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Wg,

We have just posted a WG session agenda for IETF 82. If there is any
questions or late requests please contact the chairs.

Audio/Video Transport Extensions (avtext)
======================================================================

Chairs: Magnus Westerlund <magnus.westerlund@ericsson.com>
        Keith Drage       <keith.drage@alcatel-lucent.com>


Agenda IETF 82, Taipei

15:10 - 16:10 WEDNESDAY, November 16, 2011. Room: 101C
----------------------------------------------------------------------
15:10 AVTEXT Status Update                   	          (Chairs, 10)

15:20 IEEE 1588/802.1AS Synchronisation for RTP Streams (Williams, 15)
      draft-williams-avtext-avbsync-02

15:35 RTCP SDES Item SRCNAME to Label Individual Sources  (Magnus, 15)
      draft-westerlund-avtext-rtcp-sdes-srcname-00

15:20 RTP Media Stream Pause and Resume                   (Magnus, 20)

16:10 End

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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WG,

Corrected version:

Audio/Video Transport Extensions (avtext)
======================================================================

Chairs: Magnus Westerlund <magnus.westerlund@ericsson.com>
        Keith Drage       <keith.drage@alcatel-lucent.com>


Agenda IETF 82, Taipei

15:10 - 16:10 WEDNESDAY, November 16, 2011. Room: 101C
----------------------------------------------------------------------
15:10 AVTEXT Status Update                   	          (Chairs, 10)

15:20 IEEE 1588/802.1AS Synchronisation for RTP Streams (Williams, 15)
      draft-williams-avtext-avbsync-02

15:35 RTCP SDES Item SRCNAME to Label Individual Sources  (Magnus, 15)
      draft-westerlund-avtext-rtcp-sdes-srcname-00

15:20 RTP Media Stream Pause and Resume                   (Magnus, 20)
      draft-westerlund-avtext-rtp-stream-pause-00

16:10 End

-- 

Magnus Westerlund

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


From magnus.westerlund@ericsson.com  Tue Nov  1 03:45:15 2011
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Well,

The once more corrected and in the future correct version can be
retrieved from here:

http://www.ietf.org/proceedings/82/agenda/avtext.txt

Cheers



On 2011-11-01 11:43, Magnus Westerlund wrote:
> WG,
> 
> Corrected version:
> 
> Audio/Video Transport Extensions (avtext)
> ======================================================================
> 
> Chairs: Magnus Westerlund <magnus.westerlund@ericsson.com>
>         Keith Drage       <keith.drage@alcatel-lucent.com>
> 
> 
> Agenda IETF 82, Taipei
> 
> 15:10 - 16:10 WEDNESDAY, November 16, 2011. Room: 101C
> ----------------------------------------------------------------------
> 15:10 AVTEXT Status Update                   	          (Chairs, 10)
> 
> 15:20 IEEE 1588/802.1AS Synchronisation for RTP Streams (Williams, 15)
>       draft-williams-avtext-avbsync-02
> 
> 15:35 RTCP SDES Item SRCNAME to Label Individual Sources  (Magnus, 15)
>       draft-westerlund-avtext-rtcp-sdes-srcname-00
> 
> 15:50 RTP Media Stream Pause and Resume                   (Magnus, 20)
>       draft-westerlund-avtext-rtp-stream-pause-00
> 
> 16:10 End
> 


-- 

Magnus Westerlund

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


From magnus.westerlund@ericsson.com  Thu Nov  3 08:10:40 2011
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Subject: [avtext] Pause and Resume CCM extension
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Hi,

As individual,

My colleagues and I have put together a proposal for a functionality for
pausing and resuming media streams. This is primarily intended for mixer
scenarios where a mixer would send the pause request to media senders
that currently don't need to send any media for the mixer isn't using
this source. For example video streams when the participant isn't active.

Draft:
https://datatracker.ietf.org/doc/draft-westerlund-avtext-rtp-stream-pause/


It does cover a few other usages but that is primarily to ensure that it
wouldn't break the whole system if introduced into ASM or transport
relay topologies.

This proposal is also based on that the system in general may use
something like the Media Stream Selection (see link below) to include or
exclude media streams they don't want. Thus we have in fact excluded
such usages by not allowing the end-points to use this to manipulate the
CSRC list of media stream that are mixes or selections. This as it would
only provide partial functionality compared to Media Stream Selection.

https://datatracker.ietf.org/doc/draft-westerlund-dispatch-stream-selection/

Feedback is most appreciated


Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
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From magnus.westerlund@ericsson.com  Fri Nov  4 03:25:20 2011
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Hi,

(As individual)

My colleagues and I have submitted an new Internet draft for the 00
deadline.

https://datatracker.ietf.org/doc/draft-westerlund-avtext-rtcp-sdes-srcname/

So this draft is actually split out of
https://datatracker.ietf.org/doc/draft-westerlund-avtcore-multistream-and-simulcast/

That was discussed in the AVTCORE WG in Quebec. This proposal we see as
a independent building block that becomes highly usable in RTP sessions
where end-points starting to have more than one media source being
transported in an RTP session and there is reason to have more than one
media stream (SSRC) associated with that media source, like
retransmission, FEC, duplication, layered encoding, simulcast.

We do have agenda time for the upcomming meeting so I hope you have the
time to review the draft prior to the meeting. Even more appreciated if
you want to provide feedback to the mailing list.

cheers

Magnus Westerlund

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----------------------------------------------------------------------
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: "avtext@ietf.org" <avtext@ietf.org>
Date: Sat, 5 Nov 2011 01:09:01 +0100
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Subject: [avtext] Future working group last calls
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(As working group co-chair)

It is the intent of the working group chairs to START working group last ca=
lls on the following two documents on Wednesday 16th November (strangely en=
ough very shortly after the end of the face to face meeting).

https://datatracker.ietf.org/doc/draft-ietf-avtext-rams-scenarios/
https://datatracker.ietf.org/doc/draft-ietf-avtext-splicing-for-rtp/

This will be a 3 week working group last call when it happens.

The chairs and the editors believe these two documents are complete in thei=
r current form and ready for WGLC. This is your chance as WG members to ide=
ntify any open issues or omissions before we start that WGLC, so that WGLC =
can take place on all the final material.

So we would urge you all to have a look at these two documents in respect o=
f that need.

Regards

Keith

P.S. Just to stress, this is not starting the WGLC now, it is the request f=
or you to make sure the documents are ready for that last call.=

From mzanaty@cisco.com  Fri Nov  4 20:27:37 2011
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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
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Subject: Re: [avtext] Pause and Resume CCM extension
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Magnus,
How does the proposed pause differ from tmmbr=3D0? Some implementations =
already use tmmbr=3D0 to pause peers.

Assuming there is some valuable nuance between pause and tmmbr=3D0, it =
may be useful to explicitly allow acks without requests so senders can =
notify peers when they pause (and resume) themselves autonomously. This =
is not explicitly prohibited in the current draft, just as autonomous =
tmmbn (without tmmbr) is not explicitly prohibited in RFC 5104. But it =
would be good to explicitly allow and describe this use case. Some =
implementations already use tmmbn=3D0 to notify peers they have paused =
themselves autonomously (without receiving any tmmbr).=20

Thanks,
Mo

-----Original Message-----
From: avtext-bounces@ietf.org [mailto:avtext-bounces@ietf.org] On Behalf =
Of Magnus Westerlund
Sent: Thursday, November 03, 2011 11:10 AM
To: avtext@ietf.org
Subject: [avtext] Pause and Resume CCM extension

Hi,

As individual,

My colleagues and I have put together a proposal for a functionality for
pausing and resuming media streams. This is primarily intended for mixer
scenarios where a mixer would send the pause request to media senders
that currently don't need to send any media for the mixer isn't using
this source. For example video streams when the participant isn't =
active.

Draft:
https://datatracker.ietf.org/doc/draft-westerlund-avtext-rtp-stream-pause=
/


It does cover a few other usages but that is primarily to ensure that it
wouldn't break the whole system if introduced into ASM or transport
relay topologies.

This proposal is also based on that the system in general may use
something like the Media Stream Selection (see link below) to include or
exclude media streams they don't want. Thus we have in fact excluded
such usages by not allowing the end-points to use this to manipulate the
CSRC list of media stream that are mixes or selections. This as it would
only provide partial functionality compared to Media Stream Selection.

https://datatracker.ietf.org/doc/draft-westerlund-dispatch-stream-selecti=
on/

Feedback is most appreciated


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

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

From kevin.gross@avanw.com  Sun Nov  6 20:12:31 2011
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Subject: Re: [avtext] draft-williams-avtext-avbsync updated
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--0015174785d24a966704b11d4367
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Hi Aidan,

Nice to see a new revision. Nice to see acknowledgement of RFC 6051 and
IDMS.

IDMS defines a "clocksource" SDP attribute to identify the clock
domain. The following clock distribution methods are supported by IDMS:
NTP, GPS, Galileo, IEEE 1588-2002, IEEE 1588-2008 and IEEE 802.1AS. When
IEEE 1588-2002 or IEEE 1588-2008 are used, a domain identifier is also
specified. The IEEE 802.1AS PTP profile allows use of the default domain
(0) only. You are proposing using the EUI-64 of the grandmaster instead. I
also considered this possibility for IDMS but was confounded by the
following scenario...

For fault tolerance, mission-critical media systems are configured with at
least two potential grandmasters all receiving time from a common reference
(e.g. house sync reference, GPS). Upon failure of the primary grandmaster,
another takes over and, although nothing has functionally changed about the
clock domain, the EUI-64 of the grandmaster in the network does change.
Until updated SDP advertisements reflecting the new grandmaster are
received, senders and receivers will believe they are operating on
different time domains and streams will presumably be interrupted.

It would be very useful to have a unique identifier for a clock domain. My
conclusion after considering the above scenario was that the grandmaster ID
was not that identifier and clock distribution protocol, protocol version
and domain identifier is currently the best identifier we have. I think
there is definitely room for improvement in this. It does not strike me as
an easy problem, however.

I like your idea of using a traceability indicator to help determine clock
compatibility. I am concerned however that while traceable clocks can be
assumed to be frequency synchronized (syntonized) I am not aware of any way
to gauge the accuracy of their phase synchronization. Nevertheless, adding
a traceability indicator to the SDP clock attribute would seem to be a
useful improvement.

With regards to the proposed T, M and U flags in the header extension. I
don't see a need for T if traceability information is to be available via
SDP. T is not soemthing that changes dynamically, is it? You do acknowledge
that M and U come from IEEE 1722. You should add the minimum toggle period
requirement we see in IEEE 1722 section 5.4.1 to your M discussion to be
sure these events are not missed if the clock is thrashing.

I am not convinced that the information provided by M and U will improve
system performance. No specific recommendations for how receivers should
respond to these conditions are given in this draft or in IEEE 1722.
Perhaps inclusion of such recommendations or a report from someone who has
implemented IEEE 1722 would change my opinion.

I do support reusing IDMS SDP and RFC 6051 header extensions definitions to
meet the requirements of your proposal. I encourage you to review and help
improve the IDMS draft. I'm not yet convinced that the header extension
defined in RFC 6051 needs improvement but if it does, there are already two
header options defined there; Perhaps a third could be added.

Kevin Gross

On Mon, Oct 31, 2011 at 6:50 PM, Aidan Williams <aidan.williams@audinate.com
> wrote:

> Hi All,
>
> I've updated the draft on avb sync, you can find it at:
>
>  http://tools.ietf.org/html/**draft-williams-avtext-avbsync-**02<http://tools.ietf.org/html/draft-williams-avtext-avbsync-02>
>
> The main changes are:
>
>  - more discussion about clock domains and traceable time
>  - additional header timing metadata
>  - some discussion about signalling with SDP
>
> Section 8 also describes an alternative approach using the existing RFC
> 6051 header extension.  I'd appreciate feedback on whether this is more
> desirable approach to take.
>
> regards
>            aidan
> ____
> :wq!
>
> ______________________________**_________________
> avtext mailing list
> avtext@ietf.org
> https://www.ietf.org/mailman/**listinfo/avtext<https://www.ietf.org/mailman/listinfo/avtext>
>

--0015174785d24a966704b11d4367
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Hi Aidan,<div><br></div><div>Nice to see a new revision. Nice to see acknow=
ledgement of RFC 6051 and IDMS.</div><div><br></div><div>IDMS defines a &qu=
ot;clocksource&quot; SDP attribute to identify the clock domain.=A0The foll=
owing clock distribution methods are supported by IDMS: NTP, GPS,=A0Galileo=
, IEEE 1588-2002, IEEE 1588-2008 and IEEE 802.1AS.=A0When IEEE 1588-2002 or=
 IEEE 1588-2008 are used, a domain identifier is also specified. The IEEE 8=
02.1AS PTP profile allows use of the default domain (0) only.=A0You are pro=
posing using the EUI-64 of the grandmaster=A0instead. I also considered thi=
s possibility for IDMS but was confounded by the following scenario...</div=
>
<div><br></div><div>For fault tolerance, mission-critical media systems are=
 configured with at least two potential grandmasters all receiving time fro=
m a common reference (e.g. house sync reference, GPS). Upon failure of the =
primary grandmaster, another takes over and, although nothing has functiona=
lly changed about the clock domain, the EUI-64 of the grandmaster in the ne=
twork does change. Until updated SDP advertisements reflecting the new gran=
dmaster are received, senders and receivers will believe they are operating=
 on different time domains and streams will presumably be interrupted.</div=
>
<div><br></div><div>It would be very useful to have a unique identifier for=
 a clock domain. My conclusion after considering the above scenario was tha=
t the grandmaster ID was not that identifier and clock distribution protoco=
l, protocol version and domain identifier is currently the best identifier =
we have. I think there is definitely room for improvement in this. It does =
not strike me as an easy problem, however.</div>
<div><br></div><div>I like your idea of using a traceability=A0indicator to=
 help determine clock compatibility. I am concerned however that while trac=
eable clocks can be assumed to be frequency synchronized (syntonized) I am =
not aware of any way to gauge the accuracy of their phase synchronization. =
Nevertheless, adding a traceability indicator to the SDP clock attribute wo=
uld seem to be a useful improvement.</div>
<div><br></div><div>With regards to the proposed T, M and U flags in the he=
ader extension. I don&#39;t see a need for T if traceability information is=
 to be available via SDP. T is not soemthing that changes dynamically, is i=
t? You do acknowledge that M and U come from IEEE 1722. You should add the =
minimum toggle period requirement we see in IEEE 1722 section 5.4.1 to your=
 M discussion to be sure these events are not missed if the clock is thrash=
ing.</div>
<div><br></div><div>I am not=A0convinced=A0that the information provided by=
 M and U will improve system=A0performance. No specific recommendations for=
 how receivers should respond to these=A0conditions=A0are given in this dra=
ft or in IEEE 1722. Perhaps inclusion of such recommendations or a report f=
rom someone who has implemented IEEE 1722 would change my opinion.</div>
<div><br></div><div>I do support reusing IDMS SDP and RFC 6051 header exten=
sions definitions to meet the requirements of your proposal.=A0I encourage =
you to review and help improve the IDMS draft.=A0I&#39;m not yet convinced =
that the header extension defined in RFC 6051 needs improvement but if it d=
oes, there are already two header options defined there; Perhaps a third co=
uld be added.</div>
<div><br></div><div>Kevin Gross</div><div><br><div class=3D"gmail_quote">On=
 Mon, Oct 31, 2011 at 6:50 PM, Aidan Williams <span dir=3D"ltr">&lt;<a href=
=3D"mailto:aidan.williams@audinate.com">aidan.williams@audinate.com</a>&gt;=
</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;">Hi All,<br>
<br>
I&#39;ve updated the draft on avb sync, you can find it at:<br>
<br>
 =A0<a href=3D"http://tools.ietf.org/html/draft-williams-avtext-avbsync-02"=
 target=3D"_blank">http://tools.ietf.org/html/<u></u>draft-williams-avtext-=
avbsync-<u></u>02</a><br>
<br>
The main changes are:<br>
<br>
 =A0- more discussion about clock domains and traceable time<br>
 =A0- additional header timing metadata<br>
 =A0- some discussion about signalling with SDP<br>
<br>
Section 8 also describes an alternative approach using the existing RFC 605=
1 header extension. =A0I&#39;d appreciate feedback on whether this is more =
desirable approach to take.<br>
<br>
regards<br>
 =A0 =A0 =A0 =A0 =A0 =A0aidan<br>
____<br>
:wq!<br>
<br>
______________________________<u></u>_________________<br>
avtext mailing list<br>
<a href=3D"mailto:avtext@ietf.org" target=3D"_blank">avtext@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/avtext" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/avtext</a><br>
</blockquote></div><br><br>
</div>

--0015174785d24a966704b11d4367--

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From: Roni even <Even.roni@huawei.com>
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Subject: Re: [avtext] Pause and Resume CCM extension
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Hi,
I agree with Mo, the tmmbr=3D0 is probably used by video conferencing appli=
cations to temporary stop video flows. So why use a new method.

Roni

________________________________________
From: avtext-bounces@ietf.org [avtext-bounces@ietf.org] on behalf of Mo Zan=
aty (mzanaty) [mzanaty@cisco.com]
Sent: Saturday, November 05, 2011 5:24
To: Magnus Westerlund; avtext@ietf.org
Subject: Re: [avtext] Pause and Resume CCM extension

Magnus,
How does the proposed pause differ from tmmbr=3D0? Some implementations alr=
eady use tmmbr=3D0 to pause peers.

Assuming there is some valuable nuance between pause and tmmbr=3D0, it may =
be useful to explicitly allow acks without requests so senders can notify p=
eers when they pause (and resume) themselves autonomously. This is not expl=
icitly prohibited in the current draft, just as autonomous tmmbn (without t=
mmbr) is not explicitly prohibited in RFC 5104. But it would be good to exp=
licitly allow and describe this use case. Some implementations already use =
tmmbn=3D0 to notify peers they have paused themselves autonomously (without=
 receiving any tmmbr).

Thanks,
Mo

-----Original Message-----
From: avtext-bounces@ietf.org [mailto:avtext-bounces@ietf.org] On Behalf Of=
 Magnus Westerlund
Sent: Thursday, November 03, 2011 11:10 AM
To: avtext@ietf.org
Subject: [avtext] Pause and Resume CCM extension

Hi,

As individual,

My colleagues and I have put together a proposal for a functionality for
pausing and resuming media streams. This is primarily intended for mixer
scenarios where a mixer would send the pause request to media senders
that currently don't need to send any media for the mixer isn't using
this source. For example video streams when the participant isn't active.

Draft:
https://datatracker.ietf.org/doc/draft-westerlund-avtext-rtp-stream-pause/


It does cover a few other usages but that is primarily to ensure that it
wouldn't break the whole system if introduced into ASM or transport
relay topologies.

This proposal is also based on that the system in general may use
something like the Media Stream Selection (see link below) to include or
exclude media streams they don't want. Thus we have in fact excluded
such usages by not allowing the end-points to use this to manipulate the
CSRC list of media stream that are mixes or selections. This as it would
only provide partial functionality compared to Media Stream Selection.

https://datatracker.ietf.org/doc/draft-westerlund-dispatch-stream-selection=
/

Feedback is most appreciated


Cheers

Magnus Westerlund

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

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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: "Roni even" <Even.roni@huawei.com>, "Magnus Westerlund" <magnus.westerlund@ericsson.com>, <avtext@ietf.org>
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Subject: Re: [avtext] Pause and Resume CCM extension
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Perhaps this should be reworked into an Informational RFC on how to use =
TMMBR/TMMBN for such functionality, since it is not obvious whether or =
not RFC 5104 allows this. Or, if IETF prefers to focus on standards, =
then perhaps other forums which focus on interoperability and =
implementation guidelines (like IMTC, UCIF, etc.) can codify this usage =
in their recommendations.

Thanks,
Mo

-----Original Message-----
From: Roni even [mailto:Even.roni@huawei.com]=20
Sent: Monday, November 07, 2011 6:58 AM
To: Mo Zanaty (mzanaty); Magnus Westerlund; avtext@ietf.org
Subject: RE: [avtext] Pause and Resume CCM extension

Hi,
I agree with Mo, the tmmbr=3D0 is probably used by video conferencing =
applications to temporary stop video flows. So why use a new method.

Roni

________________________________________
From: avtext-bounces@ietf.org [avtext-bounces@ietf.org] on behalf of Mo =
Zanaty (mzanaty) [mzanaty@cisco.com]
Sent: Saturday, November 05, 2011 5:24
To: Magnus Westerlund; avtext@ietf.org
Subject: Re: [avtext] Pause and Resume CCM extension

Magnus,
How does the proposed pause differ from tmmbr=3D0? Some implementations =
already use tmmbr=3D0 to pause peers.

Assuming there is some valuable nuance between pause and tmmbr=3D0, it =
may be useful to explicitly allow acks without requests so senders can =
notify peers when they pause (and resume) themselves autonomously. This =
is not explicitly prohibited in the current draft, just as autonomous =
tmmbn (without tmmbr) is not explicitly prohibited in RFC 5104. But it =
would be good to explicitly allow and describe this use case. Some =
implementations already use tmmbn=3D0 to notify peers they have paused =
themselves autonomously (without receiving any tmmbr).

Thanks,
Mo

-----Original Message-----
From: avtext-bounces@ietf.org [mailto:avtext-bounces@ietf.org] On Behalf =
Of Magnus Westerlund
Sent: Thursday, November 03, 2011 11:10 AM
To: avtext@ietf.org
Subject: [avtext] Pause and Resume CCM extension

Hi,

As individual,

My colleagues and I have put together a proposal for a functionality for
pausing and resuming media streams. This is primarily intended for mixer
scenarios where a mixer would send the pause request to media senders
that currently don't need to send any media for the mixer isn't using
this source. For example video streams when the participant isn't =
active.

Draft:
https://datatracker.ietf.org/doc/draft-westerlund-avtext-rtp-stream-pause=
/


It does cover a few other usages but that is primarily to ensure that it
wouldn't break the whole system if introduced into ASM or transport
relay topologies.

This proposal is also based on that the system in general may use
something like the Media Stream Selection (see link below) to include or
exclude media streams they don't want. Thus we have in fact excluded
such usages by not allowing the end-points to use this to manipulate the
CSRC list of media stream that are mixes or selections. This as it would
only provide partial functionality compared to Media Stream Selection.

https://datatracker.ietf.org/doc/draft-westerlund-dispatch-stream-selecti=
on/

Feedback is most appreciated


Cheers

Magnus Westerlund

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

_______________________________________________
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avtext@ietf.org
https://www.ietf.org/mailman/listinfo/avtext
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From magnus.westerlund@ericsson.com  Tue Nov  8 04:51:01 2011
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Subject: Re: [avtext] Pause and Resume CCM extension
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On 2011-11-07 21:38, Mo Zanaty (mzanaty) wrote:
> Perhaps this should be reworked into an Informational RFC on how to
> use TMMBR/TMMBN for such functionality, since it is not obvious
> whether or not RFC 5104 allows this. Or, if IETF prefers to focus on
> standards, then perhaps other forums which focus on interoperability
> and implementation guidelines (like IMTC, UCIF, etc.) can codify this
> usage in their recommendations.

Hi,

We had considered this but think there are a few issues with using TMMBR=0.

First, the semantics of TMMBR in that it cuts the bit-rate immediately
and only the increase allows other receivers to reconsider the change.
This is problematic in any scenario where you have a shared media, like
ASM or Transport Translators instead of RTP mixer topologies.

If you contrast our the PAUSE proposal with this, it takes the safer but
more complicated way of requiring all known receivers to send pause
before it occurs.

secondly, when resuming the TMMBR semantics is to not increase the
bit-rate until all participants has had the possibility to inject a new
limiation. That is not suitable at all for resume, that needs to happen
immediately.

So we think there is significant semantics issues with TMMBR/TMMBN for
pause and resume.

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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Cc: avtext@ietf.org, "'Botzko, Stephen'" <Stephen.Botzko@polycom.com>
Subject: Re: [avtext] Pause and Resume CCM extension
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Hi Magnus,
The document does not discuss the TMMBR (RFC 5104) option and section 1 =
says
that "No currently existing RTCP feedback message supports pausing and
resuming an incoming data stream."=20

The use cases in sections 3.1 and 3.2 can be addressed by TMMBR and this =
is
what is being used today by shipping products.
Section 4.4 is the only section about multicast and describes the =
behavior
of a sender in a multicast group saying that it shall not pause until =
all
the receivers that the sender knows of have requested it to be paused.  =
I am
not sure that real use cases will cause this to happen.
So the question is why add a new way to do something that is being done
today by TMMBR in the common cases.

Regards
Roni

> -----Original Message-----
> From: avtext-bounces@ietf.org [mailto:avtext-bounces@ietf.org] On
> Behalf Of Magnus Westerlund
> Sent: Tuesday, November 08, 2011 2:51 PM
> To: Mo Zanaty (mzanaty)
> Cc: avtext@ietf.org
> Subject: Re: [avtext] Pause and Resume CCM extension
>=20
> On 2011-11-07 21:38, Mo Zanaty (mzanaty) wrote:
> > Perhaps this should be reworked into an Informational RFC on how to
> > use TMMBR/TMMBN for such functionality, since it is not obvious
> > whether or not RFC 5104 allows this. Or, if IETF prefers to focus on
> > standards, then perhaps other forums which focus on interoperability
> > and implementation guidelines (like IMTC, UCIF, etc.) can codify =
this
> > usage in their recommendations.
>=20
> Hi,
>=20
> We had considered this but think there are a few issues with using
> TMMBR=3D0.
>=20
> First, the semantics of TMMBR in that it cuts the bit-rate immediately
> and only the increase allows other receivers to reconsider the change.
> This is problematic in any scenario where you have a shared media, =
like
> ASM or Transport Translators instead of RTP mixer topologies.
>=20
> If you contrast our the PAUSE proposal with this, it takes the safer
> but
> more complicated way of requiring all known receivers to send pause
> before it occurs.
>=20
> secondly, when resuming the TMMBR semantics is to not increase the
> bit-rate until all participants has had the possibility to inject a =
new
> limiation. That is not suitable at all for resume, that needs to =
happen
> immediately.
>=20
> So we think there is significant semantics issues with TMMBR/TMMBN for
> pause and resume.
>=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
> _______________________________________________
> avtext mailing list
> avtext@ietf.org
> https://www.ietf.org/mailman/listinfo/avtext


From magnus.westerlund@ericsson.com  Wed Nov  9 01:57:13 2011
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Cc: "avtext@ietf.org" <avtext@ietf.org>, "'Botzko, Stephen'" <Stephen.Botzko@polycom.com>
Subject: Re: [avtext] Pause and Resume CCM extension
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On 2011-11-08 17:03, Roni Even wrote:
> Hi Magnus,
> The document does not discuss the TMMBR (RFC 5104) option and section 1 says
> that "No currently existing RTCP feedback message supports pausing and
> resuming an incoming data stream."

Yes, we should have discussed TMMBR and its shortcomming. We will
address this in a future revision.

> 
> The use cases in sections 3.1 and 3.2 can be addressed by TMMBR and this is
> what is being used today by shipping products.
> Section 4.4 is the only section about multicast and describes the behavior
> of a sender in a multicast group saying that it shall not pause until all
> the receivers that the sender knows of have requested it to be paused.  I am
> not sure that real use cases will cause this to happen.
> So the question is why add a new way to do something that is being done
> today by TMMBR in the common cases.

Still the TMMBR semantics do create two issues, even in Point to Point
and RTP Mixer to Media sender use case.

The biggest issue is the resume semantics. TMMBR says the following:

  "If the media sender concludes that it can increase the maximum total
   media bit rate value, it SHALL wait before actually doing so, for a
   period long enough to allow a media receiver to respond to the TMMBN
   if it determines that its tuple belongs in the bounding set.  This
   delay period is estimated by the formula:

      2 * RTT + T_Dither_Max,

   where RTT is the longest round trip time known to the media sender
   and T_Dither_Max is defined in section 3.4 of [RFC4585].  Even in
   point-to-point sessions, a media sender MUST obey the aforementioned
   rule, as it is not guaranteed that a participant is able to determine
   correctly whether all the sources are co-located in a single node,
   and are coordinated."

Based on this a media receiver or mixer sending TMMBR>0 to the media
sender will not get media until two*RTT + T_Dither_Max. That is highly
problematic from our point of view. And if the above behavior should not
be done in the case of TMMBR=0 then it needs to be specified.

In most cases, the session topology a end-point is present in is hard to
determine. For example how does a end-point determine if it is in a
point to point session or a Transport Translator multi-point session
(relay) that currently only has two participants in it. Thus it becomes
hard to determine if it acceptable for the end-point to send a TMMBR=0
as it is the sole receiver, or if other end-points that was interested
of the media still needs it.

If it is a relay based topology a new participant can join at any time.
And if there is an existing TMMBR=0 limitation in place the then the new
participant will not know that this media sender is in fact paused by
another receiver.

So we think these issues with TMMBR semantics that a alternative should
be seriously considered.

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 stephen.botzko@gmail.com  Mon Nov 14 19:59:21 2011
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From: Stephen Botzko <stephen.botzko@gmail.com>
To: Roni even <Even.roni@huawei.com>
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Cc: "avtext@ietf.org" <avtext@ietf.org>
Subject: Re: [avtext] Pause and Resume CCM extension
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Our products support tmmbr; we would rather not see yet-another-mechanism
for pause/resume.

While perhaps the semantics are not perfect, providing a competing
mechanism will not help interoperability.

Stephen Botzko

On Mon, Nov 7, 2011 at 7:58 PM, Roni even <Even.roni@huawei.com> wrote:

> Hi,
> I agree with Mo, the tmmbr=3D0 is probably used by video conferencing
> applications to temporary stop video flows. So why use a new method.
>
> Roni
>
> ________________________________________
> From: avtext-bounces@ietf.org [avtext-bounces@ietf.org] on behalf of Mo
> Zanaty (mzanaty) [mzanaty@cisco.com]
> Sent: Saturday, November 05, 2011 5:24
> To: Magnus Westerlund; avtext@ietf.org
> Subject: Re: [avtext] Pause and Resume CCM extension
>
> Magnus,
> How does the proposed pause differ from tmmbr=3D0? Some implementations
> already use tmmbr=3D0 to pause peers.
>
> Assuming there is some valuable nuance between pause and tmmbr=3D0, it ma=
y
> be useful to explicitly allow acks without requests so senders can notify
> peers when they pause (and resume) themselves autonomously. This is not
> explicitly prohibited in the current draft, just as autonomous tmmbn
> (without tmmbr) is not explicitly prohibited in RFC 5104. But it would be
> good to explicitly allow and describe this use case. Some implementations
> already use tmmbn=3D0 to notify peers they have paused themselves
> autonomously (without receiving any tmmbr).
>
> Thanks,
> Mo
>
> -----Original Message-----
> From: avtext-bounces@ietf.org [mailto:avtext-bounces@ietf.org] On Behalf
> Of Magnus Westerlund
> Sent: Thursday, November 03, 2011 11:10 AM
> To: avtext@ietf.org
> Subject: [avtext] Pause and Resume CCM extension
>
> Hi,
>
> As individual,
>
> My colleagues and I have put together a proposal for a functionality for
> pausing and resuming media streams. This is primarily intended for mixer
> scenarios where a mixer would send the pause request to media senders
> that currently don't need to send any media for the mixer isn't using
> this source. For example video streams when the participant isn't active.
>
> Draft:
> https://datatracker.ietf.org/doc/draft-westerlund-avtext-rtp-stream-pause=
/
>
>
> It does cover a few other usages but that is primarily to ensure that it
> wouldn't break the whole system if introduced into ASM or transport
> relay topologies.
>
> This proposal is also based on that the system in general may use
> something like the Media Stream Selection (see link below) to include or
> exclude media streams they don't want. Thus we have in fact excluded
> such usages by not allowing the end-points to use this to manipulate the
> CSRC list of media stream that are mixes or selections. This as it would
> only provide partial functionality compared to Media Stream Selection.
>
>
> https://datatracker.ietf.org/doc/draft-westerlund-dispatch-stream-selecti=
on/
>
> Feedback is most appreciated
>
>
> 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
> ----------------------------------------------------------------------
>
> _______________________________________________
> avtext mailing list
> avtext@ietf.org
> https://www.ietf.org/mailman/listinfo/avtext
> _______________________________________________
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> _______________________________________________
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>

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Our products support tmmbr; we would rather not see yet-another-mechanism f=
or pause/resume.=A0 <br><br>While perhaps the semantics are not perfect, pr=
oviding a competing mechanism will not help interoperability.<br><br>Stephe=
n Botzko<br>
<br><div class=3D"gmail_quote">On Mon, Nov 7, 2011 at 7:58 PM, Roni even <s=
pan dir=3D"ltr">&lt;<a href=3D"mailto:Even.roni@huawei.com">Even.roni@huawe=
i.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"m=
argin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
Hi,<br>
I agree with Mo, the tmmbr=3D0 is probably used by video conferencing appli=
cations to temporary stop video flows. So why use a new method.<br>
<br>
Roni<br>
<br>
________________________________________<br>
From: <a href=3D"mailto:avtext-bounces@ietf.org">avtext-bounces@ietf.org</a=
> [<a href=3D"mailto:avtext-bounces@ietf.org">avtext-bounces@ietf.org</a>] =
on behalf of Mo Zanaty (mzanaty) [<a href=3D"mailto:mzanaty@cisco.com">mzan=
aty@cisco.com</a>]<br>

Sent: Saturday, November 05, 2011 5:24<br>
To: Magnus Westerlund; <a href=3D"mailto:avtext@ietf.org">avtext@ietf.org</=
a><br>
Subject: Re: [avtext] Pause and Resume CCM extension<br>
<div><div></div><div class=3D"h5"><br>
Magnus,<br>
How does the proposed pause differ from tmmbr=3D0? Some implementations alr=
eady use tmmbr=3D0 to pause peers.<br>
<br>
Assuming there is some valuable nuance between pause and tmmbr=3D0, it may =
be useful to explicitly allow acks without requests so senders can notify p=
eers when they pause (and resume) themselves autonomously. This is not expl=
icitly prohibited in the current draft, just as autonomous tmmbn (without t=
mmbr) is not explicitly prohibited in RFC 5104. But it would be good to exp=
licitly allow and describe this use case. Some implementations already use =
tmmbn=3D0 to notify peers they have paused themselves autonomously (without=
 receiving any tmmbr).<br>

<br>
Thanks,<br>
Mo<br>
<br>
-----Original Message-----<br>
From: <a href=3D"mailto:avtext-bounces@ietf.org">avtext-bounces@ietf.org</a=
> [mailto:<a href=3D"mailto:avtext-bounces@ietf.org">avtext-bounces@ietf.or=
g</a>] On Behalf Of Magnus Westerlund<br>
Sent: Thursday, November 03, 2011 11:10 AM<br>
To: <a href=3D"mailto:avtext@ietf.org">avtext@ietf.org</a><br>
Subject: [avtext] Pause and Resume CCM extension<br>
<br>
Hi,<br>
<br>
As individual,<br>
<br>
My colleagues and I have put together a proposal for a functionality for<br=
>
pausing and resuming media streams. This is primarily intended for mixer<br=
>
scenarios where a mixer would send the pause request to media senders<br>
that currently don&#39;t need to send any media for the mixer isn&#39;t usi=
ng<br>
this source. For example video streams when the participant isn&#39;t activ=
e.<br>
<br>
Draft:<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-westerlund-avtext-rtp-str=
eam-pause/" target=3D"_blank">https://datatracker.ietf.org/doc/draft-wester=
lund-avtext-rtp-stream-pause/</a><br>
<br>
<br>
It does cover a few other usages but that is primarily to ensure that it<br=
>
wouldn&#39;t break the whole system if introduced into ASM or transport<br>
relay topologies.<br>
<br>
This proposal is also based on that the system in general may use<br>
something like the Media Stream Selection (see link below) to include or<br=
>
exclude media streams they don&#39;t want. Thus we have in fact excluded<br=
>
such usages by not allowing the end-points to use this to manipulate the<br=
>
CSRC list of media stream that are mixes or selections. This as it would<br=
>
only provide partial functionality compared to Media Stream Selection.<br>
<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-westerlund-dispatch-strea=
m-selection/" target=3D"_blank">https://datatracker.ietf.org/doc/draft-west=
erlund-dispatch-stream-selection/</a><br>
<br>
Feedback is most appreciated<br>
<br>
<br>
Cheers<br>
<br>
Magnus Westerlund<br>
<br>
----------------------------------------------------------------------<br>
Multimedia Technologies, Ericsson Research EAB/TVM<br>
----------------------------------------------------------------------<br>
Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0<a href=3D"tel:%2B46%=
2010%207148287" value=3D"+46107148287">+46 10 7148287</a><br>
F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile <a href=3D"tel:%2B4=
6%2073%200949079" value=3D"+46730949079">+46 73 0949079</a><br>
SE-164 80 Stockholm, Sweden| mailto: <a href=3D"mailto:magnus.westerlund@er=
icsson.com">magnus.westerlund@ericsson.com</a><br>
----------------------------------------------------------------------<br>
<br>
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Subject: [avtext] I-D Action: draft-ietf-avtext-splicing-for-rtp-02.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 Extensions Work=
ing Group of the IETF.

	Title           : Content Splicing for RTP Sessions
	Author(s)       : Jinwei Xia
	Filename        : draft-ietf-avtext-splicing-for-rtp-02.txt
	Pages           : 16
	Date            : 2011-11-15

   This memo outlines RTP splicing.  Splicing is a process that replaces
   the content of the main multimedia stream with other multimedia
   content, and delivers the substitutive multimedia content to receiver
   for a period of time.  This memo provides some RTP splicing use
   cases, then we enumerate a set of requirements and analyze whether an
   existing RTP level middlebox can meet these requirements, at last we
   provide concrete guidelines for how the chosen middlebox works to
   handle RTP splicing.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avtext-splicing-for-rtp-02.t=
xt

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-avtext-splicing-for-rtp-02.txt

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From Even.roni@huawei.com  Tue Nov 15 21:51:27 2011
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Date: Wed, 16 Nov 2011 07:47:08 +0200
From: Roni Even <Even.roni@huawei.com>
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Hi,

I read the draft and I am not sure I fully understand the mechanism so there
are clarification questions

 

Looking at the examples in section 5.1 through 5.4 it looks to me that the
offer has one RTP payload type number that is being used to send two video
streams with the same parameters. I was expecting to see two payload type
numbers that will allow separate a=fmtp lines for each of the streams.
Having both with the same functionality is limiting. Looking at 5.1 I see
for example two streams (SSRCs 192392452 and 834753488)having mid=2 with the
same capability so I assume that they may come from two cameras for example.
How do you know which is from which camera, will there be another a:ssrc
parameter. It is also limited since it assumes that both have the same video
capabilities

 

I am also not sure how does it work without the SDP.  If a receiver receives
two SDES for two SSRCs with the same srcname how does he know which one is
each without getting the mapping from the SDP, like in 5.1 how does a
receiver know if this is the mid=2 or 3 stream, is it based on RTP session
multiplexing. From the current text I get it this is intended for SSRC
multiplexing but to associate streams that can be identified either by the
RTP session or by the payload like in 5.3.

 

Roni


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<o:idmap v:ext="edit" data="1" />
</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 read the draft and I am not sure I fully understand the mechanism so there are clarification questions<o:p></o:p></p><p class=MsoNormal><o:p>&nbsp;</o:p></p><p class=MsoNormal>Looking at the examples in section 5.1 through 5.4 it looks to me that the offer has one RTP payload type number that is being used to send two video streams with the same parameters. I was expecting to see two payload type numbers that will allow separate a=fmtp lines for each of the streams. Having both with the same functionality is limiting. Looking at 5.1 I see for example two streams (SSRCs 192392452 and 834753488<span style='font-size:10.0pt;font-family:Courier'>)</span>having mid=2 with the same capability so I assume that they may come from two cameras for example. How do you know which is from which camera, will there be another a:ss
 rc param
since it assumes that both have the same video capabilities<o:p></o:p></p><p class=MsoNormal><o:p>&nbsp;</o:p></p><p class=MsoNormal>I am also not sure how does it work without the SDP. &nbsp;If a receiver receives two SDES for two SSRCs with the same srcname how does he know which one is each without getting the mapping from the SDP, like in 5.1 how does a receiver know if this is the mid=2 or 3 stream, is it based on RTP session multiplexing. From the current text I get it this is intended for SSRC multiplexing but to associate streams that can be identified either by the RTP session or by the payload like in 5.3.<o:p></o:p></p><p class=MsoNormal><o:p>&nbsp;</o:p></p><p class=MsoNormal>Roni<o:p></o:p></p></div></body></html>

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From magnus.westerlund@ericsson.com  Wed Nov 16 01:11:43 2011
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WG,

In todays WG meeting I announced the start of a 3 week WG last call that
ends on the 7th of December. The WG last call is for "Content Splicing
for RTP Sessions"
https://datatracker.ietf.org/doc/draft-ietf-avtext-splicing-for-rtp/
with the intended status of Informational RFC

Please review and provide comments, even if it is just to say I read it
and I found no issues with the document.

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: [avtext] I-D Action: draft-ietf-avtext-client-to-mixer-audio-level-06.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 Extensions Work=
ing Group of the IETF.

	Title           : A Real-Time Transport Protocol (RTP) Header Extension fo=
r Client-to- Mixer Audio Level Indication
	Author(s)       : Jonathan Lennox
                          Emil Ivov
                          Enrico Marocco
	Filename        : draft-ietf-avtext-client-to-mixer-audio-level-06.txt
	Pages           : 11
	Date            : 2011-11-16

   This document defines a mechanism by which packets of Real-Time
   Transport Protocol (RTP) audio streams can indicate, in an RTP header
   extension, the audio level of the audio sample carried in the RTP
   packet.  In large conferences, this can reduce the load on an audio
   mixer or other middlebox which wants to forward only a few of the
   loudest audio streams, without requiring it to decode and measure
   every stream that is received.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avtext-client-to-mixer-audio=
-level-06.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-avtext-client-to-mixer-audio-=
level-06.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 Extensions Work=
ing Group of the IETF.

	Title           : A Real-Time Transport Protocol (RTP) Header Extension fo=
r Mixer-to- Client Audio Level Indication
	Author(s)       : Emil Ivov
                          Enrico Marocco
                          Jonathan Lennox
	Filename        : draft-ietf-avtext-mixer-to-client-audio-level-06.txt
	Pages           : 17
	Date            : 2011-11-16

   This document describes a mechanism for RTP-level mixers in audio
   conferences to deliver information about the audio level of
   individual participants.  Such audio level indicators are transported
   in the same RTP packets as the audio data they pertain to.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avtext-mixer-to-client-audio=
-level-06.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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level-06.txt


From jonathan@vidyo.com  Wed Nov 16 03:31:30 2011
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To: "internet-drafts@ietf.org" <internet-drafts@ietf.org>, "i-d-announce@ietf.org" <i-d-announce@ietf.org>
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I've just submitted new revisions of the two audio-levels drafts.  These sh=
ould address all comments received during IETF Last Call and IESG review (i=
ncluding one comment received privately).

See the drafts' changes sections for what's changed, or the diff2 available=
 here:
http://tools.ietf.org/rfcdiff?url2=3Ddraft-ietf-avtext-client-to-mixer-audi=
o-level-06.txt
http://tools.ietf.org/rfcdiff?url2=3Ddraft-ietf-avtext-mixer-to-client-audi=
o-level-06.txt

-----Original Message-----
From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]=20
Sent: Wednesday, November 16, 2011 7:04 PM
To: i-d-announce@ietf.org
Cc: avtext@ietf.org
Subject: [avtext] I-D Action: draft-ietf-avtext-client-to-mixer-audio-level=
-06.txt


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 Extensions Work=
ing Group of the IETF.

	Title           : A Real-Time Transport Protocol (RTP) Header Extension fo=
r Client-to- Mixer Audio Level Indication
	Author(s)       : Jonathan Lennox
                          Emil Ivov
                          Enrico Marocco
	Filename        : draft-ietf-avtext-client-to-mixer-audio-level-06.txt
	Pages           : 11
	Date            : 2011-11-16

   This document defines a mechanism by which packets of Real-Time
   Transport Protocol (RTP) audio streams can indicate, in an RTP header
   extension, the audio level of the audio sample carried in the RTP
   packet.  In large conferences, this can reduce the load on an audio
   mixer or other middlebox which wants to forward only a few of the
   loudest audio streams, without requiring it to decode and measure
   every stream that is received.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avtext-client-to-mixer-audio=
-level-06.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-avtext-client-to-mixer-audio-=
level-06.txt

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


From keith.drage@alcatel-lucent.com  Wed Nov 16 08:54:02 2011
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Jonathan Lennox <jonathan@vidyo.com>, "internet-drafts@ietf.org" <internet-drafts@ietf.org>, "i-d-announce@ietf.org" <i-d-announce@ietf.org>
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(as WG co chair)

The intention of the editors is that this completes all the open discusses =
and comments from the IESG.

You can find these discusses and comments at:

https://datatracker.ietf.org/doc/draft-ietf-avtext-client-to-mixer-audio-le=
vel/ballot/

https://datatracker.ietf.org/doc/draft-ietf-avtext-mixer-to-client-audio-le=
vel/ballot/

The situation at them moment is that one IESG member on each document has t=
o identify if their discusses are resolved, and out Area Director (Gonzalo)=
 will verify if all the comments have been handled in his view.

That could happen very quickly or may take a few days.

If members of the working group want to express anything on the document or=
 the comments, you have a small window for doing this, but it is very small=
. My assumption is that none of the changes require further working group c=
onsensus and that the AD will assume likewise.

Regards

Keith



> -----Original Message-----
> From: avtext-bounces@ietf.org [mailto:avtext-bounces@ietf.org] On Behalf
> Of Jonathan Lennox
> Sent: 16 November 2011 11:31
> To: internet-drafts@ietf.org; i-d-announce@ietf.org
> Cc: avtext@ietf.org
> Subject: Re: [avtext] I-D Action: draft-ietf-avtext-client-to-mixer-audio=
-
> level-06.txt
>=20
> I've just submitted new revisions of the two audio-levels drafts.  These
> should address all comments received during IETF Last Call and IESG revie=
w
> (including one comment received privately).
>=20
> See the drafts' changes sections for what's changed, or the diff2
> available here:
> http://tools.ietf.org/rfcdiff?url2=3Ddraft-ietf-avtext-client-to-mixer-
> audio-level-06.txt
> http://tools.ietf.org/rfcdiff?url2=3Ddraft-ietf-avtext-mixer-to-client-
> audio-level-06.txt
>=20
> -----Original Message-----
> From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]
> Sent: Wednesday, November 16, 2011 7:04 PM
> To: i-d-announce@ietf.org
> Cc: avtext@ietf.org
> Subject: [avtext] I-D Action: draft-ietf-avtext-client-to-mixer-audio-
> level-06.txt
>=20
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories. This draft is a work item of the Audio/Video Transport
> Extensions Working Group of the IETF.
>=20
> 	Title           : A Real-Time Transport Protocol (RTP) Header
> Extension for Client-to- Mixer Audio Level Indication
> 	Author(s)       : Jonathan Lennox
>                           Emil Ivov
>                           Enrico Marocco
> 	Filename        : draft-ietf-avtext-client-to-mixer-audio-level-
> 06.txt
> 	Pages           : 11
> 	Date            : 2011-11-16
>=20
>    This document defines a mechanism by which packets of Real-Time
>    Transport Protocol (RTP) audio streams can indicate, in an RTP header
>    extension, the audio level of the audio sample carried in the RTP
>    packet.  In large conferences, this can reduce the load on an audio
>    mixer or other middlebox which wants to forward only a few of the
>    loudest audio streams, without requiring it to decode and measure
>    every stream that is received.
>=20
>=20
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-avtext-client-to-mixer-
> audio-level-06.txt
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> This Internet-Draft can be retrieved at:
> ftp://ftp.ietf.org/internet-drafts/draft-ietf-avtext-client-to-mixer-
> audio-level-06.txt
>=20
> _______________________________________________
> avtext mailing list
> avtext@ietf.org
> https://www.ietf.org/mailman/listinfo/avtext
>=20
> _______________________________________________
> avtext mailing list
> avtext@ietf.org
> https://www.ietf.org/mailman/listinfo/avtext

From magnus.westerlund@ericsson.com  Wed Nov 16 18:30:55 2011
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Subject: Re: [avtext] I-D	Action: draft-ietf-avtext-client-to-mixer-audio-level-06.txt
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On 2011-11-17 00:53, DRAGE, Keith (Keith) wrote:
> https://datatracker.ietf.org/doc/draft-ietf-avtext-client-to-mixer-audio-level/ballot/
>
> If members of the working group want to express anything on the
> document or the comments, you have a small window for doing this, but
> it is very small. My assumption is that none of the changes require
> further working group consensus and that the AD will assume
> likewise.

WG,

I would like to point out that in client to mixer document there has
been a change of SHOULD to MUST on encrypting the header extension when
confidentiality is required.

Cheers

Magnus Westerlund

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From iesg-secretary@ietf.org  Wed Nov 16 23:15:26 2011
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Subject: [avtext] Protocol Action: 'A Real-Time Transport Protocol (RTP) Header	Extension for Client-to- Mixer Audio Level Indication' to	Proposed Standard	(draft-ietf-avtext-client-to-mixer-audio-level-06.txt)
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The IESG has approved the following document:
- 'A Real-Time Transport Protocol (RTP) Header Extension for Client-to-
   Mixer Audio Level Indication'
  (draft-ietf-avtext-client-to-mixer-audio-level-06.txt) as a Proposed
Standard

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

The IESG contact persons are Gonzalo Camarillo and Robert Sparks.

A URL of this Internet Draft is:
http://datatracker.ietf.org/doc/draft-ietf-avtext-client-to-mixer-audio-level/




Technical Summary

  This document defines a mechanism by which packets of Real-Time
  Transport Protocol (RTP) audio streams can indicate, in an RTP header
  extension, the audio level of the audio sample carried in the RTP
  packet.  In large conferences, this can reduce the load on an audio
  mixer or other middlebox which wants to forward only a few of the
  loudest audio streams, without requiring it to decode and measure
  every stream that is received.

Working Group Summary

  The document is a product of the AVTEXT working group.

Document Quality

  Vidyo has a working implementation of this internet-draft. 

Personnel

  Keith Drage is the document's shepherd.
  Gonzalo Camarillo is the responsible AD.

RFC Editor Note

  In the Security Considerations Section:
    s/relys/relies/

From iesg-secretary@ietf.org  Wed Nov 16 23:16:52 2011
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Subject: [avtext] Protocol Action: 'A Real-Time Transport Protocol (RTP) Header	Extension for Mixer-to- Client Audio Level Indication' to	Proposed Standard	(draft-ietf-avtext-mixer-to-client-audio-level-06.txt)
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The IESG has approved the following document:
- 'A Real-Time Transport Protocol (RTP) Header Extension for Mixer-to-
   Client Audio Level Indication'
  (draft-ietf-avtext-mixer-to-client-audio-level-06.txt) as a Proposed
Standard

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

The IESG contact persons are Gonzalo Camarillo and Robert Sparks.

A URL of this Internet Draft is:
http://datatracker.ietf.org/doc/draft-ietf-avtext-mixer-to-client-audio-level/




Technical Summary

This document describes a mechanism for RTP-level mixers in audio
conferences to deliver information about the audio level of
individual participants.  Such audio level indicators are transported
in the same RTP packets as the audio data they pertain to.

Working Group Summary

The document is a product of the AVTEXT working group.

Document Quality

Jitsu has a working and deployed implementation of this internet-
draft. 

Personnel

Keith Drage is the document's shepherd.
Gonzalo Camarillo is the responsible AD.

From magnus.westerlund@ericsson.com  Thu Nov 17 18:37:05 2011
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Subject: [avtext] IPR disclosures on draft-westerlund-avtext-rtcp-sdes-srcname-00 and draft-westerlund-avtext-rtp-stream-pause-00
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WG,

I have realized that I forgot to be explicit about the IPR disclosures
on the drafts in the presentations. I am sorry about this, please review
these disclosures.

RTP Media Stream Pause and Resume
draft-westerlund-avtext-rtp-stream-pause-00
IPR Disclosure:
https://datatracker.ietf.org/ipr/1641/

RTCP SDES Item SRCNAME to Label Individual Sources
draft-westerlund-avtext-rtcp-sdes-srcname-00
IPR Disclosure
https://datatracker.ietf.org/ipr/1638/

Cheers

Magnus Westerlund

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From keith.drage@alcatel-lucent.com  Tue Nov 22 09:50:53 2011
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
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(As WG cochair)

This is to announce a working group last call for draft-ietf-avtext-rams-sc=
enarios-01

https://datatracker.ietf.org/doc/draft-ietf-avtext-rams-scenarios/

Please respond to the list by December 13th 2011 (i.e. 3 weeks time) with a=
ny comments.

This is intended as an informational RFC.

It is helpful to attempt to categorise the type of your comment (including =
severity) and also to assist to provide any replacement text you feel is ne=
cessary.

If you review the document and have no comments, please tell the chairs tha=
t you have reviewed it. This is always useful information in assessing the =
degree of WG review and consensus behind the document.

Regards

Keith


From kevin.gross@avanw.com  Tue Nov 29 14:07:21 2011
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--0015175d039441e9bc04b2e6d76a
Content-Type: text/plain; charset=ISO-8859-1

Aidan and I ended up at the same conference in San Diego last weekend and
found time to discuss clock domain identification. We believe it
is desirable to use the same clock identification scheme for IDMS and
Aidan's work. Here's a proposal I've fleshed out based on our discussion.

The "clocksource" parameter should be improved to allow optional
specification of clock identifiers. Clock identifiers are derived from an
EUI-48 or EUI-64 as specified by IEEE 1588 (IEEE 1588-2008 7.5.2.2.2, IEEE
1588-2002 D.2). The listed identifiers would be for the current grandmaster
and other high-quality clocks on the IEEE 1288 domain, members of a
grandmaster cluster (IEEE 1588-2008 17.3) or an alternate master (IEEE
1588-2008 17.4).

If no clock identifiers are supplied, implementations may assume that a
match of the supplied attributes (e.g. PTP, version and domain identifier)
specifies the same clock. If one or more clock identifiers are supplied,
these should be compared to the available clock identifiers for the
specified domain. If there is at least one match, there is assumed to be a
clock match.

We both believe a traceability indication should be included in the
"clocksource" attribute for IEEE 1588-2002, IEEE 1588-2008, IEEE 802.1AS
and NTP. Galileo and GPS are implied traceable and local is presumably
never traceable. Aidan argues that  you shouldn't have to second guess
a traceable clock and therefore, no domain or clock identifiers are
required for traceable clocks.

>
Kevin Gross



> On Sun, Nov 6, 2011 at 9:12 PM, Kevin Gross <kevin.gross@avanw.com> wrote:

>
> IDMS defines a "clocksource" SDP attribute to identify the clock
> domain. The following clock distribution methods are supported by IDMS:
> NTP, GPS, Galileo, IEEE 1588-2002, IEEE 1588-2008 and IEEE 802.1AS. When
> IEEE 1588-2002 or IEEE 1588-2008 are used, a domain identifier is also
> specified. The IEEE 802.1AS PTP profile allows use of the default domain
> (0) only. You are proposing using the EUI-64 of the grandmaster instead. I
> also considered this possibility for IDMS but was confounded by the
> following scenario...
>
> For fault tolerance, mission-critical media systems are configured with at
> least two potential grandmasters all receiving time from a common reference
> (e.g. house sync reference, GPS). Upon failure of the primary grandmaster,
> another takes over and, although nothing has functionally changed about the
> clock domain, the EUI-64 of the grandmaster in the network does change.
> Until updated SDP advertisements reflecting the new grandmaster are
> received, senders and receivers will believe they are operating on
> different time domains and streams will presumably be interrupted.
>
> It would be very useful to have a unique identifier for a clock domain. My
> conclusion after considering the above scenario was that the grandmaster ID
> was not that identifier and clock distribution protocol, protocol version
> and domain identifier is currently the best identifier we have. I think
> there is definitely room for improvement in this. It does not strike me as
> an easy problem, however.
>
> I like your idea of using a traceability indicator to help determine clock
> compatibility. I am concerned however that while traceable clocks can be
> assumed to be frequency synchronized (syntonized) I am not aware of any way
> to gauge the accuracy of their phase synchronization. Nevertheless, adding
> a traceability indicator to the SDP clock attribute would seem to be a
> useful improvement.
>
>

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Content-Type: text/html; charset=ISO-8859-1
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Aidan and I ended up at the same conference in San Diego last weekend and f=
ound time to discuss clock domain identification. We=A0believe=A0it is=A0de=
sirable=A0to use the same clock identification scheme for IDMS and Aidan&#3=
9;s work. Here&#39;s a proposal I&#39;ve fleshed out based on our discussio=
n.<br>





<br><div>The &quot;clocksource&quot; parameter should be improved to allow =
optional specification of clock identifiers. Clock identifiers are derived =
from an EUI-48 or EUI-64 as specified by IEEE 1588 (IEEE 1588-2008 7.5.2.2.=
2, IEEE 1588-2002 D.2). The listed identifiers would be for the current gra=
ndmaster and other high-quality clocks on the IEEE 1288 domain, members of =
a grandmaster cluster (IEEE 1588-2008 17.3) or an alternate master (IEEE 15=
88-2008 17.4).</div>




<div><br></div><div>If no clock identifiers are supplied, implementations m=
ay assume that a match of the supplied attributes (e.g. PTP, version and do=
main identifier) specifies the same clock. If one or more clock identifiers=
 are supplied, these should be compared to the available clock identifiers =
for the specified domain. If there is at least one match, there is assumed =
to be a clock match.</div>





<div><br></div><div><div>We both=A0believe=A0a traceability indication shou=
ld be included in the &quot;clocksource&quot; attribute for IEEE 1588-2002,=
 IEEE 1588-2008, IEEE 802.1AS and NTP. Galileo=A0and GPS are implied=A0trac=
eable=A0and local is presumably never traceable.=A0Aidan argues that =A0you=
 shouldn&#39;t have to second guess a=A0traceable=A0clock and=A0therefore, =
no domain or clock identifiers are required for traceable clocks.</div>



</div><div>

<blockquote class=3D"gmail_quote" style=3D"margin-top:0px;margin-right:0px;=
margin-bottom:0px;margin-left:0.8ex;border-left-width:1px;border-left-color=
:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
</blockquote></div><div><br></div><div>Kevin Gross</div><div><br></div><div=
>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin-top:0px;margin-=
right:0px;margin-bottom:0px;margin-left:0.8ex;border-left-width:1px;border-=
left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">





</blockquote><div class=3D"gmail_quote">On Sun, Nov 6, 2011 at 9:12 PM, Kev=
in Gross <span dir=3D"ltr">&lt;<a href=3D"mailto:kevin.gross@avanw.com" tar=
get=3D"_blank">kevin.gross@avanw.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div><br></div><div>IDMS defines a &quot;clo=
cksource&quot; SDP attribute to identify the clock domain.=A0The following =
clock distribution methods are supported by IDMS: NTP, GPS,=A0Galileo, IEEE=
 1588-2002, IEEE 1588-2008 and IEEE 802.1AS.=A0When IEEE 1588-2002 or IEEE =
1588-2008 are used, a domain identifier is also specified. The IEEE 802.1AS=
 PTP profile allows use of the default domain (0) only.=A0You are proposing=
 using the EUI-64 of the grandmaster=A0instead. I also considered this poss=
ibility for IDMS but was confounded by the following scenario...</div>








<div><br></div><div>For fault tolerance, mission-critical media systems are=
 configured with at least two potential grandmasters all receiving time fro=
m a common reference (e.g. house sync reference, GPS). Upon failure of the =
primary grandmaster, another takes over and, although nothing has functiona=
lly changed about the clock domain, the EUI-64 of the grandmaster in the ne=
twork does change. Until updated SDP advertisements reflecting the new gran=
dmaster are received, senders and receivers will believe they are operating=
 on different time domains and streams will presumably be interrupted.</div=
>








<div><br></div><div>It would be very useful to have a unique identifier for=
 a clock domain. My conclusion after considering the above scenario was tha=
t the grandmaster ID was not that identifier and clock distribution protoco=
l, protocol version and domain identifier is currently the best identifier =
we have. I think there is definitely room for improvement in this. It does =
not strike me as an easy problem, however.</div>








<div><br></div></blockquote><blockquote class=3D"gmail_quote" style=3D"marg=
in:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div>I like your=
 idea of using a traceability=A0indicator to help determine clock compatibi=
lity. I am concerned however that while traceable clocks can be assumed to =
be frequency synchronized (syntonized) I am not aware of any way to gauge t=
he accuracy of their phase synchronization. Nevertheless, adding a traceabi=
lity indicator to the SDP clock attribute would seem to be a useful improve=
ment.</div>








<div><br></div></blockquote><div>=A0</div></div>

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