXCHMBSB1.dc-m.alcatel-lucent.com> <E42260F9A82F421DB84797EAA5B60653@china.huawei.com> <EC3FD58E75D43A4F8807FDE074917546182EA980@FRMRSSXCHMBSB1.dc-m.alcatel-lucent.com>
To: "Van Caenegem, Tom (Tom)" <tom.van_caenegem@alcatel-lucent.com>
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Subject: Re: [AVTCORE] Comments on draft-ietf-avtcore-feedback-supression-rtp-04
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On 30 Jun 2011, at 10:47, Van Caenegem, Tom (Tom) wrote:
> Qin,
> See below
> Tom
> -----Original Message-----
> From: Qin Wu [mailto:bill.wu@huawei.com]=20
> Sent: donderdag 30 juni 2011 11:36
> To: Van Caenegem, Tom (Tom); Colin Perkins
> Cc: 'IETF AVTCore WG'; Roni Even; Qin Wu
> Subject: Re: [AVTCORE] Comments on =
draft-ietf-avtcore-feedback-supression-rtp-04
>=20
> Hi,
> ----- Original Message -----
> From: "Van Caenegem, Tom (Tom)" <tom.van_caenegem@alcatel-lucent.com>
> To: "Qin Wu" <bill.wu@huawei.com>; "Colin Perkins" <csp@csperkins.org>
> Cc: "'IETF AVTCore WG'" <avt@ietf.org>; "Roni Even" =
<Even.roni@huawei.com>
> Sent: Thursday, June 30, 2011 5:07 PM
> Subject: RE: [AVTCORE] Comments on =
draft-ietf-avtcore-feedback-supression-rtp-04
>=20
>=20
> Qin,
> See below,
>=20
>=20
> [Qin]: Okay, maybe there is some ambiguity on describing media source =
behavior. I will add some texts to get consistent with=20
> Intermediary behavior of suppressing feedback described in the =
previous paragraph.
> The proposed text may look like:
> "
> Alternatively, the media source may directly monitor the amount of
>   feedback reports it receives from downstream.  If the media source =
notices the loss itself, then it
>   may send a NACK downstream towards the receivers to suppress =
feedback. If the media source
>   receives a NACK from another system, it should redistribute that =
NACK
>   to all other systems that would not otherwise receive it.  An =
example
>   of this is RTCP-SSM in simple feedback model [RFC5760], where the
>   distribution source reflects NACKs to other systems.  If the media =
source
>   receives a NACK from another system, but, for some
>   reason, cannot redistribute that NACK, then it may send a =
third-party
>   loss report to the systems that were unable to receive the NACK, and
>   won't receive the NACK via other means.  An example would be a
>   distribution source using RTCP-SSM in Distribution Source Feedback
>   Summary model [RFC5760].
> "
>=20
>=20
> Tom: the "for some reason" will have to be clarified. I also do not =
understand:  "may send a third-party loss report to the systems that =
were unable to receive the NACK"  Based on the rationale provided by =
Colin why one may not always simply forward/redistribute NACKs, this is =
not a matter of systems being "UNABLE" to receive the NACK. They are =
always able, it is just a decision from the intermediate to not FW the =
received NACKs.
>=20
> [Qin]:  No, basic rationale is prevent the receivers know who send =
NACK.=20
> In SSM case, the rationale is Distribution Source Feedback Summary =
model does not allow the NACK to be forwarded downstream.
>=20
> Tom: In RFC 5760 for SSM in FB summary mode, the DS might also forward =
RTCP feedback messages and transmit them in the SSM session, when this =
is explicitly signaled. It is a service/deployment decision whether =
NACKs are hence FW-ed or not, and there may be good reasons to not do =
so, but saying that such systems(i..e RTP Rx) are unable to receive the =
NACK is not justified IMO.

It might be clearer to say that the receivers don't get the NACKs, since =
they're not forwarded by the distribution source.

Colin

> [Qin]: I agree we should follow RFC5760 architecture. But I am not =
sure the example I give in the section 6.1.1 conflict with  RFC5760 =
architecture.
> The architecture I bulild looks like this:
> Suppose there are 100 RTP receivers, in these receivers, 50 RTP =
receivers belong to DS A, 50 RTP receivers belong to DS B, DS A may be =
placed at the upstream
> of DS B or DS A is adjacent to DS B at the same level, Do this kind =
architecture conflict with RFC5760? If yes, please point out.
>=20
> Tom: a DS represents the source/RTP sender of the SSM RTP session. =
Receivers send out NACKs related to packet loss in a particular SSM RTP =
session. If a receiver is not "belonging" to a DS X (I guess you mean =
..is not receiving packets from that DS X), then why would you implement =
some kind of FB suppression for such receiver(s) within the SSM RTP =
session that is sourced by that DS X  ???
>=20
> [Qin]: What I am saying here each DS has its own coverage for =
receivers. You can implement FB suppression for each DS. But you are not =
necessary to prevent interaction between DS.
> Anyway since you or other folks may think this multiple DS case =
introduce ambiguity and complexity, I will remove this case from the =
draft, Is that okay with you?
>=20
> Tom: Fine!=20
>=20
> Regards!
> -Qin
> ----- Original Message -----
> From: "Colin Perkins" <csp@csperkins.org>
> To: "Van Caenegem, Tom (Tom)" <tom.van_caenegem@alcatel-lucent.com>
> Cc: "Qin Wu" <sunseawq@huawei.com>; "'IETF AVTCore WG'" =
<avt@ietf.org>; "Roni Even" <Even.roni@huawei.com>
> Sent: Tuesday, June 21, 2011 10:07 PM
> Subject: Re: [AVTCORE] Comments on =
draft-ietf-avtcore-feedback-supression-rtp-04
>=20
>=20
> On 21 Jun 2011, at 10:29, Van Caenegem, Tom (Tom) wrote:
>> Hi,
>>=20
>> as promised, these are comments I have on the draft " RTCP Extension =
for Third-party Loss Report"
>>=20
>> -IMO the draft does not provide a good rationale on the use case for =
sending this new report. The section 3 (protocol overview) says:
>>=20
>> "If an intermediary receives a NACK from another system, it should =
redistribute that NACK to all other systems that would not otherwise =
receive it.  An example of this is RTCP-SSM in simple feedback model =
[RFC5760], where the distribution source reflects NACKs to other =
systems.  =20
>> If an intermediary receives a NACK from another system, but, for some =
reason, cannot redistribute that NACK, then it may send a third-party =
loss report to the systems that were unable to receive the NACK, and =
won't receive the NACK via other means. An example would be a =
distribution source using RTCP-SSM in Distribution Source Feedback =
Summary model [RFC5760]."
>>=20
>> My questions:=20
>>=20
>> -why would the intermediary not be able or capable to send the =
received NACK to the "systems" that were unable to receive the NACK, =
whereas it would be able or capable to send the 3rd party loss report to =
those same "systems"?=20
>=20
> Because the intermediary is trying to hide the identity or existence =
of those systems, and cannot forward the NACK from them without =
revealing that information
>=20
>> -The DS in RTCP-SSM operating in FB summary mode is mentioned as an =
example for a case where a NACK cannnot be forwarded. Why is this?  =
RFC5760 does not prevent a DS to foward the NACK (see the RFC 5760 and =
my quote from that RFC in =
"draft-vancaenegem-avtcore-retransmission-for-ssm-00"
>=20
> RFC 5760 allows the NACK to be forwarded, but this would cause the =
other receivers to become aware of, and keep state for, the participant =
that sent the NACK. If a third-party loss report is used, the other =
receivers don't become aware of the participant that signalled the loss, =
they just know that the loss occurred.=20
>=20
>> -This section 3 makes a distinction between an intermediate that does =
check the RTP stream for missing packets itself, and an intermediate =
that does not, but still the behaviour in terms of FW-ing the NACKs =
received from other "systems" is exactly the same. Why make this =
distinction then?
>>=20
>> I still have an outstanding comment which has not been resolved yet. =
The section 6.1. and 6.1.1. take RFC 5760 as use case architectures, but =
section 6.1.1. talks about two distribution sources, whereas RFC 5760 =
only defined ONE SINGLE DS. Please clarify this.
>=20
> [Qin]: two distribution sources means we have two levels of SSM =
communication, one distribution souce is placed at the upstream of the =
other distribution source.
> two distribution sources case is just one exmaple to discuss how to =
use new report when there are multiple middleboxs or intermediaries =
between media source and the receivers.=20
> The middlebox can be distribution source and can be other =
intermediary.
>=20
>>=20
>> Regards
>> Tom
>> _______________________________________________
>> Audio/Video Transport Core Maintenance
>> avt@ietf.org
>> https://www.ietf.org/mailman/listinfo/avt
>=20
>=20
>=20
> --=20
> Colin Perkins
> http://csperkins.org/
>=20
>=20
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt



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




From Even.roni@huawei.com  Sun Jul  3 09:54:44 2011
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Date: Sun, 03 Jul 2011 19:51:54 +0300
From: Roni Even <Even.roni@huawei.com>
In-reply-to: <E42260F9A82F421DB84797EAA5B60653@china.huawei.com>
To: 'Qin Wu' <bill.wu@huawei.com>, "'Van Caenegem, Tom (Tom)'" <tom.van_caenegem@alcatel-lucent.com>, 'Colin Perkins' <csp@csperkins.org>
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Cc: 'IETF AVTCore WG' <avt@ietf.org>
Subject: Re: [AVTCORE] Comments on draft-ietf-avtcore-feedback-supression-rtp-04
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Hi,
I think that we can look at the following use cases.

1. The single DS /FT in a Distribution Source Feedback Summary Model. The
case mentioned by Collin using 3rd party report to hide the topology.

2. A single DS with multiple FT. See RFC 5760 section 3 in the FT
definition. In this case I am not sure if there is a DS without any FT with
multiple FTs or one DS with a FT and multiple FTs sending unicast RTCP
feedback to the FT co-located with the DS. Maybe we need to look at both
cases.

3. RFC 5760 talks about single DS. Is there a use case for multiple DSs
where each DS is in a different SSRC space with one media source
transmitting to all. Is there such a case, maybe if there is some hierarchy
which may look like a DS which is also an RTP mixer or translator.

Does these use cases make sense?

Are there others

Regards
Roni

> -----Original Message-----
> From: Qin Wu [mailto:bill.wu@huawei.com]
> Sent: Thursday, June 30, 2011 12:36 PM
> To: Van Caenegem, Tom (Tom); Colin Perkins
> Cc: 'IETF AVTCore WG'; Roni Even; Qin Wu
> Subject: Re: [AVTCORE] Comments on draft-ietf-avtcore-feedback-
> supression-rtp-04
> 
> Hi,
> ----- Original Message -----
> From: "Van Caenegem, Tom (Tom)" <tom.van_caenegem@alcatel-lucent.com>
> To: "Qin Wu" <bill.wu@huawei.com>; "Colin Perkins" <csp@csperkins.org>
> Cc: "'IETF AVTCore WG'" <avt@ietf.org>; "Roni Even"
> <Even.roni@huawei.com>
> Sent: Thursday, June 30, 2011 5:07 PM
> Subject: RE: [AVTCORE] Comments on draft-ietf-avtcore-feedback-
> supression-rtp-04
> 
> 
> Qin,
> See below,
> 
> 
> [Qin]: Okay, maybe there is some ambiguity on describing media source
> behavior. I will add some texts to get consistent with
> Intermediary behavior of suppressing feedback described in the previous
> paragraph.
> The proposed text may look like:
> "
>  Alternatively, the media source may directly monitor the amount of
>    feedback reports it receives from downstream.  If the media source
> notices the loss itself, then it
>    may send a NACK downstream towards the receivers to suppress
> feedback. If the media source
>    receives a NACK from another system, it should redistribute that
> NACK
>    to all other systems that would not otherwise receive it.  An
> example
>    of this is RTCP-SSM in simple feedback model [RFC5760], where the
>    distribution source reflects NACKs to other systems.  If the media
> source
>    receives a NACK from another system, but, for some
>    reason, cannot redistribute that NACK, then it may send a third-
> party
>    loss report to the systems that were unable to receive the NACK, and
>    won't receive the NACK via other means.  An example would be a
>    distribution source using RTCP-SSM in Distribution Source Feedback
>    Summary model [RFC5760].
> "
> 
> 
> Tom: the "for some reason" will have to be clarified. I also do not
> understand:  "may send a third-party loss report to the systems that
> were unable to receive the NACK"  Based on the rationale provided by
> Colin why one may not always simply forward/redistribute NACKs, this is
> not a matter of systems being "UNABLE" to receive the NACK. They are
> always able, it is just a decision from the intermediate to not FW the
> received NACKs.
> 
> [Qin]:  No, basic rationale is prevent the receivers know who send
> NACK.
> In SSM case, the rationale is Distribution Source Feedback Summary
> model does not allow the NACK to be forwarded downstream.
> 
> [Qin]: I agree we should follow RFC5760 architecture. But I am not sure
> the example I give in the section 6.1.1 conflict with  RFC5760
> architecture.
> The architecture I bulild looks like this:
> Suppose there are 100 RTP receivers, in these receivers, 50 RTP
> receivers belong to DS A, 50 RTP receivers belong to DS B, DS A may be
> placed at the upstream
> of DS B or DS A is adjacent to DS B at the same level, Do this kind
> architecture conflict with RFC5760? If yes, please point out.
> 
> Tom: a DS represents the source/RTP sender of the SSM RTP session.
> Receivers send out NACKs related to packet loss in a particular SSM RTP
> session. If a receiver is not "belonging" to a DS X (I guess you mean
> ..is not receiving packets from that DS X), then why would you
> implement some kind of FB suppression for such receiver(s) within the
> SSM RTP session that is sourced by that DS X  ???
> 
> [Qin]: What I am saying here each DS has its own coverage for
> receivers. You can implement FB suppression for each DS. But you are
> not necessary to prevent interaction between DS.
> Anyway since you or other folks may think this multiple DS case
> introduce ambiguity and complexity, I will remove this case from the
> draft, Is that okay with you?
> 
> 
> Regards!
> -Qin
> ----- Original Message -----
> From: "Colin Perkins" <csp@csperkins.org>
> To: "Van Caenegem, Tom (Tom)" <tom.van_caenegem@alcatel-lucent.com>
> Cc: "Qin Wu" <sunseawq@huawei.com>; "'IETF AVTCore WG'" <avt@ietf.org>;
> "Roni Even" <Even.roni@huawei.com>
> Sent: Tuesday, June 21, 2011 10:07 PM
> Subject: Re: [AVTCORE] Comments on draft-ietf-avtcore-feedback-
> supression-rtp-04
> 
> 
> On 21 Jun 2011, at 10:29, Van Caenegem, Tom (Tom) wrote:
> > Hi,
> >
> > as promised, these are comments I have on the draft " RTCP Extension
> for Third-party Loss Report"
> >
> > -IMO the draft does not provide a good rationale on the use case for
> sending this new report. The section 3 (protocol overview) says:
> >
> > "If an intermediary receives a NACK from another system, it should
> redistribute that NACK to all other systems that would not otherwise
> receive it.  An example of this is RTCP-SSM in simple feedback model
> [RFC5760], where the distribution source reflects NACKs to other
> systems.
> > If an intermediary receives a NACK from another system, but, for some
> reason, cannot redistribute that NACK, then it may send a third-party
> loss report to the systems that were unable to receive the NACK, and
> won't receive the NACK via other means. An example would be a
> distribution source using RTCP-SSM in Distribution Source Feedback
> Summary model [RFC5760]."
> >
> > My questions:
> >
> > -why would the intermediary not be able or capable to send the
> received NACK to the "systems" that were unable to receive the NACK,
> whereas it would be able or capable to send the 3rd party loss report
> to those same "systems"?
> 
> Because the intermediary is trying to hide the identity or existence of
> those systems, and cannot forward the NACK from them without revealing
> that information
> 
> > -The DS in RTCP-SSM operating in FB summary mode is mentioned as an
> example for a case where a NACK cannnot be forwarded. Why is this?
> RFC5760 does not prevent a DS to foward the NACK (see the RFC 5760 and
> my quote from that RFC in "draft-vancaenegem-avtcore-retransmission-
> for-ssm-00"
> 
> RFC 5760 allows the NACK to be forwarded, but this would cause the
> other receivers to become aware of, and keep state for, the participant
> that sent the NACK. If a third-party loss report is used, the other
> receivers don't become aware of the participant that signalled the
> loss, they just know that the loss occurred.
> 
> > -This section 3 makes a distinction between an intermediate that does
> check the RTP stream for missing packets itself, and an intermediate
> that does not, but still the behaviour in terms of FW-ing the NACKs
> received from other "systems" is exactly the same. Why make this
> distinction then?
> >
> > I still have an outstanding comment which has not been resolved yet.
> The section 6.1. and 6.1.1. take RFC 5760 as use case architectures,
> but section 6.1.1. talks about two distribution sources, whereas RFC
> 5760 only defined ONE SINGLE DS. Please clarify this.
> 
> [Qin]: two distribution sources means we have two levels of SSM
> communication, one distribution souce is placed at the upstream of the
> other distribution source.
> two distribution sources case is just one exmaple to discuss how to use
> new report when there are multiple middleboxs or intermediaries between
> media source and the receivers.
> The middlebox can be distribution source and can be other intermediary.
> 
> >
> > Regards
> > Tom
> > _______________________________________________
> > Audio/Video Transport Core Maintenance
> > avt@ietf.org
> > https://www.ietf.org/mailman/listinfo/avt
> 
> 
> 
> --
> Colin Perkins
> http://csperkins.org/
> 



From magnus.westerlund@ericsson.com  Tue Jul  5 03:54:21 2011
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WG,

As an individual Bo Burman and I have written this internet draft that
discusses multistream and simulcast issues and proposes a set of
extensions to resolve the issues. Feedback is much appreciated.

Cheers

Magnus

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

	Title           : RTP Multiple Stream Sessions and Simulcast
	Author(s)       : Magnus Westerlund
                          Bo Burman
	Filename        : draft-westerlund-avtcore-multistream-and-simulcast-00.tx=
t
	Pages           : 60
	Date            : 2011-07-04

   RTP has always been a protocol that supports multiple participants
   each sending their own media streams in an RTP session.
   Unfortunately many implementations aimed only at 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 a problem with legacy.  This issue is analyzed,
   and RTP clarifications and signalling extensions are proposed to
   handle this issue.  A related issue is how to perform simulcast, in
   the meaning of sending multiple encodings or representations of the
   same media source, when using RTP for media transport.  This is
   further analyzed and possible solutions discussed and we arrive at a
   conclusion for session multiplexing of simulcast versions.  We also
   found a number of related issues when having multiple streams and
   simulcast.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-westerlund-avtcore-multistream-an=
d-simulcast-00.txt

Internet-Drafts are also available by anonymous FTP at:
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This Internet-Draft can be retrieved at:
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-simulcast-00.txt
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From internet-drafts@ietf.org  Wed Jul  6 05:05:05 2011
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Subject: [AVTCORE] I-D Action: draft-ietf-avtcore-idms-01.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           : RTCP for inter-destination media synchronization
	Author(s)       : Ray van Brandenburg
                          Hans M. Stokking
                          M. Oskar van Deventer
                          Omar A. Niamut
                          Fabian A. Walraven
                          Ishan Vaishnavi
                          Fernando Boronat
                          Mario Montagud
	Filename        : draft-ietf-avtcore-idms-01.txt
	Pages           : 24
	Date            : 2011-07-06

   This document gives information on an RTCP Packet Type and RTCP XR
   Block Type including associated SDP parameters for inter-destination
   media synchronization (IDMS). The RTCP XR Block Type, registered with
   IANA based on an ETSI specification, is used to collect media play-
   out information from participants in a group playing-out (watching,
   listening, etc.) a specific RTP media stream. The RTCP packet type
   specified by this document is used to distribute a summary of the
   collected information so that the participants can synchronize play-
   out.

   Typical applications for IDMS are social TV, shared service control
   (i.e. applications where two or more geographically separated users
   are watching a media stream together), distance learning, network
   quiz shows, multi-playing online games, etc.



A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avtcore-idms-01.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-idms-01.txt

From wwwrun@rfc-editor.org  Tue Jul  5 09:18:20 2011
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Subject: [AVTCORE] [Technical Errata Reported] RFC4585 (2853)
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The following errata report has been submitted for RFC4585,
"Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)".

--------------------------------------
You may review the report below and at:
http://www.rfc-editor.org/errata_search.php?rfc=4585&eid=2853

--------------------------------------
Type: Technical
Reported by: Ali Begen <abegen@cisco.com>

Section: 4.2

Original Text
-------------
OLD:
      rtcp-fb-param      = SP "app" [SP byte-string]
                         / SP token [SP byte-string]
                         / ; empty
NEW:
      rtcp-fb-param      = [ SP "app" [SP byte-string]
                         / SP token [SP byte-string] ]


OLD:
      rtcp-fb-ack-param  = SP "rpsi"
                         / SP "app" [SP byte-string]
                         / SP token [SP byte-string]
                         / ; empty
NEW:
      rtcp-fb-ack-param  = [ SP "rpsi"
                         / SP "app" [SP byte-string]
                         / SP token [SP byte-string] ]


OLD:
      rtcp-fb-nack-param = SP "pli"
                         / SP "sli"
                         / SP "rpsi"
                         / SP "app" [SP byte-string]
                         / SP token [SP byte-string]
                         / ; empty
NEW:
      rtcp-fb-nack-param = [ SP "pli"
                         / SP "sli"
                         / SP "rpsi"
                         / SP "app" [SP byte-string]
                         / SP token [SP byte-string] ]


Corrected Text
--------------


Notes
-----
During the AUTH48 of RFC 6285, we discovered a bug with the ABNF syntax in RFC 4585. The original ABNF in Section 4.2 (as noted above) is incorrect ("/ ;empty" is not a valid ABNF - RFC 5234 does not allow empty alternates).

The NEW text intends to fix it. There are 3 places where the same change should be applied (changes are identical).

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

--------------------------------------
RFC4585 (draft-ietf-avt-rtcp-feedback-11)
--------------------------------------
Title               : Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)
Publication Date    : July 2006
Author(s)           : J. Ott, S. Wenger, N. Sato, C. Burmeister, J. Rey
Category            : PROPOSED STANDARD
Source              : Audio/Video Transport
Area                : Real-time Applications and Infrastructure
Stream              : IETF
Verifying Party     : IESG

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Hi,
----- Original Message ----- 
From: "Colin Perkins" <csp@csperkins.org>
To: "Van Caenegem, Tom (Tom)" <tom.van_caenegem@alcatel-lucent.com>
Cc: "Qin Wu" <bill.wu@huawei.com>; "'IETF AVTCore WG'" <avt@ietf.org>
Sent: Saturday, July 02, 2011 8:25 PM
Subject: Re: [AVTCORE] Comments on draft-ietf-avtcore-feedback-supression-rtp-04


On 30 Jun 2011, at 10:47, Van Caenegem, Tom (Tom) wrote:
> Qin,
> See below
> Tom
> -----Original Message-----
> From: Qin Wu [mailto:bill.wu@huawei.com] 
> Sent: donderdag 30 juni 2011 11:36
> To: Van Caenegem, Tom (Tom); Colin Perkins
> Cc: 'IETF AVTCore WG'; Roni Even; Qin Wu
> Subject: Re: [AVTCORE] Comments on draft-ietf-avtcore-feedback-supression-rtp-04
> 
> Hi,
> ----- Original Message -----
> From: "Van Caenegem, Tom (Tom)" <tom.van_caenegem@alcatel-lucent.com>
> To: "Qin Wu" <bill.wu@huawei.com>; "Colin Perkins" <csp@csperkins.org>
> Cc: "'IETF AVTCore WG'" <avt@ietf.org>; "Roni Even" <Even.roni@huawei.com>
> Sent: Thursday, June 30, 2011 5:07 PM
> Subject: RE: [AVTCORE] Comments on draft-ietf-avtcore-feedback-supression-rtp-04
> 
> 
> Qin,
> See below,
> 
> 
> [Qin]: Okay, maybe there is some ambiguity on describing media source behavior. I will add some texts to get consistent with 
> Intermediary behavior of suppressing feedback described in the previous paragraph.
> The proposed text may look like:
> "
> Alternatively, the media source may directly monitor the amount of
>   feedback reports it receives from downstream.  If the media source notices the loss itself, then it
>   may send a NACK downstream towards the receivers to suppress feedback. If the media source
>   receives a NACK from another system, it should redistribute that NACK
>   to all other systems that would not otherwise receive it.  An example
>   of this is RTCP-SSM in simple feedback model [RFC5760], where the
>   distribution source reflects NACKs to other systems.  If the media source
>   receives a NACK from another system, but, for some
>   reason, cannot redistribute that NACK, then it may send a third-party
>   loss report to the systems that were unable to receive the NACK, and
>   won't receive the NACK via other means.  An example would be a
>   distribution source using RTCP-SSM in Distribution Source Feedback
>   Summary model [RFC5760].
> "
> 
> 
> Tom: the "for some reason" will have to be clarified. I also do not understand:  "may send a third-party loss report to the systems that were unable to receive the NACK"  Based on the rationale provided by Colin why one may not always simply forward/redistribute NACKs, this is not a matter of systems being "UNABLE" to receive the NACK. They are always able, it is just a decision from the intermediate to not FW the received NACKs.
> 
> [Qin]:  No, basic rationale is prevent the receivers know who send NACK. 
> In SSM case, the rationale is Distribution Source Feedback Summary model does not allow the NACK to be forwarded downstream.
> 
> Tom: In RFC 5760 for SSM in FB summary mode, the DS might also forward RTCP feedback messages and transmit them in the SSM session, when this is explicitly signaled. It is a service/deployment decision whether NACKs are hence FW-ed or not, and there may be good reasons to not do so, but saying that such systems(i..e RTP Rx) are unable to receive the NACK is not justified IMO.

It might be clearer to say that the receivers don't get the NACKs, since they're not forwarded by the distribution source.

[Qin]: Okay. I agree.

Colin

> [Qin]: I agree we should follow RFC5760 architecture. But I am not sure the example I give in the section 6.1.1 conflict with  RFC5760 architecture.
> The architecture I bulild looks like this:
> Suppose there are 100 RTP receivers, in these receivers, 50 RTP receivers belong to DS A, 50 RTP receivers belong to DS B, DS A may be placed at the upstream
> of DS B or DS A is adjacent to DS B at the same level, Do this kind architecture conflict with RFC5760? If yes, please point out.
> 
> Tom: a DS represents the source/RTP sender of the SSM RTP session. Receivers send out NACKs related to packet loss in a particular SSM RTP session. If a receiver is not "belonging" to a DS X (I guess you mean ..is not receiving packets from that DS X), then why would you implement some kind of FB suppression for such receiver(s) within the SSM RTP session that is sourced by that DS X  ???
> 
> [Qin]: What I am saying here each DS has its own coverage for receivers. You can implement FB suppression for each DS. But you are not necessary to prevent interaction between DS.
> Anyway since you or other folks may think this multiple DS case introduce ambiguity and complexity, I will remove this case from the draft, Is that okay with you?
> 
> Tom: Fine! 
> 
> Regards!
> -Qin
> ----- Original Message -----
> From: "Colin Perkins" <csp@csperkins.org>
> To: "Van Caenegem, Tom (Tom)" <tom.van_caenegem@alcatel-lucent.com>
> Cc: "Qin Wu" <sunseawq@huawei.com>; "'IETF AVTCore WG'" <avt@ietf.org>; "Roni Even" <Even.roni@huawei.com>
> Sent: Tuesday, June 21, 2011 10:07 PM
> Subject: Re: [AVTCORE] Comments on draft-ietf-avtcore-feedback-supression-rtp-04
> 
> 
> On 21 Jun 2011, at 10:29, Van Caenegem, Tom (Tom) wrote:
>> Hi,
>> 
>> as promised, these are comments I have on the draft " RTCP Extension for Third-party Loss Report"
>> 
>> -IMO the draft does not provide a good rationale on the use case for sending this new report. The section 3 (protocol overview) says:
>> 
>> "If an intermediary receives a NACK from another system, it should redistribute that NACK to all other systems that would not otherwise receive it.  An example of this is RTCP-SSM in simple feedback model [RFC5760], where the distribution source reflects NACKs to other systems.   
>> If an intermediary receives a NACK from another system, but, for some reason, cannot redistribute that NACK, then it may send a third-party loss report to the systems that were unable to receive the NACK, and won't receive the NACK via other means. An example would be a distribution source using RTCP-SSM in Distribution Source Feedback Summary model [RFC5760]."
>> 
>> My questions: 
>> 
>> -why would the intermediary not be able or capable to send the received NACK to the "systems" that were unable to receive the NACK, whereas it would be able or capable to send the 3rd party loss report to those same "systems"? 
> 
> Because the intermediary is trying to hide the identity or existence of those systems, and cannot forward the NACK from them without revealing that information
> 
>> -The DS in RTCP-SSM operating in FB summary mode is mentioned as an example for a case where a NACK cannnot be forwarded. Why is this?  RFC5760 does not prevent a DS to foward the NACK (see the RFC 5760 and my quote from that RFC in "draft-vancaenegem-avtcore-retransmission-for-ssm-00"
> 
> RFC 5760 allows the NACK to be forwarded, but this would cause the other receivers to become aware of, and keep state for, the participant that sent the NACK. If a third-party loss report is used, the other receivers don't become aware of the participant that signalled the loss, they just know that the loss occurred. 
> 
>> -This section 3 makes a distinction between an intermediate that does check the RTP stream for missing packets itself, and an intermediate that does not, but still the behaviour in terms of FW-ing the NACKs received from other "systems" is exactly the same. Why make this distinction then?
>> 
>> I still have an outstanding comment which has not been resolved yet. The section 6.1. and 6.1.1. take RFC 5760 as use case architectures, but section 6.1.1. talks about two distribution sources, whereas RFC 5760 only defined ONE SINGLE DS. Please clarify this.
> 
> [Qin]: two distribution sources means we have two levels of SSM communication, one distribution souce is placed at the upstream of the other distribution source.
> two distribution sources case is just one exmaple to discuss how to use new report when there are multiple middleboxs or intermediaries between media source and the receivers. 
> The middlebox can be distribution source and can be other intermediary.
> 
>> 
>> Regards
>> Tom
>> _______________________________________________
>> Audio/Video Transport Core Maintenance
>> avt@ietf.org
>> https://www.ietf.org/mailman/listinfo/avt
> 
> 
> 
> -- 
> Colin Perkins
> http://csperkins.org/
> 
> 
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt



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



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Subject: Re: [AVTCORE] Comments on draft-ietf-avtcore-feedback-supression-rtp-04
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Hi,
----- Original Message ----- 
From: "Roni Even" <even.roni@huawei.com>
To: "'Qin Wu'" <bill.wu@huawei.com>; "'Van Caenegem, Tom (Tom)'" <tom.van_caenegem@alcatel-lucent.com>; "'Colin Perkins'" <csp@csperkins.org>
Cc: "'IETF AVTCore WG'" <avt@ietf.org>
Sent: Monday, July 04, 2011 12:51 AM
Subject: RE: [AVTCORE] Comments on draft-ietf-avtcore-feedback-supression-rtp-04


> Hi,
> I think that we can look at the following use cases.
> 
> 1. The single DS /FT in a Distribution Source Feedback Summary Model. The
> case mentioned by Collin using 3rd party report to hide the topology.


> 2. A single DS with multiple FT. See RFC 5760 section 3 in the FT
> definition. In this case I am not sure if there is a DS without any FT with
> multiple FTs or one DS with a FT and multiple FTs sending unicast RTCP
> feedback to the FT co-located with the DS. Maybe we need to look at both
> cases.
> 
> 3. RFC 5760 talks about single DS. Is there a use case for multiple DSs
> where each DS is in a different SSRC space with one media source
> transmitting to all. Is there such a case, maybe if there is some hierarchy
> which may look like a DS which is also an RTP mixer or translator.
> 
> Does these use cases make sense?

[Qin]: I think These three cases are all SSM use case. 
Current draft covers case 1 and case 3. 
But according to  Tom's comments,
case 3 may introduce too much confusion and complexity. 
I am going to leave this case out since the focus of this draft is 
on  usage of  new message in the SSM general use case.

As for case 2, draft-vancaenegem-avtcore-retransmission-for-ssm-00
 discusses a lot about how to extend protocol 
to provide better retransmission. some part is relevant to this draft. 
But I guess updating RFC5760 or  rule changing of RFC5760 is required.

> Are there others
> 
> Regards
> Roni
> 
>> -----Original Message-----
>> From: Qin Wu [mailto:bill.wu@huawei.com]
>> Sent: Thursday, June 30, 2011 12:36 PM
>> To: Van Caenegem, Tom (Tom); Colin Perkins
>> Cc: 'IETF AVTCore WG'; Roni Even; Qin Wu
>> Subject: Re: [AVTCORE] Comments on draft-ietf-avtcore-feedback-
>> supression-rtp-04
>> 
>> Hi,
>> ----- Original Message -----
>> From: "Van Caenegem, Tom (Tom)" <tom.van_caenegem@alcatel-lucent.com>
>> To: "Qin Wu" <bill.wu@huawei.com>; "Colin Perkins" <csp@csperkins.org>
>> Cc: "'IETF AVTCore WG'" <avt@ietf.org>; "Roni Even"
>> <Even.roni@huawei.com>
>> Sent: Thursday, June 30, 2011 5:07 PM
>> Subject: RE: [AVTCORE] Comments on draft-ietf-avtcore-feedback-
>> supression-rtp-04
>> 
>> 
>> Qin,
>> See below,
>> 
>> 
>> [Qin]: Okay, maybe there is some ambiguity on describing media source
>> behavior. I will add some texts to get consistent with
>> Intermediary behavior of suppressing feedback described in the previous
>> paragraph.
>> The proposed text may look like:
>> "
>>  Alternatively, the media source may directly monitor the amount of
>>    feedback reports it receives from downstream.  If the media source
>> notices the loss itself, then it
>>    may send a NACK downstream towards the receivers to suppress
>> feedback. If the media source
>>    receives a NACK from another system, it should redistribute that
>> NACK
>>    to all other systems that would not otherwise receive it.  An
>> example
>>    of this is RTCP-SSM in simple feedback model [RFC5760], where the
>>    distribution source reflects NACKs to other systems.  If the media
>> source
>>    receives a NACK from another system, but, for some
>>    reason, cannot redistribute that NACK, then it may send a third-
>> party
>>    loss report to the systems that were unable to receive the NACK, and
>>    won't receive the NACK via other means.  An example would be a
>>    distribution source using RTCP-SSM in Distribution Source Feedback
>>    Summary model [RFC5760].
>> "
>> 
>> 
>> Tom: the "for some reason" will have to be clarified. I also do not
>> understand:  "may send a third-party loss report to the systems that
>> were unable to receive the NACK"  Based on the rationale provided by
>> Colin why one may not always simply forward/redistribute NACKs, this is
>> not a matter of systems being "UNABLE" to receive the NACK. They are
>> always able, it is just a decision from the intermediate to not FW the
>> received NACKs.
>> 
>> [Qin]:  No, basic rationale is prevent the receivers know who send
>> NACK.
>> In SSM case, the rationale is Distribution Source Feedback Summary
>> model does not allow the NACK to be forwarded downstream.
>> 
>> [Qin]: I agree we should follow RFC5760 architecture. But I am not sure
>> the example I give in the section 6.1.1 conflict with  RFC5760
>> architecture.
>> The architecture I bulild looks like this:
>> Suppose there are 100 RTP receivers, in these receivers, 50 RTP
>> receivers belong to DS A, 50 RTP receivers belong to DS B, DS A may be
>> placed at the upstream
>> of DS B or DS A is adjacent to DS B at the same level, Do this kind
>> architecture conflict with RFC5760? If yes, please point out.
>> 
>> Tom: a DS represents the source/RTP sender of the SSM RTP session.
>> Receivers send out NACKs related to packet loss in a particular SSM RTP
>> session. If a receiver is not "belonging" to a DS X (I guess you mean
>> ..is not receiving packets from that DS X), then why would you
>> implement some kind of FB suppression for such receiver(s) within the
>> SSM RTP session that is sourced by that DS X  ???
>> 
>> [Qin]: What I am saying here each DS has its own coverage for
>> receivers. You can implement FB suppression for each DS. But you are
>> not necessary to prevent interaction between DS.
>> Anyway since you or other folks may think this multiple DS case
>> introduce ambiguity and complexity, I will remove this case from the
>> draft, Is that okay with you?
>> 
>> 
>> Regards!
>> -Qin
>> ----- Original Message -----
>> From: "Colin Perkins" <csp@csperkins.org>
>> To: "Van Caenegem, Tom (Tom)" <tom.van_caenegem@alcatel-lucent.com>
>> Cc: "Qin Wu" <sunseawq@huawei.com>; "'IETF AVTCore WG'" <avt@ietf.org>;
>> "Roni Even" <Even.roni@huawei.com>
>> Sent: Tuesday, June 21, 2011 10:07 PM
>> Subject: Re: [AVTCORE] Comments on draft-ietf-avtcore-feedback-
>> supression-rtp-04
>> 
>> 
>> On 21 Jun 2011, at 10:29, Van Caenegem, Tom (Tom) wrote:
>> > Hi,
>> >
>> > as promised, these are comments I have on the draft " RTCP Extension
>> for Third-party Loss Report"
>> >
>> > -IMO the draft does not provide a good rationale on the use case for
>> sending this new report. The section 3 (protocol overview) says:
>> >
>> > "If an intermediary receives a NACK from another system, it should
>> redistribute that NACK to all other systems that would not otherwise
>> receive it.  An example of this is RTCP-SSM in simple feedback model
>> [RFC5760], where the distribution source reflects NACKs to other
>> systems.
>> > If an intermediary receives a NACK from another system, but, for some
>> reason, cannot redistribute that NACK, then it may send a third-party
>> loss report to the systems that were unable to receive the NACK, and
>> won't receive the NACK via other means. An example would be a
>> distribution source using RTCP-SSM in Distribution Source Feedback
>> Summary model [RFC5760]."
>> >
>> > My questions:
>> >
>> > -why would the intermediary not be able or capable to send the
>> received NACK to the "systems" that were unable to receive the NACK,
>> whereas it would be able or capable to send the 3rd party loss report
>> to those same "systems"?
>> 
>> Because the intermediary is trying to hide the identity or existence of
>> those systems, and cannot forward the NACK from them without revealing
>> that information
>> 
>> > -The DS in RTCP-SSM operating in FB summary mode is mentioned as an
>> example for a case where a NACK cannnot be forwarded. Why is this?
>> RFC5760 does not prevent a DS to foward the NACK (see the RFC 5760 and
>> my quote from that RFC in "draft-vancaenegem-avtcore-retransmission-
>> for-ssm-00"
>> 
>> RFC 5760 allows the NACK to be forwarded, but this would cause the
>> other receivers to become aware of, and keep state for, the participant
>> that sent the NACK. If a third-party loss report is used, the other
>> receivers don't become aware of the participant that signalled the
>> loss, they just know that the loss occurred.
>> 
>> > -This section 3 makes a distinction between an intermediate that does
>> check the RTP stream for missing packets itself, and an intermediate
>> that does not, but still the behaviour in terms of FW-ing the NACKs
>> received from other "systems" is exactly the same. Why make this
>> distinction then?
>> >
>> > I still have an outstanding comment which has not been resolved yet.
>> The section 6.1. and 6.1.1. take RFC 5760 as use case architectures,
>> but section 6.1.1. talks about two distribution sources, whereas RFC
>> 5760 only defined ONE SINGLE DS. Please clarify this.
>> 
>> [Qin]: two distribution sources means we have two levels of SSM
>> communication, one distribution souce is placed at the upstream of the
>> other distribution source.
>> two distribution sources case is just one exmaple to discuss how to use
>> new report when there are multiple middleboxs or intermediaries between
>> media source and the receivers.
>> The middlebox can be distribution source and can be other intermediary.
>> 
>> >
>> > Regards
>> > Tom
>> > _______________________________________________
>> > Audio/Video Transport Core Maintenance
>> > avt@ietf.org
>> > https://www.ietf.org/mailman/listinfo/avt
>> 
>> 
>> 
>> --
>> Colin Perkins
>> http://csperkins.org/
>> 
> 
>

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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           : Guidelines for the use of Variable Bit Rate Audio with S=
ecure RTP
	Author(s)       : Colin Perkins
                          Jean-Marc Valin
	Filename        : draft-ietf-avtcore-srtp-vbr-audio-03.txt
	Pages           : 7
	Date            : 2011-07-07

   This memo discusses potential security issues that arise when using
   variable bit rate audio with the secure RTP profile.  Guidelines to
   mitigate these issues are suggested.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avtcore-srtp-vbr-audio-03.txt

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This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-avtcore-srtp-vbr-audio-03.txt

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On 7 Jul 2011, at 10:05, Internet-Drafts@ietf.org wrote:
> A New Internet-Draft is available from the on-line Internet-Drafts =
directories. This draft is a work item of the Audio/Video Transport Core =
Maintenance Working Group of the IETF.
>=20
> 	Title           : Guidelines for the use of Variable Bit Rate =
Audio with Secure RTP
> 	Author(s)       : Colin Perkins
>                          Jean-Marc Valin
> 	Filename        : draft-ietf-avtcore-srtp-vbr-audio-03.txt
> 	Pages           : 7
> 	Date            : 2011-07-07
>=20
>   This memo discusses potential security issues that arise when using
>   variable bit rate audio with the secure RTP profile.  Guidelines to
>   mitigate these issues are suggested.
>=20
>=20
> A URL for this Internet-Draft is:
> =
http://www.ietf.org/internet-drafts/draft-ietf-avtcore-srtp-vbr-audio-03.t=
xt


This addresses some nits that came up at a result of WG last call.=20

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




From rayvbr@gmail.com  Thu Jul  7 08:06:48 2011
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Date: Thu, 7 Jul 2011 17:06:47 +0200
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From: Ray van Brandenburg <rayvbr@gmail.com>
To: avt@ietf.org
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Subject: [AVTCORE] New version of IMDS draft
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--00148536e646d6e1ce04a77c10dc
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--00148536e646d6e1c804a77c10da
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Hi all,

As you can see, I uploaded a new version of the IDMS draft. This version
includes the changes as suggested by Kevin Gross earlier but that we could't
include on time for the July 4th new document deadline.

Included are amongst others a number of clarifications regarding:
        - Use of the presentation timestamp
        - How to deal with multiple RTP packets with the same RTP timestamp
        - Use of idms in high-accuracy applications
        - Dealing with ptp/gal/gps as a time source

@Kevin: You mentioned that you would be willing to provide some additional
use cases to include in section 4. If you could write some text for this, I
can include it in the next version.

Looking forward to your comments,

Best regards,

Ray van Brandenburg

--00148536e646d6e1c804a77c10da
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div class=3D"gmail_quote"><div>Hi all,</div><div><br></div><div>As you can=
 see, I uploaded a new version of the IDMS draft. This version includes the=
 changes as suggested by Kevin Gross earlier but that we could&#39;t includ=
e on time for the July 4th new document deadline.</div>

<div>=A0</div><div>Included are amongst others a number of clarifications r=
egarding:</div><div>=A0 =A0 =A0 =A0 - Use of the presentation timestamp</di=
v><div>=A0 =A0 =A0 =A0 - How to deal with multiple RTP packets with the sam=
e RTP timestamp</div>

<div>=A0 =A0 =A0 =A0 - Use of idms in high-accuracy applications</div><div>=
=A0 =A0 =A0 =A0 - Dealing with ptp/gal/gps as a time source</div><div>=A0</=
div><div>@Kevin: You mentioned that you would be willing to provide some ad=
ditional use cases to include in section 4. If you could write some text fo=
r this, I can include it in the next version.=A0</div>

<div>=A0</div><div>Looking forward to your comments,</div><div>=A0</div><di=
v>Best regards,</div><div>=A0</div><font color=3D"#888888"><div>Ray van Bra=
ndenburg</div>
</font></div><br>

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From mmagliaro@bnisolutions.com  Thu Jul  7 15:37:36 2011
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From: Max Magliaro <mmagliaro@bnisolutions.com>
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Subject: [AVTCORE] Knowing when to use RFC 2190 vs. RFC 2429/4629 for H.263 video?
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RFC 2429 is the preferred format (actually  4629 now).
However, 2190 may still have to be used for some legacy systems.

If I am one of those legacy endpoints,  (and I am)
what, in the H.245 call signalling of an H.323 call, do I use to indicate
to the other end NOT to send me the newer-style payload header?

Specifically, I note that when I call to a Polycom RMX system, it 
exchanges H.263 with me
just fine.   However, if it opens a secondary H.239 collaboration video 
channel,
it sends that one with the newer-style headers.   When I call a Tandberg 
system,
this doesn't happen (they send 2190 style headers in the collaboration 
channel and everything
works).  If I try this with a Polycom VSX, it doesn't do this (all works 
fine).

I assume there is some "hint" in my capability set that I need to set, 
or avoid setting, so
that all endpoints (including Polycom's RMX) will know to not send the 
newer-style headers.
Or maybe it's the opposite, that there is something in the capabilities 
I need to
NOT set.

I realize that the ideal solution would be to support the current header 
structure, but I do not
have that option available to me.

If anyone has a hint as to how Polycom makes that decision to send RFC 
2190 or RFC 2429/4629,
I would appreciate it.

Thank you,

Max Magliaro
Broadband Networks, Inc


From magnus.westerlund@ericsson.com  Fri Jul  8 01:37:13 2011
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WG,

I received comments from Randell Jesup on the MMUSIC reflector, but I
think they are equally relevant to this WG. So here it he comments with
my response to them.

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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
To: Randell Jesup <randell-ietf@jesup.org>
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Hi Randell,

Thanks for the feedback. See inline for feedback on your comments.

On 2011-07-07 17:13, Randell Jesup wrote:
> On 7/7/2011 4:46 AM, Magnus Westerlund wrote:
>> Hi,
>>
>> Bo Burman and I have written an I-D that discusses a number of issues
>> around conferencing, including handling of multiple SSRCs in the same
>> RTP session. This also has brought forward an issue with the current
>> bandwidth modifiers. They are very ill suited when you have both
>> asymmetric conditions based on which direction the flows goes and how to
>> indicate individual streams versus limits for the whole aggregate. This
>> include the issue of having payload specific per stream upper bounds.
>>
>> https://datatracker.ietf.org/doc/draft-westerlund-avtcore-multistream-an=
d-simulcast/
>
> Up front: realize I'm commenting as I read, but that means that it's
> also an indication if
> something wasn't clear up-front without reading more of later parts of
> the spec.
>
>
> The purpose of the document is a little fuzzy - is it for simulcast (as
> stated and defined
> in the doc) or for both multiple-source and simulcast (as implied in the
> document and its
> title).  It  seems to be the former, but the title and some early
> statements imply it's both.


Yes, this is document that has grown while we wrote it. I actually
expect that we start splitting up the different pieces into individual
documents and tries to clarify what is applicable in what case.

But our intention from the start is to address both cases. But yes we
started with simulcast and for that we do need to resolve the multiple
SSRC in RTP session problems. So simulcast requires multiple SSRCs,
however multiple SSRC has many usages beyond simulcast.


>
> Also it should be made clear (if it isn't; I'm reading fast) that these
> mixer cases specifically
> would include one-to-one communication through a server.  Also: should
> this discuss
> non-mixer cases (point-to-point simulcast)?  I think so.

We tried to be clear that our main focus for simulcast is the
centralized case using a RTP mixer that selects which version to
forward. But Section 3 does look at the other usages of simulcast, like
point to point.

>
>     Thus there has been perceived little need for
>     handling multiple SSRCs in implementations.  This has resulted in an
>     installed legacy base that isn't fully RTP specification compliant
>     and will have different issues if they receive multiple SSRCs of
>     media, either simultaneously or in sequence.  These issues will
>     manifest themselves in various ways, either by software crashes, or
>     simply in limited functionality, like only decoding and playing back
>     the first or latest SSRC received and discarding any other SSRCs.
>
>
> As an implementer of a point-to-point videophone, we handled SSRC changes=
,
> but only were set up to decode one stream at a time (especially of
> video).  (A later
> phone could decode up to 3 and encode 1 when acting as a bridge for an
> ad-hoc video conference.)
>
> Our algorithm was to immediately react to an SSRC change, but then lock
> out changes
> for a (short) period because since we only played one stream at a time
> we didn't want to reset the
> codec on every other incoming packet.  This also helped us with a few
> other usecases involving
> various B2BUA-ish servers and server-mediated call transfers, etc.
>
> Not truly compliant per-se, of course, but worked for the usecases we
> needed to deal with.

Yes, and what you say is a confirmation of the claim we are making.
There is a number of implementations that has limited support for
multiple SSRCs but they don't express that in signalling. Therefore to
make these multiple SSRC applications work well going forward we need
explicit signalling of the number of simultanous SSRCs one support.

>
>     Simulcasting has the benefit that it is conceptually simple.  It
>     enables use of any media codec that the participants agree on,
>     allowing the mixer to be codec-agnostic.  Considering today's video
>     encoders, it is less bit-rate efficient in the path from the sending
>     client to the mixer but more efficient in the mixer to receiver path
>     compared to Scalable Video Coding.
>
>
> If you're targeting a single lower-bitrate encoding, yes.  If the
> bandwidth down to each receiver
> is adaptive to their available bandwidth (think mobile reception, or
> WiFi) then giving the mixer
> the ability to subset differently for each receiver without having to
> re-encode.  (And it avoids
> having to tell the sender to change bitrates if one of the receivers
> needs a lower rate than the
> sender is sending, and it avoids giving all the other recipients a
> lowest-common-denominator
> stream).  Of course this is part of the entire value-proposition of SVC
> - that it enables this sort of
> optimized reception with a single encoding and a low-horsepower mixer.
> Yes, I realize I'm saying
> fairly obvious things here.

Exactly, this is the whole point of simulcasting. And our research hows
that this has some advantages over SVC.

>
>     If only resolution and temporal variations
>     are needed, this can be implemented using H.264, as each simulcast
>     version provides the different resolution, and each media stream
>     within a simulcast encoding has temporal scalability using no-
>     reference frames.
>
>
> Note that this implies that the mixer parses the H.264 to the point
> where it knows if the frame
> will be used as a reference frame.  Not a problem, and probably doesn't
> require any change to the text.

Yes. you need to look at the NRI bits in the H.264 RTP payload header to
make this determination. But you need to do similar operations in an SVC
MANE if you single stream transport (SST) payload mode. So no real
difference for the temporal scalability operation.

>
> Even within SVC (and you may have stated or implied this) you could make
> multiple encodings
> with differently-targeted layers - for example a "low" set targeting
> bandwidths <500K, and a
> high set for 500K and up.  This might help avoid the problem of too many
> layers degrading
> the bit-efficiency of the encoding.

I try to make clear that I see combining SVC for medium and fine-grained
scalability while using different SVC encodings for the large steps and
send them as simulcasts to improve the efficiency of the system as an
option.

>
>      3.2.  Simulcasting to Consuming End-Point
>
>
> This section is clearly based on the assumption (up front) that the
> streams are in fact identical
> content with different encodings.  Not a problem, especially if the
> target of the document is
> a little clearer.

Yes, we always target simulcast where the original source of the content
is the same, only the encoded representation is different.

>
> 3.3 and 3.4: These assume no mixer, right?

Yes.

>
> 4.3: negotiation of transmission side - how does an endpoint indicate
> these are "equivalent"
> streams and not "alternative" streams (different camera/mic, etc).  How
> are the differing
> encodings negotiated, or is that out of scope?  (I'd think it would be
> in-scope, because the mixer
> is the one who knows who's connected to it and their capabilities.)

I hope you got this explained when you read on. We have the SCS and SCR
grouping semantics to indicate that media lines are simulcast
alternatives to each other. The actual SSRCs in the different RTP
sessions are paired as alternatives using the SRCNAME SDES item. In
practice SRCNAME that are identical for media streams of the same media
type are alternatives to each other. The negotiation of what alternative
stream is done as regular offer/answer for each media line, thus
allowing us to avoid re-inventing that mechanism.

>
> 4.5: SDP signaling and bandwidth - b=3DAS could be used in theory for
> either direction, however
> the general offer/answer semantic is that you define what you're
> willing/ready to receive, not send,
> so b=3DAS would be the downstream direction, not the highest of upstream
> or downstream - I don't
> know of any that use it that way today.  Some use upstream (perhaps
> most), some use downstream.

Yes, I think there is great confusion around how the bandwidth modifier
actually are used. Our examples in the end do use b=3DAS as receive
direction only. The discussion in this section may not be as consistent.

I think the main point from our side is that there is need for
additional bandwidth signalling attributes to allow one to express
assymetric capabilities vs desire and do O/A negotiation of it. And do
that both for individual steams and encoding proposals and for the whole
session aggregate.

> Also there should be discussion of how RTCP TMMBR fits into this.

Likely, from my perspective TMMBR has quite clear semantics as it
applies to a particular SSRC. Thus the stream and direction for a given
TMMBR limiation is clear. But, I guess it is good to understand what the
upper limit for that stream may be.

>
> nit in 6.2:
>
>     When one have multiple actual media sources in a session,
>
> have->has

Ok

>
> 7.1.2: SSRC Multiplexing
>
> Note that "legacy" receivers will often be much happier with NO SSRC chan=
ge when the
> mixer switches streams.  While in theory they should handle them, not all=
 do.  And even
> if they do, they may cause glitches at the transition (decoder resets, et=
c).  Yes, they
> shouldn't do that.

Full agree, and as this is an RTP mixer I assume that the mixer will
have its own SSRC value that it uses to send what ever stream it mixes
or selects from the incomming the content of the media, but sent as a
consistent media stream from the mixers SSRC. Thus we don't have SSRC
switching. The issue I bring up is that of the CSRC field. If one
include CSRC then that changes.

But, in fact I think this is mostly a mote argument. Also for Session
multiplexing we likely will have this issue as using different SSRCs in
the different RTP sessions have advantages.

>
>
>    NAT Traversal Failure Rate:  Due to that one need more than a single
>        flow to be established through the NAT there is some risk that one
>        succeed in establishing the first flow but fails with one or more
>        of the additional flows.  The risk that this happens are hard to
>        quantify.  However, that risk should be fairly low as one has just
>        prior successfully established one flow from the same interfaces.
>        Thus only rare events as NAT resource overload, or selecting
>        particular port numbers that are filtered etc, should be reasons
>        for failure.
>
> As the number of flows increases, the odds of a lost packet (or congestio=
n)
> increase.  The pacing timer should help with this, but even so I believe =
the
> odds of problems will increase above a linear rate.  However, this is an
> off-the-cuff assessment and I haven't followed the ICE discussions (N zil=
lion MB
> of them...).  :-)

Maybe, and I would love to know how real the issue is. Also, to a
certain degree this is an orthogonal problem if it is real. Because it
will occur also for SSRC multiplexing as even that case has multiple
underlying UDP flows, just a few less than Session multiplexed Simulcast.


>
> 8.2.  Mixer Requests of Client streams
>
> I'd guess TMMBR of 0 may not go over well with some existing clients, or =
may be ignored.

Very much possible. We plan to follow up on this topic in the future.

>
> 8.4  Multiplexing sessions
>
> This touches on the discussions currently in rtcweb

Yes, it does. It has also been voiced in discussion around CLUE interest
of multiplexing multiple sessions on top one underlying transport flow.

My main point is that it is separate problem that needs to be addressed
somehow. If you are interested in this problem I would recommend that
one do read the next version of:
https://datatracker.ietf.org/doc/draft-perkins-rtcweb-rtp-usage/

It will be out on Monday. Which tries to make clear the issues which the
Rosenberg's proposal and discuss possible ways forward for that topic.
https://datatracker.ietf.org/doc/draft-rosenberg-rtcweb-rtpmux/


>
> 8.5.1.2 Multiple sources (telepresence)
>
> Ok, this confuses me as the discussion I thought was of simultaneous enco=
dings
> of the same data, not true different sources.

Well, this is the case that tries to make it clear how you can combine
multiple real media source, like several cameras in a telepresence room,
with simulcast.

>
> At this point I'm afraid my eyes have glazed over reading the (very nicel=
y detailed
> and explained) SDP examples.  :-)

Yes, those examples are quite beefy. But, I do hope that they will help
people understand, it also helped us making clear that we had a coherent
story for many of the different usages we see for what we propose.

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
----------------------------------------------------------------------

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


--------------080405040903000402000500--

From vsingh.ietf@gmail.com  Fri Jul  8 02:29:12 2011
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Date: Fri, 8 Jul 2011 12:29:07 +0300
From: Varun Singh <vsingh.ietf@gmail.com>
To: "=?utf-8?Q?avt=40ietf.org?=" <avt@ietf.org>
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Greetings,

We have updated the MPRTP draft and the following changes have been made:
- RTCP uses only one code point (just realized that this should be 211 as 210 is taken)
- MPRTP Header extension uses the one-byte header format instead of two-byte
- Changes in SDP consideration: introduced an "interface" attribute in SDP, as an alternative to in-band interface advertisement, some O/A examples are also included.

The detailed diff can be seen at http://tools.ietf.org/rfcdiff?url2=draft-singh-avtcore-mprtp-02.txt

Regards,
Varun

Begin forwarded message:
> 
> A new version of I-D, draft-singh-avtcore-mprtp-02.txt has been successfully submitted by Varun Singh and posted to the IETF repository.
> 
> Filename: draft-singh-avtcore-mprtp
> Revision: 02
> Title: Multipath RTP (MPRTP)
> Creation date: 2011-07-08
> WG ID: Individual Submission
> Number of pages: 34
> 
> Abstract:
> The Real-time Transport Protocol (RTP) is used to deliver real-time
> content and, along with the RTP Control Protocol (RTCP), forms the
> control channel between the sender and receiver. However, RTP and
> RTCP assume a single delivery path between the sender and receiver
> and make decisions based on the measured characteristics of this
> single path. Increasingly, endpoints are becoming multi-homed, which
> means that they are connected via multiple Internet paths. Network
> utilization can be improved when endpoints use multiple parallel
> paths for communication. The resulting increase in reliability and
> throughput can also enhance the user experience. This document
> extends the Real-time Transport Protocol (RTP) so that a single
> session can take advantage of the availability of multiple paths
> between two endpoints.
> 
> 
> 
> 
> The IETF Secretariat

-- 
Varun Singh
http://www.netlab.tkk.fi/~varun



From sato652@oki.com  Fri Jul  8 01:14:26 2011
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From: "Noriyuki SATO" <sato652@oki.com>
To: "'RFC Errata System'" <rfc-editor@rfc-editor.org>, <jo@acm.org>, <stewe@stewe.org>, <carsten.burmeister@eu.panasonic.com>, <jose.rey@eu.panasonic.com>, <gonzalo.camarillo@ericsson.com>, <rjsparks@nostrum.com>, <keith.drage@alcatel-lucent.com>, <even.roni@huawei.com>
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Subject: Re: [AVTCORE] [Technical Errata Reported] RFC4585 (2853)
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Hi All,
I don't see any problem on applying these errata.

---
OKI Electric Industry Co., Ltd.
Noriyuki Sato (sato652@oki.com)
Tel +81-6-6260-0700
Fax +81-6-6260-0770

> -----Original Message-----
> From: RFC Errata System [mailto:rfc-editor@rfc-editor.org]
> Sent: Wednesday, July 06, 2011 1:18 AM
> To: jo@acm.org; stewe@stewe.org; sato652@oki.com;
> carsten.burmeister@eu.panasonic.com; jose.rey@eu.panasonic.com;
> gonzalo.camarillo@ericsson.com; rjsparks@nostrum.com;
> keith.drage@alcatel-lucent.com; even.roni@huawei.com
> Cc: abegen@cisco.com; avt@ietf.org; rfc-editor@rfc-editor.org
> Subject: [Technical Errata Reported] RFC4585 (2853)
> 
> 
> The following errata report has been submitted for RFC4585,
> "Extended RTP Profile for Real-time Transport Control Protocol
(RTCP)-Based
> Feedback (RTP/AVPF)".
> 
> --------------------------------------
> You may review the report below and at:
> http://www.rfc-editor.org/errata_search.php?rfc=4585&eid=2853
> 
> --------------------------------------
> Type: Technical
> Reported by: Ali Begen <abegen@cisco.com>
> 
> Section: 4.2
> 
> Original Text
> -------------
> OLD:
>       rtcp-fb-param      = SP "app" [SP byte-string]
>                          / SP token [SP byte-string]
>                          / ; empty
> NEW:
>       rtcp-fb-param      = [ SP "app" [SP byte-string]
>                          / SP token [SP byte-string] ]
> 
> 
> OLD:
>       rtcp-fb-ack-param  = SP "rpsi"
>                          / SP "app" [SP byte-string]
>                          / SP token [SP byte-string]
>                          / ; empty
> NEW:
>       rtcp-fb-ack-param  = [ SP "rpsi"
>                          / SP "app" [SP byte-string]
>                          / SP token [SP byte-string] ]
> 
> 
> OLD:
>       rtcp-fb-nack-param = SP "pli"
>                          / SP "sli"
>                          / SP "rpsi"
>                          / SP "app" [SP byte-string]
>                          / SP token [SP byte-string]
>                          / ; empty
> NEW:
>       rtcp-fb-nack-param = [ SP "pli"
>                          / SP "sli"
>                          / SP "rpsi"
>                          / SP "app" [SP byte-string]
>                          / SP token [SP byte-string] ]
> 
> 
> Corrected Text
> --------------
> 
> 
> Notes
> -----
> During the AUTH48 of RFC 6285, we discovered a bug with the ABNF syntax in
RFC
> 4585. The original ABNF in Section 4.2 (as noted above) is incorrect ("/
;empty"
> is not a valid ABNF - RFC 5234 does not allow empty alternates).
> 
> The NEW text intends to fix it. There are 3 places where the same change
should
> be applied (changes are identical).
> 
> Instructions:
> -------------
> This errata is currently posted as "Reported". If necessary, please
> use "Reply All" to discuss whether it should be verified or
> rejected. When a decision is reached, the verifying party (IESG)
> can log in to change the status and edit the report, if necessary.
> 
> --------------------------------------
> RFC4585 (draft-ietf-avt-rtcp-feedback-11)
> --------------------------------------
> Title               : Extended RTP Profile for Real-time Transport Control
> Protocol (RTCP)-Based Feedback (RTP/AVPF)
> Publication Date    : July 2006
> Author(s)           : J. Ott, S. Wenger, N. Sato, C. Burmeister, J. Rey
> Category            : PROPOSED STANDARD
> Source              : Audio/Video Transport
> Area                : Real-time Applications and Infrastructure
> Stream              : IETF
> Verifying Party     : IESG


From Jose.Rey@eu.panasonic.com  Fri Jul  8 01:32:14 2011
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Subject: Re: [AVTCORE] [Technical Errata Reported] RFC4585 (2853)
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Fine with me
Cheers,

Jos=E9

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On=20
> Behalf Of RFC Errata System
> Sent: Tuesday, July 05, 2011 6:18 PM
> To: jo@acm.org; stewe@stewe.org; sato652@oki.com;=20
> carsten.burmeister@eu.panasonic.com;=20
> jose.rey@eu.panasonic.com; gonzalo.camarillo@ericsson.com;=20
> rjsparks@nostrum.com; keith.drage@alcatel-lucent.com;=20
> even.roni@huawei.com
> Cc: avt@ietf.org; rfc-editor@rfc-editor.org
> Subject: [AVTCORE] [Technical Errata Reported] RFC4585 (2853)
>=20
>=20
> The following errata report has been submitted for RFC4585,
> "Extended RTP Profile for Real-time Transport Control=20
> Protocol (RTCP)-Based Feedback (RTP/AVPF)".
>=20
> --------------------------------------
> You may review the report below and at:
> http://www.rfc-editor.org/errata_search.php?rfc=3D4585&eid=3D2853
>=20
> --------------------------------------
> Type: Technical
> Reported by: Ali Begen <abegen@cisco.com>
>=20
> Section: 4.2
>=20
> Original Text
> -------------
> OLD:
>       rtcp-fb-param      =3D SP "app" [SP byte-string]
>                          / SP token [SP byte-string]
>                          / ; empty
> NEW:
>       rtcp-fb-param      =3D [ SP "app" [SP byte-string]
>                          / SP token [SP byte-string] ]
>=20
>=20
> OLD:
>       rtcp-fb-ack-param  =3D SP "rpsi"
>                          / SP "app" [SP byte-string]
>                          / SP token [SP byte-string]
>                          / ; empty
> NEW:
>       rtcp-fb-ack-param  =3D [ SP "rpsi"
>                          / SP "app" [SP byte-string]
>                          / SP token [SP byte-string] ]
>=20
>=20
> OLD:
>       rtcp-fb-nack-param =3D SP "pli"
>                          / SP "sli"
>                          / SP "rpsi"
>                          / SP "app" [SP byte-string]
>                          / SP token [SP byte-string]
>                          / ; empty
> NEW:
>       rtcp-fb-nack-param =3D [ SP "pli"
>                          / SP "sli"
>                          / SP "rpsi"
>                          / SP "app" [SP byte-string]
>                          / SP token [SP byte-string] ]
>=20
>=20
> Corrected Text
> --------------
>=20
>=20
> Notes
> -----
> During the AUTH48 of RFC 6285, we discovered a bug with the=20
> ABNF syntax in RFC 4585. The original ABNF in Section 4.2 (as=20
> noted above) is incorrect ("/ ;empty" is not a valid ABNF -=20
> RFC 5234 does not allow empty alternates).
>=20
> The NEW text intends to fix it. There are 3 places where the=20
> same change should be applied (changes are identical).
>=20
> Instructions:
> -------------
> This errata is currently posted as "Reported". If necessary, please
> use "Reply All" to discuss whether it should be verified or
> rejected. When a decision is reached, the verifying party (IESG)
> can log in to change the status and edit the report, if necessary.=20
>=20
> --------------------------------------
> RFC4585 (draft-ietf-avt-rtcp-feedback-11)
> --------------------------------------
> Title               : Extended RTP Profile for Real-time=20
> Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)
> Publication Date    : July 2006
> Author(s)           : J. Ott, S. Wenger, N. Sato, C.=20
> Burmeister, J. Rey
> Category            : PROPOSED STANDARD
> Source              : Audio/Video Transport
> Area                : Real-time Applications and Infrastructure
> Stream              : IETF
> Verifying Party     : IESG
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
>=20
>=20


Panasonic R=26D Center Germany GmbH
63225 Langen=2C Hessen=2C Germany
Reg=3A AG Offenbach =28Hessen=29 HRB 33974
Managing Director=3A Thomas Micke



From sandra.sendra.upv@gmail.com  Fri Jul  8 10:33:31 2011
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Subject: [AVTCORE] [DEADLINE EXTENDED] CFP - IEEE GLOBECOM 2011 - Smart Communication Protocols & Algorithms (SCPA 2011)
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Apologies for crossposting

------------------ CALL FOR PAPERS -----------------

Deadline extended to July 17, 2011

Smart Communication Protocols and Algorithms (SCPA 2011)
December 5-9, 2011, Huston, Texas (USA)
In conjunction with Globecom 2011
http://scpa.it.ubi.pt/

Selected papers will be invited to the Special Issue on Smart Protocols and Algorithms of the International Journal Network Protocols and Algorithms (ISSN 1943-3581)

Communication protocols and algorithms are needed to communicate network devices and exchange data between them. The appearance of new technologies usually comes with a protocol procedure and communication rules that allows data communication while taking profit of this new technology. Recent advances in hardware and communication mediums allow proposing new rules, conventions and data structures which could be used by network devices to communicate across the network. Moreover, devices with higher processing capacity let us include more complex algorithms that can be used by the network device to enhance the communication procedure.

Smart communication protocols and algorithms make use of several methods and techniques (such as machine learning techniques, decision making techniques, knowledge representation, network management, network optimization, problem solution techniques, and so on), to communicate the network devices to transfer data between them. They can be used to perceive the network conditions, or the user behavior, in order to dynamically plan, adapt, decide, take the appropriate actions, and learn from the consequences of its actions. The algorithms can make use of the information gathered from the protocol in order to sense the environment, plan actions according to the input, take consciousness of what is happening in the environment, and take the appropriate decisions using a reasoning engine. Goals such as decide which scenario fits best its end-to-end purpose, or environment prediction, can be achieved with smart protocols and algorithms. Moreover, they could learn from the past and !
 use this knowledge to improve futur
e decisions.

In this workshop, researchers are encouraged to submit papers focused on the design, development, analysis or optimization of smart communication protocols or algorithms at any communication layer. Algorithms and protocols based on artificial intelligence techniques for network management, network monitoring, quality of service enhancement, performance optimization and network secure are included in the workshop. 

We welcome technical papers presenting analytical research, simulations, practical results, position papers addressing the pros and cons of specific proposals, and papers addressing the key problems and solutions. The topics suggested by the conference can be discussed in term of concepts, state of the art, standards, deployments, implementations, running experiments and applications. 

Topics of interest:
Authors are invited to submit complete unpublished papers, which are not under review in any other conference or journal, including, but are not limited to, the following topic areas:
• Smart network protocols and algorithms for multimedia delivery
• Application layer, transport layer and network layer cognitive protocols
• Cognitive radio network protocols and algorithms
• Automatic protocols and algorithms for environment prediction.
• Algorithms and protocols to predict data network states. 
• Intelligent synchronization techniques for network protocols and algorithms
• Smart protocols and algorithms for e-health
• Software applications for smart algorithms design and development.
• Dynamic protocols based on the perception of their performance 
• Smart protocols and algorithms for Smartgrids
• Protocols and algorithms focused on building conclusions for taking the appropriate actions.
• Smart Automatic and self-autonomous ad-hoc and sensor networks.
• Artificial intelligence applied in protocols and algorithms for wireless, mobile and dynamic networks.
• Smart security protocols and algorithms
• Smart cryptographic algorithms for communication
• Artificial intelligence applied to power efficiency and energy saving protocols and algorithms
• Smart routing and switching protocols and algorithms
• Cognitive protocol and algorithm models for saving communication costs.
• Any kind of intelligent technique applied to QoS, content delivery, network Monitoring and network management.
• Smart collaborative protocols and algorithms
• Problem recognition and problem solving protocols 
• Genetic algorithms, fuzzy logic and neural networks applied to communication protocols and algorithms

Important Dates
Paper Submission: 17 July 2011 (extended deadline)
Paper Acceptance: 14 August 2011
Camera-Ready: 7 September 2011

Submission guidelines:
All submissions must be full papers in PDF format and uploaded on EDAS (http://edas.info//N10879).
They must not exceed 10 single-spaced, double-column pages using 10 pt size fonts on 8.5 x 11 inch pages in IEEE style format.


General Chairs
Jaime Lloret, Univ. Politecnica Valencia, Spain
Joel Rodrigues, Inst. of Telecommunications, Univ. of Beira Interior, Pt

TPC Chairs
Nidal Nasser, University of Guelph, Canada
Ivan Stojmenovic, University of Ottawa, Canada

Poster Chairs
Binod Vaidya, University of Ottawa, Canada
Bin Wei, AT&T, USA

Industry Chairs
Haohong Wang, Cisco
Chi-Ming Chen, AT&T, USA

Publicity Chairs
Sandra Sendra, Univ. Politecnica Valencia, Spain
Mbaye B. Sene; Cheikh Anta Diop Univ., Dakar, Senegal

Web Chair
Miguel Garcia, Univ. Politecnica Valencia, Spain

From Even.roni@huawei.com  Fri Jul  8 13:58:07 2011
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Date: Fri, 08 Jul 2011 23:55:30 +0300
From: Roni Even <Even.roni@huawei.com>
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Subject: [AVTCORE] AVTcore draft agenda for IETF81
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Hi,

Here is the proposed agenda for the AVTcore session in Quebec.

Please review

Roni Even

AVTcore co-chair


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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal>Hi,<o:p></o:p></p><p class=MsoNormal>Here is the proposed agenda for the AVTcore session in Quebec.<o:p></o:p></p><p class=MsoNormal>Please review<o:p></o:p></p><p class=MsoNormal>Roni Even<o:p></o:p></p><p class=MsoNormal>AVTcore co-chair<o:p></o:p></p></div></body></html>

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Audio/Video Transport Core Maintenance  (AVTCore) Working Group
===============================================================

CHAIRS:  Magnus Westerlund       <magnus.westerlund@ericsson.com>
         Roni Even         <even.roni@huawei.com>

AGENDA

Wednesday, 27 July 2011 at 9:00-11:30 (208 AB)
----------------------------------------------
09:00   AVTCore Status Update                               (Chairs, 20)

09:20   Payload  Status Update                      (Payload Chairs, 15)
        draft-ramalho-payload-g7110-00

09:35   Encryption of Header Extensions in SRTP             (Lennox, 15)
        draft-ietf-avtcore-srtp-encrypted-header-ext-00

09:50   RTCP Extension for Feedback Suppression Indication  (Qin Wu, 15)
        draft-ietf-avtcore-feedback-supression-rtp-00

10:05   Monitoring Architecture for RTP                     (Qin Wu, 15)
        draft-ietf-avtcore-monarch-03

10:20   RTP Multiple Stream Sessions and Simulcast(Magnus Westerlund , 30)
        draft-westerlund-avtcore-multistream-and-simulcast-00

10:50   Delayed Duplication Attribute and Duplication Grouping in SDP(Begen, 15)
        draft-begen-mmusic-temporal-interleaving-02
        draft-begen-mmusic-redundancy-grouping-01

11:05   Multipath RTP                                  (Varun Singh, 10)
        draft-singh-avtcore-mprtp-02

11:15   End

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Date: Sun, 10 Jul 2011 01:02:14 +0300
From: Roni Even <Even.roni@huawei.com>
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Hi,

I uploaded the agenda to the meeting page
http://www.ietf.org/proceedings/81/agenda/avtcore.html . 

Attached is the txt version

Roni Even

AVTcore co-chair


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<html xmlns:v="urn:schemas-microsoft-com:vml" xmlns:o="urn:schemas-microsoft-com:office:office" xmlns:w="urn:schemas-microsoft-com:office:word" xmlns:m="http://schemas.microsoft.com/office/2004/12/omml" xmlns="http://www.w3.org/TR/REC-html40"><head><META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=us-ascii"><meta name=Generator content="Microsoft Word 12 (filtered medium)"><style><!--
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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal>Hi,<o:p></o:p></p><p class=MsoNormal>I uploaded the agenda to the meeting page <a href="http://www.ietf.org/proceedings/81/agenda/avtcore.html">http://www.ietf.org/proceedings/81/agenda/avtcore.html</a> . <o:p></o:p></p><p class=MsoNormal>Attached is the txt version<o:p></o:p></p><p class=MsoNormal>Roni Even<o:p></o:p></p><p class=MsoNormal>AVTcore co-chair<o:p></o:p></p></div></body></html>

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Audio/Video Transport Core Maintenance  (AVTCore) Working Group
===============================================================

CHAIRS:  Magnus Westerlund       <magnus.westerlund@ericsson.com>
         Roni Even         <even.roni@huawei.com>

AGENDA

Wednesday, 27 July 2011 at 9:00-11:30 (208 AB)
----------------------------------------------
09:00   AVTCore Status Update                               (Chairs, 20)

09:20   Payload  Status Update                      (Payload Chairs, 15)
        draft-ramalho-payload-g7110-00

09:35   Encryption of Header Extensions in SRTP             (Lennox, 15)
        draft-ietf-avtcore-srtp-encrypted-header-ext-00

09:50   RTCP Extension for Feedback Suppression Indication  (Qin Wu, 15)
        draft-ietf-avtcore-feedback-supression-rtp-00

10:05   Monitoring Architecture for RTP                     (Qin Wu, 15)
        draft-ietf-avtcore-monarch-03

10:20   RTP Multiple Stream Sessions and Simulcast(Magnus Westerlund , 30)
        draft-westerlund-avtcore-multistream-and-simulcast-00

10:50   Retransmission for SSM Sessions                      (Begen, 10)
        draft-vancaenegem-avtcore-retransmission-for-ssm-00

11:00   Delayed Duplication Attribute and Duplication Grouping in SDP(Begen, 15)
        draft-begen-mmusic-temporal-interleaving-02
        draft-begen-mmusic-redundancy-grouping-01

11:15   Multipath RTP                                  (Varun Singh, 10)
        draft-singh-avtcore-mprtp-02

11:25   End

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Subject: [AVTCORE] I-D Action: draft-ietf-avtcore-feedback-supression-rtp-05.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           : RTCP Extension for Third-party Loss Report
	Author(s)       : Qin Wu
                          Frank Xia
                          Roni Even
	Filename        : draft-ietf-avtcore-feedback-supression-rtp-05.txt
	Pages           : 17
	Date            : 2011-07-10

   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 a
   feedback target is aware of some loss event, allowing them to
   suppress feedback.  Associated SDP signalling is also defined.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avtcore-feedback-supression-=
rtp-05.txt

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This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-avtcore-feedback-supression-r=
tp-05.txt

From Even.roni@huawei.com  Mon Jul 11 00:58:48 2011
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Date: Mon, 11 Jul 2011 10:56:04 +0300
From: Roni Even <Even.roni@huawei.com>
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Subject: Re: [AVTCORE] Fwd: I-D Action:	draft-westerlund-avtcore-multistream-and-simulcast-00.txt
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Hi Magnus,
Read the draft, I liked the analysis on the different multiplexing of RTP
streams (Session, SSRC and payload type).

Some review comments

1. The introduction talks about multistream and simulcast. I noticed that it
only talks about how to define the alternatives in terms of the stream but
is not saying how to understand the semantics of the streams, for example
which camera in Telepresence and in simulcast you assume that all
alternative provide the same view (just resolution change, but can also use
different view of the room). This is fine and is being discussed in CLUE at
the moment and should be out of scope here but it may cause the SDP to be
even larger. 

2. On the SDP syntax it may be better when you have more than one m-line to
have all the descriptions which repeats in all m-line appear st the SDP
session level with local identifier and use the identifier in the m-lines.
Maybe based on grouping.


3. In section 1.1 "Todays SDP signalling support for this is basically the
directionality attribute which indicates an end-point intend to send media
or not.  No indication of how many media streams." My understanding so far
was that the number of PT in the m-line is the maximum number of streams
that can be sent in an RTP session. This  goes also to the discussion on
support of multiple SSRCs. I think that today video conferencing endpoint is
doing a second offer answer to fix the PT it will use based on the first
offer answer exchange. 

4. The above still does not address the multiple SSRC for the same PT and I
already mentioned it in the past that I think that a lot of product today
will not handle it good. I think that this is a rare case today at least in
centralized video conferencing since the MCU (or RTP mixer if exist) will
use its own SSRC to send the mixed/switched media.

5. In section 3.2 second paragraph you talk about the receivers controlling
what the relay will send but this will make the relay something else than
RTP translator.

6. On the bandwidth support. Currently the definition is the b= is a
receiver capability. I am not sure why you need the sender one, you claim
that there is no way to control the actual bw but the receiver can use RTCP
CCM flow control to tell the sender to reduce the rate. As for the syntax,
my preference is for TIAS semantics and not the one you proposed. 

7. In the last sentence of 7.2.1 you say "This enables a simple fallback
solution to exclude a legacy client from all simulcast versions except one,
whichever is most suitable for the application". My view is that currently
when there are several m-lines of the same media type there is no definition
which one to choose if the answerer do not understand the semantics of the
offer and can support only one stream.


Roni Even



> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
> Magnus Westerlund
> Sent: Tuesday, July 05, 2011 1:54 PM
> To: IETF AVTCore WG
> Subject: [AVTCORE] Fwd: I-D Action: draft-westerlund-avtcore-
> multistream-and-simulcast-00.txt
> 
> WG,
> 
> As an individual Bo Burman and I have written this internet draft that
> discusses multistream and simulcast issues and proposes a set of
> extensions to resolve the issues. Feedback is much appreciated.
> 
> Cheers
> 
> Magnus


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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           : Explicit Congestion Notification (ECN) for RTP over UDP
	Author(s)       : Magnus Westerlund
                          Ingemar Johansson
                          Colin Perkins
                          Piers O&#39;Hanlon
                          Ken Carlberg
	Filename        : draft-ietf-avtcore-ecn-for-rtp-03.txt
	Pages           : 55
	Date            : 2011-07-11

   This memo specifies how Explicit Congestion Notification (ECN) can be
   used with the Real-time Transport Protocol (RTP) running over UDP,
   using RTP Control Protocol (RTCP) as a feedback mechanism.  It
   defines a new RTCP Extended Report (XR) block for periodic ECN
   feedback, a new RTCP transport feedback message for timely reporting
   of congestion events, and a Session Traversal Utilities for NAT
   (STUN) extension used in the optional initilization method using
   Interactive Connectivity Establishment (ICE).  Signalling and
   procedures for negotiation of capabilities and initilization methods
   are also defined.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avtcore-ecn-for-rtp-03.txt

Internet-Drafts are also available by anonymous FTP at:
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This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-avtcore-ecn-for-rtp-03.txt

From magnus.westerlund@ericsson.com  Mon Jul 11 08:07:59 2011
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Subject: Re: [AVTCORE] I-D Action: draft-ietf-avtcore-ecn-for-rtp-03.txt
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Hi,

This update only addresses some editorial minor issues we authors have
found in the time since the last update. That include actually removing
the open issues section listing issues resolved in the previous version.

The authors believe this version being ready for WG last call.

The diff against previous version:
http://tools.ietf.org/rfcdiff?url2=draft-ietf-avtcore-ecn-for-rtp-03

Cheers

Magnus

On 2011-07-11 17:03, 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           : Explicit Congestion Notification (ECN) for RTP over UDP
> 	Author(s)       : Magnus Westerlund
>                           Ingemar Johansson
>                           Colin Perkins
>                           Piers O&#39;Hanlon
>                           Ken Carlberg
> 	Filename        : draft-ietf-avtcore-ecn-for-rtp-03.txt
> 	Pages           : 55
> 	Date            : 2011-07-11
> 
>    This memo specifies how Explicit Congestion Notification (ECN) can be
>    used with the Real-time Transport Protocol (RTP) running over UDP,
>    using RTP Control Protocol (RTCP) as a feedback mechanism.  It
>    defines a new RTCP Extended Report (XR) block for periodic ECN
>    feedback, a new RTCP transport feedback message for timely reporting
>    of congestion events, and a Session Traversal Utilities for NAT
>    (STUN) extension used in the optional initilization method using
>    Interactive Connectivity Establishment (ICE).  Signalling and
>    procedures for negotiation of capabilities and initilization methods
>    are also defined.
> 
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-avtcore-ecn-for-rtp-03.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-ecn-for-rtp-03.txt
> _______________________________________________
> 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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On 07/11/2011 10:03 AM, 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           : Explicit Congestion Notification (ECN) for RTP over UDP
> 	Author(s)       : Magnus Westerlund
>                            Ingemar Johansson
>                            Colin Perkins
>                            Piers O&#39;Hanlon
>                            Ken Carlberg
> 	Filename        : draft-ietf-avtcore-ecn-for-rtp-03.txt
> 	Pages           : 55
> 	Date            : 2011-07-11
>
>     This memo specifies how Explicit Congestion Notification (ECN) can be
>     used with the Real-time Transport Protocol (RTP) running over UDP,
>     using RTP Control Protocol (RTCP) as a feedback mechanism.  It
>     defines a new RTCP Extended Report (XR) block for periodic ECN
>     feedback, a new RTCP transport feedback message for timely reporting
>     of congestion events, and a Session Traversal Utilities for NAT
>     (STUN) extension used in the optional initilization method using
>     Interactive Connectivity Establishment (ICE).  Signalling and
>     procedures for negotiation of capabilities and initilization methods
>     are also defined.

Editorial note: In two places here, 'initilization' should be 
'initialization'.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

From internet-drafts@ietf.org  Mon Jul 11 08:43:53 2011
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Subject: [AVTCORE] I-D Action: draft-ietf-avtcore-ecn-for-rtp-04.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           : Explicit Congestion Notification (ECN) for RTP over UDP
	Author(s)       : Magnus Westerlund
                          Ingemar Johansson
                          Colin Perkins
                          Piers O&#39;Hanlon
                          Ken Carlberg
	Filename        : draft-ietf-avtcore-ecn-for-rtp-04.txt
	Pages           : 55
	Date            : 2011-07-11

   This memo specifies how Explicit Congestion Notification (ECN) can be
   used with the Real-time Transport Protocol (RTP) running over UDP,
   using RTP Control Protocol (RTCP) as a feedback mechanism.  It
   defines a new RTCP Extended Report (XR) block for periodic ECN
   feedback, a new RTCP transport feedback message for timely reporting
   of congestion events, and a Session Traversal Utilities for NAT
   (STUN) extension used in the optional initialization method using
   Interactive Connectivity Establishment (ICE).  Signalling and
   procedures for negotiation of capabilities and initialization methods
   are also defined.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avtcore-ecn-for-rtp-04.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-ecn-for-rtp-04.txt

From randell-ietf@jesup.org  Mon Jul 11 08:44:15 2011
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Subject: Re: [AVTCORE] Fwd: I-D	Action:	draft-westerlund-avtcore-multistream-and-simulcast-00.txt
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On 7/11/2011 3:56 AM, Roni Even wrote:
> 3. In section 1.1 "Todays SDP signalling support for this is basically the
> directionality attribute which indicates an end-point intend to send media
> or not.  No indication of how many media streams." My understanding so far
> was that the number of PT in the m-line is the maximum number of streams
> that can be sent in an RTP session.

I've never heard of this (perhaps it's an issue with the meaning of 
stream in "number of streams
that can be sent in an RTP session").  The number of distinct payloads, 
yes.  But in theory
(though not well supported, if at all) you could have a mixer providing 
N "streams", all of the
same payload or different ones (with different SSRCs).

I think it's just a terminology issue (and given the relative positions, 
I'm probably the one who's
wrong).  :-)  But I'll note your comment isn't clear to me, and it 
should be.

> This  goes also to the discussion on
> support of multiple SSRCs. I think that today video conferencing endpoint is
> doing a second offer answer to fix the PT it will use based on the first
> offer answer exchange.

They may be; I mostly work with devices designed for point-to-point - 
but that means that
entire conferences either get locked down to the 
lowest-common-denominator codec, or
it demands the MCU transcode.  I personally prefer to leave multiple 
payloads/codecs
to allow endpoints to switch payloads in order to help adapt to various 
things, like network
conditions.  In theory, an endpoint could even switch between voice 
codecs and
better-at-music-audio-codecs when they detect music, as a random 
non-network example.
I don't know if the non-network use-cases would be compelling, but the 
network adaptation
are compelling, given the time (and possible failures) required to 
re-INVITE.

> 4. The above still does not address the multiple SSRC for the same PT and I
> already mentioned it in the past that I think that a lot of product today
> will not handle it good. I think that this is a rare case today at least in
> centralized video conferencing since the MCU (or RTP mixer if exist) will
> use its own SSRC to send the mixed/switched media.

Some devices reuse the same incoming ports for multiple sessions, both 
within a call and
between different calls they're involved in or are acting as a 
conference bridge.  As mentioned
previously, you're right that many endpoints don't deal well (or 
consistently) with multiple
SSRCs in a session.

> 5. In section 3.2 second paragraph you talk about the receivers controlling
> what the relay will send but this will make the relay something else than
> RTP translator.
>
> 6. On the bandwidth support. Currently the definition is the b= is a
> receiver capability. I am not sure why you need the sender one, you claim
> that there is no way to control the actual bw but the receiver can use RTCP
> CCM flow control to tell the sender to reduce the rate. As for the syntax,
> my preference is for TIAS semantics and not the one you proposed.

Well, knowing what rate the sender plans to use *could* be useful in 
deciding how many
resources (memory, CPU/DSP horsepower, etc) might be needed and so help 
the other side
plan or (if it's answering) help it decide how to answer (number of 
streams, codecs to accept, etc).

The fact that it could be useful doesn't always mean that it's a good 
idea to complicate the protocol
to support it directly, however.

> 7. In the last sentence of 7.2.1 you say "This enables a simple fallback
> solution to exclude a legacy client from all simulcast versions except one,
> whichever is most suitable for the application". My view is that currently
> when there are several m-lines of the same media type there is no definition
> which one to choose if the answerer do not understand the semantics of the
> offer and can support only one stream.

Agreed - there's no definition; however almost all clients that don't 
support multiple m= lines
of the same type will select the first one, or the first one that they 
can find a matching payload for.

-- 
Randell Jesup
randell-ietf@jesup.org


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Hi,

A quick update fixing the issues that Kevin noticed and a few more
spelling issues.

Cheers

Magnus

On 2011-07-11 17:43, 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           : Explicit Congestion Notification (ECN) for RTP over UDP
> 	Author(s)       : Magnus Westerlund
>                           Ingemar Johansson
>                           Colin Perkins
>                           Piers O&#39;Hanlon
>                           Ken Carlberg
> 	Filename        : draft-ietf-avtcore-ecn-for-rtp-04.txt
> 	Pages           : 55
> 	Date            : 2011-07-11
> 
>    This memo specifies how Explicit Congestion Notification (ECN) can be
>    used with the Real-time Transport Protocol (RTP) running over UDP,
>    using RTP Control Protocol (RTCP) as a feedback mechanism.  It
>    defines a new RTCP Extended Report (XR) block for periodic ECN
>    feedback, a new RTCP transport feedback message for timely reporting
>    of congestion events, and a Session Traversal Utilities for NAT
>    (STUN) extension used in the optional initialization method using
>    Interactive Connectivity Establishment (ICE).  Signalling and
>    procedures for negotiation of capabilities and initialization methods
>    are also defined.
> 
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-avtcore-ecn-for-rtp-04.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-ecn-for-rtp-04.txt
> _______________________________________________
> I-D-Announce mailing list
> I-D-Announce@ietf.org
> https://www.ietf.org/mailman/listinfo/i-d-announce
> Internet-Draft directories: http://www.ietf.org/shadow.html
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> 


-- 

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
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SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
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From Even.roni@huawei.com  Mon Jul 11 09:58:40 2011
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Date: Mon, 11 Jul 2011 19:55:54 +0300
From: Roni Even <Even.roni@huawei.com>
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Subject: Re: [AVTCORE] Fwd:	I-D	Action:	draft-westerlund-avtcore-multistream-and-simulcast-00.txt
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Inline
Roni

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
> Randell Jesup
> Sent: Monday, July 11, 2011 6:44 PM
> To: avt@ietf.org
> Subject: Re: [AVTCORE] Fwd: I-D Action: draft-westerlund-avtcore-
> multistream-and-simulcast-00.txt
> 
> On 7/11/2011 3:56 AM, Roni Even wrote:
> > 3. In section 1.1 "Todays SDP signalling support for this is
> basically the
> > directionality attribute which indicates an end-point intend to send
> media
> > or not.  No indication of how many media streams." My understanding
> so far
> > was that the number of PT in the m-line is the maximum number of
> streams
> > that can be sent in an RTP session.
> 
> I've never heard of this (perhaps it's an issue with the meaning of
> stream in "number of streams
> that can be sent in an RTP session").  The number of distinct payloads,
> yes.  But in theory
> (though not well supported, if at all) you could have a mixer providing
> N "streams", all of the
> same payload or different ones (with different SSRCs).
You are if you do ssrc multiplex, I assumed that if you do session multiplex
you do not do ssrc multiplex but of course this is allowed.
> 
> I think it's just a terminology issue (and given the relative
> positions,
> I'm probably the one who's
> wrong).  :-)  But I'll note your comment isn't clear to me, and it
> should be.
> 
> > This  goes also to the discussion on
> > support of multiple SSRCs. I think that today video conferencing
> endpoint is
> > doing a second offer answer to fix the PT it will use based on the
> first
> > offer answer exchange.
> 
> They may be; I mostly work with devices designed for point-to-point -
> but that means that
> entire conferences either get locked down to the
> lowest-common-denominator codec, or
> it demands the MCU transcode.  I personally prefer to leave multiple
> payloads/codecs
> to allow endpoints to switch payloads in order to help adapt to various
> things, like network
> conditions.  In theory, an endpoint could even switch between voice
> codecs and
> better-at-music-audio-codecs when they detect music, as a random
> non-network example.
> I don't know if the non-network use-cases would be compelling, but the
> network adaptation
> are compelling, given the time (and possible failures) required to
> re-INVITE.

This is the same point that EP will try to keep one PT in a point to point
or MCU that terminates the call leg, and they do not expect SSRC
multiplexing as mentioned in the next point
> 


> > 4. The above still does not address the multiple SSRC for the same PT
> and I
> > already mentioned it in the past that I think that a lot of product
> today
> > will not handle it good. I think that this is a rare case today at
> least in
> > centralized video conferencing since the MCU (or RTP mixer if exist)
> will
> > use its own SSRC to send the mixed/switched media.
> 
> Some devices reuse the same incoming ports for multiple sessions, both
> within a call and
> between different calls they're involved in or are acting as a
> conference bridge.  As mentioned
> previously, you're right that many endpoints don't deal well (or
> consistently) with multiple
> SSRCs in a session.
> 
> > 5. In section 3.2 second paragraph you talk about the receivers
> controlling
> > what the relay will send but this will make the relay something else
> than
> > RTP translator.
> >
> > 6. On the bandwidth support. Currently the definition is the b= is a
> > receiver capability. I am not sure why you need the sender one, you
> claim
> > that there is no way to control the actual bw but the receiver can
> use RTCP
> > CCM flow control to tell the sender to reduce the rate. As for the
> syntax,
> > my preference is for TIAS semantics and not the one you proposed.
> 
> Well, knowing what rate the sender plans to use *could* be useful in
> deciding how many
> resources (memory, CPU/DSP horsepower, etc) might be needed and so help
> the other side
> plan or (if it's answering) help it decide how to answer (number of
> streams, codecs to accept, etc).
> 
> The fact that it could be useful doesn't always mean that it's a good
> idea to complicate the protocol
> to support it directly, however.
> 
> > 7. In the last sentence of 7.2.1 you say "This enables a simple
> fallback
> > solution to exclude a legacy client from all simulcast versions
> except one,
> > whichever is most suitable for the application". My view is that
> currently
> > when there are several m-lines of the same media type there is no
> definition
> > which one to choose if the answerer do not understand the semantics
> of the
> > offer and can support only one stream.
> 
> Agreed - there's no definition; however almost all clients that don't
> support multiple m= lines
> of the same type will select the first one, or the first one that they
> can find a matching payload for.
> 
> --
> Randell Jesup
> randell-ietf@jesup.org
> 
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt


From Even.roni@huawei.com  Tue Jul 12 01:32:28 2011
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Date: Tue, 12 Jul 2011 11:27:49 +0300
From: Roni Even <Even.roni@huawei.com>
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Subject: [AVTCORE] comments on draft-ietf-avtcore-srtp-encrypted-header-ext-00.txt
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Hi,
I read the draft, some comments:

1. I think it needs an offer answer section. We need to specify the behavior
when an offer to encrypt the header is sent and the answerer do not support
the encryption header extension. Can alternatives, one with encryption and
one without be offered. 

2. Extmap can be a session level attribute. What about encryption, is it
only a media level.

In general are the offer answer rules from RFC 5285 applicable here and how.


Regards
Roni Even


From jonathan@vidyo.com  Tue Jul 12 11:37:51 2011
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From: Jonathan Lennox <jonathan@vidyo.com>
To: Roni Even <Even.roni@huawei.com>
Date: Tue, 12 Jul 2011 14:37:46 -0400
Thread-Topic: comments on draft-ietf-avtcore-srtp-encrypted-header-ext-00.txt
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The intention is that from the point of view of an answerer that doesn't un=
derstand this spec, urn:ietf:params:rtp-hdrext:encrypt is just another unkn=
own header extension element URI, so it will negotiate not to use it using =
the standard RFC 5285 mechanisms.

It's thus also possible to offer encrypted and non-encrypted versions of th=
e same header extension element, if that's acceptable for security reasons.

I can add some explicit text explaining this in the next revision of the dr=
aft.

On Jul 12, 2011, at 4:27 AM, Roni Even wrote:

> Hi,
> I read the draft, some comments:
>=20
> 1. I think it needs an offer answer section. We need to specify the behav=
ior
> when an offer to encrypt the header is sent and the answerer do not suppo=
rt
> the encryption header extension. Can alternatives, one with encryption an=
d
> one without be offered.=20
>=20
> 2. Extmap can be a session level attribute. What about encryption, is it
> only a media level.
>=20
> In general are the offer answer rules from RFC 5285 applicable here and h=
ow.
>=20
>=20
> Regards
> Roni Even
>=20

--
Jonathan Lennox
jonathan@vidyo.com



From Even.roni@huawei.com  Tue Jul 12 12:10:45 2011
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Date: Tue, 12 Jul 2011 22:07:44 +0300
From: Roni Even <Even.roni@huawei.com>
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Subject: Re: [AVTCORE] comments on draft-ietf-avtcore-srtp-encrypted-header-ext-00.txt
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Hi Jonathan,
I think it will be good to have some such text.
What about the question about using it at the session level?

Thanks
Roni

> -----Original Message-----
> From: Jonathan Lennox [mailto:jonathan@vidyo.com]
> Sent: Tuesday, July 12, 2011 9:38 PM
> To: Roni Even
> Cc: avt@ietf.org
> Subject: Re: comments on draft-ietf-avtcore-srtp-encrypted-header-ext-
> 00.txt
> 
> The intention is that from the point of view of an answerer that
> doesn't understand this spec, urn:ietf:params:rtp-hdrext:encrypt is
> just another unknown header extension element URI, so it will negotiate
> not to use it using the standard RFC 5285 mechanisms.
> 
> It's thus also possible to offer encrypted and non-encrypted versions
> of the same header extension element, if that's acceptable for security
> reasons.
> 
> I can add some explicit text explaining this in the next revision of
> the draft.
> 
> On Jul 12, 2011, at 4:27 AM, Roni Even wrote:
> 
> > Hi,
> > I read the draft, some comments:
> >
> > 1. I think it needs an offer answer section. We need to specify the
> behavior
> > when an offer to encrypt the header is sent and the answerer do not
> support
> > the encryption header extension. Can alternatives, one with
> encryption and
> > one without be offered.
> >
> > 2. Extmap can be a session level attribute. What about encryption, is
> it
> > only a media level.
> >
> > In general are the offer answer rules from RFC 5285 applicable here
> and how.
> >
> >
> > Regards
> > Roni Even
> >
> 
> --
> Jonathan Lennox
> jonathan@vidyo.com



From jonathan@vidyo.com  Tue Jul 12 12:45:54 2011
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From: Jonathan Lennox <jonathan@vidyo.com>
To: Roni Even <Even.roni@huawei.com>
Date: Tue, 12 Jul 2011 15:45:48 -0400
Thread-Topic: comments on draft-ietf-avtcore-srtp-encrypted-header-ext-00.txt
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Subject: Re: [AVTCORE] comments on draft-ietf-avtcore-srtp-encrypted-header-ext-00.txt
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Hi, Roni --

If extensions are specified at the session level, it's straightforward if a=
ll the media streams are using SRTP -- the standard RFC 5285 rules apply.

If there's a mix of SRTP and non-SRTP media streams, either we can say that=
 encrypted extension headers MUST be ignored for non-SRTP streams, or we ca=
n forbid specifying encrypted extension headers at the session level in thi=
s case.  I'm inclined towards the former, but would be fine with either opt=
ion.

On Jul 12, 2011, at 3:07 PM, Roni Even wrote:

> Hi Jonathan,
> I think it will be good to have some such text.
> What about the question about using it at the session level?
>=20
> Thanks
> Roni
>=20
>> -----Original Message-----
>> From: Jonathan Lennox [mailto:jonathan@vidyo.com]
>> Sent: Tuesday, July 12, 2011 9:38 PM
>> To: Roni Even
>> Cc: avt@ietf.org
>> Subject: Re: comments on draft-ietf-avtcore-srtp-encrypted-header-ext-
>> 00.txt
>>=20
>> The intention is that from the point of view of an answerer that
>> doesn't understand this spec, urn:ietf:params:rtp-hdrext:encrypt is
>> just another unknown header extension element URI, so it will negotiate
>> not to use it using the standard RFC 5285 mechanisms.
>>=20
>> It's thus also possible to offer encrypted and non-encrypted versions
>> of the same header extension element, if that's acceptable for security
>> reasons.
>>=20
>> I can add some explicit text explaining this in the next revision of
>> the draft.
>>=20
>> On Jul 12, 2011, at 4:27 AM, Roni Even wrote:
>>=20
>>> Hi,
>>> I read the draft, some comments:
>>>=20
>>> 1. I think it needs an offer answer section. We need to specify the
>> behavior
>>> when an offer to encrypt the header is sent and the answerer do not
>> support
>>> the encryption header extension. Can alternatives, one with
>> encryption and
>>> one without be offered.
>>>=20
>>> 2. Extmap can be a session level attribute. What about encryption, is
>> it
>>> only a media level.
>>>=20
>>> In general are the offer answer rules from RFC 5285 applicable here
>> and how.
>>>=20
>>>=20
>>> Regards
>>> Roni Even
>>>=20
>>=20
>> --
>> Jonathan Lennox
>> jonathan@vidyo.com
>=20
>=20

--
Jonathan Lennox
jonathan@vidyo.com



From Even.roni@huawei.com  Tue Jul 12 14:13:35 2011
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Date: Wed, 13 Jul 2011 00:10:51 +0300
From: Roni Even <Even.roni@huawei.com>
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Subject: Re: [AVTCORE] comments on draft-ietf-avtcore-srtp-encrypted-header-ext-00.txt
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Jonathan,
I am OK with the first option.
Roni

> -----Original Message-----
> From: Jonathan Lennox [mailto:jonathan@vidyo.com]
> Sent: Tuesday, July 12, 2011 10:46 PM
> To: Roni Even
> Cc: avt@ietf.org
> Subject: Re: comments on draft-ietf-avtcore-srtp-encrypted-header-ext-
> 00.txt
> 
> Hi, Roni --
> 
> If extensions are specified at the session level, it's straightforward
> if all the media streams are using SRTP -- the standard RFC 5285 rules
> apply.
> 
> If there's a mix of SRTP and non-SRTP media streams, either we can say
> that encrypted extension headers MUST be ignored for non-SRTP streams,
> or we can forbid specifying encrypted extension headers at the session
> level in this case.  I'm inclined towards the former, but would be fine
> with either option.
> 
> On Jul 12, 2011, at 3:07 PM, Roni Even wrote:
> 
> > Hi Jonathan,
> > I think it will be good to have some such text.
> > What about the question about using it at the session level?
> >
> > Thanks
> > Roni
> >
> >> -----Original Message-----
> >> From: Jonathan Lennox [mailto:jonathan@vidyo.com]
> >> Sent: Tuesday, July 12, 2011 9:38 PM
> >> To: Roni Even
> >> Cc: avt@ietf.org
> >> Subject: Re: comments on draft-ietf-avtcore-srtp-encrypted-header-
> ext-
> >> 00.txt
> >>
> >> The intention is that from the point of view of an answerer that
> >> doesn't understand this spec, urn:ietf:params:rtp-hdrext:encrypt is
> >> just another unknown header extension element URI, so it will
> negotiate
> >> not to use it using the standard RFC 5285 mechanisms.
> >>
> >> It's thus also possible to offer encrypted and non-encrypted
> versions
> >> of the same header extension element, if that's acceptable for
> security
> >> reasons.
> >>
> >> I can add some explicit text explaining this in the next revision of
> >> the draft.
> >>
> >> On Jul 12, 2011, at 4:27 AM, Roni Even wrote:
> >>
> >>> Hi,
> >>> I read the draft, some comments:
> >>>
> >>> 1. I think it needs an offer answer section. We need to specify the
> >> behavior
> >>> when an offer to encrypt the header is sent and the answerer do not
> >> support
> >>> the encryption header extension. Can alternatives, one with
> >> encryption and
> >>> one without be offered.
> >>>
> >>> 2. Extmap can be a session level attribute. What about encryption,
> is
> >> it
> >>> only a media level.
> >>>
> >>> In general are the offer answer rules from RFC 5285 applicable here
> >> and how.
> >>>
> >>>
> >>> Regards
> >>> Roni Even
> >>>
> >>
> >> --
> >> Jonathan Lennox
> >> jonathan@vidyo.com
> >
> >
> 
> --
> Jonathan Lennox
> jonathan@vidyo.com



From bill.wu@huawei.com  Thu Jul 14 01:16:35 2011
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Date: Thu, 14 Jul 2011 16:15:50 +0800
From: Qin Wu <bill.wu@huawei.com>
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Cc: 'Jonathan Lennox' <jonathan@vidyo.com>
Subject: Re: [AVTCORE] comments on draft-ietf-avtcore-srtp-encrypted-header-ext-00.txt
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One more comment to draft-ietf-avtcore-srtp-encrypted-header-ext:
This draft allocates two new lable value from 0x06 to 0x07 for k_he and k_hs, I am wondering
Is there a need in the IANA section to regiserter these values and if yes, who are responsible for regisering
 thes new values? Does it rely on key management protocol or SRTP protcol or this draft?

Regards!
-Qin
----- Original Message ----- 
From: "Roni Even" <Even.roni@huawei.com>
To: <avt@ietf.org>
Cc: "'Jonathan Lennox'" <jonathan@vidyo.com>
Sent: Tuesday, July 12, 2011 4:27 PM
Subject: [AVTCORE] comments ondraft-ietf-avtcore-srtp-encrypted-header-ext-00.txt


> Hi,
> I read the draft, some comments:
> 
> 1. I think it needs an offer answer section. We need to specify the behavior
> when an offer to encrypt the header is sent and the answerer do not support
> the encryption header extension. Can alternatives, one with encryption and
> one without be offered. 
> 
> 2. Extmap can be a session level attribute. What about encryption, is it
> only a media level.
> 
> In general are the offer answer rules from RFC 5285 applicable here and how.
> 
> 
> Regards
> Roni Even
> 
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt

From jonathan@vidyo.com  Thu Jul 14 08:15:30 2011
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From: Jonathan Lennox <jonathan@vidyo.com>
To: Qin Wu <bill.wu@huawei.com>
Date: Thu, 14 Jul 2011 11:15:22 -0400
Thread-Topic: [AVTCORE] comments on draft-ietf-avtcore-srtp-encrypted-header-ext-00.txt
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Hi, Qin --

My inclination would be that label values are used rarely enough that there=
 doesn't need to be an IANA registry for them -- they should only be alloca=
ted by standards action, and coordination is done by the usual working grou=
p consensus processes.

Is there some use case where these values would need to be allocated for no=
n-standards-track purposes, or by working groups other than AVTCore?

On Jul 14, 2011, at 4:15 AM, Qin Wu wrote:

> One more comment to draft-ietf-avtcore-srtp-encrypted-header-ext:
> This draft allocates two new lable value from 0x06 to 0x07 for k_he and k=
_hs, I am wondering
> Is there a need in the IANA section to regiserter these values and if yes=
, who are responsible for regisering
> thes new values? Does it rely on key management protocol or SRTP protcol =
or this draft?

--
Jonathan Lennox
jonathan@vidyo.com



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From: Jonathan Lennox <jonathan@vidyo.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Date: Thu, 14 Jul 2011 18:35:29 -0400
Thread-Topic: [AVTCORE] Fwd: I-D Action: draft-westerlund-avtcore-multistream-and-simulcast-00.txt
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Subject: Re: [AVTCORE] Fwd: I-D Action: draft-westerlund-avtcore-multistream-and-simulcast-00.txt
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On Jul 5, 2011, at 6:54 AM, Magnus Westerlund wrote:

> WG,
>=20
> As an individual Bo Burman and I have written this internet draft that
> discusses multistream and simulcast issues and proposes a set of
> extensions to resolve the issues. Feedback is much appreciated.


Hi, Magnus.

Thank you for writing this draft; I think multi-stream is a very important =
area of RTP, one that's hasn't had enough attention paid to it.

The draft discusses a number of different issues: multi-stream, simulcast, =
RTX, FEC, etc., but seems to assume that these are separate-but-related pro=
blems that can be solved independently, possibly with different solutions t=
o the different problems.

I don't believe that this is the case; instead, I believe that in practice =
you will end up with sessions carrying multiple sources, each of which coul=
d be be simulcast, with RTX and FEC being used independently for any or all=
 of the simulcast streams.  Thus we need one solution that can handle all t=
hese problems at once, including the interactions among them.  (Vidyo has b=
een doing multi-stream with scalable coding for many years.)

As the draft concludes, the multi-stream scenario is best handled by SSRC-m=
ultiplexing, as was originally envisioned by RTP.  Given that this is the c=
ase, it's not clear to me that it's a good idea to use a separate (and in m=
y opinion, less flexible) mechanism for the specific special case of simulc=
ast.  Instead, given the pressure from groups such as RTCweb to reduce the =
port consumption of RTP, I'd recommend SSRC-multiplexing across the board.

Note also that in the case of multiple sources which are each independently=
 simulcast, the simulcast structure of different sources could well be diff=
erent.   (E.g., source A could be available in two resolutions, while sourc=
e B has only one and source C has three.)  It'd be extremely difficult to m=
ap this in session-multiplexed simulcast, whereas for SSRC-multiplexed simu=
lcast it should be no more complex than the single-simulcast-source scenari=
o.=20

--
Jonathan Lennox
jonathan@vidyo.com



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Subject: Re: [AVTCORE] comments on draft-ietf-avtcore-srtp-encrypted-header-ext-00.txt
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Hi,:
The old or new label value is used to calculate the encryption and salting keys that encrypt the either header or payload. 
It seems they should be registered somewhere. However I am not aware of any other SDO or organization use these values.
Anyway, I am okay with your clarification.

Regards!
-Qin
----- Original Message ----- 
From: "Jonathan Lennox" <jonathan@vidyo.com>
To: "Qin Wu" <bill.wu@huawei.com>
Cc: <avt@ietf.org>
Sent: Thursday, July 14, 2011 11:15 PM
Subject: Re: [AVTCORE] comments on draft-ietf-avtcore-srtp-encrypted-header-ext-00.txt


> Hi, Qin --
> 
> My inclination would be that label values are used rarely enough that there doesn't need to be an IANA registry for them -- they should only be allocated by standards action, and coordination is done by the usual working group consensus processes.
> 
> Is there some use case where these values would need to be allocated for non-standards-track purposes, or by working groups other than AVTCore?
> 
> On Jul 14, 2011, at 4:15 AM, Qin Wu wrote:
> 
>> One more comment to draft-ietf-avtcore-srtp-encrypted-header-ext:
>> This draft allocates two new lable value from 0x06 to 0x07 for k_he and k_hs, I am wondering
>> Is there a need in the IANA section to regiserter these values and if yes, who are responsible for regisering
>> thes new values? Does it rely on key management protocol or SRTP protcol or this draft?
> 
> --
> Jonathan Lennox
> jonathan@vidyo.com
> 
> 
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt

From bill.wu@huawei.com  Thu Jul 14 21:14:47 2011
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From: Qin Wu <bill.wu@huawei.com>
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Subject: Re: [AVTCORE] Fwd: I-D Action:draft-westerlund-avtcore-multistream-and-simulcast-00.txt
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Hi,
I have read this draft and think it is well written. Several comments below:
Section 1.2.  Simulcast
"
   Simulcast is the act of simultaneously sending multiple different
   versions of a media content.  This can be done in several ways and
   for different purposes.  This document focuses on the case where one
   wants to provide multiple different encodings towards a intermediary
   so that the intermediary can select which version to forward to other
   participants in the session.  More discussion on the different ways
   of doing simulcast, which is the focus of this document in "Simulcast
   Usage and Applicability" (Section 3).
"

   [Qin]:Suppose the multiple different version of the same media content
   is transmitted simultaneously to the receivers, how do you make sure
   these multiple streams can be recieved by the intermediary in between 
   at the same time and processed at the same time?

Section 1.2:

[Qin]: Have you considered the case where multiple streams are sent simultaneously to the same 
RTP receiver and Intermediary does not involve in simulcast processing. In other words,
the intermediary is transparent to simulcasting and the RTP receiver is powerful and intelligent enough to
deal with simucasting multiple streams.


Section 3.2.  Simulcasting to Consuming End-Point
"
   This usage is based on an RTP Transport Translator (Section 3.3:
   Topo-Trn-Translator) [RFC5117]. ........................
"

[Qin] It seems to me translator in the case described in section 3.2 
      also can send same encoding to multiple destinations or send
      different encoding to independent destination, It look like to me
      the case described in section 3.3 and section 3.4 are just two sub-cases
      under the case described in section 3.2.
      Also the mixer case described in section 3.1 has the same issue.
      would you like to clarify a little bit about this?

Section 3.3 Simulcasting to Consuming End-Point
"
  ...................... Thus some
   consideration and mechanism are needed to support such a use case so
   that it doesn't break RTCP reception reporting.

"
    [Qin] Do you have solution to the translator case since the translator is 
    transparent to the RTP receivers?

Section 3.3 Same Encoding to Multiple Destinations
"
  ... As this type of simulcast is analog to ASM usage and RTP has good
   support for ASM sessions, no further consideration for this case is
   done.
"

   [Qin]: Have you considered SSM case or ASM case which are also simucasted in this document?
Is the solution for SSM case different from the case you described in this document?

Regards!
-Qin
----- Original Message ----- 
From: "Magnus Westerlund" <magnus.westerlund@ericsson.com>
To: "IETF AVTCore WG" <avt@ietf.org>
Sent: Tuesday, July 05, 2011 6:54 PM
Subject: [AVTCORE] Fwd: I-D Action:draft-westerlund-avtcore-multistream-and-simulcast-00.txt


> WG,
> 
> As an individual Bo Burman and I have written this internet draft that
> discusses multistream and simulcast issues and proposes a set of
> extensions to resolve the issues. Feedback is much appreciated.
> 
> Cheers
> 
> Magnus
>


--------------------------------------------------------------------------------


> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
>

From bill.wu@huawei.com  Fri Jul 15 01:39:28 2011
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Subject: Re: [AVTCORE] Upadated AVTcore agenda for Quebec
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Hi,
I saw there was quite a lot of traffic in the XRBlock mailing list.
As one parpcipant of XRBlock WG, I am wondering why chairs
don't arrange a timeslot to discss the update of XRBlock work at this time.

Regards!
-Qin
  ----- Original Message ----- 
  From: Roni Even 
  To: 'IETF AVTCore WG' 
  Sent: Sunday, July 10, 2011 6:02 AM
  Subject: [AVTCORE] Upadated AVTcore agenda for Quebec


  Hi,

  I uploaded the agenda to the meeting page http://www.ietf.org/proceedings/81/agenda/avtcore.html . 

  Attached is the txt version

  Roni Even

  AVTcore co-chair



------------------------------------------------------------------------------


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<DIV><FONT size=2 face=&#23435;&#20307;>Hi,</FONT></DIV>
<DIV>I saw there was quite a&nbsp;lot of&nbsp;traffic in the XRBlock mailing 
list.</DIV>
<DIV>As one parpcipant of XRBlock WG, I am wondering why&nbsp;chairs</DIV>
<DIV>don't arrange a&nbsp;timeslot to discss the update of XRBlock work at this 
time.</DIV>
<DIV><FONT size=2 face=&#23435;&#20307;></FONT>&nbsp;</DIV>
<DIV>Regards!</DIV>
<DIV>-Qin</DIV>
<BLOCKQUOTE 
style="BORDER-LEFT: #000000 2px solid; PADDING-LEFT: 5px; PADDING-RIGHT: 0px; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px">
  <DIV style="FONT: 9pt &#23435;&#20307;">----- Original Message ----- </DIV>
  <DIV style="FONT: 9pt &#23435;&#20307;; BACKGROUND: #e4e4e4; font-color: black"><B>From:</B> 
  <A title=Even.roni@huawei.com href="mailto:Even.roni@huawei.com">Roni Even</A> 
  </DIV>
  <DIV style="FONT: 9pt &#23435;&#20307;"><B>To:</B> <A title=avt@ietf.org 
  href="mailto:avt@ietf.org">'IETF AVTCore WG'</A> </DIV>
  <DIV style="FONT: 9pt &#23435;&#20307;"><B>Sent:</B> Sunday, July 10, 2011 6:02 AM</DIV>
  <DIV style="FONT: 9pt &#23435;&#20307;"><B>Subject:</B> [AVTCORE] Upadated AVTcore agenda 
  for Quebec</DIV>
  <DIV><BR></DIV>
  <DIV class=WordSection1>
  <P class=MsoNormal>Hi,<o:p></o:p></P>
  <P class=MsoNormal>I uploaded the agenda to the meeting page <A 
  href="http://www.ietf.org/proceedings/81/agenda/avtcore.html">http://www.ietf.org/proceedings/81/agenda/avtcore.html</A> 
  . <o:p></o:p></P>
  <P class=MsoNormal>Attached is the txt version<o:p></o:p></P>
  <P class=MsoNormal>Roni Even<o:p></o:p></P>
  <P class=MsoNormal>AVTcore co-chair<o:p></o:p></P></DIV>
  <P>
  <HR>

  <P></P>_______________________________________________<BR>Audio/Video 
  Transport Core Maintenance<BR><A 
  href="mailto:avt@ietf.org">avt@ietf.org</A><BR><A 
  href="https://www.ietf.org/mailman/listinfo/avt">https://www.ietf.org/mailman/listinfo/avt</A><BR></BLOCKQUOTE></BODY></HTML>

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From magnus.westerlund@ericsson.com  Tue Jul 19 05:31:52 2011
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Thanks for the comments,

Inline responses.

On 2011-07-11 09:56, Roni Even wrote:
> Hi Magnus,
> Read the draft, I liked the analysis on the different multiplexing of RTP
> streams (Session, SSRC and payload type).
> 
> Some review comments
> 
> 1. The introduction talks about multistream and simulcast. I noticed that it
> only talks about how to define the alternatives in terms of the stream but
> is not saying how to understand the semantics of the streams, for example
> which camera in Telepresence and in simulcast you assume that all
> alternative provide the same view (just resolution change, but can also use
> different view of the room). This is fine and is being discussed in CLUE at
> the moment and should be out of scope here but it may cause the SDP to be
> even larger. 

Exactly, I think the actual meta information about which streams is
captured in relation to another is for the external application to take
care of, or at least another part of the system. As you say, CLUE WG's
work is highly relevant in this.

> 
> 2. On the SDP syntax it may be better when you have more than one m-line to
> have all the descriptions which repeats in all m-line appear st the SDP
> session level with local identifier and use the identifier in the m-lines.
> Maybe based on grouping.

I don't quite understand you. Which parts repeats? An example may look
like it repeats a lot, but there is actually not a lot of information
that are redundant, or exactly the same.

CNAMEs are by design the same. However that repetition comes due to us
reusing existing attribute which is actually one of our main points. We
want to reuse the existing signalling attributes rather than redefine
with some alteration.

> 
> 
> 3. In section 1.1 "Todays SDP signalling support for this is basically the
> directionality attribute which indicates an end-point intend to send media
> or not.  No indication of how many media streams." My understanding so far
> was that the number of PT in the m-line is the maximum number of streams
> that can be sent in an RTP session. This  goes also to the discussion on
> support of multiple SSRCs. I think that today video conferencing endpoint is
> doing a second offer answer to fix the PT it will use based on the first
> offer answer exchange. 

Personally I have seen no binding between the number of streams and the
number of PTs. A m= line with a single PT can have 100 SSRSs sending
media in the RTP session. If you believe such a rule exist, then I would
like to press you for quoting that text that specifies that.


> 
> 4. The above still does not address the multiple SSRC for the same PT and I
> already mentioned it in the past that I think that a lot of product today
> will not handle it good. I think that this is a rare case today at least in
> centralized video conferencing since the MCU (or RTP mixer if exist) will
> use its own SSRC to send the mixed/switched media.

Exactly, I think the reasonable assumption with a legacy terminal is
that it handles one stream in and one stream out per RTP session. That
is why we want to introduce these new SDP attributes to indicate
capability of receiving or intention to send multiple streams in the
same RTP session.

> 
> 5. In section 3.2 second paragraph you talk about the receivers controlling
> what the relay will send but this will make the relay something else than
> RTP translator.

Yes, but I think it is a reality that already is in use in some RTP
derivate systems. You want a simple non media processing box, at the
same time you want to limit the bandwidth consume somehow. Thus you need
to do dynamic selection of the streams. At the same time you don't want
the concept of the RTP session with a joint SSRC space and reporting to
break. Thus I think there is need for work towards this model also.

However, this is not something that is Bo's and my intention to focus
on. I just identify that in this usage we have short commings.


> 
> 6. On the bandwidth support. Currently the definition is the b= is a
> receiver capability. I am not sure why you need the sender one, you claim
> that there is no way to control the actual bw but the receiver can use RTCP
> CCM flow control to tell the sender to reduce the rate. As for the syntax,
> my preference is for TIAS semantics and not the one you proposed. 

If you with TIAS semantics you mean the "media bitrate" without the
header overhead? The draft actually has this an open issue in the
document, if one should include IP level or not.

Page 19-20:
   It is an open
   question exactly what this token bucket is measuring, if it is RTP
   payload only, like TIAS, or if it includes all headers down to the IP
   level as most of the other bandwidth modifiers do.

We need a send direction for various reasons actually. For example in
Send only streams, one like to indicate what the highest bandwidth one
intended to send. The receiver may reduce it further, but he should have
some value of the desired maximum bit-rate the sender invite to use. Or
answering a SDP offer. If the offerer has offered to receive 4 payload
types with various bit-rates it makes sense to declare for the sender
the highest it intends to generate for any of the payload types it
supports. Also, without improved directionality capabilities one can't
express that you can receive 3 mbps, but only intended to send 1.5 mbps.

It is a matter of making the capabilities and the actual configured
usage being known to the parties.


> 
> 7. In the last sentence of 7.2.1 you say "This enables a simple fallback
> solution to exclude a legacy client from all simulcast versions except one,
> whichever is most suitable for the application". My view is that currently
> when there are several m-lines of the same media type there is no definition
> which one to choose if the answerer do not understand the semantics of the
> offer and can support only one stream.

Yes, there might be interoperability issues there, but the most likely
behavior for a client that doesn't know what to do with streams is to
set the port to 0 for anything but the first, or maybe the one with
a=content:main.

Cheers

Magnus Westerlund

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From magnus.westerlund@ericsson.com  Tue Jul 19 05:52:38 2011
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Jonathan,

Thanks for the comments.

See inline for further comments.

On 2011-07-15 00:35, Jonathan Lennox wrote:
> On Jul 5, 2011, at 6:54 AM, Magnus Westerlund wrote:
> 
>> WG,
>> 
>> As an individual Bo Burman and I have written this internet draft
>> that discusses multistream and simulcast issues and proposes a set
>> of extensions to resolve the issues. Feedback is much appreciated.
> 
> 
> Hi, Magnus.
> 
> Thank you for writing this draft; I think multi-stream is a very
> important area of RTP, one that's hasn't had enough attention paid to
> it.
> 
> The draft discusses a number of different issues: multi-stream,
> simulcast, RTX, FEC, etc., but seems to assume that these are
> separate-but-related problems that can be solved independently,
> possibly with different solutions to the different problems.
> 
> I don't believe that this is the case; instead, I believe that in
> practice you will end up with sessions carrying multiple sources,
> each of which could be be simulcast, with RTX and FEC being used
> independently for any or all of the simulcast streams.  Thus we need
> one solution that can handle all these problems at once, including
> the interactions among them.  (Vidyo has been doing multi-stream with
> scalable coding for many years.)

I try to take as encompassing view of this as possible. Yes, we don't
include RTX or FEC in the discussion currently. I think mostly because
we didn't need to. RTX clearly has two different modes, either SSRC or
Session multiplexing. FEC depends on which type of FEC one intend to use
for what multiplexing that is available.

I think we should try to make clear indications going forward for how
these should be structured in future extensions.


> 
> As the draft concludes, the multi-stream scenario is best handled by
> SSRC-multiplexing, as was originally envisioned by RTP.  Given that
> this is the case, it's not clear to me that it's a good idea to use a
> separate (and in my opinion, less flexible) mechanism for the
> specific special case of simulcast.  Instead, given the pressure from
> groups such as RTCweb to reduce the port consumption of RTP, I'd
> recommend SSRC-multiplexing across the board.

Jonathan lets me ask you a question. Is any hesitation to use several
RTP sessions when most suitable due to the port consumption issue or for
some other reason? And here we can debate what is most suitable.

My view is that multiple SSRCs in an RTP session is a very good
mechanism for streams that one expect to be treated the same way in both
central nodes or in end-points. When it comes to sets of streams that
needs a different treatment, I think it is logical and a powerful
construct to use the RTP session for separating these sets of streams.

Yes, the number of underlying transport flows is an important concern in
some use cases. But, I think we shouldn't use that as a reason for
removing an important concept in RTP. If the issue we have is that one
RTP session results in one UDP flow, then lets do something about that
instead of removing the RTP session concept.

> 
> Note also that in the case of multiple sources which are each
> independently simulcast, the simulcast structure of different sources
> could well be different.   (E.g., source A could be available in two
> resolutions, while source B has only one and source C has three.)
> It'd be extremely difficult to map this in session-multiplexed
> simulcast, whereas for SSRC-multiplexed simulcast it should be no
> more complex than the single-simulcast-source scenario.

This is a valid concern, but which I think we have a reasonable straight
story for. In the direction you have a certain number of simulcast
versions you negotiate that number of session and their individual
configurations.

If one looks at the example in Section 8.5.2.2, you see that we have
there 3 simulcast versions going towards a central mixer. The return
channel from the mixer are only two different sessions, supporting to
receive all the possible payload types we can get from the other
participants. And the difference between these two return sessions are
if they are content:main or content:alternative. In other words what we
do with the streams locally when rendering them.

The two main points we have for choosing Session multiplexing is:
A) RTP sessions provides a context for what purpose the streams in them
have.

B) We don't need to define any new signalling to negotiate the different
simulcast levels. Only a bit of binding together the streams and we are
done. The SCS and SCR grouping semantics and the SRCNAME is really all
we require for getting session multiplexed simulcast to work. The rest
is to provide improvements over todays order when it comes to bandwidth
signalling and media control in centralized conferencing.

cheers

Magnus Westerlund

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----------------------------------------------------------------------
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From magnus.westerlund@ericsson.com  Tue Jul 19 06:08:24 2011
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Qin,

Thanks for your comments.

See inline for follow-up.

On 2011-07-15 06:11, Qin Wu wrote:
> Hi,
> I have read this draft and think it is well written. Several comments below:
> Section 1.2.  Simulcast
> "
>    Simulcast is the act of simultaneously sending multiple different
>    versions of a media content.  This can be done in several ways and
>    for different purposes.  This document focuses on the case where one
>    wants to provide multiple different encodings towards a intermediary
>    so that the intermediary can select which version to forward to other
>    participants in the session.  More discussion on the different ways
>    of doing simulcast, which is the focus of this document in "Simulcast
>    Usage and Applicability" (Section 3).
> "
> 
>    [Qin]:Suppose the multiple different version of the same media content
>    is transmitted simultaneously to the receivers, how do you make sure
>    these multiple streams can be recieved by the intermediary in between 
>    at the same time and processed at the same time?

If you have an conference session using a centralized mixer then you
normally negotiate the setup with the conference server, that is the
controling entity of the mixer. Thus, that side ensures that the mixer
and the client really are agreeing on using simulcast.

> 
> Section 1.2:
> 
> [Qin]: Have you considered the case where multiple streams are sent simultaneously to the same 
> RTP receiver and Intermediary does not involve in simulcast processing. In other words,
> the intermediary is transparent to simulcasting and the RTP receiver is powerful and intelligent enough to
> deal with simucasting multiple streams.

Yes, that is what Section 3.2, really tries to discuss. We are
considering it, but it is not our main case.

> 
> 
> Section 3.2.  Simulcasting to Consuming End-Point
> "
>    This usage is based on an RTP Transport Translator (Section 3.3:
>    Topo-Trn-Translator) [RFC5117]. ........................
> "
> 
> [Qin] It seems to me translator in the case described in section 3.2 
>       also can send same encoding to multiple destinations or send
>       different encoding to independent destination, It look like to me
>       the case described in section 3.3 and section 3.4 are just two sub-cases
>       under the case described in section 3.2.
>       Also the mixer case described in section 3.1 has the same issue.
>       would you like to clarify a little bit about this?

To certain degree I think you are correct. Clearly a relay and a full
mesh transport is equivalent from a high level functionality
perspective. Their impact on parts of the system is clearly different.

However, that is not the main difference between 3.2 and 3.3. In 3.3 all
peers receive the same encoding, while the 3.2 client receives multiple
encodings of the same version. That is the more fundamental difference.

When it comes to 3.4 compared to 3.2. In 3.4 party A receives one
version of the encoding, but it is different from what is neighbor,
party B, receives. In addition there is no need in that case for A and B
to be aware of each other. If A and B needs to be aware when receiving
content C then I think that usage maps better onto 3.1.

> 
> Section 3.3 Simulcasting to Consuming End-Point
> "
>   ...................... Thus some
>    consideration and mechanism are needed to support such a use case so
>    that it doesn't break RTCP reception reporting.
> 
> "
>     [Qin] Do you have solution to the translator case since the translator is 
>     transparent to the RTP receivers?

No, we don't have a solution. We identify this solution because I have
heard about conferencing solutions that operates under very similar
premises with end-points or a central control selecting which streams to
deliver and where it sounds much more like a basic transport relay than
an RTP mixer, selecting streams onto the Mixer's own SSRC values.

If there is desire to support this use case, then the proponents for
this should propose something.

> 
> Section 3.3 Same Encoding to Multiple Destinations
> "
>   ... As this type of simulcast is analog to ASM usage and RTP has good
>    support for ASM sessions, no further consideration for this case is
>    done.
> "
> 
>    [Qin]: Have you considered SSM case or ASM case which are also simucasted in this document?
> Is the solution for SSM case different from the case you described in this document?

We haven't considered SSM or ASM specifically. It was not our focus.
However, I think Session multiplexed simulcast is a given in any cases
where the sender wants to use simulcast. That as sessions can be mapped
onto different multicast groups, thus allowing a receiver to jump
between the simulcast representations based on each receivers need. That
is not possible for SSRC multiplexed simulcast.

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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For your information.

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From magnus.westerlund@ericsson.com  Tue Jul 19 08:35:15 2011
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Subject: [AVTCORE] RTP Internal Multiplexing of RTP sessions
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WG,

As Individual contributor.

I don't know if people in the WG are aware of the discussion ongoing in
RTCWEB around multiplexing of protocols over the same underlying
transport flow, like UDP flow. That include RTP sessions.

There is a proposal:
"Multiplexing of Real-Time Transport Protocol (RTP) Traffic for Browser
                  based Real-Time Communications (RTC)"
https://datatracker.ietf.org/doc/draft-rosenberg-rtcweb-rtpmux/?include_text=1

This proposal is criticized by the following draft.
https://datatracker.ietf.org/doc/draft-perkins-rtcweb-rtp-usage/?include_text=1

There is an ongoing discussion on several aspects of these proposals on
the RTCWEB WG. You are welcome to go join the discussion of the other
questions raised on that mailing list.

Here I primarily want to raise the issue I see with multiplexing several
RTP sessions onto the same lower layer transport without something
explicitly identifying these separate sessions. And in particular the
RTP SSRC semantics changes that is proposed by the Rosenberg draft.

Personally I think it is time we tackle the issue of how to efficiently
run multiple RTP sessions over one lower layer transport. This as there
are several wanting to have that functionality. However, I think this
must be done in a very careful way. As we don't want to create an
alternative version of RTP which is what the Rosenberg proposal in fact
does. I do want us to gather the requirements on this and think of how
we can accomplish this.

I think there is a clear risk that we will not be able to accomplish it
without either adding some type of shim header which has it sets of
drawback or in fact fork or change the RTP protocol in such a way that
it will not be fully compatible with what exist today. But if that is to
be done, we should have a very clear consensus on that and ensure that
we then develop future extensions to RTP within a single version of the
protocol, not two.

As WG chair:

We are considering re-arranging the agenda to facilitate a debate of
this topic in Quebec.

So please indicate both your view of the topic in general and if you
think we should re-prioritize the agenda.

cheers

Magnus Westerlund

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Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
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From Even.roni@huawei.com  Tue Jul 19 11:30:46 2011
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Hi Magnus,

> > 2. On the SDP syntax it may be better when you have more than one m-
> line to
> > have all the descriptions which repeats in all m-line appear st the
> SDP
> > session level with local identifier and use the identifier in the m-
> lines.
> > Maybe based on grouping.
> 
> I don't quite understand you. Which parts repeats? An example may look
> like it repeats a lot, but there is actually not a lot of information
> that are redundant, or exactly the same.
> 
> CNAMEs are by design the same. However that repetition comes due to us
> reusing existing attribute which is actually one of our main points. We
> want to reuse the existing signalling attributes rather than redefine
> with some alteration.
> 
> >
Three options using your example from 8.5.1.2, the second I am not sure
since it may change the SDP SSRC syntax

1.   v=0
   o=alice 2890844526 2890842807 IN IP4 192.0.2.156
   s=Telepresence Alice Invite
   c=IN IP4 192.0.2.156
   b=AS:8965
   t=0 0
   
   param = 1 cname:alice@foo.example.com
   param = 2 srcname:a3:d3:4b:f1:22:12
   param = 3 srcname:12:3f:ab:d2:ec:32
   param = 4 srcname:7f:12:db:87:2d:52



   m=audio 49200 RTP/AVP 97 96
   b=AS:725
   a=rtpmap:96 G719/48000/2
   a=rtpmap:97 G719/48000
   a=bw:send pt=96 SMT:tb=128800:1500
   a=bw:send pt=97 SMT:tb=64800:1500
   a=bw:recv pt=* AMT:tb=644000:1500
   a=max-recv-ssrc:* 5
   a=max-send-ssrc:97 3
   a=max-send-ssrc:96 1
   a=ssrc:239245219 param:1
   a=ssrc:239245219 param:2
   a=ssrc:986545121 param:1
   a=ssrc:986545121 param:3
   a=ssrc:2199983234 param:1
   a=ssrc:2199983234 param=4
   m=video 49300 RTP/AVP 96
   b=AS:8240
   a=rtpmap:96 H264/90000
   a=fmtp:96 profile-level-id=42c01e
   a=imageattr:* send [x=1280,y=720] recv [x=1280,y=720]
   a=bw:send pt=96 SMT:tb=2500000:8192
   a=bw:recv pt=96 SMT:tb=3000000:8192
   a=bw:send pt=* AMT:tb=8000000:16384
   a=max-recv-ssrc:* 5
   a=max-send-ssrc:* 3
   a=ssrc:245219239 param:1
   a=ssrc:245219239 param:2
   a=ssrc:545121986 param:1
   a=ssrc:545121986 param:3
   a=ssrc:199983234 param:1
   a=ssrc:199983234 param:4
   a=content:main




2.   v=0
   o=alice 2890844526 2890842807 IN IP4 192.0.2.156
   s=Telepresence Alice Invite
   c=IN IP4 192.0.2.156
   b=AS:8965
   t=0 0
   
   param = 1 cname:alice@foo.example.com rcname:a3:d3:4b:f1:22:12
   param = 2 name:alice@foo.example.com srcname:12:3f:ab:d2:ec:32
   param = 3 cname:alice@foo.example.com srcname:7f:12:db:87:2d:52



   m=audio 49200 RTP/AVP 97 96
   b=AS:725
   a=rtpmap:96 G719/48000/2
   a=rtpmap:97 G719/48000
   a=bw:send pt=96 SMT:tb=128800:1500
   a=bw:send pt=97 SMT:tb=64800:1500
   a=bw:recv pt=* AMT:tb=644000:1500
   a=max-recv-ssrc:* 5
   a=max-send-ssrc:97 3
   a=max-send-ssrc:96 1
   a=ssrc:239245219 param:1
   a=ssrc:986545121 param:2
   a=ssrc:2199983234 param:3
   m=video 49300 RTP/AVP 96
   b=AS:8240
   a=rtpmap:96 H264/90000
   a=fmtp:96 profile-level-id=42c01e
   a=imageattr:* send [x=1280,y=720] recv [x=1280,y=720]
   a=bw:send pt=96 SMT:tb=2500000:8192
   a=bw:recv pt=96 SMT:tb=3000000:8192
   a=bw:send pt=* AMT:tb=8000000:16384
   a=max-recv-ssrc:* 5
   a=max-send-ssrc:* 3
   a=ssrc:245219239 param:1
   
   a=ssrc:545121986 param:2
   
   a=ssrc:199983234 param:3
   
   a=content:main



3. the shortest will be the ssrc-id=id:ssrc-value, the id will identify the
id to use in the media level with all the parameters except the other
ssrc-id's in the multi line. We can probably give a better syntax but it is
shorter.


   v=0
   o=alice 2890844526 2890842807 IN IP4 192.0.2.156
   s=Telepresence Alice Invite
   c=IN IP4 192.0.2.156
   b=AS:8965
   t=0 0

multi = cname:alice@foo.example.com srcname:a3:d3:4b:f1:22:12
ssrc-id=1:239245219 ssrc-id=2: 245219239

multi= cname:alice@foo.example.com srcname:12:3f:ab:d2:ec:32
ssrc-id=3:986545121 ssrc-id=4:545121986

 multi= cname:alice@foo.example.com srcname:7f:12:db:87:2d:52
ssrc-id=5:2199983234 ssrc-id=6:199983234  



m=audio 49200 RTP/AVP 97 96
   b=AS:725
   a=rtpmap:96 G719/48000/2
   a=rtpmap:97 G719/48000
   a=bw:send pt=96 SMT:tb=128800:1500
   a=bw:send pt=97 SMT:tb=64800:1500
   a=bw:recv pt=* AMT:tb=644000:1500
   a=max-recv-ssrc:* 5
   a=max-send-ssrc:97 3
   a=max-send-ssrc:96 1
   a=multi:1
   a= multi:3
   a= multi:5
   
   m=video 49300 RTP/AVP 96
   b=AS:8240
   a=rtpmap:96 H264/90000
   a=fmtp:96 profile-level-id=42c01e
   a=imageattr:* send [x=1280,y=720] recv [x=1280,y=720]
   a=bw:send pt=96 SMT:tb=2500000:8192
   a=bw:recv pt=96 SMT:tb=3000000:8192
   a=bw:send pt=* AMT:tb=8000000:16384
   a=max-recv-ssrc:* 5
   a=max-send-ssrc:* 3
   a= multi:2 
   a= multi:4 
   a= multi:6
   a=content:main


Roni


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To: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Cc: IETF AVTCore WG <avt@ietf.org>
Subject: Re: [AVTCORE] RTP Internal Multiplexing of RTP sessions
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I'm very interested in this work and willing to help.

On Jul 19, 2011, at 11:35 AM, Magnus Westerlund wrote:

> WG,
>=20
> As Individual contributor.
>=20
> I don't know if people in the WG are aware of the discussion ongoing =
in
> RTCWEB around multiplexing of protocols over the same underlying
> transport flow, like UDP flow. That include RTP sessions.
>=20
> There is a proposal:
> "Multiplexing of Real-Time Transport Protocol (RTP) Traffic for =
Browser
>                  based Real-Time Communications (RTC)"
> =
https://datatracker.ietf.org/doc/draft-rosenberg-rtcweb-rtpmux/?include_te=
xt=3D1
>=20
> This proposal is criticized by the following draft.
> =
https://datatracker.ietf.org/doc/draft-perkins-rtcweb-rtp-usage/?include_t=
ext=3D1
>=20
> There is an ongoing discussion on several aspects of these proposals =
on
> the RTCWEB WG. You are welcome to go join the discussion of the other
> questions raised on that mailing list.
>=20
> Here I primarily want to raise the issue I see with multiplexing =
several
> RTP sessions onto the same lower layer transport without something
> explicitly identifying these separate sessions. And in particular the
> RTP SSRC semantics changes that is proposed by the Rosenberg draft.
>=20
> Personally I think it is time we tackle the issue of how to =
efficiently
> run multiple RTP sessions over one lower layer transport. This as =
there
> are several wanting to have that functionality. However, I think this
> must be done in a very careful way. As we don't want to create an
> alternative version of RTP which is what the Rosenberg proposal in =
fact
> does. I do want us to gather the requirements on this and think of how
> we can accomplish this.
>=20
> I think there is a clear risk that we will not be able to accomplish =
it
> without either adding some type of shim header which has it sets of
> drawback or in fact fork or change the RTP protocol in such a way that
> it will not be fully compatible with what exist today. But if that is =
to
> be done, we should have a very clear consensus on that and ensure that
> we then develop future extensions to RTP within a single version of =
the
> protocol, not two.
>=20
> As WG chair:
>=20
> We are considering re-arranging the agenda to facilitate a debate of
> this topic in Quebec.
>=20
> So please indicate both your view of the topic in general and if you
> think we should re-prioritize the agenda.
>=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


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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>, "IETF AVTCore WG" <avt@ietf.org>
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Subject: Re: [AVTCORE] RTP Internal Multiplexing of RTP sessions
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> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of =
Magnus Westerlund
> Sent: Tuesday, July 19, 2011 5:35 PM
> To: IETF AVTCore WG
> Subject: [AVTCORE] RTP Internal Multiplexing of RTP sessions
>=20
> WG,
>=20
> As Individual contributor.
>=20
> I don't know if people in the WG are aware of the discussion ongoing =
in
> RTCWEB around multiplexing of protocols over the same underlying
> transport flow, like UDP flow. That include RTP sessions.
>=20
> There is a proposal:
> "Multiplexing of Real-Time Transport Protocol (RTP) Traffic for =
Browser
>                   based Real-Time Communications (RTC)"
> =
https://datatracker.ietf.org/doc/draft-rosenberg-rtcweb-rtpmux/?include_t=
ext=3D1
>=20
> This proposal is criticized by the following draft.
> =
https://datatracker.ietf.org/doc/draft-perkins-rtcweb-rtp-usage/?include_=
text=3D1
>=20
> There is an ongoing discussion on several aspects of these proposals =
on
> the RTCWEB WG. You are welcome to go join the discussion of the other
> questions raised on that mailing list.
>=20
> Here I primarily want to raise the issue I see with multiplexing =
several
> RTP sessions onto the same lower layer transport without something
> explicitly identifying these separate sessions. And in particular the
> RTP SSRC semantics changes that is proposed by the Rosenberg draft.
>=20
> Personally I think it is time we tackle the issue of how to =
efficiently
> run multiple RTP sessions over one lower layer transport. This as =
there
> are several wanting to have that functionality. However, I think this
> must be done in a very careful way. As we don't want to create an
> alternative version of RTP which is what the Rosenberg proposal in =
fact
> does. I do want us to gather the requirements on this and think of how
> we can accomplish this.
>=20
> I think there is a clear risk that we will not be able to accomplish =
it
> without either adding some type of shim header which has it sets of
> drawback or in fact fork or change the RTP protocol in such a way that
> it will not be fully compatible with what exist today. But if that is =
to
> be done, we should have a very clear consensus on that and ensure that
> we then develop future extensions to RTP within a single version of =
the
> protocol, not two.
>=20
> As WG chair:
>=20
> We are considering re-arranging the agenda to facilitate a debate of
> this topic in Quebec.
>=20
> So please indicate both your view of the topic in general and if you
> think we should re-prioritize the agenda.

If that would help, I can donate some minutes from my presentations ;)

-acbegen
=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 olivier.crete@collabora.com  Tue Jul 19 14:57:51 2011
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Hello,

In RFC 4585, the RPSI message is defined as containing a codec specific
FCI. H.263 defines something to be used in such a case, but I see
nothing similar for H.264. Is it not used at all? Do people just use
H.271 back channel messages wrapped in RFC 5104 message ?

Also, I assume the PictureID used for RFC 4585 AVPF SLI messages is the
same as in H.271 ?

--=20
Olivier Cr=C3=AAte
olivier.crete@collabora.com

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Date: Wed, 20 Jul 2011 10:19:00 +0300
From: Roni Even <Even.roni@huawei.com>
In-reply-to: <D0A278D5-9D05-4DA7-AFC8-EE5E995A0090@csperkins.org>
To: 'Colin Perkins' <csp@csperkins.org>, 'Qin Wu' <bill.wu@huawei.com>
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Cc: avt@ietf.org
Subject: Re: [AVTCORE] I-D Action: draft-ietf-avtcore-monarch-03.txt
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Hi,
Looking at the tag mechanism I agree with Colin that using it just to reduce
the size of the packet is not the important. 
Still I can see a different use case for the tag in order to group XR
reports as one entity. The monitoring architecture suggest dividing large
reports to small blocks and the assumption is the by the SSRC you define a
group of blocks per an application. Using the tag may provide another level
of freedom allowing more than one tag per SSRC and the tag defines a
reporting unit. This is a change from the current usage of tags and I am
bringing it to see if this usage is useful.

As for the correlation with non-rtp protocols I agree that SDES may be used
for it.

Regards
Roni Even
As individual

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
> Colin Perkins
> Sent: Tuesday, June 21, 2011 5:56 PM
> To: Qin Wu
> Cc: avt@ietf.org
> Subject: Re: [AVTCORE] I-D Action: draft-ietf-avtcore-monarch-03.txt
> 
> Qin,
> 
> This does address many of my comments on the earlier versions. Some
> quick comments on this draft:
> 
> - In Section 5.2, there is no need to report the CNAME, since this is
> redundant with RTCP SDES information.
> 
> - Section 5.2 suggests reporting "the sequence number of the first
> packet of the RTP session". A receiver cannot know what is the first
> packet of the session, only what was the first packet it received, so
> this should be clarified to "the sequence number of the first packet
> received in the RTP session". The first sequence number received also
> becomes less useful after the sequence number space wraps around, so it
> might make sense to include a count of sequence number cycles since
> then.
> 
> - Section 5.3 discusses sending information that can be used to
> correlate an RTP source with non-RTP protocols. This seems like a
> reasonable thing to add. However, since it is describing the source,
> rather than providing a quality report, should this be an new RTCP SDES
> block type, rather than an RTCP XR block?
> 
> Colin
> 
> 
> On 16 Jun 2011, at 10:29, Qin Wu wrote:
> 
> > Hi,:
> > In response to the recent comments and suggestion on the list, a new
> version of draft-ietf-avtcore-monarch is just submitted.
> > http://www.ietf.org/internet-drafts/draft-ietf-avtcore-monarch-03.txt
> > The diff is:
> > http://tools.ietf.org/rfcdiff?url2=draft-ietf-avtcore-monarch-03
> >
> > Please check if your comments are addressed in this version.
> >
> > Regards!
> > -Qin
> >
> > ----- Original Message -----
> > From: <internet-drafts@ietf.org>
> > To: <i-d-announce@ietf.org>
> > Cc: <avt@ietf.org>
> > Sent: Thursday, June 16, 2011 5:21 PM
> > Subject: [AVTCORE] I-D Action: draft-ietf-avtcore-monarch-03.txt
> >
> >
> >> 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 Architectures for RTP
> >> Author(s)       : Qin Wu
> >>                         Geoff Hunt
> >>                         Philip Arden
> >> Filename        : draft-ietf-avtcore-monarch-03.txt
> >> Pages           : 23
> >> Date            : 2011-06-16
> >>
> >>  This memo proposes an architecture for extending RTCP with a new
> RTCP
> >>  XR (RFC3611) block type to report new metrics regarding media
> >>  transmission or reception quality, as proposed 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 &quot;mix and
> match&quot; 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-
> 03.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-03.txt
> >> _______________________________________________
> >> 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
> 
> 
> 
> --
> Colin Perkins
> http://csperkins.org/
> 
> 
> 
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt


From john.elwell@siemens-enterprise.com  Wed Jul 20 02:30:49 2011
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>, IETF AVTCore WG <avt@ietf.org>
Date: Wed, 20 Jul 2011 11:30:46 +0200
Thread-Topic: [AVTCORE] RTP Internal Multiplexing of RTP sessions
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I think this is an important topic and should receive some agenda time in Q=
uebec.

John=20

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On=20
> Behalf Of Magnus Westerlund
> Sent: 19 July 2011 16:35
> To: IETF AVTCore WG
> Subject: [AVTCORE] RTP Internal Multiplexing of RTP sessions
>=20
> WG,
>=20
> As Individual contributor.
>=20
> I don't know if people in the WG are aware of the discussion=20
> ongoing in
> RTCWEB around multiplexing of protocols over the same underlying
> transport flow, like UDP flow. That include RTP sessions.
>=20
> There is a proposal:
> "Multiplexing of Real-Time Transport Protocol (RTP) Traffic=20
> for Browser
>                   based Real-Time Communications (RTC)"
> https://datatracker.ietf.org/doc/draft-rosenberg-rtcweb-rtpmux
> /?include_text=3D1
>=20
> This proposal is criticized by the following draft.
> https://datatracker.ietf.org/doc/draft-perkins-rtcweb-rtp-usag
> e/?include_text=3D1
>=20
> There is an ongoing discussion on several aspects of these=20
> proposals on
> the RTCWEB WG. You are welcome to go join the discussion of the other
> questions raised on that mailing list.
>=20
> Here I primarily want to raise the issue I see with=20
> multiplexing several
> RTP sessions onto the same lower layer transport without something
> explicitly identifying these separate sessions. And in particular the
> RTP SSRC semantics changes that is proposed by the Rosenberg draft.
>=20
> Personally I think it is time we tackle the issue of how to=20
> efficiently
> run multiple RTP sessions over one lower layer transport.=20
> This as there
> are several wanting to have that functionality. However, I think this
> must be done in a very careful way. As we don't want to create an
> alternative version of RTP which is what the Rosenberg=20
> proposal in fact
> does. I do want us to gather the requirements on this and think of how
> we can accomplish this.
>=20
> I think there is a clear risk that we will not be able to=20
> accomplish it
> without either adding some type of shim header which has it sets of
> drawback or in fact fork or change the RTP protocol in such a way that
> it will not be fully compatible with what exist today. But if=20
> that is to
> be done, we should have a very clear consensus on that and ensure that
> we then develop future extensions to RTP within a single=20
> version of the
> protocol, not two.
>=20
> As WG chair:
>=20
> We are considering re-arranging the agenda to facilitate a debate of
> this topic in Quebec.
>=20
> So please indicate both your view of the topic in general and if you
> think we should re-prioritize the agenda.
>=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  Wed Jul 20 02:52:34 2011
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Hi Roni,

I understand what you are after. But, this looks very much like a
general change to the a=ssrc attribute, than something specific to our
proposed extension. That we propose a new SDES item seems to make it
clearer that the current structure isn't optimized for size. But the
question does it really matter. It is not like these SDP anyway results
in SIP packets that are below 1500 bytes.

It also creates a level of indirection, which isn't that problematic for
the machine interpretations but for humans.

Lets get back to it when we have agreed on the high level issues and
actually are doing the extension.

Cheers

Magnus


On 2011-07-19 20:27, Roni Even wrote:
> Hi Magnus,
> 
>>> 2. On the SDP syntax it may be better when you have more than one m-
>> line to
>>> have all the descriptions which repeats in all m-line appear st the
>> SDP
>>> session level with local identifier and use the identifier in the m-
>> lines.
>>> Maybe based on grouping.
>>
>> I don't quite understand you. Which parts repeats? An example may look
>> like it repeats a lot, but there is actually not a lot of information
>> that are redundant, or exactly the same.
>>
>> CNAMEs are by design the same. However that repetition comes due to us
>> reusing existing attribute which is actually one of our main points. We
>> want to reuse the existing signalling attributes rather than redefine
>> with some alteration.
>>
>>>
> Three options using your example from 8.5.1.2, the second I am not sure
> since it may change the SDP SSRC syntax
> 
> 1.   v=0
>    o=alice 2890844526 2890842807 IN IP4 192.0.2.156
>    s=Telepresence Alice Invite
>    c=IN IP4 192.0.2.156
>    b=AS:8965
>    t=0 0
>    
>    param = 1 cname:alice@foo.example.com
>    param = 2 srcname:a3:d3:4b:f1:22:12
>    param = 3 srcname:12:3f:ab:d2:ec:32
>    param = 4 srcname:7f:12:db:87:2d:52
> 
> 
> 
>    m=audio 49200 RTP/AVP 97 96
>    b=AS:725
>    a=rtpmap:96 G719/48000/2
>    a=rtpmap:97 G719/48000
>    a=bw:send pt=96 SMT:tb=128800:1500
>    a=bw:send pt=97 SMT:tb=64800:1500
>    a=bw:recv pt=* AMT:tb=644000:1500
>    a=max-recv-ssrc:* 5
>    a=max-send-ssrc:97 3
>    a=max-send-ssrc:96 1
>    a=ssrc:239245219 param:1
>    a=ssrc:239245219 param:2
>    a=ssrc:986545121 param:1
>    a=ssrc:986545121 param:3
>    a=ssrc:2199983234 param:1
>    a=ssrc:2199983234 param=4
>    m=video 49300 RTP/AVP 96
>    b=AS:8240
>    a=rtpmap:96 H264/90000
>    a=fmtp:96 profile-level-id=42c01e
>    a=imageattr:* send [x=1280,y=720] recv [x=1280,y=720]
>    a=bw:send pt=96 SMT:tb=2500000:8192
>    a=bw:recv pt=96 SMT:tb=3000000:8192
>    a=bw:send pt=* AMT:tb=8000000:16384
>    a=max-recv-ssrc:* 5
>    a=max-send-ssrc:* 3
>    a=ssrc:245219239 param:1
>    a=ssrc:245219239 param:2
>    a=ssrc:545121986 param:1
>    a=ssrc:545121986 param:3
>    a=ssrc:199983234 param:1
>    a=ssrc:199983234 param:4
>    a=content:main
> 
> 
> 
> 
> 2.   v=0
>    o=alice 2890844526 2890842807 IN IP4 192.0.2.156
>    s=Telepresence Alice Invite
>    c=IN IP4 192.0.2.156
>    b=AS:8965
>    t=0 0
>    
>    param = 1 cname:alice@foo.example.com rcname:a3:d3:4b:f1:22:12
>    param = 2 name:alice@foo.example.com srcname:12:3f:ab:d2:ec:32
>    param = 3 cname:alice@foo.example.com srcname:7f:12:db:87:2d:52
> 
> 
> 
>    m=audio 49200 RTP/AVP 97 96
>    b=AS:725
>    a=rtpmap:96 G719/48000/2
>    a=rtpmap:97 G719/48000
>    a=bw:send pt=96 SMT:tb=128800:1500
>    a=bw:send pt=97 SMT:tb=64800:1500
>    a=bw:recv pt=* AMT:tb=644000:1500
>    a=max-recv-ssrc:* 5
>    a=max-send-ssrc:97 3
>    a=max-send-ssrc:96 1
>    a=ssrc:239245219 param:1
>    a=ssrc:986545121 param:2
>    a=ssrc:2199983234 param:3
>    m=video 49300 RTP/AVP 96
>    b=AS:8240
>    a=rtpmap:96 H264/90000
>    a=fmtp:96 profile-level-id=42c01e
>    a=imageattr:* send [x=1280,y=720] recv [x=1280,y=720]
>    a=bw:send pt=96 SMT:tb=2500000:8192
>    a=bw:recv pt=96 SMT:tb=3000000:8192
>    a=bw:send pt=* AMT:tb=8000000:16384
>    a=max-recv-ssrc:* 5
>    a=max-send-ssrc:* 3
>    a=ssrc:245219239 param:1
>    
>    a=ssrc:545121986 param:2
>    
>    a=ssrc:199983234 param:3
>    
>    a=content:main
> 
> 
> 
> 3. the shortest will be the ssrc-id=id:ssrc-value, the id will identify the
> id to use in the media level with all the parameters except the other
> ssrc-id's in the multi line. We can probably give a better syntax but it is
> shorter.
> 
> 
>    v=0
>    o=alice 2890844526 2890842807 IN IP4 192.0.2.156
>    s=Telepresence Alice Invite
>    c=IN IP4 192.0.2.156
>    b=AS:8965
>    t=0 0
> 
> multi = cname:alice@foo.example.com srcname:a3:d3:4b:f1:22:12
> ssrc-id=1:239245219 ssrc-id=2: 245219239
> 
> multi= cname:alice@foo.example.com srcname:12:3f:ab:d2:ec:32
> ssrc-id=3:986545121 ssrc-id=4:545121986
> 
>  multi= cname:alice@foo.example.com srcname:7f:12:db:87:2d:52
> ssrc-id=5:2199983234 ssrc-id=6:199983234  
> 
> 
> 
> m=audio 49200 RTP/AVP 97 96
>    b=AS:725
>    a=rtpmap:96 G719/48000/2
>    a=rtpmap:97 G719/48000
>    a=bw:send pt=96 SMT:tb=128800:1500
>    a=bw:send pt=97 SMT:tb=64800:1500
>    a=bw:recv pt=* AMT:tb=644000:1500
>    a=max-recv-ssrc:* 5
>    a=max-send-ssrc:97 3
>    a=max-send-ssrc:96 1
>    a=multi:1
>    a= multi:3
>    a= multi:5
>    
>    m=video 49300 RTP/AVP 96
>    b=AS:8240
>    a=rtpmap:96 H264/90000
>    a=fmtp:96 profile-level-id=42c01e
>    a=imageattr:* send [x=1280,y=720] recv [x=1280,y=720]
>    a=bw:send pt=96 SMT:tb=2500000:8192
>    a=bw:recv pt=96 SMT:tb=3000000:8192
>    a=bw:send pt=* AMT:tb=8000000:16384
>    a=max-recv-ssrc:* 5
>    a=max-send-ssrc:* 3
>    a= multi:2 
>    a= multi:4 
>    a= multi:6
>    a=content:main
> 
> 
> Roni
> 
> 


-- 

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] Comment on draft-westerlund-avtcore-multistream-and-simulcast
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In the case were we want to use SDP to indicate video streams are =
simulcast of the same thing, I think it would be better to use the SDP =
grouping framework instead of using SSRC to indicate this.=20

I also wonder, why not just use layered codecs, seems like outperform =
the simulcast approach. I read section 3.1 but don't really find it =
convincing that this simulcast is better than layered codecs.=20


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> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On =
Behalf Of Cullen Jennings (fluffy)
> Sent: Wednesday, July 20, 2011 4:36 PM
> To: IETF AVT WG; IETF MMUSIC WG
> Subject: [MMUSIC] Comment =
ondraft-westerlund-avtcore-multistream-and-simulcast
>=20
>=20
> In the case were we want to use SDP to indicate video streams are =
simulcast of the same thing, I think it would be better to
> use the SDP grouping framework instead of using SSRC to indicate this.

As in:
https://datatracker.ietf.org/doc/draft-begen-mmusic-redundancy-grouping/ =


> I also wonder, why not just use layered codecs, seems like outperform =
the simulcast approach. I read section 3.1 but don't
> really find it convincing that this simulcast is better than layered =
codecs.

The draft above provides some motivation why people would wanna transmit =
the same thing twice or more.
=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

From stewe@stewe.org  Wed Jul 20 08:26:50 2011
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On 7.20.2011 07:36 , "Cullen Jennings" <fluffy@cisco.com> wrote:

> 
> In the case were we want to use SDP to indicate video streams are simulcast of
> the same thing, I think it would be better to use the SDP grouping framework
> instead of using SSRC to indicate this.
> 
> I also wonder, why not just use layered codecs, seems like outperform the
> simulcast approach. I read section 3.1 but don't really find it convincing
> that this simulcast is better than layered codecs.
> 


I read the word "scalable" -- can't help to chip in...

>From Magnus' draft, section 3.1, second-to-last paragraph characterizing
scalable coding, with inline comments:

> The scalable video encoding requires a more complex encoder compared
>    to non-scalable encoding.
> 
This is true from an implementation complexity viewpoint.  It is not
necessarily true from a cycle viewpoint, at least not in the encoder.  I
would argue that, in most practical cases, highly optimized scalable and
simulcast encoders require about the same number of cycles to code similar
resolutions.  Perhaps within a +/- 10% range.
In the decoder, modern standards such as H.264 SVC allow for something
called "single loop decoding", which keep the computational complexity at
only marginally higher levels than for non-scalable decoding of the highest
simulcasted solution.  There are cases where layered decoding actually takes
fewer cycles (at very high bitrates, where the overhead due to entropy
decoding of the less efficient simulcasted stream outweighs the complexity
through cross-layer prediction).
> 
> Also, if the resolution difference is big,
>    the scalable codec may in fact be only marginally more bandwidth
>    efficient, between the encoding client and the mixer, than a
>    simulcast that sends the resolutions in separate streams, assuming
>    equivalent video quality.
> 
This is correct, but it's only half of the story.  One operation point for
which most scalable codecs today are optimized is 2:1 spatial scalability,
meaning that the base layer is half the resolution in both dimensions or 1/4
of the pixel count.  Another one is (roughly) 1.5 scalability, with 720p to
1080p in mind.  SVC works still reasonably well at 1:3, although this is not
a common operation point.
The second half of the story is that SVC (as well as previous layered coding
standards) do allow for multilayer scalability.  If you know how to
implement this (it's not trivial), you get quite a bit of the gain, with the
additional advantage of having intermediate operation points that you can
use when video pictures are zoomed under user control.
In either case, modern scalable video standards, well implemented, are
always better than simulcast from a bandwidth viewpoint.  And should be.
That's information theory 101.

> 
> At the same time, with scalable video
>    encoding, the transmission of all but the lowest resolution will
>    definitely consume more bandwidth from the mixer to the other
>    participants than a non-scalable encoding, again assuming equivalent
>    video quality.

Again, this is correct.  Once again, though, only part of the story.
First, he overhead of scalable coding over non-scalable coding of the
highest resolution has decreased to levels that may not quite be background
noise, but not much higher either.  You may wish to take a look at a
document Alex Eleftheriadis and me wrote for rtcweb,
http://tools.ietf.org/html/draft-wenger-rtcweb-layered-codec-00.  It
contains a reference to the characterization tests conducted by MPEG.  Most
tests performed in academia roughly mirror those results.  Second, most of
us sit behind highly asymmetric links, such as cable modem or DSL.  In such
environments, an overhead (even a large one) for the largest picture
resolution is less penalizing than an overhead on the sending side.  Sending
bits are more expensive than receiving bits.

What the draft has omitted are SVC's superior error resilience properties
and congestion control options, when being used in an appropriately designed
system environment.  Some of that stems from temporal scalability (which
modern non-scalable video coding standards include in an occasionally
crippled form), but a lot more comes from spatial scalability.  Alex and my
draft cited above provides some input.

Regards,
Stephan
  




> ___________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
> 



--B_3393995204_1741896
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	charset="US-ASCII"
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<html><head></head><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: s=
pace; -webkit-line-break: after-white-space; color: rgb(0, 0, 0); font-size:=
 14px; font-family: Calibri, sans-serif; "><div>On 7.20.2011 07:36 , "Cullen=
 Jennings" &lt;<a href=3D"mailto:fluffy@cisco.com">fluffy@cisco.com</a>&gt; wr=
ote:</div><div><br></div><blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE"=
 style=3D"border-left-color: rgb(181, 196, 223); border-left-width: 5px; borde=
r-left-style: solid; padding-top: 0px; padding-right: 0px; padding-bottom: 0=
px; padding-left: 5px; margin-top: 0px; margin-right: 0px; margin-bottom: 0p=
x; margin-left: 5px; "><div><br></div><div>In the case were we want to use S=
DP to indicate video streams are simulcast of the same thing, I think it wou=
ld be better to use the SDP grouping framework instead of using SSRC to indi=
cate this.</div><div><br></div><div>I also wonder, why not just use layered =
codecs, seems like outperform the simulcast approach. I read section 3.1 but=
 don't really find it convincing that this simulcast is better than layered =
codecs.</div><div><br></div></blockquote><div><hr align=3D"center" size=3D"&quot=
;3&quot;" width=3D"&quot;95%&quot;"></div><div><br></div><div>I read the word =
"scalable" -- can't help to chip in...</div><div><br></div><div>From Magnus'=
 draft, section 3.1, second-to-last paragraph characterizing scalable coding=
, with inline comments:</div><div><br></div><div><div><blockquote style=3D"mar=
gin:0 0 0 40px; border:none; padding:0px;"><div>The scalable video encoding =
requires a more complex encoder compared</div><div>   to non-scalable encodi=
ng.&nbsp;</div><div><br></div></blockquote><div>This is true from an impleme=
ntation complexity viewpoint. &nbsp;It is not necessarily true from a cycle =
viewpoint, at least not in the encoder. &nbsp;I would argue that, in most pr=
actical cases, highly optimized scalable and simulcast encoders require abou=
t the same number of cycles to code similar resolutions. &nbsp;Perhaps withi=
n a +/- 10% range.</div><div>In the decoder, modern standards such as H.264 =
SVC allow for something called "single loop decoding", which keep the comput=
ational complexity at only marginally higher levels than for non-scalable de=
coding of the highest simulcasted solution. &nbsp;There are cases where laye=
red decoding actually takes fewer cycles (at very high bitrates, where the o=
verhead due to entropy decoding of the less efficient simulcasted stream out=
weighs the complexity through cross-layer prediction).</div><blockquote styl=
e=3D"margin:0 0 0 40px; border:none; padding:0px;"><div><br></div><div>Also, i=
f the resolution difference is big,</div><div>   the scalable codec may in f=
act be only marginally more bandwidth</div><div>   efficient, between the en=
coding client and the mixer, than a</div><div>   simulcast that sends the re=
solutions in separate streams, assuming</div><div>   equivalent video qualit=
y.&nbsp;</div><div><br></div></blockquote><div>This is correct, but it's onl=
y half of the story. &nbsp;One operation point for which most scalable codec=
s today are optimized is 2:1 spatial scalability, meaning that the base laye=
r is half the resolution in both dimensions or 1/4 of the pixel count. &nbsp=
;Another one is (roughly) 1.5 scalability, with 720p to 1080p in mind. &nbsp=
;SVC works still reasonably well at 1:3, although this is not a common opera=
tion point.</div><div>The second half of the story is that SVC (as well as p=
revious layered coding standards) do allow for multilayer scalability. &nbsp=
;If you know how to implement this (it's not trivial), you get quite a bit o=
f the gain, with the additional advantage of having intermediate operation p=
oints that you can use when video pictures are zoomed under user control.</d=
iv><div>In either case, modern scalable video standards, well implemented, a=
re always better than simulcast from a bandwidth viewpoint. &nbsp;And should=
 be. &nbsp;That's information theory 101.</div><div><br></div><blockquote st=
yle=3D"margin:0 0 0 40px; border:none; padding:0px;"><div><br></div><div>At th=
e same time, with scalable video</div><div>   encoding, the transmission of =
all but the lowest resolution will</div><div>   definitely consume more band=
width from the mixer to the other</div><div>   participants than a non-scala=
ble encoding, again assuming equivalent</div><div>   video quality.</div></b=
lockquote><div><br></div><div>Again, this is correct. &nbsp;Once again, thou=
gh, only part of the story.</div><div>First, he overhead of scalable coding =
over non-scalable coding of the highest resolution has decreased to levels t=
hat may not quite be background noise, but not much higher either. &nbsp;You=
 may wish to take a look at a document Alex Eleftheriadis and me wrote for r=
tcweb,&nbsp;http://tools.ietf.org/html/draft-wenger-rtcweb-layered-codec-00.=
 &nbsp;It contains a reference to the characterization tests conducted by MP=
EG. &nbsp;Most tests performed in academia roughly mirror those results. &nb=
sp;Second, most of us sit behind highly asymmetric links, such as cable mode=
m or DSL. &nbsp;In such environments, an overhead (even a large one) for the=
 largest picture resolution is less penalizing than an overhead on the sendi=
ng side. &nbsp;Sending bits are more expensive than receiving bits.</div><di=
v><br></div><div>What the draft has omitted are SVC's superior error resilie=
nce properties and congestion control options, when being used in an appropr=
iately designed system environment. &nbsp;Some of that stems from temporal s=
calability (which modern non-scalable video coding standards include in an o=
ccasionally crippled form), but a lot more comes from spatial scalability. &=
nbsp;Alex and my draft cited above provides some input.</div><div><br></div>=
<div>Regards,</div><div>Stephan</div><div>&nbsp;&nbsp;</div></div><div><br><=
/div></div><div><br></div><div><br></div><div><br></div><blockquote id=3D"MAC_=
OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT: #b5c4df 5 solid; PADDING=
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From jonathan@vidyo.com  Wed Jul 20 10:44:47 2011
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From: Jonathan Lennox <jonathan@vidyo.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Date: Wed, 20 Jul 2011 13:44:43 -0400
Thread-Topic: [AVTCORE] Fwd: I-D Action: draft-westerlund-avtcore-multistream-and-simulcast-00.txt
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Hi, Magnus -- thanks for the reply.

On Jul 19, 2011, at 8:52 AM, Magnus Westerlund wrote:

>>=20
>> As the draft concludes, the multi-stream scenario is best handled by
>> SSRC-multiplexing, as was originally envisioned by RTP.  Given that
>> this is the case, it's not clear to me that it's a good idea to use a
>> separate (and in my opinion, less flexible) mechanism for the
>> specific special case of simulcast.  Instead, given the pressure from
>> groups such as RTCweb to reduce the port consumption of RTP, I'd
>> recommend SSRC-multiplexing across the board.
>=20
> Jonathan lets me ask you a question. Is any hesitation to use several
> RTP sessions when most suitable due to the port consumption issue or for
> some other reason? And here we can debate what is most suitable.

Part of it is port consumption, but more importantly, architecturally, is t=
he problem that architectures that are designed around the "one source (in =
each direction) per session" assumption break badly in a whole lot of scena=
rios that are more complicated than the simple point-to-point one-source-pe=
r-media-type model.

> My view is that multiple SSRCs in an RTP session is a very good
> mechanism for streams that one expect to be treated the same way in both
> central nodes or in end-points. When it comes to sets of streams that
> needs a different treatment, I think it is logical and a powerful
> construct to use the RTP session for separating these sets of streams.

This is valid so long as you can make the simplifying assumption that you *=
can* treat every source in a session identically, blindly, with no further =
information about the content of each one.  But there are plenty of scenari=
os in which

>> Note also that in the case of multiple sources which are each
>> independently simulcast, the simulcast structure of different sources
>> could well be different.   (E.g., source A could be available in two
>> resolutions, while source B has only one and source C has three.)
>> It'd be extremely difficult to map this in session-multiplexed
>> simulcast, whereas for SSRC-multiplexed simulcast it should be no
>> more complex than the single-simulcast-source scenario.
>=20
> This is a valid concern, but which I think we have a reasonable straight
> story for. In the direction you have a certain number of simulcast
> versions you negotiate that number of session and their individual
> configurations.

I'm not thinking about asymmetric sources in each direction -- I'm thinking=
 about multiple, heterogeneous simulcast sources in the *same* direction.  =
In this case, it's really unclear how one would map the simulcast channels =
to multiple sessions.=20

--
Jonathan Lennox
jonathan@vidyo.com



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Hi,

I had a wrong pointer to the feedback-supression-rtp pointing at 00 version
inside 05. It is now updated.

Regards

Roni Even


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class=3DMsoNormal>Hi,<o:p></o:p></p><p class=3DMsoNormal>I had a wrong =
pointer to the feedback-supression-rtp pointing at 00 version inside 05. =
It is now updated.<o:p></o:p></p><p =
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Subject: Re: [AVTCORE] RTP Internal Multiplexing of RTP sessions
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I agree that there is a clear perception of a requirement here, and that =
we should discuss the issue in Quebec to determine if it's a real issue, =
and if so, how we should address the requirement.

Colin


On 20 Jul 2011, at 10:30, Elwell, John wrote:
> I think this is an important topic and should receive some agenda time =
in Quebec.
>=20
> John=20
>=20
>> -----Original Message-----
>> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On=20
>> Behalf Of Magnus Westerlund
>> Sent: 19 July 2011 16:35
>> To: IETF AVTCore WG
>> Subject: [AVTCORE] RTP Internal Multiplexing of RTP sessions
>>=20
>> WG,
>>=20
>> As Individual contributor.
>>=20
>> I don't know if people in the WG are aware of the discussion=20
>> ongoing in
>> RTCWEB around multiplexing of protocols over the same underlying
>> transport flow, like UDP flow. That include RTP sessions.
>>=20
>> There is a proposal:
>> "Multiplexing of Real-Time Transport Protocol (RTP) Traffic=20
>> for Browser
>>                  based Real-Time Communications (RTC)"
>> https://datatracker.ietf.org/doc/draft-rosenberg-rtcweb-rtpmux
>> /?include_text=3D1
>>=20
>> This proposal is criticized by the following draft.
>> https://datatracker.ietf.org/doc/draft-perkins-rtcweb-rtp-usag
>> e/?include_text=3D1
>>=20
>> There is an ongoing discussion on several aspects of these=20
>> proposals on
>> the RTCWEB WG. You are welcome to go join the discussion of the other
>> questions raised on that mailing list.
>>=20
>> Here I primarily want to raise the issue I see with=20
>> multiplexing several
>> RTP sessions onto the same lower layer transport without something
>> explicitly identifying these separate sessions. And in particular the
>> RTP SSRC semantics changes that is proposed by the Rosenberg draft.
>>=20
>> Personally I think it is time we tackle the issue of how to=20
>> efficiently
>> run multiple RTP sessions over one lower layer transport.=20
>> This as there
>> are several wanting to have that functionality. However, I think this
>> must be done in a very careful way. As we don't want to create an
>> alternative version of RTP which is what the Rosenberg=20
>> proposal in fact
>> does. I do want us to gather the requirements on this and think of =
how
>> we can accomplish this.
>>=20
>> I think there is a clear risk that we will not be able to=20
>> accomplish it
>> without either adding some type of shim header which has it sets of
>> drawback or in fact fork or change the RTP protocol in such a way =
that
>> it will not be fully compatible with what exist today. But if=20
>> that is to
>> be done, we should have a very clear consensus on that and ensure =
that
>> we then develop future extensions to RTP within a single=20
>> version of the
>> protocol, not two.
>>=20
>> As WG chair:
>>=20
>> We are considering re-arranging the agenda to facilitate a debate of
>> this topic in Quebec.
>>=20
>> So please indicate both your view of the topic in general and if you
>> think we should re-prioritize the agenda.
>>=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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>>=20
> _______________________________________________
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From: Qin Wu <bill.wu@huawei.com>
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Hi,
I think using tag has both disadvantage and advantage.
please see my below.
----- Original Message ----- 
From: "Roni Even" <even.roni@huawei.com>
To: "'Colin Perkins'" <csp@csperkins.org>; "'Qin Wu'" <sunseawq@huawei.com>
Cc: <avt@ietf.org>
Sent: Wednesday, July 20, 2011 3:19 PM
Subject: RE: [AVTCORE] I-D Action: draft-ietf-avtcore-monarch-03.txt


> Hi,
> Looking at the tag mechanism I agree with Colin that using it just to reduce
> the size of the packet is not the important. 
[Qin]: 
Yes, Using tag to associate identity block with other small blocks doesn't reduce size of RTCP if
1. Each RTCP XR packet contains one or more identity block 
2. Each identity block contains SSRC.
Therefore using tag may introduce extra overhead when it is used to distinguish different stream to be reported
On the other hand,, using tag  is redundant comparing with using SSRC since SSRC also can be used to distinguish 
the different stream to be reported.

If the identity block is not carried in each RTCP XR packet, e.g., at the start of session, you still need to parse each 
small bock with tag included and mapping  the tag to SSRC when receiving RTCP XR packet. It introduces processing
complexity. This is Con part of using tag I know.

> Still I can see a different use case for the tag in order to group XR
> reports as one entity. The monitoring architecture suggest dividing large
> reports to small blocks and the assumption is the by the SSRC you define a
> group of blocks per an application. Using the tag may provide another level
> of freedom allowing more than one tag per SSRC and the tag defines a
> reporting unit. This is a change from the current usage of tags and I am
> bringing it to see if this usage is useful.

[Qin]: Good point. Using SSRC enables grouping per stream
However the Pro of Using tag is may provide more subtle granularity for grouping e.g., 
a. distinguish the video stream and audio stream with the same SSRC by using different tag 
b. distinguish the base layer video stream and enhancement layer video stream with the same SSRC by using different tag.
Does these two cases mentioned above make sense?

> 
> As for the correlation with non-rtp protocols I agree that SDES may be used
> for it.
> 
> Regards
> Roni Even
> As individual
> 
>> -----Original Message-----
>> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
>> Colin Perkins
>> Sent: Tuesday, June 21, 2011 5:56 PM
>> To: Qin Wu
>> Cc: avt@ietf.org
>> Subject: Re: [AVTCORE] I-D Action: draft-ietf-avtcore-monarch-03.txt
>> 
>> Qin,
>> 
>> This does address many of my comments on the earlier versions. Some
>> quick comments on this draft:
>> 
>> - In Section 5.2, there is no need to report the CNAME, since this is
>> redundant with RTCP SDES information.
>> 
>> - Section 5.2 suggests reporting "the sequence number of the first
>> packet of the RTP session". A receiver cannot know what is the first
>> packet of the session, only what was the first packet it received, so
>> this should be clarified to "the sequence number of the first packet
>> received in the RTP session". The first sequence number received also
>> becomes less useful after the sequence number space wraps around, so it
>> might make sense to include a count of sequence number cycles since
>> then.
>> 
>> - Section 5.3 discusses sending information that can be used to
>> correlate an RTP source with non-RTP protocols. This seems like a
>> reasonable thing to add. However, since it is describing the source,
>> rather than providing a quality report, should this be an new RTCP SDES
>> block type, rather than an RTCP XR block?
>> 
>> Colin
>> 
>> 
>> On 16 Jun 2011, at 10:29, Qin Wu wrote:
>> 
>> > Hi,:
>> > In response to the recent comments and suggestion on the list, a new
>> version of draft-ietf-avtcore-monarch is just submitted.
>> > http://www.ietf.org/internet-drafts/draft-ietf-avtcore-monarch-03.txt
>> > The diff is:
>> > http://tools.ietf.org/rfcdiff?url2=draft-ietf-avtcore-monarch-03
>> >
>> > Please check if your comments are addressed in this version.
>> >
>> > Regards!
>> > -Qin
>> >
>> > ----- Original Message -----
>> > From: <internet-drafts@ietf.org>
>> > To: <i-d-announce@ietf.org>
>> > Cc: <avt@ietf.org>
>> > Sent: Thursday, June 16, 2011 5:21 PM
>> > Subject: [AVTCORE] I-D Action: draft-ietf-avtcore-monarch-03.txt
>> >
>> >
>> >> 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 Architectures for RTP
>> >> Author(s)       : Qin Wu
>> >>                         Geoff Hunt
>> >>                         Philip Arden
>> >> Filename        : draft-ietf-avtcore-monarch-03.txt
>> >> Pages           : 23
>> >> Date            : 2011-06-16
>> >>
>> >>  This memo proposes an architecture for extending RTCP with a new
>> RTCP
>> >>  XR (RFC3611) block type to report new metrics regarding media
>> >>  transmission or reception quality, as proposed 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 &quot;mix and
>> match&quot; 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-
>> 03.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-03.txt
>> >> _______________________________________________
>> >> 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
>> 
>> 
>> 
>> --
>> 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  Thu Jul 21 02:24:48 2011
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On 2011-07-20 16:36, Cullen Jennings wrote:
> 
> In the case were we want to use SDP to indicate video streams are
> simulcast of the same thing, I think it would be better to use the
> SDP grouping framework instead of using SSRC to indicate this.

Cullen, I recommend that you go read the draft again.

We are proposing to use SDP grouping framework. We suggest that one
defines two grouping semantics SCS and SCR. One for grouping media lines
where one want to send different simulcast versions, another for
indicating the willingness to receive simulcast over a group of media
lines.

This is to enable simulcast only in one direction and to ensure that we
get offer/answer behavior. The grouping semantics is what is used to
negotiate the actual usage of simulcast or not.

We do have the SRCNAME but that is for actually indicating how different
streams map to different sources. This becomes important when an
end-point has more than one source of the same media type it want to
simulcast, like for example multiple cameras in a telepresence room.

The SRCNAME SDES item doesn't need to be in the SDP, it is only included
in the examples to show that it is possible to have them to speed up
acquisition of this information.

> 
> I also wonder, why not just use layered codecs, seems like outperform
> the simulcast approach. I read section 3.1 but don't really find it
> convincing that this simulcast is better than layered codecs.

Layered coding is less efficient to reach a particular quality level
than a stream that is encoded for that particular level without any
additional quality levels.

This means that layered encoding is guaranteed to be less efficient in
the leg from a central mixer to a receiver for selecting a given media
quality.

Layered encoding is however a bit more efficient in the leg from the
media source to the mixer than simulcasting.

The exact levels of gain varies with the target points, the number of
qualities etc.

We see that a clear use case for combining simulcast with codecs that
has certain scalability features. For example using H.264/AVC or SVC to
provide temporal scalability, but use simulcast for resolution changes.


Cheers

Magnus Westerlund

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From magnus.westerlund@ericsson.com  Thu Jul 21 02:55:30 2011
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Hi Jonathan,

Good points although I am not certain I gotten your point in all the cases.

On 2011-07-20 19:44, Jonathan Lennox wrote:
> 
> Hi, Magnus -- thanks for the reply.
> 
> On Jul 19, 2011, at 8:52 AM, Magnus Westerlund wrote:
> 
>>> 
>>> As the draft concludes, the multi-stream scenario is best handled
>>> by SSRC-multiplexing, as was originally envisioned by RTP.  Given
>>> that this is the case, it's not clear to me that it's a good idea
>>> to use a separate (and in my opinion, less flexible) mechanism
>>> for the specific special case of simulcast.  Instead, given the
>>> pressure from groups such as RTCweb to reduce the port
>>> consumption of RTP, I'd recommend SSRC-multiplexing across the
>>> board.
>> 
>> Jonathan lets me ask you a question. Is any hesitation to use
>> several RTP sessions when most suitable due to the port consumption
>> issue or for some other reason? And here we can debate what is most
>> suitable.
> 
> Part of it is port consumption, but more importantly,
> architecturally, is the problem that architectures that are designed
> around the "one source (in each direction) per session" assumption
> break badly in a whole lot of scenarios that are more complicated
> than the simple point-to-point one-source-per-media-type model.
> 

I fully agree, but fails to understand how multiple streams in each
direction in multiple sessions in one form is different from multiple
streams in multiple session in another way is different.

Session based simulcast will use m RTP sessions per media type with N
streams in them.

SSRC multiplexed simulcast will use 1 RTP session per media type with
m*N streams in them.

In both cases we both have multiple streams and multiple RTP sessions.
It is primarily the number of RTP sessions that is the varying factor.

>> My view is that multiple SSRCs in an RTP session is a very good 
>> mechanism for streams that one expect to be treated the same way in
>> both central nodes or in end-points. When it comes to sets of
>> streams that needs a different treatment, I think it is logical and
>> a powerful construct to use the RTP session for separating these
>> sets of streams.
> 
> This is valid so long as you can make the simplifying assumption that
> you *can* treat every source in a session identically, blindly, with
> no further information about the content of each one.  But there are
> plenty of scenarios in which

Yes, I agree that there are scenarios where you will need to have
detailed understanding of what exactly each stream is. But
SSRC-multiplexed simulcast needs one to always have this control
independently if one needs it or not just to get it working. Thus
preventing applications that can be done simpler to be simpler.

> 
>>> Note also that in the case of multiple sources which are each 
>>> independently simulcast, the simulcast structure of different
>>> sources could well be different.   (E.g., source A could be
>>> available in two resolutions, while source B has only one and
>>> source C has three.) It'd be extremely difficult to map this in
>>> session-multiplexed simulcast, whereas for SSRC-multiplexed
>>> simulcast it should be no more complex than the
>>> single-simulcast-source scenario.
>> 
>> This is a valid concern, but which I think we have a reasonable
>> straight story for. In the direction you have a certain number of
>> simulcast versions you negotiate that number of session and their
>> individual configurations.
> 
> I'm not thinking about asymmetric sources in each direction -- I'm
> thinking about multiple, heterogeneous simulcast sources in the
> *same* direction.  In this case, it's really unclear how one would
> map the simulcast channels to multiple sessions.

That is exactly one of the case we are tackling, unless I misunderstand
you.

An end-point has 2 cameras in a telepresence room capturing different
parts of the participant table so they are of the same purpose in the
application. Those cameras (sources) we want to simulcast to a mixer in
3 qualities.

Then the upstream RTP Session usage would consist of 3 RTP sessions, one
for each quality tier. Within each of these RTP sessions there will be 2
streams (SSRCs) one for each camera sending media from the end-point to
the mixer.

The usage in the other direction can either be independent RTP sessions
or for example be overlapped with one the quality tiers (as we do in the
advanced examples in the draft).

When you talk about multiple sources, are they from the same end-point
of different end-points? If they are from the same end-point I think we
do have a working solution in this document.

If they are from different end-points I think we also have a solution as
long as we are talking mixer based sessions. Because in that case we can
negotiate each upstream end-point to the central mixer independently.
However, the mixer clearly needs to ensure that these streams match the
downstream capabilities setup to the other participants.

If we are talking relay based based then I think we have issues as we
have identified.

Cheers

Magnus Westerlund

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From stephen.botzko@gmail.com  Thu Jul 21 04:12:46 2011
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Date: Thu, 21 Jul 2011 07:12:44 -0400
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From: Stephen Botzko <stephen.botzko@gmail.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Cc: Cullen Jennings <fluffy@cisco.com>, IETF AVT WG <avt@ietf.org>, IETF MMUSIC WG <mmusic@ietf.org>
Subject: Re: [AVTCORE] [MMUSIC] Comment on draft-westerlund-avtcore-multistream-and-simulcast
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in-line
Stephen Botzko

On Thu, Jul 21, 2011 at 5:24 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> On 2011-07-20 16:36, Cullen Jennings wrote:
> >
> > In the case were we want to use SDP to indicate video streams are
> > simulcast of the same thing, I think it would be better to use the
> > SDP grouping framework instead of using SSRC to indicate this.
>
> Cullen, I recommend that you go read the draft again.
>
> We are proposing to use SDP grouping framework. We suggest that one
> defines two grouping semantics SCS and SCR. One for grouping media lines
> where one want to send different simulcast versions, another for
> indicating the willingness to receive simulcast over a group of media
> lines.
>
> This is to enable simulcast only in one direction and to ensure that we
> get offer/answer behavior. The grouping semantics is what is used to
> negotiate the actual usage of simulcast or not.
>
> We do have the SRCNAME but that is for actually indicating how different
> streams map to different sources. This becomes important when an
> end-point has more than one source of the same media type it want to
> simulcast, like for example multiple cameras in a telepresence room.
>
> The SRCNAME SDES item doesn't need to be in the SDP, it is only included
> in the examples to show that it is possible to have them to speed up
> acquisition of this information.
>
> >
> > I also wonder, why not just use layered codecs, seems like outperform
> > the simulcast approach. I read section 3.1 but don't really find it
> > convincing that this simulcast is better than layered codecs.
>
> Layered coding is less efficient to reach a particular quality level
> than a stream that is encoded for that particular level without any
> additional quality levels.
>
> This means that layered encoding is guaranteed to be less efficient in
> the leg from a central mixer to a receiver for selecting a given media
> quality.
>
> Layered encoding is however a bit more efficient in the leg from the
> media source to the mixer than simulcasting.
>
> The exact levels of gain varies with the target points, the number of
> qualities etc.
>
> We see that a clear use case for combining simulcast with codecs that
> has certain scalability features. For example using H.264/AVC or SVC to
> provide temporal scalability, but use simulcast for resolution changes.
>
> I agree that simulcast is important, including the case where two differe=
nt
codecs are used.  In that scenario, the resolution might be the same.

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

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in-line<br>Stephen Botzko<br><br><div class=3D"gmail_quote">On Thu, Jul 21,=
 2011 at 5:24 AM, Magnus Westerlund <span dir=3D"ltr">&lt;<a href=3D"mailto=
:magnus.westerlund@ericsson.com">magnus.westerlund@ericsson.com</a>&gt;</sp=
an> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;"><div class=3D"im">On 2011-07-20 16:36, Cull=
en Jennings wrote:<br>
&gt;<br>
&gt; In the case were we want to use SDP to indicate video streams are<br>
&gt; simulcast of the same thing, I think it would be better to use the<br>
&gt; SDP grouping framework instead of using SSRC to indicate this.<br>
<br>
</div>Cullen, I recommend that you go read the draft again.<br>
<br>
We are proposing to use SDP grouping framework. We suggest that one<br>
defines two grouping semantics SCS and SCR. One for grouping media lines<br=
>
where one want to send different simulcast versions, another for<br>
indicating the willingness to receive simulcast over a group of media<br>
lines.<br>
<br>
This is to enable simulcast only in one direction and to ensure that we<br>
get offer/answer behavior. The grouping semantics is what is used to<br>
negotiate the actual usage of simulcast or not.<br>
<br>
We do have the SRCNAME but that is for actually indicating how different<br=
>
streams map to different sources. This becomes important when an<br>
end-point has more than one source of the same media type it want to<br>
simulcast, like for example multiple cameras in a telepresence room.<br>
<br>
The SRCNAME SDES item doesn&#39;t need to be in the SDP, it is only include=
d<br>
in the examples to show that it is possible to have them to speed up<br>
acquisition of this information.<br>
<div class=3D"im"><br>
&gt;<br>
&gt; I also wonder, why not just use layered codecs, seems like outperform<=
br>
&gt; the simulcast approach. I read section 3.1 but don&#39;t really find i=
t<br>
&gt; convincing that this simulcast is better than layered codecs.<br>
<br>
</div>Layered coding is less efficient to reach a particular quality level<=
br>
than a stream that is encoded for that particular level without any<br>
additional quality levels.<br>
<br>
This means that layered encoding is guaranteed to be less efficient in<br>
the leg from a central mixer to a receiver for selecting a given media<br>
quality.<br>
<br>
Layered encoding is however a bit more efficient in the leg from the<br>
media source to the mixer than simulcasting.<br>
<br>
The exact levels of gain varies with the target points, the number of<br>
qualities etc.<br>
<br>
We see that a clear use case for combining simulcast with codecs that<br>
has certain scalability features. For example using H.264/AVC or SVC to<br>
provide temporal scalability, but use simulcast for resolution changes.<br>
<br></blockquote><div>I agree that simulcast is important, including the ca=
se where two different codecs are used.=A0 In that scenario, the resolution=
 might be the same. <br></div><blockquote class=3D"gmail_quote" style=3D"ma=
rgin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding=
-left: 1ex;">

<br>
Cheers<br>
<br>
Magnus Westerlund<br>
<br>
----------------------------------------------------------------------<br>
Multimedia Technologies, Ericsson Research EAB/TVM<br>
----------------------------------------------------------------------<br>
Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0<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>
<div class=3D"im">Audio/Video Transport Core Maintenance<br>
</div><a href=3D"mailto:avt@ietf.org">avt@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/avt" target=3D"_blank">htt=
ps://www.ietf.org/mailman/listinfo/avt</a><br>
</blockquote></div><br>

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Date: Thu, 21 Jul 2011 16:21:01 +0300
From: Roni Even <Even.roni@huawei.com>
To: avt@ietf.org, rtcweb@ietf.org
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Subject: [AVTCORE] Updated AVTcore agenda - new slot for multiplexing RTP for broweser based RTC
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Hi,

We apologize for the last minute change. 

The topic of "Multiplexing of Real-Time Transport Protocol (RTP) Traffic for
Browser based Real-Time Communications (RTC)" proposed in
draft-rosenberg-rtcweb-rtpmux-00 caused a long discussion in RTCweb mailing
list and the feeling is that it should be also discussed in AVTcore.  Magnus
has started a thread on the AVTcore list.
In order to have a time slot we moved the payload WG status presentation to
the AVText slot which will be on Wednesday afternoon. 
We had to remove the mprtp presentation based on low discussion on the
mailing list. We may have one slide on mprtp during the chair status
 
Please review the above draft, I recommend also to look at
http://tools.ietf.org/html/draft-perkins-rtcweb-rtp-usage-02 
 
Thanks
Roni Even

 

 


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vlink=3Dpurple><div class=3DWordSection1><p =
class=3DMsoNormal>Hi,<o:p></o:p></p><p class=3DMsoNormal>We apologize =
for the last minute change. <o:p></o:p></p><pre><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'>The topic =
of &quot;Multiplexing of Real-Time Transport Protocol (RTP) Traffic for =
Browser based Real-Time Communications (RTC)&quot; proposed in =
&nbsp;&nbsp;draft-rosenberg-rtcweb-rtpmux-00 caused a long discussion in =
RTCweb mailing list and the feeling is that it should be also discussed =
in AVTcore. &nbsp;Magnus has started a thread on the AVTcore =
list.<o:p></o:p></span></pre><pre><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'>In order =
to have a time slot we moved the payload WG status presentation to the =
AVText slot which will be on Wednesday afternoon. =
<o:p></o:p></span></pre><pre><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'>We had to =
remove the mprtp presentation based on low discussion on the mailing =
list. We may have one slide on mprtp during the chair =
status<o:p></o:p></span></pre><pre><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'><o:p>&nbsp;=
</o:p></span></pre><pre><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'>Please =
review the above draft, I recommend also to look at <a =
href=3D"http://tools.ietf.org/html/draft-perkins-rtcweb-rtp-usage-02">htt=
p://tools.ietf.org/html/draft-perkins-rtcweb-rtp-usage-02</a> =
<o:p></o:p></span></pre><pre><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'><o:p>&nbsp;=
</o:p></span></pre><pre><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'>Thanks<o:p>=
</o:p></span></pre><pre><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'>Roni =
Even<o:p></o:p></span></pre><p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></body></html>=

--Boundary_(ID_1T7UMdrcuhl+hRF3i/yHzw)--

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Audio/Video Transport Core Maintenance  (AVTCore) Working Group
===============================================================

CHAIRS:  Magnus Westerlund       <magnus.westerlund@ericsson.com>
         Roni Even         <even.roni@huawei.com>

AGENDA

Wednesday, 27 July 2011 at 9:00-11:30 (208 AB)
----------------------------------------------
09:00   AVTCore Status Update                               (Chairs, 20)

09:20   Encryption of Header Extensions in SRTP             (Lennox, 15)
        draft-ietf-avtcore-srtp-encrypted-header-ext-00

09:35   RTCP Extension for Feedback Suppression Indication  (Qin Wu, 15)
        draft-ietf-avtcore-feedback-supression-rtp-05

09:50   Monitoring Architecture for RTP                     (Qin Wu, 15)
        draft-ietf-avtcore-monarch-03

10:05   RTP Multiple Stream Sessions and Simulcast(Magnus Westerlund , 30)
        draft-westerlund-avtcore-multistream-and-simulcast-00

10:35   Delayed Duplication Attribute and Duplication Grouping in SDP(Begen, 15)
        draft-begen-mmusic-temporal-interleaving-02
        draft-begen-mmusic-redundancy-grouping-01

10:50   Retransmission for SSM Sessions                      (Begen, 10)
        draft-vancaenegem-avtcore-retransmission-for-ssm-00

11:00   Multiplexing of RTP trafic for Browser based RTC (Rosenberg, 30)
        draft-rosenberg-rtcweb-rtpmux-00

11:30   End

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<html>
<head>
<title>Audio/Video Transport Core Maintenance  (AVTCore) Working Group Agenda</title>
</head>
<body bgcolor="#ffffff"><h1>Audio/Video Transport Core Maintenance  (AVTCore) Working Group Agenda</h1>
<hr>
<p>
<h3>
Chairs: Magnus Westerlund       <magnus.westerlund@ericsson.com>, Roni Even         <even.roni@huawei.com>
</h3>
<p>
<h2>Wednesday, 27 July 2011 at 9:00-11:30 (208 AB)</h2>
<p>
<table border="0" cellpadding="5">
<tr>
<th align="left">09:00
<th align="left">AVTCore Status Update<th align="left">Chairs
<tr>
<th align="left">09:20
<th align="left">Encryption of Header Extensions in SRTP<th align="left">Lennox
<tr>
<td align="left">
<td align="left"><a href="http://tools.ietf.org/id/draft-ietf-avtcore-srtp-encrypted-header-ext-00">draft-ietf-avtcore-srtp-encrypted-header-ext-00</a>
<tr>
<th align="left">09:35
<th align="left">RTCP Extension for Feedback Suppression Indication<th align="left">Qin Wu
<tr>
<td align="left">
<td align="left"><a href="http://tools.ietf.org/id/draft-ietf-avtcore-feedback-supression-rtp-05">draft-ietf-avtcore-feedback-supression-rtp-05</a>
<tr>
<th align="left">09:50
<th align="left">Monitoring Architecture for RTP<th align="left">Qin Wu
<tr>
<td align="left">
<td align="left"><a href="http://tools.ietf.org/id/draft-ietf-avtcore-monarch-03">draft-ietf-avtcore-monarch-03</a>
<tr>
<th align="left">10:05
<th align="left">RTP Multiple Stream Sessions and Simulcast<th align="left">Magnus Westerlund 
<tr>
<td align="left">
<td align="left"><a href="http://tools.ietf.org/id/draft-westerlund-avtcore-multistream-and-simulcast-00">draft-westerlund-avtcore-multistream-and-simulcast-00</a>
<tr>
<th align="left">10:35
<th align="left">Delayed Duplication Attribute and Duplication Grouping in SDP<th align="left">Begen
<tr>
<td align="left">
<td align="left"><a href="http://tools.ietf.org/id/draft-begen-mmusic-temporal-interleaving-02">draft-begen-mmusic-temporal-interleaving-02</a>
<tr>
<td align="left">
<td align="left"><a href="http://tools.ietf.org/id/draft-begen-mmusic-redundancy-grouping-01">draft-begen-mmusic-redundancy-grouping-01</a>
<tr>
<th align="left">10:50
<th align="left">Retransmission for SSM Sessions<th align="left">Begen
<tr>
<td align="left">
<td align="left"><a href="http://tools.ietf.org/id/draft-vancaenegem-avtcore-retransmission-for-ssm-00">draft-vancaenegem-avtcore-retransmission-for-ssm-00</a>
<tr>
<th align="left">11:00
<th align="left">Multiplexing of RTP trafic for Browser based RTC<th align="left">Rosenberg
<tr>
<td align="left">
<td align="left"><a href="http://tools.ietf.org/id/draft-rosenberg-rtcweb-rtpmux-00">draft-rosenberg-rtcweb-rtpmux-00</a>
<tr>
<th align="left">11:30
<th align="left">End</table>

--Boundary_(ID_JK5DzPDoqh6oPJ3Nh+Btzw)--

From harald@alvestrand.no  Thu Jul 21 07:33:18 2011
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Date: Thu, 21 Jul 2011 16:33:11 +0200
From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [AVTCORE] RTP Internal Multiplexing of RTP sessions
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Speaking as RTCWEB contributor....

I think the discussion in RTCWEB has shown clearly that there isn't a=20
clear consensus in the community on what an RTP session is, what the=20
existence of RTP sessions requires from the applications, and what the=20
RTP session is an useful tool for.

The debate has been framed in terms of the cost (in both complexity and=20
network resources) of running multiple transports when no difference in=20
treatment is desired from the network - but I think the confusion goes=20
deeper than that; in the discussions in the W3C WEBRTC WG, which deals=20
with the API requirements for the same thing, there doesn't seem to be=20
any sign that anyone sees the need to represent an "RTP session" at the=20
API offered to the Web applications using RTCWEB.

I'm looking forward to the discussion, both inside and outside the=20
AVTCORE session, in Quebec.

                      Harald

On 07/19/11 17:35, Magnus Westerlund wrote:
> WG,
>
> As Individual contributor.
>
> I don't know if people in the WG are aware of the discussion ongoing in=

> RTCWEB around multiplexing of protocols over the same underlying
> transport flow, like UDP flow. That include RTP sessions.
>
> There is a proposal:
> "Multiplexing of Real-Time Transport Protocol (RTP) Traffic for Browser=

>                    based Real-Time Communications (RTC)"
> https://datatracker.ietf.org/doc/draft-rosenberg-rtcweb-rtpmux/?include=
_text=3D1
>
> This proposal is criticized by the following draft.
> https://datatracker.ietf.org/doc/draft-perkins-rtcweb-rtp-usage/?includ=
e_text=3D1
>
> There is an ongoing discussion on several aspects of these proposals on=

> the RTCWEB WG. You are welcome to go join the discussion of the other
> questions raised on that mailing list.
>
> Here I primarily want to raise the issue I see with multiplexing severa=
l
> RTP sessions onto the same lower layer transport without something
> explicitly identifying these separate sessions. And in particular the
> RTP SSRC semantics changes that is proposed by the Rosenberg draft.
>
> Personally I think it is time we tackle the issue of how to efficiently=

> run multiple RTP sessions over one lower layer transport. This as there=

> are several wanting to have that functionality. However, I think this
> must be done in a very careful way. As we don't want to create an
> alternative version of RTP which is what the Rosenberg proposal in fact=

> does. I do want us to gather the requirements on this and think of how
> we can accomplish this.
>
> I think there is a clear risk that we will not be able to accomplish it=

> without either adding some type of shim header which has it sets of
> drawback or in fact fork or change the RTP protocol in such a way that
> it will not be fully compatible with what exist today. But if that is t=
o
> be done, we should have a very clear consensus on that and ensure that
> we then develop future extensions to RTP within a single version of the=

> protocol, not two.
>
> As WG chair:
>
> We are considering re-arranging the agenda to facilitate a debate of
> this topic in Quebec.
>
> So please indicate both your view of the topic in general and if you
> think we should re-prioritize the agenda.
>
> 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
>



From tom111.taylor@bell.net  Fri Jul 22 13:02:18 2011
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From: Tom Taylor <tom111.taylor@bell.net>
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Subject: [AVTCORE] Multicast transition BoF (MULTRANS)
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A BoF on multicast operation in scenarios where some components 
(endpoints, network) support IPv4 and others support IPv6 is being held 
Monday afternoon from 1-3 pm in 205 ABC. Some people on this list may be 
interested. Starting points are
  - draft-tsou-multicast-transition-taxonomy-01, which gives some
    background on older work in the area but is out of date in some
    aspects

  - draft-jaclee-behave-v4v6-mcast-ps-02, which is the primary content
    being placed before the BoF.

Tom Taylor

From magnus.westerlund@ericsson.com  Sat Jul 23 03:58:32 2011
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Subject: [AVTCORE] Comments on draft-begen-mmusic-redundancy-grouping-01
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Hi,

(Primarily as individual, but the WG home question is as WG chair)

I have reviewed this draft and some comments on it.

1. If this solution proposal is to be taken forward I think it needs to
be discussed in AVTCORE. The reason is that it has several RTP details
that needs to be figured out to make stream duplication issue free in a
general usage. I will take up a few of these issues in the below comments.

2. One aspect this document has is a relation to what is discussed in
draft-westerlund-avtcore-multistream-and-simulcast-00. Stream
duplication both results in a certain form of multi-stream and also has
a similar but slightly different semantics to simulcast in that the
streams are two representations of a single original media source. In
this case they are clones, but not true duplicates due to that RTP
doesn't handle try duplicates well.

3. Another issue that also the authors has identified is the question of
what one do when one has multiple media streams from the same end-point.
And in a general case one can't use the CNAME to separate the sources.
The reason is that two media sources that has a common timing
(synchronization) context do needs to indicate that by having the same
CNAME. Thus there are need for some additional mechanism to ensure that
SSRC carrying RTP media streams that are duplicates of each other do in
fact be labeled explicitly. I also note that a SDP based solution is not
sufficient to resolve this issue. I think we need something in RTP, so
that SSRC collisions can be handled.

4. In addition there is a few questions how this duplication really
should set some RTP fields. Is the duplication only working on Sequence
number level? There seem to be additional questions, for example in the
need to have RTP timestamp considerations between the streams. They need
to point identically into the joint timeline. Otherwise outages, or
receiver taking synch from one and not the other stream can end up with
different synchronizations. Becomes especially interesting to discuss
when the time shifted transmission.

5. The draft also discussed Payload Type differences and have them in
the SDP examples. I don't see how that is appropriate. If the streams
are true encoding duplicates then they should use the same payload type
configuration, and thus a single payload type should be sufficient and
in fact recommended, unless the source encoding truly changes.

6. I think it is worth discussing also the benefits of SSRC multiplexed
versus the session multiplexed. In multicast context there is clear
benefits from a control and congestion perspective to have session
multiplexed versions and these mapped onto different multicast groups
enabling a receiver to choose one or both streams.

7. Thus I think the signalling aspects of this is the minor part.
Although there clearly are some aspects of this needing discussion. For
example clearer offer/answer rules for handling these.

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 gonzalo.camarillo@ericsson.com  Sat Jul 23 09:51:42 2011
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Date: Sat, 23 Jul 2011 12:51:38 -0400
From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: [AVTCORE] Request to review a SIPREC draft related to RTP
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Folks,

the SIPREC WG is working on a description about how RTP will be used in
SIPREC environments. We would like to get someone from AVTCORE to have a
look at the document to make sure it is OK from an RTP architecture
perspective.

http://tools.ietf.org/html/draft-eckel-siprec-rtp-rec-01

Thanks,

Gonzalo

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Date: Sat, 23 Jul 2011 19:27:23 +0000
From: Qin Wu <bill.wu@huawei.com>
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From Even.roni@huawei.com  Sun Jul 24 04:34:02 2011
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Date: Sun, 24 Jul 2011 14:30:43 +0300
From: Roni Even <Even.roni@huawei.com>
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Subject: Re: [AVTCORE] Request to review a SIPREC draft related to RTP - please repond to SIPrec list
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Hi,
If you review this work please respond to sipr@ietf.org and not avtcore list

Roni Even
AVTcore  co-chair

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
> Gonzalo Camarillo
> Sent: Saturday, July 23, 2011 7:52 PM
> To: avt@ietf.org
> Subject: [AVTCORE] Request to review a SIPREC draft related to RTP
> 
> Folks,
> 
> the SIPREC WG is working on a description about how RTP will be used in
> SIPREC environments. We would like to get someone from AVTCORE to have
> a
> look at the document to make sure it is OK from an RTP architecture
> perspective.
> 
> http://tools.ietf.org/html/draft-eckel-siprec-rtp-rec-01
> 
> Thanks,
> 
> Gonzalo
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt


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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: "Qin Wu" <bill.wu@huawei.com>, "Gonzalo Camarillo" <Gonzalo.Camarillo@ericsson.com>, <avt@ietf.org>
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Hi Qin,

Thank you for reviewing the draft and providing comments. Please see =
inline for my responses.

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of =
Qin Wu
> Sent: Saturday, July 23, 2011 12:27 PM
> To: Gonzalo Camarillo; avt@ietf.org
> Subject: [AVTCORE] =B4=F0=B8=B4: Request to review a SIPREC draft =
related to RTP
>=20
> Hi,
> I have read this draft and seems interesting and on the right track, =
However I have some comments
> below:
> 1. According to figure 1, the CS(Communication Session) is running =
between UA and SRS. However
> according to
> draft-ietf-siprec-architecture, CS is running between two UAs. So it =
seems there is a little
> inconsistency
> with this. Does it indicate SRS should be sitting at the UA? How to =
deal with the case where the SRS
> is sitting
> between two UA, in this case, CS is just forwarded by SRS.

You are correct. In this case, the SRC is acting as the other UA within =
the CS. This is similar to Figure 2 of draft-ietf-siprec-architecture. =
The case you mention is similar to Figure 1 of =
draft-ietf-siprec-architecture. This draft is meant to apply to both =
cases. Do you think it would be help if I added another diagram and =
additional descriptions regarding this?

> 2.In section 9, it said:
> "   A mixer sends one SDES packet containing a
>    chunk for each contributing source from which it is receiving SDES
>    information, or multiple complete SDES packets.
> "
> So the question is in which case,the Mixer should send multiple =
complete SDES packets instead of
> sending only on SDES packet containing all
> SDES information?

My understanding is that this is up to implementation of the mixer. As =
such, an SRC needs to be able to handle both cases when dealing with a =
CS involving a mixer. Similarly, a SRC may act as a mixer toward the =
SRS, so the SRS needs to be able to handle both cases as well.
Let me know if that addresses your concern, and if so, I will add it as =
clarification.

> 3. In section 12.1.1, in which circumstance the xml schema for video =
fast update will be used instead
> of RTCP feedback method?

Its use is discouraged; however, it is mentioned here such that an =
SRC/SRS is aware of it and able to send/receive it when necessary. For =
example, if I UA does not support RTCP feedback, a SRC may want to use =
SIP INFO.
Let me know if that addresses your concern, and if so, I will add it as =
clarification.
=20
> 4. In section 12.2, why Using the FIR command to recover from errors =
is explicitly
>    disallowed, instead, using PLI?

I am merely restating the restrictions specified in RFC 5104 in order to =
highlight them. My intention is not to add any additional restrictions =
that are SIPREC specific.

Cheers,
Charles
=20
> Regards!
> -Qin
> ________________________________________
> =B7=A2=BC=FE=C8=CB: avt-bounces@ietf.org [avt-bounces@ietf.org] =
=B4=FA=B1=ED Gonzalo Camarillo
> [Gonzalo.Camarillo@ericsson.com]
> =B7=A2=CB=CD=CA=B1=BC=E4: 2011=C4=EA7=D4=C224=C8=D5 0:51
> =B5=BD: avt@ietf.org
> =D6=F7=CC=E2: [AVTCORE] Request to review a SIPREC draft related to =
RTP
>=20
> Folks,
>=20
> the SIPREC WG is working on a description about how RTP will be used =
in
> SIPREC environments. We would like to get someone from AVTCORE to have =
a
> look at the document to make sure it is OK from an RTP architecture
> perspective.
>=20
> http://tools.ietf.org/html/draft-eckel-siprec-rtp-rec-01
>=20
> Thanks,
>=20
> Gonzalo
> _______________________________________________
> 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

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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>, "mmusic (E-mail)" <mmusic@ietf.org>, "IETF AVTCore WG" <avt@ietf.org>
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Subject: Re: [AVTCORE] Comments on draft-begen-mmusic-redundancy-grouping-01
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> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of =
Magnus Westerlund
> Sent: Friday, July 22, 2011 8:39 AM
> To: mmusic (E-mail); IETF AVTCore WG
> Subject: [AVTCORE] Comments on =
draft-begen-mmusic-redundancy-grouping-01
>=20
> Hi,
>=20
> (Primarily as individual, but the WG home question is as WG chair)
>=20
> I have reviewed this draft and some comments on it.

Thanks for the review.
=20
> 1. If this solution proposal is to be taken forward I think it needs =
to
> be discussed in AVTCORE. The reason is that it has several RTP details
> that needs to be figured out to make stream duplication issue free in =
a
> general usage. I will take up a few of these issues in the below =
comments.

That is why we asked for a slot in avtcore on wed.
=20
> 2. One aspect this document has is a relation to what is discussed in
> draft-westerlund-avtcore-multistream-and-simulcast-00. Stream
> duplication both results in a certain form of multi-stream and also =
has
> a similar but slightly different semantics to simulcast in that the
> streams are two representations of a single original media source. In
> this case they are clones, but not true duplicates due to that RTP
> doesn't handle try duplicates well.

The data carried in the redundant streams are identical but the rtp =
header differs due to the different SSRCs. That is all. SSRC muxing =
nicely fits the use case we are tackling with. Simulcasting is quite =
different from our use case because in simulcasting different RTP =
streams most likely carry different data.
=20
> 3. Another issue that also the authors has identified is the question =
of
> what one do when one has multiple media streams from the same =
end-point.

>From the same end-point (e.g., source) there could of course of multiple =
media streams. You can use SSRC muxing for example to identify your =
streams and use ssrc-group to group them if needed.

The concern in that editorial note was what if there were multiple =
streams in each m line but what if you wanted grouping/association among =
only a subset.

For example,

        m=3Dvideo 30000 RTP/AVP 101 102
        ....
        m=3Dvideo 30000 RTP/AVP 103 104

I can easily associate 101 and 102 if I want to, but what if 101 and 103 =
were associated with each other. Can I use a=3Dgroup line? Probably not. =
But, I don't know why you would need that kind of relation in the first =
place (even if you need, there are probably less messy ways to express =
them). If that had been a very desirable feature, SDP and grouping =
semantics along with RTP would not have defined in their current form I =
guess.

> And in a general case one can't use the CNAME to separate the sources.
> The reason is that two media sources that has a common timing
> (synchronization) context do needs to indicate that by having the same
> CNAME. Thus there are need for some additional mechanism to ensure =
that

If 101 and 103 are associated, why can't I use the same CNAME for those =
streams? If 102 and 104 are associated/related, I would use a common =
CNAME for them, too (but theirs would be separate from 101 and 103's).

> SSRC carrying RTP media streams that are duplicates of each other do =
in
> fact be labeled explicitly. I also note that a SDP based solution is =
not
> sufficient to resolve this issue. I think we need something in RTP, so
> that SSRC collisions can be handled.

I can't say I am following.

> 4. In addition there is a few questions how this duplication really
> should set some RTP fields. Is the duplication only working on =
Sequence

That is the easiest way for a middle box or endpoint doing the stream =
merging (duplicate suppression). If seqnums are different, what would =
one do? look into the payload?

> number level? There seem to be additional questions, for example in =
the
> need to have RTP timestamp considerations between the streams. They =
need
> to point identically into the joint timeline. Otherwise outages, or
> receiver taking synch from one and not the other stream can end up =
with
> different synchronizations. Becomes especially interesting to discuss
> when the time shifted transmission.

The duplicate streams are coming from the same source, so they have the =
same timestamp as well as seqnums. It has to be this way because the =
node merging the streams will not re-write anything at all. It is a =
simple suppression operation. Beauty of it is that things stay pretty =
simple to use and implement without any changes to RTP/RTCP plane at =
all.

As for the time-shifted transmission, I suppose you are referring to the =
temporal diversity/delayed duplication =
(draft-begen-mmusic-temporal-interleaving-02). Well, in that case, =
timestamps are still the same for the reasons above. Changing the =
timestamps for the delayed packets does not bring any benefit (actually =
it complicates things) since the ultimate RTP application will still =
receive one and only one RTP stream.

> 5. The draft also discussed Payload Type differences and have them in
> the SDP examples. I don't see how that is appropriate. If the streams

When you are doing SSRC muxing, you put multiple streams in the same m =
line and they end up having different PT numbers. What is wrong with =
that? If they are in separate m lines, they could have the same PT =
number.

> are true encoding duplicates then they should use the same payload =
type
> configuration, and thus a single payload type should be sufficient and
> in fact recommended, unless the source encoding truly changes.

See above.

> 6. I think it is worth discussing also the benefits of SSRC =
multiplexed
> versus the session multiplexed. In multicast context there is clear
> benefits from a control and congestion perspective to have session
> multiplexed versions and these mapped onto different multicast groups
> enabling a receiver to choose one or both streams.

This is what we have in section 5.2. Different streams can be sent in =
different multicast sessions but it does not mean that they are separate =
RTP sessions. I discussed this in detail with Colin. Thanks to him for =
trying to teach me over and over what an RTP session actually means.
=20
> 7. Thus I think the signalling aspects of this is the minor part.
> Although there clearly are some aspects of this needing discussion. =
For
> example clearer offer/answer rules for handling these.

We have some text around O/A stuff. But frankly, I don=92t see any use =
case where duplication will be used in a unicast application. I mean if =
redundancy is needed, one should use FEC instead for unicast transport. =
Duplication is a nice use case for multicast transport where the bw =
overhead is easier to justify and the implementation is easier. And one =
would naturally implement this spec in networks where he already knows =
that the network elements support this spec. So, maybe we should remove =
the whole O/A text anyway and write that this will only be used =
declaratively.

-acbegen
=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  Mon Jul 25 04:54:35 2011
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Subject: Re: [AVTCORE] Comments on draft-begen-mmusic-redundancy-grouping-01
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On 2011-07-24 23:21, Ali C. Begen (abegen) wrote:
> 
> 
>> -----Original Message----- From: avt-bounces@ietf.org
>> [mailto:avt-bounces@ietf.org] On Behalf Of Magnus Westerlund Sent:
>> Friday, July 22, 2011 8:39 AM To: mmusic (E-mail); IETF AVTCore WG 
>> Subject: [AVTCORE] Comments on
>> draft-begen-mmusic-redundancy-grouping-01
>> 
>> Hi,
>> 
>> (Primarily as individual, but the WG home question is as WG chair)
>> 
>> I have reviewed this draft and some comments on it.
> 
> Thanks for the review.
> 
>> 1. If this solution proposal is to be taken forward I think it
>> needs to be discussed in AVTCORE. The reason is that it has several
>> RTP details that needs to be figured out to make stream duplication
>> issue free in a general usage. I will take up a few of these issues
>> in the below comments.
> 
> That is why we asked for a slot in avtcore on wed.
> 
>> 2. One aspect this document has is a relation to what is discussed
>> in draft-westerlund-avtcore-multistream-and-simulcast-00. Stream 
>> duplication both results in a certain form of multi-stream and also
>> has a similar but slightly different semantics to simulcast in that
>> the streams are two representations of a single original media
>> source. In this case they are clones, but not true duplicates due
>> to that RTP doesn't handle try duplicates well.
> 
> The data carried in the redundant streams are identical but the rtp
> header differs due to the different SSRCs. That is all. SSRC muxing
> nicely fits the use case we are tackling with. Simulcasting is quite
> different from our use case because in simulcasting different RTP
> streams most likely carry different data.

I agree that the purpose for multiple RTP streams from the same original
media source are different from the primary case I set out to solve. But
as Jonathan Lennox pointed out, redundancy and FEC have very similar
behaviors to simulcast with different encodings of the same original
source.

Thus, I do see a need to discuss for example how you identify the
streams that are a duplicate of each other.

> 
>> 3. Another issue that also the authors has identified is the
>> question of what one do when one has multiple media streams from
>> the same end-point.
> 
>> From the same end-point (e.g., source) there could of course of
>> multiple media streams. You can use SSRC muxing for example to
>> identify your streams and use ssrc-group to group them if needed.
> 
> The concern in that editorial note was what if there were multiple
> streams in each m line but what if you wanted grouping/association
> among only a subset.
> 
> For example,
> 
> m=video 30000 RTP/AVP 101 102 .... m=video 30000 RTP/AVP 103 104
> 
> I can easily associate 101 and 102 if I want to, but what if 101 and
> 103 were associated with each other. Can I use a=group line? Probably
> not. But, I don't know why you would need that kind of relation in
> the first place (even if you need, there are probably less messy ways
> to express them). If that had been a very desirable feature, SDP and
> grouping semantics along with RTP would not have defined in their
> current form I guess.

Yes, that is a very valid question also for simulcast.

And I think we can provide a story for doing that. But I don't think it
should be based on using the RTP payload type to identify either of the
versions.

> 
>> And in a general case one can't use the CNAME to separate the
>> sources. The reason is that two media sources that has a common
>> timing (synchronization) context do needs to indicate that by
>> having the same CNAME. Thus there are need for some additional
>> mechanism to ensure that
> 
> If 101 and 103 are associated, why can't I use the same CNAME for
> those streams? If 102 and 104 are associated/related, I would use a
> common CNAME for them, too (but theirs would be separate from 101 and
> 103's).

If SSRC A has a duplciation in SSRC C and SSRC B has a duplication in
SSRC D and both A and B comes from the same end-point and have a common
synchronization context, then all of A-D needs to have the same CNAME.

And I get the impression that you are using a payload type based
association scheme here. And I think that is a bad idea. For example it
can't support particular many streams. It also looks semantically wrong,
as if Stream A uses PT 96, then SSRC C should use PT 96 also for the
corresponding packet duplicates. And if A switches from one PT to
another the other stream needs to follow.

Thus I think a SDES item based approach for indicating that it is in
fact A and C that are duplicates, rather than A and B or A and D.

> 
>> SSRC carrying RTP media streams that are duplicates of each other
>> do in fact be labeled explicitly. I also note that a SDP based
>> solution is not sufficient to resolve this issue. I think we need
>> something in RTP, so that SSRC collisions can be handled.
> 
> I can't say I am following.

I suggest that you use something like the SRCNAME SDES item extension I
propose in my draft.

> 
>> 4. In addition there is a few questions how this duplication
>> really should set some RTP fields. Is the duplication only working
>> on Sequence
> 
> That is the easiest way for a middle box or endpoint doing the stream
> merging (duplicate suppression). If seqnums are different, what would
> one do? look into the payload?

It was simply that you had not documented what needs to be the same.
Sequence number being the same is likely the easiest, but one do have to
raise some warning flags if one has processing that changes these
streams. For example a translator modifying the packetization must do
the exact same manipulation on the other stream.

But, I don't think anything else really works.

> 
>> number level? There seem to be additional questions, for example in
>> the need to have RTP timestamp considerations between the streams.
>> They need to point identically into the joint timeline. Otherwise
>> outages, or receiver taking synch from one and not the other stream
>> can end up with different synchronizations. Becomes especially
>> interesting to discuss when the time shifted transmission.
> 
> The duplicate streams are coming from the same source, so they have
> the same timestamp as well as seqnums. It has to be this way because
> the node merging the streams will not re-write anything at all. It is
> a simple suppression operation. Beauty of it is that things stay
> pretty simple to use and implement without any changes to RTP/RTCP
> plane at all.

Yes, but you hadn't made it clear, and you have both SSRC and implied
change of PT. So I didn't want to assume anything.

The timestamp field also have implcations due to its binding with RTCP
SR information.

> 
> As for the time-shifted transmission, I suppose you are referring to
> the temporal diversity/delayed duplication
> (draft-begen-mmusic-temporal-interleaving-02). Well, in that case,
> timestamps are still the same for the reasons above. Changing the
> timestamps for the delayed packets does not bring any benefit
> (actually it complicates things) since the ultimate RTP application
> will still receive one and only one RTP stream.

Ok, but you will need to make clear on how one should set the RTCP SR
timestamp NTP mapping to be clear. And what that implies for the RTCP
jitter calculation.

> 
>> 5. The draft also discussed Payload Type differences and have them
>> in the SDP examples. I don't see how that is appropriate. If the
>> streams
> 
> When you are doing SSRC muxing, you put multiple streams in the same
> m line and they end up having different PT numbers. What is wrong
> with that? If they are in separate m lines, they could have the same
> PT number.

I am still asking why, there is no reason to use different PTs, unless
you are doing the association between the different streams by them
having different PTs. Which I believe is a Bad Idea (TM). The encoding
in the packet are the same, so why don't they have the same PT.


> 
>> 6. I think it is worth discussing also the benefits of SSRC
>> multiplexed versus the session multiplexed. In multicast context
>> there is clear benefits from a control and congestion perspective
>> to have session multiplexed versions and these mapped onto
>> different multicast groups enabling a receiver to choose one or
>> both streams.
> 
> This is what we have in section 5.2. Different streams can be sent in
> different multicast sessions but it does not mean that they are
> separate RTP sessions. I discussed this in detail with Colin. Thanks
> to him for trying to teach me over and over what an RTP session
> actually means.

Yes, you have an example, but not a discussion of why an implementor
should choose Session multiplexing over SSRC. I think that should be
included, or I think you can wait for the architectural document I think
will result from my multi-stream and simulcast draft.

> 
>> 7. Thus I think the signalling aspects of this is the minor part. 
>> Although there clearly are some aspects of this needing discussion.
>> For example clearer offer/answer rules for handling these.
> 
> We have some text around O/A stuff. But frankly, I don’t see any use
> case where duplication will be used in a unicast application. I mean
> if redundancy is needed, one should use FEC instead for unicast
> transport. Duplication is a nice use case for multicast transport
> where the bw overhead is easier to justify and the implementation is
> easier. And one would naturally implement this spec in networks where
> he already knows that the network elements support this spec. So,
> maybe we should remove the whole O/A text anyway and write that this
> will only be used declaratively.

Then, I think we should slap an application statement to the beginning
of the draft to ensure that it may only be used in multicast. For
example b=TIAS was only intended to used in declarative SDPs, but it has
become common in O/A, but it doesn't have clear rules.

So I think it imporant that one don't assume to much. Yes, it might not
be the best idea for unicast, but that is not going to prevent it from
being used.

Cheers

Magnus Westerlund

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


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Subject: Re: [AVTCORE] Comments on draft-begen-mmusic-redundancy-grouping-01
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One small comment, basically agreeing with Magnus point about payload =
types.

Usually when people use payload identifiers to do demultiplexing they =
later regret it. Payload identifiers (in RTP or any other protocol for =
that matter) should only be used to decide which parser to hand the data =
to, not which state variable to bind the data in the packet to.=20

(further aside: this is one of the few things the OSI guys actually got =
right when they had both payload ids and SAPs at every layer).

DaveO.


On Jul 25, 2011, at 6:54 AM, Magnus Westerlund wrote:

> On 2011-07-24 23:21, Ali C. Begen (abegen) wrote:
>>=20
>>=20
>>> -----Original Message----- From: avt-bounces@ietf.org
>>> [mailto:avt-bounces@ietf.org] On Behalf Of Magnus Westerlund Sent:
>>> Friday, July 22, 2011 8:39 AM To: mmusic (E-mail); IETF AVTCore WG=20=

>>> Subject: [AVTCORE] Comments on
>>> draft-begen-mmusic-redundancy-grouping-01
>>>=20
>>> Hi,
>>>=20
>>> (Primarily as individual, but the WG home question is as WG chair)
>>>=20
>>> I have reviewed this draft and some comments on it.
>>=20
>> Thanks for the review.
>>=20
>>> 1. If this solution proposal is to be taken forward I think it
>>> needs to be discussed in AVTCORE. The reason is that it has several
>>> RTP details that needs to be figured out to make stream duplication
>>> issue free in a general usage. I will take up a few of these issues
>>> in the below comments.
>>=20
>> That is why we asked for a slot in avtcore on wed.
>>=20
>>> 2. One aspect this document has is a relation to what is discussed
>>> in draft-westerlund-avtcore-multistream-and-simulcast-00. Stream=20
>>> duplication both results in a certain form of multi-stream and also
>>> has a similar but slightly different semantics to simulcast in that
>>> the streams are two representations of a single original media
>>> source. In this case they are clones, but not true duplicates due
>>> to that RTP doesn't handle try duplicates well.
>>=20
>> The data carried in the redundant streams are identical but the rtp
>> header differs due to the different SSRCs. That is all. SSRC muxing
>> nicely fits the use case we are tackling with. Simulcasting is quite
>> different from our use case because in simulcasting different RTP
>> streams most likely carry different data.
>=20
> I agree that the purpose for multiple RTP streams from the same =
original
> media source are different from the primary case I set out to solve. =
But
> as Jonathan Lennox pointed out, redundancy and FEC have very similar
> behaviors to simulcast with different encodings of the same original
> source.
>=20
> Thus, I do see a need to discuss for example how you identify the
> streams that are a duplicate of each other.
>=20
>>=20
>>> 3. Another issue that also the authors has identified is the
>>> question of what one do when one has multiple media streams from
>>> the same end-point.
>>=20
>>> =46rom the same end-point (e.g., source) there could of course of
>>> multiple media streams. You can use SSRC muxing for example to
>>> identify your streams and use ssrc-group to group them if needed.
>>=20
>> The concern in that editorial note was what if there were multiple
>> streams in each m line but what if you wanted grouping/association
>> among only a subset.
>>=20
>> For example,
>>=20
>> m=3Dvideo 30000 RTP/AVP 101 102 .... m=3Dvideo 30000 RTP/AVP 103 104
>>=20
>> I can easily associate 101 and 102 if I want to, but what if 101 and
>> 103 were associated with each other. Can I use a=3Dgroup line? =
Probably
>> not. But, I don't know why you would need that kind of relation in
>> the first place (even if you need, there are probably less messy ways
>> to express them). If that had been a very desirable feature, SDP and
>> grouping semantics along with RTP would not have defined in their
>> current form I guess.
>=20
> Yes, that is a very valid question also for simulcast.
>=20
> And I think we can provide a story for doing that. But I don't think =
it
> should be based on using the RTP payload type to identify either of =
the
> versions.
>=20
>>=20
>>> And in a general case one can't use the CNAME to separate the
>>> sources. The reason is that two media sources that has a common
>>> timing (synchronization) context do needs to indicate that by
>>> having the same CNAME. Thus there are need for some additional
>>> mechanism to ensure that
>>=20
>> If 101 and 103 are associated, why can't I use the same CNAME for
>> those streams? If 102 and 104 are associated/related, I would use a
>> common CNAME for them, too (but theirs would be separate from 101 and
>> 103's).
>=20
> If SSRC A has a duplciation in SSRC C and SSRC B has a duplication in
> SSRC D and both A and B comes from the same end-point and have a =
common
> synchronization context, then all of A-D needs to have the same CNAME.
>=20
> And I get the impression that you are using a payload type based
> association scheme here. And I think that is a bad idea. For example =
it
> can't support particular many streams. It also looks semantically =
wrong,
> as if Stream A uses PT 96, then SSRC C should use PT 96 also for the
> corresponding packet duplicates. And if A switches from one PT to
> another the other stream needs to follow.
>=20
> Thus I think a SDES item based approach for indicating that it is in
> fact A and C that are duplicates, rather than A and B or A and D.
>=20
>>=20
>>> SSRC carrying RTP media streams that are duplicates of each other
>>> do in fact be labeled explicitly. I also note that a SDP based
>>> solution is not sufficient to resolve this issue. I think we need
>>> something in RTP, so that SSRC collisions can be handled.
>>=20
>> I can't say I am following.
>=20
> I suggest that you use something like the SRCNAME SDES item extension =
I
> propose in my draft.
>=20
>>=20
>>> 4. In addition there is a few questions how this duplication
>>> really should set some RTP fields. Is the duplication only working
>>> on Sequence
>>=20
>> That is the easiest way for a middle box or endpoint doing the stream
>> merging (duplicate suppression). If seqnums are different, what would
>> one do? look into the payload?
>=20
> It was simply that you had not documented what needs to be the same.
> Sequence number being the same is likely the easiest, but one do have =
to
> raise some warning flags if one has processing that changes these
> streams. For example a translator modifying the packetization must do
> the exact same manipulation on the other stream.
>=20
> But, I don't think anything else really works.
>=20
>>=20
>>> number level? There seem to be additional questions, for example in
>>> the need to have RTP timestamp considerations between the streams.
>>> They need to point identically into the joint timeline. Otherwise
>>> outages, or receiver taking synch from one and not the other stream
>>> can end up with different synchronizations. Becomes especially
>>> interesting to discuss when the time shifted transmission.
>>=20
>> The duplicate streams are coming from the same source, so they have
>> the same timestamp as well as seqnums. It has to be this way because
>> the node merging the streams will not re-write anything at all. It is
>> a simple suppression operation. Beauty of it is that things stay
>> pretty simple to use and implement without any changes to RTP/RTCP
>> plane at all.
>=20
> Yes, but you hadn't made it clear, and you have both SSRC and implied
> change of PT. So I didn't want to assume anything.
>=20
> The timestamp field also have implcations due to its binding with RTCP
> SR information.
>=20
>>=20
>> As for the time-shifted transmission, I suppose you are referring to
>> the temporal diversity/delayed duplication
>> (draft-begen-mmusic-temporal-interleaving-02). Well, in that case,
>> timestamps are still the same for the reasons above. Changing the
>> timestamps for the delayed packets does not bring any benefit
>> (actually it complicates things) since the ultimate RTP application
>> will still receive one and only one RTP stream.
>=20
> Ok, but you will need to make clear on how one should set the RTCP SR
> timestamp NTP mapping to be clear. And what that implies for the RTCP
> jitter calculation.
>=20
>>=20
>>> 5. The draft also discussed Payload Type differences and have them
>>> in the SDP examples. I don't see how that is appropriate. If the
>>> streams
>>=20
>> When you are doing SSRC muxing, you put multiple streams in the same
>> m line and they end up having different PT numbers. What is wrong
>> with that? If they are in separate m lines, they could have the same
>> PT number.
>=20
> I am still asking why, there is no reason to use different PTs, unless
> you are doing the association between the different streams by them
> having different PTs. Which I believe is a Bad Idea (TM). The encoding
> in the packet are the same, so why don't they have the same PT.
>=20
>=20
>>=20
>>> 6. I think it is worth discussing also the benefits of SSRC
>>> multiplexed versus the session multiplexed. In multicast context
>>> there is clear benefits from a control and congestion perspective
>>> to have session multiplexed versions and these mapped onto
>>> different multicast groups enabling a receiver to choose one or
>>> both streams.
>>=20
>> This is what we have in section 5.2. Different streams can be sent in
>> different multicast sessions but it does not mean that they are
>> separate RTP sessions. I discussed this in detail with Colin. Thanks
>> to him for trying to teach me over and over what an RTP session
>> actually means.
>=20
> Yes, you have an example, but not a discussion of why an implementor
> should choose Session multiplexing over SSRC. I think that should be
> included, or I think you can wait for the architectural document I =
think
> will result from my multi-stream and simulcast draft.
>=20
>>=20
>>> 7. Thus I think the signalling aspects of this is the minor part.=20
>>> Although there clearly are some aspects of this needing discussion.
>>> For example clearer offer/answer rules for handling these.
>>=20
>> We have some text around O/A stuff. But frankly, I don=92t see any =
use
>> case where duplication will be used in a unicast application. I mean
>> if redundancy is needed, one should use FEC instead for unicast
>> transport. Duplication is a nice use case for multicast transport
>> where the bw overhead is easier to justify and the implementation is
>> easier. And one would naturally implement this spec in networks where
>> he already knows that the network elements support this spec. So,
>> maybe we should remove the whole O/A text anyway and write that this
>> will only be used declaratively.
>=20
> Then, I think we should slap an application statement to the beginning
> of the draft to ensure that it may only be used in multicast. For
> example b=3DTIAS was only intended to used in declarative SDPs, but it =
has
> become common in O/A, but it doesn't have clear rules.
>=20
> So I think it imporant that one don't assume to much. Yes, it might =
not
> be the best idea for unicast, but that is not going to prevent it from
> being used.
>=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


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> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Monday, July 25, 2011 6:54 AM
> To: Ali C. Begen (abegen)
> Cc: mmusic (E-mail); IETF AVTCore WG
> Subject: Re: [AVTCORE] Comments on =
draft-begen-mmusic-redundancy-grouping-01
>=20
> On 2011-07-24 23:21, Ali C. Begen (abegen) wrote:
> >
> >
> >> -----Original Message----- From: avt-bounces@ietf.org
> >> [mailto:avt-bounces@ietf.org] On Behalf Of Magnus Westerlund Sent:
> >> Friday, July 22, 2011 8:39 AM To: mmusic (E-mail); IETF AVTCore WG
> >> Subject: [AVTCORE] Comments on
> >> draft-begen-mmusic-redundancy-grouping-01
> >>
> >> Hi,
> >>
> >> (Primarily as individual, but the WG home question is as WG chair)
> >>
> >> I have reviewed this draft and some comments on it.
> >
> > Thanks for the review.
> >
> >> 1. If this solution proposal is to be taken forward I think it
> >> needs to be discussed in AVTCORE. The reason is that it has several
> >> RTP details that needs to be figured out to make stream duplication
> >> issue free in a general usage. I will take up a few of these issues
> >> in the below comments.
> >
> > That is why we asked for a slot in avtcore on wed.
> >
> >> 2. One aspect this document has is a relation to what is discussed
> >> in draft-westerlund-avtcore-multistream-and-simulcast-00. Stream
> >> duplication both results in a certain form of multi-stream and also
> >> has a similar but slightly different semantics to simulcast in that
> >> the streams are two representations of a single original media
> >> source. In this case they are clones, but not true duplicates due
> >> to that RTP doesn't handle try duplicates well.
> >
> > The data carried in the redundant streams are identical but the rtp
> > header differs due to the different SSRCs. That is all. SSRC muxing
> > nicely fits the use case we are tackling with. Simulcasting is quite
> > different from our use case because in simulcasting different RTP
> > streams most likely carry different data.
>=20
> I agree that the purpose for multiple RTP streams from the same =
original
> media source are different from the primary case I set out to solve. =
But
> as Jonathan Lennox pointed out, redundancy and FEC have very similar
> behaviors to simulcast with different encodings of the same original
> source.
>=20
> Thus, I do see a need to discuss for example how you identify the
> streams that are a duplicate of each other.

Ok, lets discuss it. My current proposal is to use different SSRCs. PTs =
turn out to be different since I designate these streams in the same m =
line. Unless I am mistakes, you cannot put two different streams with =
the same PT in a single m line. Remember, we do not use PT to demux the =
RTP streams, we use SSRC for that purpose.
=20
> >
> >> 3. Another issue that also the authors has identified is the
> >> question of what one do when one has multiple media streams from
> >> the same end-point.
> >
> >> From the same end-point (e.g., source) there could of course of
> >> multiple media streams. You can use SSRC muxing for example to
> >> identify your streams and use ssrc-group to group them if needed.
> >
> > The concern in that editorial note was what if there were multiple
> > streams in each m line but what if you wanted grouping/association
> > among only a subset.
> >
> > For example,
> >
> > m=3Dvideo 30000 RTP/AVP 101 102 .... m=3Dvideo 30000 RTP/AVP 103 104
> >
> > I can easily associate 101 and 102 if I want to, but what if 101 and
> > 103 were associated with each other. Can I use a=3Dgroup line? =
Probably
> > not. But, I don't know why you would need that kind of relation in
> > the first place (even if you need, there are probably less messy =
ways
> > to express them). If that had been a very desirable feature, SDP and
> > grouping semantics along with RTP would not have defined in their
> > current form I guess.
>=20
> Yes, that is a very valid question also for simulcast.
>=20
> And I think we can provide a story for doing that. But I don't think =
it
> should be based on using the RTP payload type to identify either of =
the
> versions.
>=20
> >
> >> And in a general case one can't use the CNAME to separate the
> >> sources. The reason is that two media sources that has a common
> >> timing (synchronization) context do needs to indicate that by
> >> having the same CNAME. Thus there are need for some additional
> >> mechanism to ensure that
> >
> > If 101 and 103 are associated, why can't I use the same CNAME for
> > those streams? If 102 and 104 are associated/related, I would use a
> > common CNAME for them, too (but theirs would be separate from 101 =
and
> > 103's).
>=20
> If SSRC A has a duplciation in SSRC C and SSRC B has a duplication in
> SSRC D and both A and B comes from the same end-point and have a =
common
> synchronization context, then all of A-D needs to have the same CNAME.

Well, above implies that all streams are coming from the same source and =
yes, they will end up having the same CNAME since you required A will be =
synched with B. But, still I think this is workable without any =
additional mechanism. A and C would be in the same m line and B and D =
would be in the same (but separate from the first) m line. That would =
give me what I want.
=20
> And I get the impression that you are using a payload type based
> association scheme here. And I think that is a bad idea. For example =
it
> can't support particular many streams. It also looks semantically =
wrong,
> as if Stream A uses PT 96, then SSRC C should use PT 96 also for the
> corresponding packet duplicates. And if A switches from one PT to
> another the other stream needs to follow.

No, I am not suggesting to use PT for demuxing. I use SSRC for that =
purpose. As I asked above, if I can use the same PT for multiple streams =
in the same m line, then I will. But I don=92t think I can.

> Thus I think a SDES item based approach for indicating that it is in
> fact A and C that are duplicates, rather than A and B or A and D.
>=20
> >
> >> SSRC carrying RTP media streams that are duplicates of each other
> >> do in fact be labeled explicitly. I also note that a SDP based
> >> solution is not sufficient to resolve this issue. I think we need
> >> something in RTP, so that SSRC collisions can be handled.
> >
> > I can't say I am following.
>=20
> I suggest that you use something like the SRCNAME SDES item extension =
I
> propose in my draft.
>=20
> >
> >> 4. In addition there is a few questions how this duplication
> >> really should set some RTP fields. Is the duplication only working
> >> on Sequence
> >
> > That is the easiest way for a middle box or endpoint doing the =
stream
> > merging (duplicate suppression). If seqnums are different, what =
would
> > one do? look into the payload?
>=20
> It was simply that you had not documented what needs to be the same.

I think I did, but I will check.

> Sequence number being the same is likely the easiest, but one do have =
to
> raise some warning flags if one has processing that changes these
> streams. For example a translator modifying the packetization must do
> the exact same manipulation on the other stream.

Yes, obviously. We can put this warning.
=20
> But, I don't think anything else really works.

Right, seqnum is the easiest and possibly the only way in practice.
=20
> >
> >> number level? There seem to be additional questions, for example in
> >> the need to have RTP timestamp considerations between the streams.
> >> They need to point identically into the joint timeline. Otherwise
> >> outages, or receiver taking synch from one and not the other stream
> >> can end up with different synchronizations. Becomes especially
> >> interesting to discuss when the time shifted transmission.
> >
> > The duplicate streams are coming from the same source, so they have
> > the same timestamp as well as seqnums. It has to be this way because
> > the node merging the streams will not re-write anything at all. It =
is
> > a simple suppression operation. Beauty of it is that things stay
> > pretty simple to use and implement without any changes to RTP/RTCP
> > plane at all.
>=20
> Yes, but you hadn't made it clear, and you have both SSRC and implied
> change of PT. So I didn't want to assume anything.

OK, will make this clear, as well.
=20
> The timestamp field also have implcations due to its binding with RTCP
> SR information.
>=20
> >
> > As for the time-shifted transmission, I suppose you are referring to
> > the temporal diversity/delayed duplication
> > (draft-begen-mmusic-temporal-interleaving-02). Well, in that case,
> > timestamps are still the same for the reasons above. Changing the
> > timestamps for the delayed packets does not bring any benefit
> > (actually it complicates things) since the ultimate RTP application
> > will still receive one and only one RTP stream.
>=20
> Ok, but you will need to make clear on how one should set the RTCP SR
> timestamp NTP mapping to be clear. And what that implies for the RTCP
> jitter calculation.

The fact that you are using delayed duplication or spatial diversity =
means that you will likely introduce some jitter into your stream unless =
the node merging the streams smooths out that jitter in the first place. =
There is not nothing special for the ultimate endpoint for preparing the =
reports, it just does  what it is supposed to do. In most systems, the =
ultimate RTP receivers might not be even aware of the fact that they are =
receiving a stream merged from two identical streams. That is a design =
feature.

> >
> >> 5. The draft also discussed Payload Type differences and have them
> >> in the SDP examples. I don't see how that is appropriate. If the
> >> streams
> >
> > When you are doing SSRC muxing, you put multiple streams in the same
> > m line and they end up having different PT numbers. What is wrong
> > with that? If they are in separate m lines, they could have the same
> > PT number.
>=20
> I am still asking why, there is no reason to use different PTs, unless
> you are doing the association between the different streams by them
> having different PTs. Which I believe is a Bad Idea (TM). The encoding
> in the packet are the same, so why don't they have the same PT.

Well again, I am not using it for demuxing purposes. I just believe that =
is how the m line works. Tell me otherwise and I will be happy to make =
PTs the same.
=20
>=20
> >
> >> 6. I think it is worth discussing also the benefits of SSRC
> >> multiplexed versus the session multiplexed. In multicast context
> >> there is clear benefits from a control and congestion perspective
> >> to have session multiplexed versions and these mapped onto
> >> different multicast groups enabling a receiver to choose one or
> >> both streams.
> >
> > This is what we have in section 5.2. Different streams can be sent =
in
> > different multicast sessions but it does not mean that they are
> > separate RTP sessions. I discussed this in detail with Colin. Thanks
> > to him for trying to teach me over and over what an RTP session
> > actually means.
>=20
> Yes, you have an example, but not a discussion of why an implementor
> should choose Session multiplexing over SSRC. I think that should be
> included, or I think you can wait for the architectural document I =
think
> will result from my multi-stream and simulcast draft.

Lets discuss in the session.
=20
> >
> >> 7. Thus I think the signalling aspects of this is the minor part.
> >> Although there clearly are some aspects of this needing discussion.
> >> For example clearer offer/answer rules for handling these.
> >
> > We have some text around O/A stuff. But frankly, I don=92t see any =
use
> > case where duplication will be used in a unicast application. I mean
> > if redundancy is needed, one should use FEC instead for unicast
> > transport. Duplication is a nice use case for multicast transport
> > where the bw overhead is easier to justify and the implementation is
> > easier. And one would naturally implement this spec in networks =
where
> > he already knows that the network elements support this spec. So,
> > maybe we should remove the whole O/A text anyway and write that this
> > will only be used declaratively.
>=20
> Then, I think we should slap an application statement to the beginning
> of the draft to ensure that it may only be used in multicast. For

Wfm.

> example b=3DTIAS was only intended to used in declarative SDPs, but it =
has
> become common in O/A, but it doesn't have clear rules.
>=20
> So I think it imporant that one don't assume to much. Yes, it might =
not
> be the best idea for unicast, but that is not going to prevent it from
> being used.

If one really uses this for unicast, let them deal with all the =
potential O/A problems that might surface. It is more like they will =
deserve it :)

Cheers,
-acbegen
=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
> ----------------------------------------------------------------------


From magnus.westerlund@ericsson.com  Mon Jul 25 09:12:36 2011
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On 2011-07-25 10:07, Ali C. Begen (abegen) wrote:

> Ok, lets discuss it. My current proposal is to use different SSRCs.
PTs turn out to be different since I designate these streams in the same
m line. Unless I am mistakes, you cannot put two different streams with
the same PT in a single m line. Remember, we do not use PT to demux the
RTP streams, we use SSRC for that purpose.

No, that is wrong. The PT is completely unrelated to to which SSRC that
use them. The set of configured PTs can be used by any of the sources
that transmit in the direction for which they are configured. I do
include the direction because of the O/A aspects of having different
configurations for receving in different directions.


>> If SSRC A has a duplciation in SSRC C and SSRC B has a duplication
>> in SSRC D and both A and B comes from the same end-point and have a
>> common synchronization context, then all of A-D needs to have the
>> same CNAME.
> 
> Well, above implies that all streams are coming from the same source
> and yes, they will end up having the same CNAME since you required A
> will be synched with B. But, still I think this is workable without
> any additional mechanism. A and C would be in the same m line and B
> and D would be in the same (but separate from the first) m line. That
> would give me what I want.

I am not agreeing that you would put A and B in different sessions. This
doesn't make sense if you actually like to provide two related media
streams of the same type, for example two cameras covering different
parts of a telepresence room, or two different camera angels in a movie.

What you are suggesting seems to be a hack to just get around the issue
that if you have streams that are some form of alternatives of the
actual same original source then you need to identify, like the SRCNAME
proposal.


> 
>> And I get the impression that you are using a payload type based 
>> association scheme here. And I think that is a bad idea. For
>> example it can't support particular many streams. It also looks
>> semantically wrong, as if Stream A uses PT 96, then SSRC C should
>> use PT 96 also for the corresponding packet duplicates. And if A
>> switches from one PT to another the other stream needs to follow.
> 
> No, I am not suggesting to use PT for demuxing. I use SSRC for that
> purpose. As I asked above, if I can use the same PT for multiple
> streams in the same m line, then I will. But I don’t think I can.
> 

Yes, understand this is a different type of missunderstanding of RTP.


>> 
>> Ok, but you will need to make clear on how one should set the RTCP
>> SR timestamp NTP mapping to be clear. And what that implies for the
>> RTCP jitter calculation.
> 
> The fact that you are using delayed duplication or spatial diversity
> means that you will likely introduce some jitter into your stream
> unless the node merging the streams smooths out that jitter in the
> first place. There is not nothing special for the ultimate endpoint
> for preparing the reports, it just does  what it is supposed to do.
> In most systems, the ultimate RTP receivers might not be even aware
> of the fact that they are receiving a stream merged from two
> identical streams. That is a design feature.
> 

When you talk about jitter here, you are talking about small transport
level jitter, rather than jitter on the level of frame intervals.

Sure, if you have a translator that merge these, then you clearly should
be hidden to the end-point behind translator.

>>>> 7. Thus I think the signalling aspects of this is the minor
>>>> part. Although there clearly are some aspects of this needing
>>>> discussion. For example clearer offer/answer rules for handling
>>>> these.
>>> 
>>> We have some text around O/A stuff. But frankly, I don’t see any
>>> use case where duplication will be used in a unicast application.
>>> I mean if redundancy is needed, one should use FEC instead for
>>> unicast transport. Duplication is a nice use case for multicast
>>> transport where the bw overhead is easier to justify and the
>>> implementation is easier. And one would naturally implement this
>>> spec in networks where he already knows that the network elements
>>> support this spec. So, maybe we should remove the whole O/A text
>>> anyway and write that this will only be used declaratively.
>> 
>> Then, I think we should slap an application statement to the
>> beginning of the draft to ensure that it may only be used in
>> multicast. For
> 
> Wfm.
> 
>> example b=TIAS was only intended to used in declarative SDPs, but
>> it has become common in O/A, but it doesn't have clear rules.
>> 
>> So I think it imporant that one don't assume to much. Yes, it might
>> not be the best idea for unicast, but that is not going to prevent
>> it from being used.
> 
> If one really uses this for unicast, let them deal with all the
> potential O/A problems that might surface. It is more like they will
> deserve it :)
> 

:-)

Cheers

Magnus Westerlund

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


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[Apologies for cross posting AVT lists - reason below]

=20

All,

=20

I will be discussing issues contained in draft-ramalho-payload g7110-00
at the AVT extensions timeslot on Wednesday afternoon.

=20

This draft is a payload WG draft, but there isn't a payload meeting at
IETF 81. And this draft was previously scheduled to be discussed  the
AVT core timeslot prior to its being moved to AVTEXT (thus the reason
for the cross post to all AVT groups). Roni and Ali encouraged me to say
a few words prior to the discussion this week.

=20

G.711.0 is a stateless and lossless compression of G.711 VoIP RTP
payloads. Thus it is a compression algorithm tailored for G.711. When
negotiated end-to-end it is negotiated "as if it were a codec".

=20

However, being lossless and stateless, G.711.0 can be applied anywhere
within an end-to-end G.711 negotiated session. When applied in this
manner, it suffers no voice quality degradation relative to G.711 (duh,
its lossless).

=20

Most of the open issues in the draft relate to the use of G.711.0 within
the context of an and-to-end G.711 negotiated session.

=20

I look forward to comments generated from the discussion of this draft
for those cases.

=20

Regards,

=20

Michael Ramalho

=20

=20

Michael Ramalho
Technical Leader
Product Development
mramalho@cisco.com <mailto:mramalho@cisco.com>=20
Phone: +1 919 476 2038
Mobile: +1 941 544 2844



Cisco Systems, Inc.
4564 Tuscana Drive
Sarasota, FL 34241-4201
United States
http://ramalho.webhop.info
Skype: mramalho_mar42

=20

=20

=09
=09

=20

=20


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<p class=3DMsoNormal>[Apologies for cross posting AVT lists &#8211; =
reason below]<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>All,<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>I will be discussing issues contained in =
draft-ramalho-payload
g7110-00 at the AVT extensions timeslot on Wednesday =
afternoon.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>This draft is a payload WG draft, but there =
isn&#8217;t a
payload meeting at IETF 81. And this draft was previously scheduled to =
be
discussed &nbsp;the AVT core timeslot prior to its being moved to AVTEXT =
(thus the
reason for the cross post to all AVT groups). Roni and Ali encouraged me =
to say
a few words prior to the discussion this week.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>G.711.0 is a stateless and lossless compression of =
G.711 VoIP
RTP payloads. Thus it is a compression algorithm tailored for G.711. =
When
negotiated end-to-end it is negotiated &#8220;as if it were a =
codec&#8221;.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>However, being lossless and stateless, G.711.0 can =
be
applied anywhere within an end-to-end G.711 negotiated session. When =
applied in
this manner, it suffers no voice quality degradation relative to G.711 =
(duh,
its lossless).<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Most of the open issues in the draft relate to the =
use of
G.711.0 within the context of an and-to-end G.711 negotiated =
session.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>I look forward to comments generated from the =
discussion of
this draft for those cases.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Regards,<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Michael Ramalho<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<table class=3DMsoNormalTable border=3D0 cellspacing=3D0 cellpadding=3D0 =
width=3D543
 style=3D'width:407.25pt'>
 <tr>
  <td style=3D'padding:0in 0in 0in 0in'>
  <table class=3DMsoNormalTable border=3D0 cellspacing=3D0 =
cellpadding=3D0 width=3D543
   style=3D'width:407.25pt'>
   <tr>
    <td colspan=3D3 style=3D'padding:0in 0in 0in 0in'>
    <p class=3DMsoNormal><span =
style=3D'font-size:12.0pt;font-family:"Times New Roman","serif"'><img
    width=3D110 height=3D73 id=3D"Picture_x0020_1"
    src=3D"cid:image001.gif@01CC4AD1.08107BA0"
    alt=3D"http://www.cisco.com/swa/i/logo.gif"><o:p></o:p></span></p>
    </td>
   </tr>
   <tr>
    <td nowrap valign=3Dtop style=3D'padding:0in 0in 11.25pt .25in'>
    <p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;margin-bottom:12.0pt'><b><span
    =
style=3D'font-size:8.5pt;font-family:"Arial","sans-serif";color:#666666'>=
Michael
    Ramalho</span></b><span =
style=3D'font-size:8.5pt;font-family:"Arial","sans-serif";
    color:#666666'><br>
    </span><b><span =
style=3D'font-size:8.5pt;font-family:"Arial","sans-serif";
    color:#666666'>Technical Leader</span></b><span =
style=3D'font-size:8.5pt;
    font-family:"Arial","sans-serif";color:#666666'><br>
    </span><b><span =
style=3D'font-size:8.5pt;font-family:"Arial","sans-serif";
    color:#666666'>Product Development</span></b><span =
style=3D'font-size:8.5pt;
    font-family:"Arial","sans-serif";color:#666666'><br>
    <a href=3D"mailto:mramalho@cisco.com"><span =
style=3D'color:#666666'>mramalho@cisco.com</span></a><br>
    Phone: </span><b><span =
style=3D'font-size:8.5pt;font-family:"Arial","sans-serif";
    color:#666666'>+1 919 476 2038</span></b><span =
style=3D'font-size:8.5pt;
    font-family:"Arial","sans-serif";color:#666666'><br>
    Mobile: </span><b><span =
style=3D'font-size:8.5pt;font-family:"Arial","sans-serif";
    color:#666666'>+1 941 544 2844</span></b><span =
style=3D'font-size:8.5pt;
    font-family:"Arial","sans-serif";color:#666666'><br>
    <br>
    <o:p></o:p></span></p>
    </td>
    <td nowrap valign=3Dtop style=3D'padding:0in 0in 7.5pt 15.0pt'>
    <p class=3DMsoNormal =
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    =
style=3D'font-size:8.5pt;font-family:"Arial","sans-serif";color:#666666'>=
Cisco
    Systems, Inc.</span></b><span =
style=3D'font-size:8.5pt;font-family:"Arial","sans-serif";
    color:#666666'><br>
    4564 Tuscana Drive<br>
    Sarasota, FL 34241-4201<br>
    United States<br>
    <b>http://ramalho.webhop.info</b><br>
    <b>Skype:</b> mramalho_mar42<o:p></o:p></span></p>
    </td>
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    <p class=3DMsoNormal><span =
style=3D'font-size:12.0pt;font-family:"Times New =
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From ray.vanbrandenburg@tno.nl  Tue Jul 26 15:31:54 2011
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From: "Brandenburg, R. (Ray) van" <ray.vanbrandenburg@tno.nl>
To: "'magnus.westerlund@ericsson.com'" <magnus.westerlund@ericsson.com>
Thread-Topic: Comments on draft-westerlund-avtcore-multistream-and-simulcast-00
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Date: Tue, 26 Jul 2011 22:31:51 +0000
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Subject: [AVTCORE] Comments on draft-westerlund-avtcore-multistream-and-simulcast-00
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Hi Magnus,

Reading your draft 'RTP Multiple Stream Sessions and Simulcast', I was wond=
ering whether the proposed simulcast extensions could allow for RTP to be u=
sed for the delivery of spatially segmented content.

Spatial segmentation is the concept of spatially dividing a video up into m=
ultiple segments, each being independently decodable. It can for example be=
 used to efficiently deliver video to clients who are only interested in sp=
ecific areas of a video. Such bandwidth-saving are especially important whe=
n dealing with extremely high resolutions.

Depending on the application, the recombination of segments into a single v=
ideo might take place on the client or on a network node (such as an RTP mi=
xer). Your simulcast extensions might be a good way to more effectively dea=
l with the large number of segment streams being received by such a node.

I think it would be interesting to see whether we can expand the scope of y=
our draft to include the specific properties of spatially segmented content.

Cheers,

Ray van Brandenburg
This e-mail and its contents are subject to the DISCLAIMER at http://www.tn=
o.nl/emaildisclaimer

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<div class=3D"WordSection1">
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span lang=3D"EN-US" s=
tyle=3D"font-size:12.0pt;font-family:&quot;Helvetica&quot;,&quot;sans-serif=
&quot;">Hi Magnus,<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span lang=3D"EN-US" s=
tyle=3D"font-size:12.0pt;font-family:&quot;Helvetica&quot;,&quot;sans-serif=
&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span lang=3D"EN-US" s=
tyle=3D"font-size:12.0pt;font-family:&quot;Helvetica&quot;,&quot;sans-serif=
&quot;">Reading your draft 'RTP Multiple Stream Sessions and Simulcast', I =
was wondering whether the proposed simulcast extensions could
 allow for RTP to be used for the delivery of spatially segmented content. =
<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span lang=3D"EN-US" s=
tyle=3D"font-size:12.0pt;font-family:&quot;Helvetica&quot;,&quot;sans-serif=
&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span lang=3D"EN-US" s=
tyle=3D"font-size:12.0pt;font-family:&quot;Helvetica&quot;,&quot;sans-serif=
&quot;">Spatial segmentation is the concept of spatially dividing a video u=
p into multiple segments, each being independently decodable.
 It can for example be used to efficiently deliver video to clients who are=
 only interested in specific areas of a video. Such bandwidth-saving are es=
pecially important when dealing with extremely high resolutions.
<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span lang=3D"EN-US" s=
tyle=3D"font-size:12.0pt;font-family:&quot;Helvetica&quot;,&quot;sans-serif=
&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span lang=3D"EN-US" s=
tyle=3D"font-size:12.0pt;font-family:&quot;Helvetica&quot;,&quot;sans-serif=
&quot;">Depending on the application, the recombination of segments into a =
single video might take place on the client or on a network
 node (such as an RTP mixer). Your simulcast extensions might be a good way=
 to more effectively deal with the large number of segment streams being re=
ceived by such a node.<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span lang=3D"EN-US" s=
tyle=3D"font-size:12.0pt;font-family:&quot;Helvetica&quot;,&quot;sans-serif=
&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span lang=3D"EN-US" s=
tyle=3D"font-size:12.0pt;font-family:&quot;Helvetica&quot;,&quot;sans-serif=
&quot;">I think it would be interesting to see whether we can expand the sc=
ope of your draft to include the specific properties of spatially
 segmented content.<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span lang=3D"EN-US" s=
tyle=3D"font-size:12.0pt;font-family:&quot;Helvetica&quot;,&quot;sans-serif=
&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:12.0pt;font-family:&quot;Helvetica&quot;,&quot;sans-serif&quot;">Cheers,=
<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:12.0pt;font-family:&quot;Helvetica&quot;,&quot;sans-serif&quot;"><o:p>&n=
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Subject: Re: [AVTCORE] Comments on draft-westerlund-avtcore-multistream-and-simulcast-00
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Such spatial relationships are CLUE's scope.

On Tue, Jul 26, 2011 at 6:31 PM, Brandenburg, R. (Ray) van <
ray.vanbrandenburg@tno.nl> wrote:

>  Hi Magnus,****
>
> ** **
>
> Reading your draft 'RTP Multiple Stream Sessions and Simulcast', I was
> wondering whether the proposed simulcast extensions could allow for RTP to
> be used for the delivery of spatially segmented content. ****
>
> ** **
>
> Spatial segmentation is the concept of spatially dividing a video up into
> multiple segments, each being independently decodable. It can for example be
> used to efficiently deliver video to clients who are only interested in
> specific areas of a video. Such bandwidth-saving are especially important
> when dealing with extremely high resolutions. ****
>
> ** **
>
> Depending on the application, the recombination of segments into a single
> video might take place on the client or on a network node (such as an RTP
> mixer). Your simulcast extensions might be a good way to more effectively
> deal with the large number of segment streams being received by such a node.
> ****
>
> ** **
>
> I think it would be interesting to see whether we can expand the scope of
> your draft to include the specific properties of spatially segmented
> content.****
>
> ** **
>
> Cheers,****
>
> ** **
>
> Ray van Brandenburg****
>  This e-mail and its contents are subject to the DISCLAIMER at
> http://www.tno.nl/emaildisclaimer
>
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
>
>

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Such spatial relationships are CLUE&#39;s scope.<br><br><div class=3D"gmail=
_quote">On Tue, Jul 26, 2011 at 6:31 PM, Brandenburg, R. (Ray) van <span di=
r=3D"ltr">&lt;<a href=3D"mailto:ray.vanbrandenburg@tno.nl">ray.vanbrandenbu=
rg@tno.nl</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;">





<div link=3D"blue" vlink=3D"purple" lang=3D"NL">
<div>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:12.0pt" lang=3D"EN-US">Hi Magnus,<u></u><u></u></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:12.0pt" lang=3D"EN-US"><u></u>=A0<u></u></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:12.0pt" lang=3D"EN-US">Reading your draft &#39;RTP Multiple Stream Sessi=
ons and Simulcast&#39;, I was wondering whether the proposed simulcast exte=
nsions could
 allow for RTP to be used for the delivery of spatially segmented content. =
<u></u><u></u></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:12.0pt" lang=3D"EN-US"><u></u>=A0<u></u></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:12.0pt" lang=3D"EN-US">Spatial segmentation is the concept of spatially =
dividing a video up into multiple segments, each being independently decoda=
ble.
 It can for example be used to efficiently deliver video to clients who are=
 only interested in specific areas of a video. Such bandwidth-saving are es=
pecially important when dealing with extremely high resolutions.
<u></u><u></u></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:12.0pt" lang=3D"EN-US"><u></u>=A0<u></u></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:12.0pt" lang=3D"EN-US">Depending on the application, the recombination o=
f segments into a single video might take place on the client or on a netwo=
rk
 node (such as an RTP mixer). Your simulcast extensions might be a good way=
 to more effectively deal with the large number of segment streams being re=
ceived by such a node.<u></u><u></u></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:12.0pt" lang=3D"EN-US"><u></u>=A0<u></u></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:12.0pt" lang=3D"EN-US">I think it would be interesting to see whether we=
 can expand the scope of your draft to include the specific properties of s=
patially
 segmented content.<u></u><u></u></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:12.0pt" lang=3D"EN-US"><u></u>=A0<u></u></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:12.0pt">Cheers,<u></u><u></u></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:12.0pt"><u></u>=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:12.0pt">Ray van Brandenburg=
</span><u></u><u></u></p>
</div>
<font face=3D"monospace">This e-mail and its contents are subject to the DI=
SCLAIMER at <a href=3D"http://www.tno.nl/emaildisclaimer" target=3D"_blank"=
>http://www.tno.nl/emaildisclaimer</a></font></div>

<br>_______________________________________________<br>
Audio/Video Transport Core Maintenance<br>
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ps://www.ietf.org/mailman/listinfo/avt</a><br>
<br></blockquote></div><br>

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From adam@nostrum.com  Wed Jul 27 06:37:59 2011
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Date: Wed, 27 Jul 2011 09:37:56 -0400
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Subject: [AVTCORE] Media Types Registry
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I'm not entirely sure what consensus the chairs took away from the room 
regarding the "RTP Payload Format media types" IANA registry, but I do 
want to make sure we consider the following before we call the topic 
resolved:

Has anyone asked Steve Casner?

I would appear that he put in the work to get an RFC published on the 
topic as recently as four years ago. This isn't an easy task, so I would 
assume he had some pretty good reasons for putting in the effort. I 
imagine that those reasons would be very relevant input to this 
conversation.

/a

From ray.vanbrandenburg@tno.nl  Wed Jul 27 07:38:33 2011
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From: "Brandenburg, R. (Ray) van" <ray.vanbrandenburg@tno.nl>
To: Stephen Botzko <stephen.botzko@gmail.com>
Thread-Topic: [AVTCORE] Comments on draft-westerlund-avtcore-multistream-and-simulcast-00
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Let me further clarify what I mean by spatial segmentation.

In the most general case, you would have a single camera source that record=
s in a very high resolution (e.g. 4K or higher). Instead of encoding the re=
corded images as a single video stream, the camera spatially segments each =
video frame and encodes the results into multiple video streams (e.g. 4 or =
16) each representing a specific area of the recorded video. The result wou=
ld be an RTP sender that makes available 16 separate streams.

Depending on the area a user is interested in (the region-of-interest), an =
RTP receiver (or mixer) would set up one or more sessions to receive the on=
e or more streams that are necessary to reconstruct the desired region-of-i=
nterest. The result would be that different RTP receivers would simultaneou=
sly receive different subsets of the available segment streams.

While some of the desired functionality might lie in the CLUE area (which I=
 must admit I am not that familiar with), I think there is a considerable o=
verlap with the use cases described in the Multistream/Simulcast draft.

Cheers,

Ray van Brandenburg


From: Stephen Botzko [mailto:stephen.botzko@gmail.com]
Sent: woensdag 27 juli 2011 4:19
To: Brandenburg, R. (Ray) van
Cc: magnus.westerlund@ericsson.com; avt@ietf.org
Subject: Re: [AVTCORE] Comments on draft-westerlund-avtcore-multistream-and=
-simulcast-00

Such spatial relationships are CLUE's scope.
On Tue, Jul 26, 2011 at 6:31 PM, Brandenburg, R. (Ray) van <ray.vanbrandenb=
urg@tno.nl<mailto:ray.vanbrandenburg@tno.nl>> wrote:
Hi Magnus,

Reading your draft 'RTP Multiple Stream Sessions and Simulcast', I was wond=
ering whether the proposed simulcast extensions could allow for RTP to be u=
sed for the delivery of spatially segmented content.

Spatial segmentation is the concept of spatially dividing a video up into m=
ultiple segments, each being independently decodable. It can for example be=
 used to efficiently deliver video to clients who are only interested in sp=
ecific areas of a video. Such bandwidth-saving are especially important whe=
n dealing with extremely high resolutions.

Depending on the application, the recombination of segments into a single v=
ideo might take place on the client or on a network node (such as an RTP mi=
xer). Your simulcast extensions might be a good way to more effectively dea=
l with the large number of segment streams being received by such a node.

I think it would be interesting to see whether we can expand the scope of y=
our draft to include the specific properties of spatially segmented content=
.

Cheers,

Ray van Brandenburg
This e-mail and its contents are subject to the DISCLAIMER at http://www.tn=
o.nl/emaildisclaimer

_______________________________________________
Audio/Video Transport Core Maintenance
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family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Let me fur=
ther clarify what I mean by spatial segmentation.
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">In the mos=
t general case, you would have a single camera source that records in a ver=
y high resolution (e.g. 4K or higher). Instead of encoding
 the recorded images as a single video stream, the camera spatially segment=
s each video frame and encodes the results into multiple video streams (e.g=
. 4 or 16) each representing a specific area of the recorded video. The res=
ult would be an RTP sender that
 makes available 16 separate streams.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Depending =
on the area a user is interested in (the region-of-interest), an RTP receiv=
er (or mixer) would set up one or more sessions to receive
 the one or more streams that are necessary to reconstruct the desired regi=
on-of-interest. The result would be that different RTP receivers would simu=
ltaneously receive different subsets of the available segment streams.<o:p>=
</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">While some=
 of the desired functionality might lie in the CLUE area (which I must admi=
t I am not that familiar with), I think there is a considerable
 overlap with the use cases described in the Multistream/Simulcast draft. <=
o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Cheers,<o:=
p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Ray van Br=
andenburg<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><b><span lang=3D"EN-US" style=3D"font-size:10.0pt;fo=
nt-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span =
lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&qu=
ot;sans-serif&quot;"> Stephen Botzko [mailto:stephen.botzko@gmail.com]
<br>
<b>Sent:</b> woensdag 27 juli 2011 4:19<br>
<b>To:</b> Brandenburg, R. (Ray) van<br>
<b>Cc:</b> magnus.westerlund@ericsson.com; avt@ietf.org<br>
<b>Subject:</b> Re: [AVTCORE] Comments on draft-westerlund-avtcore-multistr=
eam-and-simulcast-00<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">Such spatial relation=
ships are CLUE's scope.<o:p></o:p></p>
<div>
<p class=3D"MsoNormal">On Tue, Jul 26, 2011 at 6:31 PM, Brandenburg, R. (Ra=
y) van &lt;<a href=3D"mailto:ray.vanbrandenburg@tno.nl">ray.vanbrandenburg@=
tno.nl</a>&gt; wrote:<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;text-autospace:none">
<span lang=3D"EN-US">Hi Magnus,</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;text-autospace:none">
<span lang=3D"EN-US">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;text-autospace:none">
<span lang=3D"EN-US">Reading your draft 'RTP Multiple Stream Sessions and S=
imulcast', I was wondering whether the proposed simulcast extensions could =
allow for RTP to be used for the delivery of spatially segmented content.
</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;text-autospace:none">
<span lang=3D"EN-US">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;text-autospace:none">
<span lang=3D"EN-US">Spatial segmentation is the concept of spatially divid=
ing a video up into multiple segments, each being independently decodable. =
It can for example be used to efficiently deliver video to clients who are =
only interested in specific areas
 of a video. Such bandwidth-saving are especially important when dealing wi=
th extremely high resolutions.
</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;text-autospace:none">
<span lang=3D"EN-US">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;text-autospace:none">
<span lang=3D"EN-US">Depending on the application, the recombination of seg=
ments into a single video might take place on the client or on a network no=
de (such as an RTP mixer). Your simulcast extensions might be a good way to=
 more effectively deal with the large
 number of segment streams being received by such a node.</span><o:p></o:p>=
</p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;text-autospace:none">
<span lang=3D"EN-US">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;text-autospace:none">
<span lang=3D"EN-US">I think it would be interesting to see whether we can =
expand the scope of your draft to include the specific properties of spatia=
lly segmented content.</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;text-autospace:none">
<span lang=3D"EN-US">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;text-autospace:none">
Cheers,<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;text-autospace:none">
&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">Ray van Brandenburg<o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Courier New&quot;">=
This e-mail and its contents are subject to the DISCLAIMER at
<a href=3D"http://www.tno.nl/emaildisclaimer" target=3D"_blank">http://www.=
tno.nl/emaildisclaimer</a></span><o:p></o:p></p>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><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><o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</body>
</html>

--_000_FCC100FC8D6B034CB88CD8173B2DA1580AF45651EXCMBX03tsntnon_--

From casner@acm.org  Wed Jul 27 08:36:46 2011
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Date: Wed, 27 Jul 2011 08:36:35 -0700 (PDT)
From: Stephen Casner <casner@acm.org>
To: Adam Roach <adam@nostrum.com>
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Cc: "AVT \(Audio/Video Transport\) WG" <avt@ietf.org>
Subject: Re: [AVTCORE] Media Types Registry
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What was the question?

                      -- Steve

On Wed, 27 Jul 2011, Adam Roach wrote:

> I'm not entirely sure what consensus the chairs took away from the room
> regarding the "RTP Payload Format media types" IANA registry, but I do want to
> make sure we consider the following before we call the topic resolved:
>
> Has anyone asked Steve Casner?
>
> I would appear that he put in the work to get an RFC published on the topic as
> recently as four years ago. This isn't an easy task, so I would assume he had
> some pretty good reasons for putting in the effort. I imagine that those
> reasons would be very relevant input to this conversation.
>
> /a
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt

From Even.roni@huawei.com  Wed Jul 27 09:26:47 2011
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Date: Wed, 27 Jul 2011 19:23:56 +0300
From: Roni Even <Even.roni@huawei.com>
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Subject: Re: [AVTCORE] Media Types Registry
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Hi Steve,
We got a question from IANA about RTP Payload Format media types
(http://www.iana.org/assignments/rtp-parameters ) registry which is not
current and is not well maintained. 

The payload subtype are registered in the main registry that avoids
collision etc. http://www.iana.org/assignments/media-types/index.html .

The question was if we see value in keeping it since the uniqueness is based
on the main registry. We were not sure why it was created and how it was
supposed to be updated.
We have two choices, update it and define the procedure to keep it current
or cancel it.
Regards
Roni Even



RTP Payload Types are identified by Media Types:
The main registry that avoids collision etc. 
http://www.iana.org/assignments/media-types/index.html
Registration in this registry is required, and works as intended
There is a second registry:
RTP Payload Format media types
http://www.iana.org/assignments/rtp-parameters
This registry is intended to contain all media types that have RTP Payload
formats
The issue with the second registry:
Poorly maintained, and a lot of formats are missing
Limited extra value, only usable if you want to know all RTP payload formats
Unknown that it exist

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
> Stephen Casner
> Sent: Wednesday, July 27, 2011 6:37 PM
> To: Adam Roach
> Cc: AVT (Audio/Video Transport) WG
> Subject: Re: [AVTCORE] Media Types Registry
> 
> What was the question?
> 
>                       -- Steve
> 
> On Wed, 27 Jul 2011, Adam Roach wrote:
> 
> > I'm not entirely sure what consensus the chairs took away from the
> room
> > regarding the "RTP Payload Format media types" IANA registry, but I
> do want to
> > make sure we consider the following before we call the topic
> resolved:
> >
> > Has anyone asked Steve Casner?
> >
> > I would appear that he put in the work to get an RFC published on the
> topic as
> > recently as four years ago. This isn't an easy task, so I would
> assume he had
> > some pretty good reasons for putting in the effort. I imagine that
> those
> > reasons would be very relevant input to this conversation.
> >
> > /a
> > _______________________________________________
> > 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


From stephen.botzko@gmail.com  Wed Jul 27 10:14:46 2011
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Subject: Re: [AVTCORE] Comments on draft-westerlund-avtcore-multistream-and-simulcast-00
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--bcaec5014ba93fcd4604a9102f0b
Content-Type: text/plain; charset=ISO-8859-1

Defining spatial relationships between RTP video sources is one reason CLUE
was formed.

Although Magnus explicitly used a telepresence system as one of his use
cases this morning, he made it clear that the side information needed to
define the spatial relationships was not part of his intended scope.

Stephen Botzko

On Wed, Jul 27, 2011 at 10:38 AM, Brandenburg, R. (Ray) van <
ray.vanbrandenburg@tno.nl> wrote:

>  Let me further clarify what I mean by spatial segmentation. ****
>
> ** **
>
> In the most general case, you would have a single camera source that
> records in a very high resolution (e.g. 4K or higher). Instead of encoding
> the recorded images as a single video stream, the camera spatially segments
> each video frame and encodes the results into multiple video streams (e.g. 4
> or 16) each representing a specific area of the recorded video. The result
> would be an RTP sender that makes available 16 separate streams.****
>
> ** **
>
> Depending on the area a user is interested in (the region-of-interest), an
> RTP receiver (or mixer) would set up one or more sessions to receive the one
> or more streams that are necessary to reconstruct the desired
> region-of-interest. The result would be that different RTP receivers would
> simultaneously receive different subsets of the available segment streams.
> ****
>
> ** **
>
> While some of the desired functionality might lie in the CLUE area (which I
> must admit I am not that familiar with), I think there is a considerable
> overlap with the use cases described in the Multistream/Simulcast draft. *
> ***
>
> ** **
>
> Cheers,****
>
> ** **
>
> Ray van Brandenburg****
>
> ** **
>
> ** **
>
> *From:* Stephen Botzko [mailto:stephen.botzko@gmail.com]
> *Sent:* woensdag 27 juli 2011 4:19
> *To:* Brandenburg, R. (Ray) van
> *Cc:* magnus.westerlund@ericsson.com; avt@ietf.org
> *Subject:* Re: [AVTCORE] Comments on
> draft-westerlund-avtcore-multistream-and-simulcast-00****
>
> ** **
>
> Such spatial relationships are CLUE's scope.****
>
> On Tue, Jul 26, 2011 at 6:31 PM, Brandenburg, R. (Ray) van <
> ray.vanbrandenburg@tno.nl> wrote:****
>
> Hi Magnus,****
>
>  ****
>
> Reading your draft 'RTP Multiple Stream Sessions and Simulcast', I was
> wondering whether the proposed simulcast extensions could allow for RTP to
> be used for the delivery of spatially segmented content. ****
>
>  ****
>
> Spatial segmentation is the concept of spatially dividing a video up into
> multiple segments, each being independently decodable. It can for example be
> used to efficiently deliver video to clients who are only interested in
> specific areas of a video. Such bandwidth-saving are especially important
> when dealing with extremely high resolutions. ****
>
>  ****
>
> Depending on the application, the recombination of segments into a single
> video might take place on the client or on a network node (such as an RTP
> mixer). Your simulcast extensions might be a good way to more effectively
> deal with the large number of segment streams being received by such a node.
> ****
>
>  ****
>
> I think it would be interesting to see whether we can expand the scope of
> your draft to include the specific properties of spatially segmented
> content.****
>
>  ****
>
> Cheers,****
>
>  ****
>
> Ray van Brandenburg****
>
> This e-mail and its contents are subject to the DISCLAIMER at
> http://www.tno.nl/emaildisclaimer****
>
>
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt****
>
> ** **
>

--bcaec5014ba93fcd4604a9102f0b
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Defining spatial relationships between RTP video sources is one reason CLUE=
 was formed.=A0 <br><br>Although Magnus explicitly used a telepresence syst=
em as one of his use cases this morning, he made it clear that the side inf=
ormation needed to define the spatial relationships was not part of his int=
ended scope.<br>
<br>Stephen Botzko<br><br><div class=3D"gmail_quote">On Wed, Jul 27, 2011 a=
t 10:38 AM, Brandenburg, R. (Ray) van <span dir=3D"ltr">&lt;<a href=3D"mail=
to:ray.vanbrandenburg@tno.nl">ray.vanbrandenburg@tno.nl</a>&gt;</span> wrot=
e:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;">





<div link=3D"blue" vlink=3D"purple" lang=3D"NL">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#1F497D" lang=
=3D"EN-US">Let me further clarify what I mean by spatial segmentation.
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#1F497D" lang=
=3D"EN-US"><u></u>=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#1F497D" lang=
=3D"EN-US">In the most general case, you would have a single camera source =
that records in a very high resolution (e.g. 4K or higher). Instead of enco=
ding
 the recorded images as a single video stream, the camera spatially segment=
s each video frame and encodes the results into multiple video streams (e.g=
. 4 or 16) each representing a specific area of the recorded video. The res=
ult would be an RTP sender that
 makes available 16 separate streams.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#1F497D" lang=
=3D"EN-US"><u></u>=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#1F497D" lang=
=3D"EN-US">Depending on the area a user is interested in (the region-of-int=
erest), an RTP receiver (or mixer) would set up one or more sessions to rec=
eive
 the one or more streams that are necessary to reconstruct the desired regi=
on-of-interest. The result would be that different RTP receivers would simu=
ltaneously receive different subsets of the available segment streams.<u></=
u><u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#1F497D" lang=
=3D"EN-US"><u></u>=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#1F497D" lang=
=3D"EN-US">While some of the desired functionality might lie in the CLUE ar=
ea (which I must admit I am not that familiar with), I think there is a con=
siderable
 overlap with the use cases described in the Multistream/Simulcast draft. <=
u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#1F497D" lang=
=3D"EN-US"><u></u>=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#1F497D" lang=
=3D"EN-US">Cheers,<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#1F497D" lang=
=3D"EN-US"><u></u>=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#1F497D" lang=
=3D"EN-US">Ray van Brandenburg<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#1F497D" lang=
=3D"EN-US"><u></u>=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#1F497D" lang=
=3D"EN-US"><u></u>=A0<u></u></span></p>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt" lang=3D"EN-US">F=
rom:</span></b><span style=3D"font-size:10.0pt" lang=3D"EN-US"> Stephen Bot=
zko [mailto:<a href=3D"mailto:stephen.botzko@gmail.com" target=3D"_blank">s=
tephen.botzko@gmail.com</a>]
<br>
<b>Sent:</b> woensdag 27 juli 2011 4:19<br>
<b>To:</b> Brandenburg, R. (Ray) van<br>
<b>Cc:</b> <a href=3D"mailto:magnus.westerlund@ericsson.com" target=3D"_bla=
nk">magnus.westerlund@ericsson.com</a>; <a href=3D"mailto:avt@ietf.org" tar=
get=3D"_blank">avt@ietf.org</a><br>
<b>Subject:</b> Re: [AVTCORE] Comments on draft-westerlund-avtcore-multistr=
eam-and-simulcast-00<u></u><u></u></span></p><div><div></div><div class=3D"=
h5">
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>=A0<u></u></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">Such spatial relation=
ships are CLUE&#39;s scope.<u></u><u></u></p>
<div>
<p class=3D"MsoNormal">On Tue, Jul 26, 2011 at 6:31 PM, Brandenburg, R. (Ra=
y) van &lt;<a href=3D"mailto:ray.vanbrandenburg@tno.nl" target=3D"_blank">r=
ay.vanbrandenburg@tno.nl</a>&gt; wrote:<u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"text-autospace:none">
<span lang=3D"EN-US">Hi Magnus,</span><u></u><u></u></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none">
<span lang=3D"EN-US">=A0</span><u></u><u></u></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none">
<span lang=3D"EN-US">Reading your draft &#39;RTP Multiple Stream Sessions a=
nd Simulcast&#39;, I was wondering whether the proposed simulcast extension=
s could allow for RTP to be used for the delivery of spatially segmented co=
ntent.
</span><u></u><u></u></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none">
<span lang=3D"EN-US">=A0</span><u></u><u></u></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none">
<span lang=3D"EN-US">Spatial segmentation is the concept of spatially divid=
ing a video up into multiple segments, each being independently decodable. =
It can for example be used to efficiently deliver video to clients who are =
only interested in specific areas
 of a video. Such bandwidth-saving are especially important when dealing wi=
th extremely high resolutions.
</span><u></u><u></u></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none">
<span lang=3D"EN-US">=A0</span><u></u><u></u></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none">
<span lang=3D"EN-US">Depending on the application, the recombination of seg=
ments into a single video might take place on the client or on a network no=
de (such as an RTP mixer). Your simulcast extensions might be a good way to=
 more effectively deal with the large
 number of segment streams being received by such a node.</span><u></u><u><=
/u></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none">
<span lang=3D"EN-US">=A0</span><u></u><u></u></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none">
<span lang=3D"EN-US">I think it would be interesting to see whether we can =
expand the scope of your draft to include the specific properties of spatia=
lly segmented content.</span><u></u><u></u></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none">
<span lang=3D"EN-US">=A0</span><u></u><u></u></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none">
Cheers,<u></u><u></u></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none">
=A0<u></u><u></u></p>
<p class=3D"MsoNormal">Ray van Brandenburg<u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Courier New&quot;">=
This e-mail and its contents are subject to the DISCLAIMER at
<a href=3D"http://www.tno.nl/emaildisclaimer" target=3D"_blank">http://www.=
tno.nl/emaildisclaimer</a></span><u></u><u></u></p>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
_______________________________________________<br>
Audio/Video Transport Core Maintenance<br>
<a href=3D"mailto:avt@ietf.org" target=3D"_blank">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><u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
</div></div></div>
</div>

</blockquote></div><br>

--bcaec5014ba93fcd4604a9102f0b--

From magnus.westerlund@ericsson.com  Wed Jul 27 10:24:13 2011
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Date: Wed, 27 Jul 2011 13:23:38 -0400
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [AVTCORE] Comments on draft-westerlund-avtcore-multistream-and-simulcast-00
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Hi Ray,

As you describe this case, I would say that it would be very similar to
having for example 3 cameras covering adjacent fields of view. The
spatial relation between the streams are clearly a CLUE question, not a
AVTCORE one.

Cheers

Magnus

On 2011-07-27 10:38, Brandenburg, R. (Ray) van wrote:
> 
> In the most general case, you would have a single camera source that
> records in a very high resolution (e.g. 4K or higher). Instead of
> encoding the recorded images as a single video stream, the camera
> spatially segments each video frame and encodes the results into
> multiple video streams (e.g. 4 or 16) each representing a specific area
> of the recorded video. The result would be an RTP sender that makes
> available 16 separate streams.
> 
>  
> 
> Depending on the area a user is interested in (the region-of-interest),
> an RTP receiver (or mixer) would set up one or more sessions to receive
> the one or more streams that are necessary to reconstruct the desired
> region-of-interest. The result would be that different RTP receivers
> would simultaneously receive different subsets of the available segment
> streams.
> 
>  
> 
> While some of the desired functionality might lie in the CLUE area
> (which I must admit I am not that familiar with), I think there is a
> considerable overlap with the use cases described in the
> Multistream/Simulcast draft.
> 

-- 

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 casner@acm.org  Wed Jul 27 10:44:30 2011
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Date: Wed, 27 Jul 2011 10:44:14 -0700 (PDT)
From: Stephen Casner <casner@acm.org>
To: Roni Even <Even.roni@huawei.com>
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Subject: Re: [AVTCORE] Media Types Registry
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Roni and AVT,

I created the RTP Payload Format Media Types table initially because I
felt that the limited table of static payload types needed to be
accompanied by the more extensive list of payload formats that were
available using dynamic payload types.  As noted below, the expected
value was that it identified the subset of registered media subtypes
that were defined for transport via RTP.

For audio and video, many of the subtypes in the main registry are
defined as part of an RTP payload format, but not all.  And for the
text subtypes in the RTP Payload Format Media Types table, finding
those in the main table or realizing that they exist would be harder.

Clearly the value of this table depends upon it being maintained.  Is
IANA willing to establish a practice that when new media subtypes are
defined for RTP they would automatically go into this table?  I would
hope so.  Or would IANA require that the IANA Considerations section
of each payload format RFC explicitly mention this?

                                                        -- Steve

On Wed, 27 Jul 2011, Roni Even wrote:

> Hi Steve,
> We got a question from IANA about RTP Payload Format media types
> (http://www.iana.org/assignments/rtp-parameters ) registry which is not
> current and is not well maintained.
>
> The payload subtype are registered in the main registry that avoids
> collision etc. http://www.iana.org/assignments/media-types/index.html .
>
> The question was if we see value in keeping it since the uniqueness is based
> on the main registry. We were not sure why it was created and how it was
> supposed to be updated.
> We have two choices, update it and define the procedure to keep it current
> or cancel it.
> Regards
> Roni Even
>
>
>
> RTP Payload Types are identified by Media Types:
> The main registry that avoids collision etc.
> http://www.iana.org/assignments/media-types/index.html
> Registration in this registry is required, and works as intended
> There is a second registry:
> RTP Payload Format media types
> http://www.iana.org/assignments/rtp-parameters
> This registry is intended to contain all media types that have RTP Payload
> formats
> The issue with the second registry:
> Poorly maintained, and a lot of formats are missing
> Limited extra value, only usable if you want to know all RTP payload formats
> Unknown that it exist
>
> > -----Original Message-----
> > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
> > Stephen Casner
> > Sent: Wednesday, July 27, 2011 6:37 PM
> > To: Adam Roach
> > Cc: AVT (Audio/Video Transport) WG
> > Subject: Re: [AVTCORE] Media Types Registry
> >
> > What was the question?
> >
> >                       -- Steve
> >
> > On Wed, 27 Jul 2011, Adam Roach wrote:
> >
> > > I'm not entirely sure what consensus the chairs took away from the
> > room
> > > regarding the "RTP Payload Format media types" IANA registry, but I
> > do want to
> > > make sure we consider the following before we call the topic
> > resolved:
> > >
> > > Has anyone asked Steve Casner?
> > >
> > > I would appear that he put in the work to get an RFC published on the
> > topic as
> > > recently as four years ago. This isn't an easy task, so I would
> > assume he had
> > > some pretty good reasons for putting in the effort. I imagine that
> > those
> > > reasons would be very relevant input to this conversation.
> > >
> > > /a
> > > _______________________________________________
> > > 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
>

From ray.vanbrandenburg@tno.nl  Wed Jul 27 10:53:05 2011
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From: "Brandenburg, R. (Ray) van" <ray.vanbrandenburg@tno.nl>
To: 'Magnus Westerlund' <magnus.westerlund@ericsson.com>
Thread-Topic: [AVTCORE] Comments on draft-westerlund-avtcore-multistream-and-simulcast-00
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Subject: Re: [AVTCORE] Comments on draft-westerlund-avtcore-multistream-and-simulcast-00
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I agree.

What I meant was that in the case of spatially segmented content and thus r=
eceivers receiving multiple streams, you run into the same kind of difficul=
ties as you described in your draft: when to use multiple sessions, when to=
 multiplex multiple streams into one session.=20

Furthermore, for spatially segmented content it might be especially useful =
to have multiple versions of the same stream available (in different resolu=
tions for example) and some form of RTP Simulcast might thus be useful.

Ray

-----Original Message-----
From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]=20
Sent: woensdag 27 juli 2011 19:24
To: Brandenburg, R. (Ray) van
Cc: Stephen Botzko; avt@ietf.org
Subject: Re: [AVTCORE] Comments on draft-westerlund-avtcore-multistream-and=
-simulcast-00

Hi Ray,

As you describe this case, I would say that it would be very similar to hav=
ing for example 3 cameras covering adjacent fields of view. The spatial rel=
ation between the streams are clearly a CLUE question, not a AVTCORE one.

Cheers

Magnus

On 2011-07-27 10:38, Brandenburg, R. (Ray) van wrote:
>=20
> In the most general case, you would have a single camera source that=20
> records in a very high resolution (e.g. 4K or higher). Instead of=20
> encoding the recorded images as a single video stream, the camera=20
> spatially segments each video frame and encodes the results into=20
> multiple video streams (e.g. 4 or 16) each representing a specific=20
> area of the recorded video. The result would be an RTP sender that=20
> makes available 16 separate streams.
>=20
> =20
>=20
> Depending on the area a user is interested in (the=20
> region-of-interest), an RTP receiver (or mixer) would set up one or=20
> more sessions to receive the one or more streams that are necessary to=20
> reconstruct the desired region-of-interest. The result would be that=20
> different RTP receivers would simultaneously receive different subsets=20
> of the available segment streams.
>=20
> =20
>=20
> While some of the desired functionality might lie in the CLUE area=20
> (which I must admit I am not that familiar with), I think there is a=20
> considerable overlap with the use cases described in the=20
> Multistream/Simulcast draft.
>=20

--=20

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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This e-mail and its contents are subject to the DISCLAIMER at http://www.tn=
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Hi,

I think one of the reason this registry are missing so much is that we
don't have put this into the IANA consideration section or had clear
process established with IANA.

I have talked with Michelle and it might be possible to a default
process into place. RFC 4288 is apparently under way to do revisions.

https://datatracker.ietf.org/doc/draft-freed-media-type-regs/

Something we should discuss around this document is if there should be a
question in the registration template "Does the media type have an RTP
payload type" that thus should result in the media type being added to
this registry automatically.

But, I think the low work solution is to instruct IANA to add this to
their checklist. Have the PAYLOAD WG chairs try to ensure that the IANA
section contains the right instruction. And that someone spends the 2-4
hours going through the documents to find all missing RTP payload types.

I do think there is some value with this registry, as it is difficult to
find the sub-set of media types that has RTP payload formats.

Cheers

Magnus


On 2011-07-27 13:44, Stephen Casner wrote:
> Roni and AVT,
> 
> I created the RTP Payload Format Media Types table initially because I
> felt that the limited table of static payload types needed to be
> accompanied by the more extensive list of payload formats that were
> available using dynamic payload types.  As noted below, the expected
> value was that it identified the subset of registered media subtypes
> that were defined for transport via RTP.
> 
> For audio and video, many of the subtypes in the main registry are
> defined as part of an RTP payload format, but not all.  And for the
> text subtypes in the RTP Payload Format Media Types table, finding
> those in the main table or realizing that they exist would be harder.
> 
> Clearly the value of this table depends upon it being maintained.  Is
> IANA willing to establish a practice that when new media subtypes are
> defined for RTP they would automatically go into this table?  I would
> hope so.  Or would IANA require that the IANA Considerations section
> of each payload format RFC explicitly mention this?
> 
>                                                         -- Steve
> 
> On Wed, 27 Jul 2011, Roni Even wrote:
> 
>> Hi Steve,
>> We got a question from IANA about RTP Payload Format media types
>> (http://www.iana.org/assignments/rtp-parameters ) registry which is not
>> current and is not well maintained.
>>
>> The payload subtype are registered in the main registry that avoids
>> collision etc. http://www.iana.org/assignments/media-types/index.html .
>>
>> The question was if we see value in keeping it since the uniqueness is based
>> on the main registry. We were not sure why it was created and how it was
>> supposed to be updated.
>> We have two choices, update it and define the procedure to keep it current
>> or cancel it.
>> Regards
>> Roni Even
>>
>>
>>
>> RTP Payload Types are identified by Media Types:
>> The main registry that avoids collision etc.
>> http://www.iana.org/assignments/media-types/index.html
>> Registration in this registry is required, and works as intended
>> There is a second registry:
>> RTP Payload Format media types
>> http://www.iana.org/assignments/rtp-parameters
>> This registry is intended to contain all media types that have RTP Payload
>> formats
>> The issue with the second registry:
>> Poorly maintained, and a lot of formats are missing
>> Limited extra value, only usable if you want to know all RTP payload formats
>> Unknown that it exist
>>
>>> -----Original Message-----
>>> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
>>> Stephen Casner
>>> Sent: Wednesday, July 27, 2011 6:37 PM
>>> To: Adam Roach
>>> Cc: AVT (Audio/Video Transport) WG
>>> Subject: Re: [AVTCORE] Media Types Registry
>>>
>>> What was the question?
>>>
>>>                       -- Steve
>>>
>>> On Wed, 27 Jul 2011, Adam Roach wrote:
>>>
>>>> I'm not entirely sure what consensus the chairs took away from the
>>> room
>>>> regarding the "RTP Payload Format media types" IANA registry, but I
>>> do want to
>>>> make sure we consider the following before we call the topic
>>> resolved:
>>>>
>>>> Has anyone asked Steve Casner?
>>>>
>>>> I would appear that he put in the work to get an RFC published on the
>>> topic as
>>>> recently as four years ago. This isn't an easy task, so I would
>>> assume he had
>>>> some pretty good reasons for putting in the effort. I imagine that
>>> those
>>>> reasons would be very relevant input to this conversation.
>>>>
>>>> /a
>>>> _______________________________________________
>>>> 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
>>
> _______________________________________________
> 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 casner@acm.org  Wed Jul 27 15:50:42 2011
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Cc: "'AVT \(Audio/Video Transport\) WG'" <avt@ietf.org>
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Magnus,

> I think one of the reason this registry are missing so much is that we
> don't have put this into the IANA consideration section or had clear
> process established with IANA.
>
> I have talked with Michelle and it might be possible to a default
> process into place. RFC 4288 is apparently under way to do revisions.
>
> https://datatracker.ietf.org/doc/draft-freed-media-type-regs/
>
> Something we should discuss around this document is if there should be a
> question in the registration template "Does the media type have an RTP
> payload type" that thus should result in the media type being added to
> this registry automatically.

Please be careful with terminology.  An "RTP Payload Type" is a
number, for which a small set has been defined statically, but most
are assigned dynamically.  So the question you suggested for the
template is not correct.

In RFC 3555, it was specified that the encoding considerations should
be explicit about RTP transport:

    2. Procedure For Registering MIME Types for RTP Payload Types

       [snip]

	  Encoding considerations
	     The fact that the type can be transferred via RTP MUST be
	     noted.

In RFC 4855, that instruction is changed, but I don't remember why.
Instead, the same fact is noted in under "Restrictions on usage":

     Restrictions on usage:
	  The fact that the media type is defined for transfer via RTP
	  MUST be noted, in particular, if the transfer depends on RTP
	  framing and hence the media type is only defined for transfer
	  via RTP.

Thus, if these procedures have been followed, there should already be
sufficient information for IANA to take the additional step.

> But, I think the low work solution is to instruct IANA to add this to
> their checklist. Have the PAYLOAD WG chairs try to ensure that the IANA
> section contains the right instruction. And that someone spends the 2-4
> hours going through the documents to find all missing RTP payload types.
>
> I do think there is some value with this registry, as it is difficult to
> find the sub-set of media types that has RTP payload formats.

That is my opinion as well.

                                                        -- Steve

> On 2011-07-27 13:44, Stephen Casner wrote:
> > Roni and AVT,
> >
> > I created the RTP Payload Format Media Types table initially because I
> > felt that the limited table of static payload types needed to be
> > accompanied by the more extensive list of payload formats that were
> > available using dynamic payload types.  As noted below, the expected
> > value was that it identified the subset of registered media subtypes
> > that were defined for transport via RTP.
> >
> > For audio and video, many of the subtypes in the main registry are
> > defined as part of an RTP payload format, but not all.  And for the
> > text subtypes in the RTP Payload Format Media Types table, finding
> > those in the main table or realizing that they exist would be harder.
> >
> > Clearly the value of this table depends upon it being maintained.  Is
> > IANA willing to establish a practice that when new media subtypes are
> > defined for RTP they would automatically go into this table?  I would
> > hope so.  Or would IANA require that the IANA Considerations section
> > of each payload format RFC explicitly mention this?
> >
> >                                                         -- Steve
> >
> > On Wed, 27 Jul 2011, Roni Even wrote:
> >
> >> Hi Steve,
> >> We got a question from IANA about RTP Payload Format media types
> >> (http://www.iana.org/assignments/rtp-parameters ) registry which is not
> >> current and is not well maintained.
> >>
> >> The payload subtype are registered in the main registry that avoids
> >> collision etc. http://www.iana.org/assignments/media-types/index.html .
> >>
> >> The question was if we see value in keeping it since the uniqueness is based
> >> on the main registry. We were not sure why it was created and how it was
> >> supposed to be updated.
> >> We have two choices, update it and define the procedure to keep it current
> >> or cancel it.
> >> Regards
> >> Roni Even
> >>
> >>
> >>
> >> RTP Payload Types are identified by Media Types:
> >> The main registry that avoids collision etc.
> >> http://www.iana.org/assignments/media-types/index.html
> >> Registration in this registry is required, and works as intended
> >> There is a second registry:
> >> RTP Payload Format media types
> >> http://www.iana.org/assignments/rtp-parameters
> >> This registry is intended to contain all media types that have RTP Payload
> >> formats
> >> The issue with the second registry:
> >> Poorly maintained, and a lot of formats are missing
> >> Limited extra value, only usable if you want to know all RTP payload formats
> >> Unknown that it exist
> >>
> >>> -----Original Message-----
> >>> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
> >>> Stephen Casner
> >>> Sent: Wednesday, July 27, 2011 6:37 PM
> >>> To: Adam Roach
> >>> Cc: AVT (Audio/Video Transport) WG
> >>> Subject: Re: [AVTCORE] Media Types Registry
> >>>
> >>> What was the question?
> >>>
> >>>                       -- Steve
> >>>
> >>> On Wed, 27 Jul 2011, Adam Roach wrote:
> >>>
> >>>> I'm not entirely sure what consensus the chairs took away from the
> >>> room
> >>>> regarding the "RTP Payload Format media types" IANA registry, but I
> >>> do want to
> >>>> make sure we consider the following before we call the topic
> >>> resolved:
> >>>>
> >>>> Has anyone asked Steve Casner?
> >>>>
> >>>> I would appear that he put in the work to get an RFC published on the
> >>> topic as
> >>>> recently as four years ago. This isn't an easy task, so I would
> >>> assume he had
> >>>> some pretty good reasons for putting in the effort. I imagine that
> >>> those
> >>>> reasons would be very relevant input to this conversation.
> >>>>
> >>>> /a
> >>>> _______________________________________________
> >>>> 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
> >>
> > _______________________________________________
> > 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?r?gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>

From magnus.westerlund@ericsson.com  Thu Jul 28 08:19:47 2011
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Date: Thu, 28 Jul 2011 11:19:42 -0400
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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To: Stephen Casner <casner@acm.org>
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Cc: "'AVT \(Audio/Video Transport\) WG'" <avt@ietf.org>
Subject: Re: [AVTCORE] Media Types Registry
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On 2011-07-27 18:50, Stephen Casner wrote:
> Magnus,
> 
>> I think one of the reason this registry are missing so much is that we
>> don't have put this into the IANA consideration section or had clear
>> process established with IANA.
>>
>> I have talked with Michelle and it might be possible to a default
>> process into place. RFC 4288 is apparently under way to do revisions.
>>
>> https://datatracker.ietf.org/doc/draft-freed-media-type-regs/
>>
>> Something we should discuss around this document is if there should be a
>> question in the registration template "Does the media type have an RTP
>> payload type" that thus should result in the media type being added to
>> this registry automatically.
> 
> Please be careful with terminology.  An "RTP Payload Type" is a
> number, for which a small set has been defined statically, but most
> are assigned dynamically.  So the question you suggested for the
> template is not correct.
> 

Yes, you are correct. it should say "RTP Payload Format".


> In RFC 3555, it was specified that the encoding considerations should
> be explicit about RTP transport:
> 
>     2. Procedure For Registering MIME Types for RTP Payload Types
> 
>        [snip]
> 
> 	  Encoding considerations
> 	     The fact that the type can be transferred via RTP MUST be
> 	     noted.
> 
> In RFC 4855, that instruction is changed, but I don't remember why.
> Instead, the same fact is noted in under "Restrictions on usage":
> 
>      Restrictions on usage:
> 	  The fact that the media type is defined for transfer via RTP
> 	  MUST be noted, in particular, if the transfer depends on RTP
> 	  framing and hence the media type is only defined for transfer
> 	  via RTP.
> 
> Thus, if these procedures have been followed, there should already be
> sufficient information for IANA to take the additional step.

Yes, I think it has been followed. I think who ever gets to do a bit of
digging through the RFCs will find if it is solid or have been missed
some times.

But, I think it is probably safer to have this instruction in the IANA
consideration section to avoid that IANA misses this.

Cheers

Magnus Westerlund

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