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From: Colin Perkins <csp@csperkins.org>
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To: Ali C. Begen (abegen) <abegen@cisco.com>
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Subject: Re: [AVT] FW: New Version Notification for draft-ietf-avt-ports-for-ucast-mcast-rtp-10
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Ali,

On 21 Dec 2010, at 07:10, Ali C. Begen (abegen) wrote:
> This revision includes the changes we agreed on in the last week's =
call and after the feedback on the list. It also addresses Glen's =
comments, makes some clarifications here and there.


This is looking good. Some minor comments only:

- Section 3:  =93Therefore, the server will need the client to also =
include the Token along with the RTCP messages that are different from =
the RTCP message that triggerred the unicast session establishment.=94 - =
I don=92t understand what this means. Can you clarify in the draft?

- Section 4.1: =93Here, the SSRC value is not necessarily linked to the =
one used by the client in the multicast session.=94 - what do you mean =
by =93not necessarily=94 here? They=92re either part of the same RTP =
session, in which case the client should use the same SSRC; or they=92re =
not, in which case the client should choose its SSRC randomly.

- Section 4.2: The addition of the packet types element is good. =
However, for clarify, it would be useful to include a packet diagram =
showing the format of such an element with enough entries to fill more =
than one 32 bit block and require padding to the next.

- Figure 6: =93PT of request=94 - the RTCP packet that causes this =
respond to be generated might not be a request to the server. This field =
might be clearer called =93Failed PT=94?

I didn't look at the SDP sections, sorry.

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




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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Colin Perkins" <csp@csperkins.org>
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Subject: Re: [AVT] FW: New Version Notification for draft-ietf-avt-ports-for-ucast-mcast-rtp-10
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Hi Colin,

> -----Original Message-----
> From: Colin Perkins [mailto:csp@csperkins.org]
> Sent: Monday, January 03, 2011 9:22 AM
> To: Ali C. Begen (abegen)
> Cc: avt@ietf.org
> Subject: Re: [AVT] FW: New Version Notification for =
draft-ietf-avt-ports-for-ucast-mcast-rtp-10
>=20
> Ali,
>=20
> On 21 Dec 2010, at 07:10, Ali C. Begen (abegen) wrote:
> > This revision includes the changes we agreed on in the last week's =
call and after the feedback on the list. It also addresses
> Glen's comments, makes some clarifications here and there.
>=20
>=20
> This is looking good. Some minor comments only:
>=20
> - Section 3:  =93Therefore, the server will need the client to also =
include the Token along with the RTCP messages that are
> different from the RTCP message that triggerred the unicast session =
establishment.=94 - I don=92t understand what this means.
> Can you clarify in the draft?

Certain messages don=92t establish a unicast session but control it, and =
they can be required to be bundled with a token as well as described in =
4.3.1.
=20
> - Section 4.1: =93Here, the SSRC value is not necessarily linked to =
the one used by the client in the multicast session.=94 - what do
> you mean by =93not necessarily=94 here? They=92re either part of the =
same RTP session, in which case the client should use the
> same SSRC; or they=92re not, in which case the client should choose =
its SSRC randomly.

When requesting a Token, the SSRC used or chosen by the client does not =
matter. Think about a client requesting a token during booting. It can =
use that token with any ssm session later on. We could say "Here the =
SSRC value can be chosen randomly by the client since the Port Mapping =
Request message is not necessarily linked to a specific multicast =
session."

Is this better?
=20
> - Section 4.2: The addition of the packet types element is good. =
However, for clarify, it would be useful to include a packet
> diagram showing the format of such an element with enough entries to =
fill more than one 32 bit block and require padding
> to the next.

OK, I will give it a shot.
=20
> - Figure 6: =93PT of request=94 - the RTCP packet that causes this =
respond to be generated might not be a request to the server.
> This field might be clearer called =93Failed PT=94?

Sounds good.
=20
> I didn't look at the SDP sections, sorry.

Thanks anyway.
-acbegen
=20
> --
> Colin Perkins
> http://csperkins.org/
>=20
>=20


From csp@csperkins.org  Mon Jan  3 08:38:50 2011
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To: Ali C. Begen (abegen) <abegen@cisco.com>
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Subject: Re: [AVT] FW: New Version Notification for draft-ietf-avt-ports-for-ucast-mcast-rtp-10
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Hi Ali,

On 3 Jan 2011, at 16:24, Ali C. Begen (abegen) wrote:
> Hi Colin,
>=20
>> -----Original Message-----
>> From: Colin Perkins [mailto:csp@csperkins.org]
>> Sent: Monday, January 03, 2011 9:22 AM
>> To: Ali C. Begen (abegen)
>> Cc: avt@ietf.org
>> Subject: Re: [AVT] FW: New Version Notification for =
draft-ietf-avt-ports-for-ucast-mcast-rtp-10
>>=20
>> Ali,
>>=20
>> On 21 Dec 2010, at 07:10, Ali C. Begen (abegen) wrote:
>>> This revision includes the changes we agreed on in the last week's =
call and after the feedback on the list. It also addresses
>> Glen's comments, makes some clarifications here and there.
>>=20
>>=20
>> This is looking good. Some minor comments only:
>>=20
>> - Section 3:  =93Therefore, the server will need the client to also =
include the Token along with the RTCP messages that are
>> different from the RTCP message that triggerred the unicast session =
establishment.=94 - I don=92t understand what this means.
>> Can you clarify in the draft?
>=20
> Certain messages don=92t establish a unicast session but control it, =
and they can be required to be bundled with a token as well as described =
in 4.3.1.

Sure, I understand that. But the text in the draft is unclear, and needs =
updating.

>> - Section 4.1: =93Here, the SSRC value is not necessarily linked to =
the one used by the client in the multicast session.=94 - what do
>> you mean by =93not necessarily=94 here? They=92re either part of the =
same RTP session, in which case the client should use the
>> same SSRC; or they=92re not, in which case the client should choose =
its SSRC randomly.
>=20
> When requesting a Token, the SSRC used or chosen by the client does =
not matter. Think about a client requesting a token during booting. It =
can use that token with any ssm session later on. We could say "Here the =
SSRC value can be chosen randomly by the client since the Port Mapping =
Request message is not necessarily linked to a specific multicast =
session."
>=20
> Is this better?

Not really, since it's just as vague. Saying the SSRC "can" be chosen =
randomly doesn't help: either it is chosen randomly, or it isn't. Which =
is it?

>> - Section 4.2: The addition of the packet types element is good. =
However, for clarify, it would be useful to include a packet
>> diagram showing the format of such an element with enough entries to =
fill more than one 32 bit block and require padding
>> to the next.
>=20
> OK, I will give it a shot.
>=20
>> - Figure 6: =93PT of request=94 - the RTCP packet that causes this =
respond to be generated might not be a request to the server.
>> This field might be clearer called =93Failed PT=94?
>=20
> Sounds good.
>=20
>> I didn't look at the SDP sections, sorry.
>=20
> Thanks anyway.
> -acbegen
>=20
>> --
>> Colin Perkins
>> http://csperkins.org/
>>=20
>>=20
>=20
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt



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




From abegen@cisco.com  Mon Jan  3 08:47:45 2011
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Colin Perkins" <csp@csperkins.org>
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Subject: Re: [AVT] FW: New Version Notification for draft-ietf-avt-ports-for-ucast-mcast-rtp-10
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> >> This is looking good. Some minor comments only:
> >>
> >> - Section 3:  =93Therefore, the server will need the client to also =
include the Token along with the RTCP messages that are
> >> different from the RTCP message that triggerred the unicast session =
establishment.=94 - I don=92t understand what this means.
> >> Can you clarify in the draft?
> >
> > Certain messages don=92t establish a unicast session but control it, =
and they can be required to be bundled with a token as
> well as described in 4.3.1.
>=20
> Sure, I understand that. But the text in the draft is unclear, and =
needs updating.

What about:

   During the unicast session lifetime, client's certain actions and
   messages sent by the client might need to be authorized by the server =
by
   requiring a valid Token.  Therefore, the client needs
   to also include the Token along with such RTCP messages.=20
   These are explained later in this document.
=20
> >> - Section 4.1: =93Here, the SSRC value is not necessarily linked to =
the one used by the client in the multicast session.=94 - what
> do
> >> you mean by =93not necessarily=94 here? They=92re either part of =
the same RTP session, in which case the client should use the
> >> same SSRC; or they=92re not, in which case the client should choose =
its SSRC randomly.
> >
> > When requesting a Token, the SSRC used or chosen by the client does =
not matter. Think about a client requesting a token
> during booting. It can use that token with any ssm session later on. =
We could say "Here the SSRC value can be chosen
> randomly by the client since the Port Mapping Request message is not =
necessarily linked to a specific multicast session."
> >
> > Is this better?
>=20
> Not really, since it's just as vague. Saying the SSRC "can" be chosen =
randomly doesn't help: either it is chosen randomly, or it
> isn't. Which is it?

"Is chosen" then. I was reluctant to say this explicitly since a client =
who asks for a token right on the instant it needed one could still use =
the SSRC from the multicast session if it wants to. But, doing so will =
not break anything.

-acbegen

=20
> >> - Section 4.2: The addition of the packet types element is good. =
However, for clarify, it would be useful to include a packet
> >> diagram showing the format of such an element with enough entries =
to fill more than one 32 bit block and require
> padding
> >> to the next.
> >
> > OK, I will give it a shot.
> >
> >> - Figure 6: =93PT of request=94 - the RTCP packet that causes this =
respond to be generated might not be a request to the
> server.
> >> This field might be clearer called =93Failed PT=94?
> >
> > Sounds good.
> >
> >> I didn't look at the SDP sections, sorry.
> >
> > Thanks anyway.
> > -acbegen
> >
> >> --
> >> Colin Perkins
> >> http://csperkins.org/
> >>
> >>
> >
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www.ietf.org/mailman/listinfo/avt
>=20
>=20
>=20
> --
> Colin Perkins
> http://csperkins.org/
>=20
>=20


From oran@cisco.com  Mon Jan  3 08:55:06 2011
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To: "Ali C. Begen (abegen)" <abegen@cisco.com>
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Subject: Re: [AVT] FW: New Version Notification for draft-ietf-avt-ports-for-ucast-mcast-rtp-10
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On Jan 3, 2011, at 11:49 AM, Ali C. Begen (abegen) wrote:

>>>> This is looking good. Some minor comments only:
>>>>=20
>>>> - Section 3:  =93Therefore, the server will need the client to also =
include the Token along with the RTCP messages that are
>>>> different from the RTCP message that triggerred the unicast session =
establishment.=94 - I don=92t understand what this means.
>>>> Can you clarify in the draft?
>>>=20
>>> Certain messages don=92t establish a unicast session but control it, =
and they can be required to be bundled with a token as
>> well as described in 4.3.1.
>>=20
>> Sure, I understand that. But the text in the draft is unclear, and =
needs updating.
>=20
> What about:
>=20
>   During the unicast session lifetime, client's certain actions and
>   messages sent by the client might need to be authorized by the =
server by
>   requiring a valid Token.  Therefore, the client needs
>   to also include the Token along with such RTCP messages.=20
>   These are explained later in this document.
>=20
How about some word-smithing:
"During the lifetime of a unicast session, a client may need to send =
RTCP messages that require server authorization. Since such messages =
require a valid Token for authorization, the client needs to include the =
token along with such RTCP messages as explained in detail in later =
sections of the document"


>>>> - Section 4.1: =93Here, the SSRC value is not necessarily linked to =
the one used by the client in the multicast session.=94 - what
>> do
>>>> you mean by =93not necessarily=94 here? They=92re either part of =
the same RTP session, in which case the client should use the
>>>> same SSRC; or they=92re not, in which case the client should choose =
its SSRC randomly.
>>>=20
>>> When requesting a Token, the SSRC used or chosen by the client does =
not matter. Think about a client requesting a token
>> during booting. It can use that token with any ssm session later on. =
We could say "Here the SSRC value can be chosen
>> randomly by the client since the Port Mapping Request message is not =
necessarily linked to a specific multicast session."
>>>=20
>>> Is this better?
>>=20
>> Not really, since it's just as vague. Saying the SSRC "can" be chosen =
randomly doesn't help: either it is chosen randomly, or it
>> isn't. Which is it?
>=20
> "Is chosen" then. I was reluctant to say this explicitly since a =
client who asks for a token right on the instant it needed one could =
still use the SSRC from the multicast session if it wants to. But, doing =
so will not break anything.
>=20
> -acbegen
>=20
>=20
>>>> - Section 4.2: The addition of the packet types element is good. =
However, for clarify, it would be useful to include a packet
>>>> diagram showing the format of such an element with enough entries =
to fill more than one 32 bit block and require
>> padding
>>>> to the next.
>>>=20
>>> OK, I will give it a shot.
>>>=20
>>>> - Figure 6: =93PT of request=94 - the RTCP packet that causes this =
respond to be generated might not be a request to the
>> server.
>>>> This field might be clearer called =93Failed PT=94?
>>>=20
>>> Sounds good.
>>>=20
>>>> I didn't look at the SDP sections, sorry.
>>>=20
>>> Thanks anyway.
>>> -acbegen
>>>=20
>>>> --
>>>> Colin Perkins
>>>> http://csperkins.org/
>>>>=20
>>>>=20
>>>=20
>>> _______________________________________________
>>> Audio/Video Transport Working Group
>>> avt@ietf.org
>>> https://www.ietf.org/mailman/listinfo/avt
>>=20
>>=20
>>=20
>> --
>> Colin Perkins
>> http://csperkins.org/
>>=20
>>=20
>=20
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt


From csp@csperkins.org  Mon Jan  3 09:04:38 2011
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References: <20101221063801.32FE03A69DD@core3.amsl.com> <04CAD96D4C5A3D48B1919248A8FE0D540DEF4023@xmb-sjc-215.amer.cisco.com> <2C992ED0-C498-4268-99DB-8AC361FCF74A@csperkins.org> <04CAD96D4C5A3D48B1919248A8FE0D540DFAF3D6@xmb-sjc-215.amer.cisco.com> <CAB92DF9-754F-4020-A1B6-01C5DAC2A4E5@csperkins.org> <04CAD96D4C5A3D48B1919248A8FE0D540DFAF3F3@xmb-sjc-215.amer.cisco.com> <3BF3E0F5-2588-41C4-9C0B-5B105E8AE61F@cisco.com>
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Subject: Re: [AVT] FW: New Version Notification for draft-ietf-avt-ports-for-ucast-mcast-rtp-10
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Looks good.
Colin


On 3 Jan 2011, at 16:57, David R Oran wrote:

>=20
> On Jan 3, 2011, at 11:49 AM, Ali C. Begen (abegen) wrote:
>=20
>>>>> This is looking good. Some minor comments only:
>>>>>=20
>>>>> - Section 3:  =93Therefore, the server will need the client to =
also include the Token along with the RTCP messages that are
>>>>> different from the RTCP message that triggerred the unicast =
session establishment.=94 - I don=92t understand what this means.
>>>>> Can you clarify in the draft?
>>>>=20
>>>> Certain messages don=92t establish a unicast session but control =
it, and they can be required to be bundled with a token as
>>> well as described in 4.3.1.
>>>=20
>>> Sure, I understand that. But the text in the draft is unclear, and =
needs updating.
>>=20
>> What about:
>>=20
>>  During the unicast session lifetime, client's certain actions and
>>  messages sent by the client might need to be authorized by the =
server by
>>  requiring a valid Token.  Therefore, the client needs
>>  to also include the Token along with such RTCP messages.=20
>>  These are explained later in this document.
>>=20
> How about some word-smithing:
> "During the lifetime of a unicast session, a client may need to send =
RTCP messages that require server authorization. Since such messages =
require a valid Token for authorization, the client needs to include the =
token along with such RTCP messages as explained in detail in later =
sections of the document"
>=20
>=20
>>>>> - Section 4.1: =93Here, the SSRC value is not necessarily linked =
to the one used by the client in the multicast session.=94 - what
>>> do
>>>>> you mean by =93not necessarily=94 here? They=92re either part of =
the same RTP session, in which case the client should use the
>>>>> same SSRC; or they=92re not, in which case the client should =
choose its SSRC randomly.
>>>>=20
>>>> When requesting a Token, the SSRC used or chosen by the client does =
not matter. Think about a client requesting a token
>>> during booting. It can use that token with any ssm session later on. =
We could say "Here the SSRC value can be chosen
>>> randomly by the client since the Port Mapping Request message is not =
necessarily linked to a specific multicast session."
>>>>=20
>>>> Is this better?
>>>=20
>>> Not really, since it's just as vague. Saying the SSRC "can" be =
chosen randomly doesn't help: either it is chosen randomly, or it
>>> isn't. Which is it?
>>=20
>> "Is chosen" then. I was reluctant to say this explicitly since a =
client who asks for a token right on the instant it needed one could =
still use the SSRC from the multicast session if it wants to. But, doing =
so will not break anything.
>>=20
>> -acbegen
>>=20
>>=20
>>>>> - Section 4.2: The addition of the packet types element is good. =
However, for clarify, it would be useful to include a packet
>>>>> diagram showing the format of such an element with enough entries =
to fill more than one 32 bit block and require
>>> padding
>>>>> to the next.
>>>>=20
>>>> OK, I will give it a shot.
>>>>=20
>>>>> - Figure 6: =93PT of request=94 - the RTCP packet that causes this =
respond to be generated might not be a request to the
>>> server.
>>>>> This field might be clearer called =93Failed PT=94?
>>>>=20
>>>> Sounds good.
>>>>=20
>>>>> I didn't look at the SDP sections, sorry.
>>>>=20
>>>> Thanks anyway.
>>>> -acbegen
>>>>=20
>>>>> --
>>>>> Colin Perkins
>>>>> http://csperkins.org/
>>>>>=20
>>>>>=20
>>>>=20
>>>> _______________________________________________
>>>> Audio/Video Transport Working Group
>>>> avt@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/avt
>>>=20
>>>=20
>>>=20
>>> --
>>> Colin Perkins
>>> http://csperkins.org/
>>>=20
>>>=20
>>=20
>> _______________________________________________
>> Audio/Video Transport Working Group
>> avt@ietf.org
>> https://www.ietf.org/mailman/listinfo/avt
>=20
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt



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




From abegen@cisco.com  Mon Jan  3 12:39:35 2011
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
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Subject: [AVT] Call for Participation for ACM MMSys 2011 - San Jose, CA
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Sorry if you see this call multiple times, but many of the regular =
sessions and two special sessions are particularly of interest for the =
avt and httpstreaming communities. Technical program and registration =
are now available on the Web site.
http://www.mmsys.org/?q=3Dnode/41=20

-acbegen

=3D=3D=3D=3D MMSys 2011 Call For Participation =3D=3D=3D=3D

        ACM Multimedia Systems 2011
            Feb. 23 - 25, 2011
               San Jose, CA
               www.mmsys.org

Conference registration now open at: http://www.regonline.com/mmsys2011=20

New This Year:
  * Low registration fees made possible by Cisco's hosting support.
  * Keynote Speaker: Mark Watson, Netflix
  * Special Session: Modern Media Transport
  * Dataset Track
  * Full program at www.mmsys.org

The ACM Multimedia Systems conference provides a forum for researchers, =
engineers, and scientist to present and share their latest research =
findings in multimedia systems. While research about specific aspects of =
multimedia systems is regularly published in the various proceedings and =
transactions of the networking, operating system, real-time system, and =
database communities, MMSys aims to cut across these domains in the =
context of multimedia data types. This provides a unique opportunity to =
view the intersections and interplay of the various approaches and =
solutions developed across these domains to deal with multimedia data =
types. Furthermore, MMSys provides an avenue for communicating research =
that addresses multimedia systems holistically.

Industrial participants are highly encouraged to attend. Students are =
invited to take advantage of special student pricing.=20

Sponsored by ACM SIG Multimedia
In-cooperation with SIGCOMM and SIGOPS

General Co-Chairs
   Ketan Mayer-Patel, Univ. of North Carolina, USA
   Ali C. Begen, Cisco, Canada

Technical Program Chair
   Mark Claypool, Worcester Polytechnic Institute, USA

Technical Program Committee

Luca Abeni, University of Trento, Italy
Kevin Almeroth, University of California at Santa Barbara, USA
Grenville Armitage, Swinburne University of Technology, Australia
Ali Begen, Cisco, Canada
Azer Bestavros, Boston University, USA
Dick Bulterman, CWI, Netherlands
Surendar Chandra, FXPAL, USA
Bruce Davie, Cisco, USA
Wu-Chang Feng, Portland State University, USA
Wu-Chi Feng, Portland State University, USA
Carsten Griwodz, University of Oslo, Norway
P=E5l Halvorsen, University of Oslo, Norway
Mohamed Hefeeda, Simon Fraser University, Canada
Tristan Henderson, University of St. Andrews, UK
Sugih Jamin, University of Michigan, USA
Charles "Buck" Krasic, University of British Columbia, Canada
Baochun Li, University of Toronto, Canada
Kang Li, University of Georgia, USA
Tom Little, Boston University, USA
Dwight Makaroff, University of Saskatchewan, Canada
Andreas Mauthe, Lancaster University, UK
John Miller, Microsoft, UK
Klara Nahrstedt, University of Illinois Urbana-Champaign, USA
Wei Tsang Ooi, National University of Singapore, Singapore
Reza Rejaie, University of Oregon, USA
Christoph Rensing, University of Darmstadt, Germany
Larry Rowe, FXPAL, USA
Nabil Sarhan, Wayne State University, USA
Travis Schluessler, Intel, USA
Prashant Shenoy, University of Massachusetts, USA
Shervin Shirmohammadi, University of Ottawa, Canada
Michael Vernick, Avaya, USA
Carey Williamson, University of Calgary, Canada
Huahui Wu, Google, USA
Roger Zimmerman, National University of Singapore, Singapore
Michael Zink, University of Massachusetts, USA

From keith.drage@alcatel-lucent.com  Tue Jan  4 03:38:13 2011
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: "avt@ietf.org" <avt@ietf.org>
Date: Tue, 4 Jan 2011 12:40:16 +0100
Thread-Topic: Gen-ART Telechat review of draft-ietf-avt-rtp-cnames-03.txt
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Subject: [AVT] FW: Gen-ART Telechat review of draft-ietf-avt-rtp-cnames-03.txt
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(As WG co-chair)

As part of the publication request process to become an RFC, i-ds go throug=
h a number of separate reviews.=20

I'm forwarding this review to the list as correcting it has impact on the b=
its on the wire, and AVT may have some input on what change they would like=
 to see.

If so, please express.

regards

Keith

-----Original Message-----
From: Suresh Krishnan [mailto:suresh.krishnan@ericsson.com]=20
Sent: Monday, January 03, 2011 3:27 AM
To: General Area Review Team; draft-ietf-avt-rtp-cnames.all@tools.ietf.org
Subject: Gen-ART Telechat review of draft-ietf-avt-rtp-cnames-03.txt

I have been selected as the General Area Review Team (Gen-ART) reviewer for=
 this draft (for background on Gen-ART, please see http://www.alvestrand.no=
/ietf/gen/art/gen-art-FAQ.html).

Please wait for direction from your document shepherd or AD before posting =
a new version of the draft.

Document: draft-ietf-avt-rtp-cnames-03.txt
Reviewer: Suresh Krishnan
Review Date: 2011/01/02
IESG Telechat date: 2011/01/06

Summary: This draft has addressed my comments from the previous Last Call r=
eview but has a new issue that needs to be fixed before publication as a Pr=
oposed Standard.

* Section 5 Step 2

This step describes the use of an EUI-64 identifier and how to generate one=
 if it does not exists. The procedure for the creation of the EUI-64 from a=
 48 bit MAC address is incorrect.

RFC4291 describes how to create a **modified EUI-64** identifier (and not a=
 EUI-64 identifier) from a MAC address.

A MAC address of xx:78:90:12:34:56 will be transformed to a modified
EUI-64 of

yy7890FFFE123456

where yy is the value of xx with the u/l bit (bit 6) flipped.

The IEEE reference for creation of an EUI-64 from a 48 bit MAC address

http://standards.ieee.org/develop/regauth/tut/eui64.pdf

would result in an EUI-64 of

xx7890FFFF123456 for the same MAC address.

I understand that there will be no interoperability issues due to this (sin=
ce the cname is opaque) but there may be issues with testing compliance (si=
nce the expected hash values will be way off).

I recommend either using the term "modified EUI-64" instead of EUI-64 or ch=
anging the reference from RFC4291 to the IEEE document in order to fix this=
 inconsistency.

Thanks
Suresh











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Forwarding to the AVT list in case folks there aren't watching =
ietf@ietf.org.

The recently released -03 reflects what's below

RjS

Begin forwarded message:

> From: Colin Perkins <csp@csperkins.org>
> Date: December 17, 2010 11:59:51 AM CST
> To: Allison Mankin <allison.mankin@gmail.com>
> Cc: Allison Mankin <mankin@psg.com>, Ali Begen <abegen@cisco.com>, =
IETF Discussion <ietf@ietf.org>, Transport Directorate =
<tsv-dir@ietf.org>
> Subject: Re: TSV-Dir Last Call review of draft-ietf-avt-rtp-cnames-02
>=20
> Allison,
>=20
> Thanks for the review. These all seem like reasonable suggestions, =
which we should incorporate.=20
>=20
> Colin
>=20
>=20
>=20
> On 14 Dec 2010, at 02:56, Allison Mankin wrote:
>> TSV-DIR Last Call review of "Guidelines for Chosing RTP Control =
Protocol (RTCP) Canonical Names (CNAMEs)
>>=20
>> I've reviewed this document as part of the transport area =
directorate's ongoing effort to review key IETF documents. These =
comments were written primarily for the transport area directors, but =
are copied to the document's authors for their information and to allow =
them to address any issues raised. The authors should consider this =
review together with any other last-call comments they receive.  Please =
always cc tsv-dir@ietf.org if you reply to or forward this review.
>>=20
>> Summary: This draft is on the right track, but has open issues =
described in the review.
>>=20
>> The review below gives my basis for three recommendations:
>> 1. For long-term persistent CNAMEs, differentiate among the UUID =
"Versions" in RFC 4122
>> 2. For short-term persistent CNAMEs, allow IPv4-only endpoints to =
generate a pseudo-random alternative to their MAC, if desired
>> 3. For per-session CNAMEs, provide some guidance for the random value =
and the key, which are currently left underspecified.
>>=20
>> Section 3 sets up the major requirement to be met, which is better =
uniqueness properties than those of the CNAME generation in RFC 3550. =
There's a hint in the last sentence of the section about an additional =
requirement to afford privacy support.  This sentence only mentions =
avoiding exposure of private addresses.  Avoidance of linkage among RTP =
streams due to the use of fixed CNAMEs comes up later.
>>=20
>> Section 4 sets up the CNAME types, persistent versus per-session, and =
within persistent types, short-term IPv4-only, short-term IPv6-capable, =
and long-term.  The methods of generating these CNAMEs are distinct and =
I understand from the WG mailing list discussion of their AD review that =
they are to be specified as mandatory.   On December 2, Ali wrote to AVT =
that the next revision will change language in Section 4 from the =
existing:
>>=20
>>   Other methods, beyond the three methods listed above, are not
>>   compliant with this specification and SHOULD NOT be used.
>>=20
>> To the following:
>>=20
>>   It is believed that obtaining uniqueness is an important property =
that
>>   requires careful evaluation of the method. This document provides a
>>   number of methods, for which it is believed that at least one of =
them
>>   would meet all deployment scenarios. This document therefore does =
not
>>   provide for the implementor to define and select an alternative
>>   method.
>>=20
>>   A future specification might define an alternative method for
>>   generating RTCP CNAMEs as long as the proposed method has =
appropriate
>>   uniqueness, and there is consistency between the methods used for
>>   multiple RTP sessions that are to be correlated. However, such a
>>   specification needs to be reviewed and approved before deployment.
>>=20
>> The second sentence of the first paragraph could be written better:
>>=20
>>   This document provides a number of methods, at least one of which
>>   should be suitable for all deployment scenarios.
>=20
> Agreed.
>=20
>> My remaining discussion is about making this statement more true.
>>=20
>> First, all the methods should afford some access to privacy support, =
but as written, there is little or none for the long-term persistent and =
the IPv4-only short-term persistent CNAMEs.
>>=20
>>                                     An implementation that wishes to
>>  discourage this type of third-party monitoring can generate a unique
>>  RTCP CNAME for each RTP session, or group of related RTP sessions,
>>  that it joins.  Such a per-session RTCP CNAME cannot be used for
>>  traffic analysis, and so provides some limited form of privacy
>>=20
>> The above statement implies that a long-term persistent CNAME =
inherently cannot have privacy support.  It's not necessarily so: if the =
CNAME doesn't embed identifying information, observers get connections =
among multiple streams but they only don't have a fixed host identity, =
only a pseudonym.
>>=20
>> Specific issues per CNAME method:
>>=20
>> 1) Long-term persistent CNAMEs are required to be generated by an =
adaptation of RFC 4122 UUIDs (the adaptation is to leave off the string =
"urn:uuid:"). The draft references Section 3 of RFC 4122.  It should =
reference Section 4 instead because Section 3 does not detail the UUID =
components. =20
>=20
> Agreed. This looks like a typo that should be fixed.
>=20
>> Moreover, it would be good for the draft to advise on usage among the =
four different versions of UUID that RFC 4122 covers:
>>=20
>> + Version 1: generated from clock + MAC, or + MAC-format random =
number
>> + Version 3: generated from namespace:name
>> + Version 4: generated from "truly random or pseudo-random number"
>> + Version 5: generated from hash of namespace:name
>>=20
>> There needs to be some guidance because the use of the UUID Versions =
3/5, derived from namespace:name, has the same problem with =
non-uniqueness of FQDNs as the RFC 3550 FQDN-based CNAMEs.  Perhaps this =
document should advise against using 3/5.  In addition, right now the =
draft is silent on mitigating identity exposure, but it could offer some =
guidance towards using the variant of Version 1 that does not expose the =
MAC or towards using Version 4.
>=20
> This guidance seems appropriate, thanks.
>=20
>> 2) Short-term persistent CNAMEs  are allowed to use a pseudo-randomly =
generated RFC 4941 IPv6 privacy address if the device is IPv6-capable, =
but are required to exposetheir MACs if the device is IPv4-only.  Why =
not allow IPv4-only endpoints to to generate pseudo-random MACs using =
the specification of doing so provided for the Version 1 variant in RFC =
4122?
>=20
> That seems like a reasonable addition.
>=20
>> 3) Per-session CNAMEs are generated using SHA1-HMAC over the =
concatenation of the initially-used SSRC, the source and destination =
addresses and ports, and "a randomly generated value".  RFC 4086 is =
given as a reference for the randomly generated value.  I think an =
implementer would like to know how many bytes of randomly generated =
value should be used; this doesn't come out clearly from reading RFC =
4086. Further, what are the requirements for the key for HMAC here?  In =
reading RFC 2104, the reference for HMAC, I end up with some ideas about =
the key length, but not about how long/where I can use the same key or =
whether there are problematic keys - to mention two questions that come =
to mind.  Keying would not normally be easy to under-specify this way =
because a security application would call for some keying details, but =
in this application there is no security use for the keys.  Either it =
would be good to add some guidance or perhaps SHA1-HMAC could be =
replaced with an "inse
> cure hash."
>=20
> We'll add some guidance in -03.
>=20
> Thanks,
> Colin
>=20
> --=20
> Colin Perkins
> http://csperkins.org/
>=20
>=20
>=20
> _______________________________________________
> Ietf mailing list
> Ietf@ietf.org
> https://www.ietf.org/mailman/listinfo/ietf


--Apple-Mail-39-179686159
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<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
">Forwarding to the AVT list in case folks there aren't watching <a =
href=3D"mailto:ietf@ietf.org">ietf@ietf.org</a>.<div><br><div>The =
recently released -03 reflects what's =
below</div><div><br></div><div>RjS<br><div><br><div>Begin forwarded =
message:</div><br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px;"><span =
style=3D"font-family:'Helvetica'; font-size:medium; color:rgba(0, 0, 0, =
1);"><b>From: </b></span><span style=3D"font-family:'Helvetica'; =
font-size:medium;">Colin Perkins &lt;<a =
href=3D"mailto:csp@csperkins.org">csp@csperkins.org</a>&gt;<br></span></di=
v><div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px;"><span style=3D"font-family:'Helvetica'; =
font-size:medium; color:rgba(0, 0, 0, 1);"><b>Date: </b></span><span =
style=3D"font-family:'Helvetica'; font-size:medium;">December 17, 2010 =
11:59:51 AM CST<br></span></div><div style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span =
style=3D"font-family:'Helvetica'; font-size:medium; color:rgba(0, 0, 0, =
1);"><b>To: </b></span><span style=3D"font-family:'Helvetica'; =
font-size:medium;">Allison Mankin &lt;<a =
href=3D"mailto:allison.mankin@gmail.com">allison.mankin@gmail.com</a>&gt;<=
br></span></div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px;"><span =
style=3D"font-family:'Helvetica'; font-size:medium; color:rgba(0, 0, 0, =
1);"><b>Cc: </b></span><span style=3D"font-family:'Helvetica'; =
font-size:medium;">Allison Mankin &lt;<a =
href=3D"mailto:mankin@psg.com">mankin@psg.com</a>&gt;, Ali Begen &lt;<a =
href=3D"mailto:abegen@cisco.com">abegen@cisco.com</a>&gt;, IETF =
Discussion &lt;<a href=3D"mailto:ietf@ietf.org">ietf@ietf.org</a>&gt;, =
Transport Directorate &lt;<a =
href=3D"mailto:tsv-dir@ietf.org">tsv-dir@ietf.org</a>&gt;<br></span></div>=
<div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px;"><span style=3D"font-family:'Helvetica'; =
font-size:medium; color:rgba(0, 0, 0, 1);"><b>Subject: </b></span><span =
style=3D"font-family:'Helvetica'; font-size:medium;"><b>Re: TSV-Dir Last =
Call review of =
draft-ietf-avt-rtp-cnames-02</b><br></span></div><br><div>Allison,<br><br>=
Thanks for the review. These all seem like reasonable suggestions, which =
we should incorporate. <br><br>Colin<br><br><br><br>On 14 Dec 2010, at =
02:56, Allison Mankin wrote:<br><blockquote type=3D"cite">TSV-DIR Last =
Call review of "Guidelines for Chosing RTP Control Protocol (RTCP) =
Canonical Names (CNAMEs)<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">I've reviewed =
this document as part of the transport area directorate's ongoing effort =
to review key IETF documents. These comments were written primarily for =
the transport area directors, but are copied to the document's authors =
for their information and to allow them to address any issues raised. =
The authors should consider this review together with any other =
last-call comments they receive. &nbsp;Please always cc <a =
href=3D"mailto:tsv-dir@ietf.org">tsv-dir@ietf.org</a> if you reply to or =
forward this review.<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">Summary: This =
draft is on the right track, but has open issues described in the =
review.<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">The review =
below gives my basis for three =
recommendations:<br></blockquote><blockquote type=3D"cite">1. For =
long-term persistent CNAMEs, differentiate among the UUID "Versions" in =
RFC 4122<br></blockquote><blockquote type=3D"cite">2. For short-term =
persistent CNAMEs, allow IPv4-only endpoints to generate a pseudo-random =
alternative to their MAC, if desired<br></blockquote><blockquote =
type=3D"cite">3. For per-session CNAMEs, provide some guidance for the =
random value and the key, which are currently left =
underspecified.<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">Section 3 sets =
up the major requirement to be met, which is better uniqueness =
properties than those of the CNAME generation in RFC 3550. There's a =
hint in the last sentence of the section about an additional requirement =
to afford privacy support. &nbsp;This sentence only mentions avoiding =
exposure of private addresses. &nbsp;Avoidance of linkage among RTP =
streams due to the use of fixed CNAMEs comes up =
later.<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">Section 4 sets =
up the CNAME types, persistent versus per-session, and within persistent =
types, short-term IPv4-only, short-term IPv6-capable, and long-term. =
&nbsp;The methods of generating these CNAMEs are distinct and I =
understand from the WG mailing list discussion of their AD review that =
they are to be specified as mandatory. &nbsp;&nbsp;On December 2, Ali =
wrote to AVT that the next revision will change language in Section 4 =
from the existing:<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite"> =
&nbsp;&nbsp;Other methods, beyond the three methods listed above, are =
not<br></blockquote><blockquote type=3D"cite"> &nbsp;&nbsp;compliant =
with this specification and SHOULD NOT be =
used.<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">To the =
following:<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite"> &nbsp;&nbsp;It =
is believed that obtaining uniqueness is an important property =
that<br></blockquote><blockquote type=3D"cite"> &nbsp;&nbsp;requires =
careful evaluation of the method. This document provides =
a<br></blockquote><blockquote type=3D"cite"> &nbsp;&nbsp;number of =
methods, for which it is believed that at least one of =
them<br></blockquote><blockquote type=3D"cite"> &nbsp;&nbsp;would meet =
all deployment scenarios. This document therefore does =
not<br></blockquote><blockquote type=3D"cite"> &nbsp;&nbsp;provide for =
the implementor to define and select an =
alternative<br></blockquote><blockquote type=3D"cite"> =
&nbsp;&nbsp;method.<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite"> &nbsp;&nbsp;A =
future specification might define an alternative method =
for<br></blockquote><blockquote type=3D"cite"> &nbsp;&nbsp;generating =
RTCP CNAMEs as long as the proposed method has =
appropriate<br></blockquote><blockquote type=3D"cite"> =
&nbsp;&nbsp;uniqueness, and there is consistency between the methods =
used for<br></blockquote><blockquote type=3D"cite"> &nbsp;&nbsp;multiple =
RTP sessions that are to be correlated. However, such =
a<br></blockquote><blockquote type=3D"cite"> &nbsp;&nbsp;specification =
needs to be reviewed and approved before =
deployment.<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">The second =
sentence of the first paragraph could be written =
better:<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite"> =
&nbsp;&nbsp;This document provides a number of methods, at least one of =
which<br></blockquote><blockquote type=3D"cite"> &nbsp;&nbsp;should be =
suitable for all deployment =
scenarios.<br></blockquote><br>Agreed.<br><br><blockquote type=3D"cite">My=
 remaining discussion is about making this statement more =
true.<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">First, all the =
methods should afford some access to privacy support, but as written, =
there is little or none for the long-term persistent and the IPv4-only =
short-term persistent CNAMEs.<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite"> =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;An =
implementation that wishes to<br></blockquote><blockquote type=3D"cite"> =
&nbsp;discourage this type of third-party monitoring can generate a =
unique<br></blockquote><blockquote type=3D"cite"> &nbsp;RTCP CNAME for =
each RTP session, or group of related RTP =
sessions,<br></blockquote><blockquote type=3D"cite"> &nbsp;that it =
joins. &nbsp;Such a per-session RTCP CNAME cannot be used =
for<br></blockquote><blockquote type=3D"cite"> &nbsp;traffic analysis, =
and so provides some limited form of privacy<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">The above =
statement implies that a long-term persistent CNAME inherently cannot =
have privacy support. &nbsp;It's not necessarily so: if the CNAME =
doesn't embed identifying information, observers get connections among =
multiple streams but they only don't have a fixed host identity, only a =
pseudonym.<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">Specific issues =
per CNAME method:<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">1) Long-term =
persistent CNAMEs are required to be generated by an adaptation of RFC =
4122 UUIDs (the adaptation is to leave off the string "urn:uuid:"). The =
draft references Section 3 of RFC 4122. &nbsp;It should reference =
Section 4 instead because Section 3 does not detail the UUID components. =
&nbsp;<br></blockquote><br>Agreed. This looks like a typo that should be =
fixed.<br><br><blockquote type=3D"cite">Moreover, it would be good for =
the draft to advise on usage among the four different versions of UUID =
that RFC 4122 covers:<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">+ Version 1: =
generated from clock + MAC, or + MAC-format random =
number<br></blockquote><blockquote type=3D"cite">+ Version 3: generated =
from namespace:name<br></blockquote><blockquote type=3D"cite">+ Version =
4: generated from "truly random or pseudo-random =
number"<br></blockquote><blockquote type=3D"cite">+ Version 5: generated =
from hash of namespace:name<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">There needs to =
be some guidance because the use of the UUID Versions 3/5, derived from =
namespace:name, has the same problem with non-uniqueness of FQDNs as the =
RFC 3550 FQDN-based CNAMEs. &nbsp;Perhaps this document should advise =
against using 3/5. &nbsp;In addition, right now the draft is silent on =
mitigating identity exposure, but it could offer some guidance towards =
using the variant of Version 1 that does not expose the MAC or towards =
using Version 4.<br></blockquote><br>This guidance seems appropriate, =
thanks.<br><br><blockquote type=3D"cite">2) Short-term persistent CNAMEs =
&nbsp;are allowed to use a pseudo-randomly generated RFC 4941 IPv6 =
privacy address if the device is IPv6-capable, but are required to =
exposetheir MACs if the device is IPv4-only. &nbsp;Why not allow =
IPv4-only endpoints to to generate pseudo-random MACs using the =
specification of doing so provided for the Version 1 variant in RFC =
4122?<br></blockquote><br>That seems like a reasonable =
addition.<br><br><blockquote type=3D"cite">3) Per-session CNAMEs are =
generated using SHA1-HMAC over the concatenation of the initially-used =
SSRC, the source and destination addresses and ports, and "a randomly =
generated value". &nbsp;RFC 4086 is given as a reference for the =
randomly generated value. &nbsp;I think an implementer would like to =
know how many bytes of randomly generated value should be used; this =
doesn't come out clearly from reading RFC 4086. Further, what are the =
requirements for the key for HMAC here? &nbsp;In reading RFC 2104, the =
reference for HMAC, I end up with some ideas about the key length, but =
not about how long/where I can use the same key or whether there are =
problematic keys - to mention two questions that come to mind. =
&nbsp;Keying would not normally be easy to under-specify this way =
because a security application would call for some keying details, but =
in this application there is no security use for the keys. &nbsp;Either =
it would be good to add some guidance or perhaps SHA1-HMAC could be =
replaced with an "inse<br></blockquote> cure hash."<br><br>We'll add =
some guidance in -03.<br><br>Thanks,<br>Colin<br><br>-- <br>Colin =
Perkins<br><a =
href=3D"http://csperkins.org/">http://csperkins.org/</a><br><br><br><br>__=
_____________________________________________<br>Ietf mailing =
list<br>Ietf@ietf.org<br>https://www.ietf.org/mailman/listinfo/ietf<br></d=
iv></blockquote></div><br></div></div></body></html>=

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From abegen@cisco.com  Tue Jan  4 12:44:01 2011
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Colin Perkins" <csp@csperkins.org>, "Dave Oran (oran)" <oran@cisco.com>
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I asked the chairs yesterday when they will start WGLC. Just before they =
do that we will submit a revision to address these minor changes.

Chairs, please provide a timeline.

-acbegen

> -----Original Message-----
> From: Colin Perkins [mailto:csp@csperkins.org]
> Sent: Monday, January 03, 2011 12:07 PM
> To: Dave Oran (oran)
> Cc: Ali C. Begen (abegen); avt@ietf.org
> Subject: Re: [AVT] FW: New Version Notification for =
draft-ietf-avt-ports-for-ucast-mcast-rtp-10
>=20
> Looks good.
> Colin
>=20
>=20
> On 3 Jan 2011, at 16:57, David R Oran wrote:
>=20
> >
> > On Jan 3, 2011, at 11:49 AM, Ali C. Begen (abegen) wrote:
> >
> >>>>> This is looking good. Some minor comments only:
> >>>>>
> >>>>> - Section 3:  =93Therefore, the server will need the client to =
also include the Token along with the RTCP messages that
> are
> >>>>> different from the RTCP message that triggerred the unicast =
session establishment.=94 - I don=92t understand what this
> means.
> >>>>> Can you clarify in the draft?
> >>>>
> >>>> Certain messages don=92t establish a unicast session but control =
it, and they can be required to be bundled with a token as
> >>> well as described in 4.3.1.
> >>>
> >>> Sure, I understand that. But the text in the draft is unclear, and =
needs updating.
> >>
> >> What about:
> >>
> >>  During the unicast session lifetime, client's certain actions and
> >>  messages sent by the client might need to be authorized by the =
server by
> >>  requiring a valid Token.  Therefore, the client needs
> >>  to also include the Token along with such RTCP messages.
> >>  These are explained later in this document.
> >>
> > How about some word-smithing:
> > "During the lifetime of a unicast session, a client may need to send =
RTCP messages that require server authorization. Since
> such messages require a valid Token for authorization, the client =
needs to include the token along with such RTCP messages
> as explained in detail in later sections of the document"
> >
> >
> >>>>> - Section 4.1: =93Here, the SSRC value is not necessarily linked =
to the one used by the client in the multicast session.=94 -
> what
> >>> do
> >>>>> you mean by =93not necessarily=94 here? They=92re either part of =
the same RTP session, in which case the client should use
> the
> >>>>> same SSRC; or they=92re not, in which case the client should =
choose its SSRC randomly.
> >>>>
> >>>> When requesting a Token, the SSRC used or chosen by the client =
does not matter. Think about a client requesting a
> token
> >>> during booting. It can use that token with any ssm session later =
on. We could say "Here the SSRC value can be chosen
> >>> randomly by the client since the Port Mapping Request message is =
not necessarily linked to a specific multicast session."
> >>>>
> >>>> Is this better?
> >>>
> >>> Not really, since it's just as vague. Saying the SSRC "can" be =
chosen randomly doesn't help: either it is chosen randomly,
> or it
> >>> isn't. Which is it?
> >>
> >> "Is chosen" then. I was reluctant to say this explicitly since a =
client who asks for a token right on the instant it needed one
> could still use the SSRC from the multicast session if it wants to. =
But, doing so will not break anything.
> >>
> >> -acbegen
> >>
> >>
> >>>>> - Section 4.2: The addition of the packet types element is good. =
However, for clarify, it would be useful to include a
> packet
> >>>>> diagram showing the format of such an element with enough =
entries to fill more than one 32 bit block and require
> >>> padding
> >>>>> to the next.
> >>>>
> >>>> OK, I will give it a shot.
> >>>>
> >>>>> - Figure 6: =93PT of request=94 - the RTCP packet that causes =
this respond to be generated might not be a request to the
> >>> server.
> >>>>> This field might be clearer called =93Failed PT=94?
> >>>>
> >>>> Sounds good.
> >>>>
> >>>>> I didn't look at the SDP sections, sorry.
> >>>>
> >>>> Thanks anyway.
> >>>> -acbegen
> >>>>
> >>>>> --
> >>>>> Colin Perkins
> >>>>> http://csperkins.org/
> >>>>>
> >>>>>
> >>>>
> >>>> _______________________________________________
> >>>> Audio/Video Transport Working Group
> >>>> avt@ietf.org
> >>>> https://www.ietf.org/mailman/listinfo/avt
> >>>
> >>>
> >>>
> >>> --
> >>> Colin Perkins
> >>> http://csperkins.org/
> >>>
> >>>
> >>
> >> _______________________________________________
> >> Audio/Video Transport Working Group
> >> avt@ietf.org
> >> https://www.ietf.org/mailman/listinfo/avt
> >
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www.ietf.org/mailman/listinfo/avt
>=20
>=20
>=20
> --
> Colin Perkins
> http://csperkins.org/
>=20
>=20


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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.


	Title           : Port Mapping Between Unicast and Multicast RTP Sessions
	Author(s)       : A. Begen, et al.
	Filename        : draft-ietf-avt-ports-for-ucast-mcast-rtp-11.txt
	Pages           : 35
	Date            : 2011-01-05

This document presents a port mapping solution that allows RTP
receivers to choose their own ports for an auxiliary unicast session
in RTP applications using both unicast and multicast services.  The
solution provides protection against denial-of-service or packet
amplification attacks that could be used to cause one or more RTP
packets to be sent to a victim client.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avt-ports-for-ucast-mcast-rtp-11.txt

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Cg==

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Date: Wed, 5 Jan 2011 17:56:15 +0100
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(As WG co-chair)

The WG chairs believe this document now represents working group consensus =
on the technical solution so...

This is to announce a working group last call for draft-ietf-avt-ports-for-=
ucast-mcast-rtp-11 with comments to be provided by 19th January.

http://www.ietf.org/internet-drafts/draft-ietf-avt-ports-for-ucast-mcast-rt=
p-11.txt

In providing comments, please identify the level you believe the comment is=
 (editorial, minor technical, major technical) in order to give some guidan=
ce to prioritizing the show stoppers.=20

Please provide your comments in reply to this message.

If you read it and have no comments, and believe the document is therefore =
OK, then please send a message to the WG chairs - this is extremely helpful=
 in determining level of review the document has had.

regards

Keith=

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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>, "avt@ietf.org" <avt@ietf.org>
Date: Wed, 5 Jan 2011 18:11:25 +0100
Thread-Topic: WGLC for draft-ietf-avt-ports-for-ucast-mcast-rtp-11
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Subject: Re: [AVT] WGLC for draft-ietf-avt-ports-for-ucast-mcast-rtp-11
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(As WG co-chair)

I should add that the document is intended as proposed standard, so please =
take that into account in your review.

Keith=20

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On=20
> Behalf Of DRAGE, Keith (Keith)
> Sent: Wednesday, January 05, 2011 4:56 PM
> To: avt@ietf.org
> Cc: draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
> Subject: [AVT] WGLC for draft-ietf-avt-ports-for-ucast-mcast-rtp-11
>=20
> (As WG co-chair)
>=20
> The WG chairs believe this document now represents working=20
> group consensus on the technical solution so...
>=20
> This is to announce a working group last call for=20
> draft-ietf-avt-ports-for-ucast-mcast-rtp-11 with comments to=20
> be provided by 19th January.
>=20
> http://www.ietf.org/internet-drafts/draft-ietf-avt-ports-for-u
> cast-mcast-rtp-11.txt
>=20
> In providing comments, please identify the level you believe=20
> the comment is (editorial, minor technical, major technical)=20
> in order to give some guidance to prioritizing the show stoppers.=20
>=20
> Please provide your comments in reply to this message.
>=20
> If you read it and have no comments, and believe the document=20
> is therefore OK, then please send a message to the WG chairs=20
> - this is extremely helpful in determining level of review=20
> the document has had.
>=20
> regards
>=20
> Keith
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
> =

From fruchtalle@googlemail.com  Fri Jan  7 01:42:37 2011
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From: Siga <fruchtalle@googlemail.com>
To: avt@ietf.org
Content-Type: multipart/alternative; boundary=20cf3054a549b68f5704993e7701
Subject: [AVT] RTP Protocol cannot be started
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--20cf3054a549b68f5704993e7701
Content-Type: text/plain; charset=ISO-8859-1

Hi,
I have implemented a SIP Client and to extend my project I wrote a RTP
Protocol and try to send the RTP packets and when I notice it using
wireshark I get a strange  message, let me show it so that you can have
better idea

 *CLIENT                                      SERVER*

        SIP Request:INVITE
      ------------------------------------*>*


       SIP Status: 100 Trying
    *<*------------------------------------


       SIP/SDP Status: 200 OK
    *<*------------------------------------


       SIP Request:ACK
    ------------------------------------*>*


      *UDP Source port: bre Destination port*
    ---------------------------------------*>*                    Here I am
planning to send RTP Packets but instead I get this strange message on my
wireshark.

Do anybody have an idea where I am making my mistake.


Regards,
Siga

--20cf3054a549b68f5704993e7701
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Hi,<br><div class=3D"gmail_quote">I have implemented a SIP Client and to ex=
tend my project I wrote a RTP Protocol and try to send the RTP packets and =
when I notice it using wireshark I get a strange=A0 message, let me show it=
 so that you can have better idea<br>

<br><span style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-fam=
ily: verdana,arial,helvetica,sans-serif; font-size: 11px; font-style: norma=
l; font-variant: normal; font-weight: normal; letter-spacing: normal; line-=
height: 14px; text-indent: 0px; text-transform: none; white-space: normal; =
word-spacing: 0px;"><span style=3D"line-height: 15px;"><span>=A0</span><b>C=
LIENT =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 SERVER</b><br>

<br style=3D"margin: 0px; padding: 0px;">=A0=A0=A0=A0=A0=A0=A0 SIP Request:=
INVITE<br style=3D"margin: 0px; padding: 0px;">=A0=A0=A0=A0=A0 ------------=
------------------------<b>&gt;</b><br><br><br style=3D"margin: 0px; paddin=
g: 0px;">=A0=A0=A0=A0=A0=A0 SIP Status: 100 Trying<br style=3D"margin: 0px;=
 padding: 0px;">

=A0=A0=A0 <b>&lt;</b>------------------------------------<br><br><br style=
=3D"margin: 0px; padding: 0px;">=A0=A0=A0=A0=A0=A0 SIP/SDP Status: 200 OK<b=
r style=3D"margin: 0px; padding: 0px;">=A0=A0=A0 <b>&lt;</b>---------------=
---------------------<br>

<br><br style=3D"margin: 0px; padding: 0px;">=A0=A0=A0=A0=A0=A0 SIP Request=
:ACK<br style=3D"margin: 0px; padding: 0px;">=A0=A0=A0 --------------------=
----------------<b>&gt;</b><br><br><br style=3D"margin: 0px; padding: 0px;"=
>=A0=A0=A0=A0=A0 <b>UDP Source port: bre Destination port</b><br style=3D"m=
argin: 0px; padding: 0px;">

=A0=A0=A0 ---------------------------------------<b>&gt;</b>=A0=A0=A0=A0=A0=
=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 Here I am planning to send RTP P=
ackets but instead I get this strange message on my wireshark.<br><br>Do an=
ybody have an idea where I am making my mistake.<br>

<br><br style=3D"margin: 0px; padding: 0px;"></span></span>Regards,<br><fon=
t color=3D"#888888">Siga<br><br><br>
</font></div><br>

--20cf3054a549b68f5704993e7701--

From keith.drage@alcatel-lucent.com  Fri Jan  7 05:23:20 2011
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Siga <fruchtalle@googlemail.com>, "avt@ietf.org" <avt@ietf.org>
Date: Fri, 7 Jan 2011 14:25:22 +0100
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Subject: Re: [AVT] RTP Protocol cannot be started
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I would suggest you post your query to:

sip-implementors@cs.columbia.edu=20

I suspect you will need to provide details of the SDP offer and answer in t=
he SIP messages you show before anyone can provide any meaningful response.=
=20

regards

Keith


________________________________

	From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Siga
	Sent: Friday, January 07, 2011 9:45 AM
	To: avt@ietf.org
	Subject: [AVT] RTP Protocol cannot be started
=09
=09
	Hi,
=09
	I have implemented a SIP Client and to extend my project I wrote a RTP Pro=
tocol and try to send the RTP packets and when I notice it using wireshark =
I get a strange  message, let me show it so that you can have better idea
=09
	 CLIENT                                      SERVER
=09
	        SIP Request:INVITE
	      ------------------------------------>
=09
=09
	       SIP Status: 100 Trying
	    <------------------------------------
=09
=09
	       SIP/SDP Status: 200 OK
	    <------------------------------------
=09
=09
	       SIP Request:ACK
	    ------------------------------------>
=09
=09
	      UDP Source port: bre Destination port
	    --------------------------------------->                    Here I am =
planning to send RTP Packets but instead I get this strange message on my w=
ireshark.
=09
	Do anybody have an idea where I am making my mistake.
=09
=09
	Regards,
	Siga
=09
=09
=09



From keith.drage@alcatel-lucent.com  Sat Jan  8 10:06:38 2011
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
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Date: Sat, 8 Jan 2011 19:08:38 +0100
Thread-Topic: Guidelines for the use of Variable Bit Rate Audio with Secure RTP
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Subject: [AVT] Guidelines for the use of Variable Bit Rate Audio with Secure RTP
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(As WG co-chair)

In support of the milestone:

Feb 2011  Guidelines for the use of Variable Bit Rate Audio with Secure RTP=
 as informational (or possibly BCP)

This is a WG consensus call to adopt all (or some) of the text of:

http://www.ietf.org/id/draft-perkins-avt-srtp-vbr-audio-05.txt

to fulfil that milestone. At this stage the text does not have to be regard=
ed perfect, the question is more "Does this text form a better basis for co=
mpleting the work in the working group that some text yet to be submitted?"

Please respond by Monday 18th January with your positions.

While the prime intention of this call is to adopt the entire text of the a=
uthor draft, if you consider that some of the text is appropriate, but that=
 some of the text does not form a reasonable start for the work, then pleas=
e indicate that text and we will make a separate call on it (and adopt the =
rest).


regards

Keith=

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[Apologies if you receive this more than once]
+++++++++++++++++++++ [ NOSSDAV 2011 Call for Papers ] =
+++++++++++++++++++++++


    The 21th International Workshop on Network and Operating Systems
                     Support for Digital Audio and Video

                               June 2-3, 2011
                     Vancouver, British Columbia, Canada

                      http://nss.cs.ubc.ca/nossdav2011/


NOSSDAV 2011 is the 21th anniversary of SIGMM's leading workshop on =
network and
operating systems support for digital audio and video. The workshop, =
hosted at
the University of British Columbia (UBC), will continue to focus on =
emerging
research topics, controversial ideas, and future research directions in =
the area
of multimedia systems research.

As in previous years, we will maintain the focused single-track format a =
setting
that stimulates lively discussions among the senior and junior =
participants.
NOSSDAV encourages experimental research based on real systems and real =
data sets.
Public availability of the source code and data sets discussed in papers =
presented
at NOSSDAV is highly encouraged.

For NOSSDAV 2011, we will accept papers on broad ranges of topics =
related to the
transmission and presentation of digital audio/video objects. We are =
particularly
interested in soliciting articles that discuss system-level support for =
distributed
social media, as well as papers that focus on enabling multimedia =
applications in
distributed cloud. Other topics of interest include (but are not =
restricted to):

* OS, middleware and network support
* Overlay networks
* Media streaming, distribution and storage support
* Web 2.0 systems and social networks
* Media sensor and ad hoc networks / embedded systems
* Multicore architecture support
* Wireless and mobile multimedia systems / network processor support
* Networked GPUs, graphics and virtual environments
* Networked games / real-time immersive systems
* Multimedia communications and system security
* Grid/Cloud computing support

Please contact the workshop co-chairs to check if your topic is within =
the scope of
NOSSDAV. Papers will be judged on their relevance, technical content and =
correctness,
and the clarity of presentation of the research. Papers should not be =
under review
at another venue nor previously published elsewhere.

Submissions should be at most SIX pages, using standard ACM proceedings =
style.
We expect these submissions to be the kernel of what will eventually =
lead to full-length
papers at high-quality conferences or journals.

Important Dates

Paper Submission Deadline: 24 Feb 2011
Decision Notification:     24 Mar 2011
Camera Ready Due:          7 Apr 2011

For more information, please visit the workshop website at
http://nss.cs.ubc.ca/nossdav2011/


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Subject: [AVT] NOSSDAV 2011 Call for Papers
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[Apologies if you receive this more than once]
+++++++++++++++++++++ [ NOSSDAV 2011 Call for Papers ] =
+++++++++++++++++++++++


    The 21th International Workshop on Network and Operating Systems
                     Support for Digital Audio and Video

                               June 2-3, 2011
                     Vancouver, British Columbia, Canada

                      http://nss.cs.ubc.ca/nossdav2011/


NOSSDAV 2011 is the 21th anniversary of SIGMM's leading workshop on =
network and
operating systems support for digital audio and video. The workshop, =
hosted at
the University of British Columbia (UBC), will continue to focus on =
emerging
research topics, controversial ideas, and future research directions in =
the area
of multimedia systems research.

As in previous years, we will maintain the focused single-track format a =
setting
that stimulates lively discussions among the senior and junior =
participants.
NOSSDAV encourages experimental research based on real systems and real =
data sets.
Public availability of the source code and data sets discussed in papers =
presented
at NOSSDAV is highly encouraged.

For NOSSDAV 2011, we will accept papers on broad ranges of topics =
related to the
transmission and presentation of digital audio/video objects. We are =
particularly
interested in soliciting articles that discuss system-level support for =
distributed
social media, as well as papers that focus on enabling multimedia =
applications in
distributed cloud. Other topics of interest include (but are not =
restricted to):

* OS, middleware and network support
* Overlay networks
* Media streaming, distribution and storage support
* Web 2.0 systems and social networks
* Media sensor and ad hoc networks / embedded systems
* Multicore architecture support
* Wireless and mobile multimedia systems / network processor support
* Networked GPUs, graphics and virtual environments
* Networked games / real-time immersive systems
* Multimedia communications and system security
* Grid/Cloud computing support

Please contact the workshop co-chairs to check if your topic is within =
the scope of
NOSSDAV. Papers will be judged on their relevance, technical content and =
correctness,
and the clarity of presentation of the research. Papers should not be =
under review
at another venue nor previously published elsewhere.

Submissions should be at most SIX pages, using standard ACM proceedings =
style.
We expect these submissions to be the kernel of what will eventually =
lead to full-length
papers at high-quality conferences or journals.

Important Dates

Paper Submission Deadline: 24 Feb 2011
Decision Notification:     24 Mar 2011
Camera Ready Due:          7 Apr 2011

For more information, please visit the workshop website at
http://nss.cs.ubc.ca/nossdav2011/


From iesg-secretary@ietf.org  Mon Jan 10 11:43:31 2011
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Date: Mon, 10 Jan 2011 11:43:29 -0800
Cc: Internet Architecture Board <iab@iab.org>, avt mailing list <avt@ietf.org>, avt chair <avt-chairs@tools.ietf.org>, RFC Editor <rfc-editor@rfc-editor.org>
Subject: [AVT] Protocol Action: 'RTP Control Protocol (RTCP) Port for	Source-Specific Multicast (SSM) Sessions' to Proposed Standard	(draft-ietf-avt-rtcp-port-for-ssm-04.txt)
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The IESG has approved the following document:
- 'RTP Control Protocol (RTCP) Port for Source-Specific Multicast (SSM)
   Sessions'
  (draft-ietf-avt-rtcp-port-for-ssm-04.txt) as a Proposed Standard

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

The IESG contact persons are Robert Sparks and Gonzalo Camarillo.

A URL of this Internet Draft is:
http://datatracker.ietf.org/doc/draft-ietf-avt-rtcp-port-for-ssm/



Technical Summary

The Session Description Protocol (SDP) has an attribute that allows RTP
applications to specify an address and a port associated with the RTP
Control Protocol (RTCP) traffic.  In RTP-based source-specific
multicast (SSM) sessions, the same attribute is used to designate the
address and the RTCP port of the Feedback Target in the SDP description. 
However, the RTCP port associated with the SSM session itself cannot be
specified by the same attribute to avoid ambiguity, and thus, is
required to be derived from the "m=" line of the media description. 
Deriving the RTCP port from the "m=" line imposes an unnecessary
restriction.  This document removes this restriction by introducing a
new SDP attribute.


Working Group Summary

The document achieved consensus in the AVT working group. 

Document Quality

The document was reviewed by both the AVT and MMUSIC working groups.

Personnel

Keith Drage is the document shepherd. Robert Sparks is the responsible AD.


From csp@csperkins.org  Mon Jan 10 13:44:08 2011
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Thanks, these address my comments.
Colin


On 5 Jan 2011, at 15:36, Ali C. Begen (abegen) wrote:

> This has minor fixes Colin pointed out a few days ago. We are not =
aware of any remaining open issue and the draft should be ready for the =
chairs' review.
>=20
> Diff:
> =
https://datatracker.ietf.org/doc/draft-ietf-avt-ports-for-ucast-mcast-rtp/=
history/
>=20
> -acbegen
>=20
> -----Original Message-----
> From: IETF I-D Submission Tool [mailto:idsubmission@ietf.org]=20
> Sent: Wednesday, January 05, 2011 10:29 AM
> To: Ali C. Begen (abegen)
> Cc: Dan Wing (dwing); Tom.Van_Caenegem@alcatel-lucent.com
> Subject: New Version Notification for =
draft-ietf-avt-ports-for-ucast-mcast-rtp-11=20
>=20
>=20
> A new version of I-D, draft-ietf-avt-ports-for-ucast-mcast-rtp-11.txt =
has been successfully submitted by Ali Begen and posted to the IETF =
repository.
>=20
> Filename:	 draft-ietf-avt-ports-for-ucast-mcast-rtp
> Revision:	 11
> Title:		 Port Mapping Between Unicast and Multicast RTP =
Sessions
> Creation_date:	 2011-01-05
> WG ID:		 avt
> Number_of_pages: 35
>=20
> Abstract:
> This document presents a port mapping solution that allows RTP
> receivers to choose their own ports for an auxiliary unicast session
> in RTP applications using both unicast and multicast services.  The
> solution provides protection against denial-of-service or packet
> amplification attacks that could be used to cause one or more RTP
> packets to be sent to a victim client.
>=20
>=20
>=20
> The IETF Secretariat.
>=20
>=20
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt



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




From petithug@acm.org  Mon Jan 10 14:04:57 2011
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I just released a new version of this draft:

http://www.ietf.org/internet-drafts/draft-petithuguenin-avt-multiple-clock-rates-03.txt

I presented this draft during IETF 77, seeking WG adoption for this work.  There
was a positive response, but unfortunately the WG workload was too high at the
time for this.  So would the WG be interested to adopt this (I can present again
in Prague) this time, or should I try another way to have this published?

Thanks.

- -- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.


	Title           : Guidelines for Choosing RTP Control Protocol (RTCP) Canonical Names (CNAMEs)
	Author(s)       : A. Begen, et al.
	Filename        : draft-ietf-avt-rtp-cnames-04.txt
	Pages           : 11
	Date            : 2011-01-10

The RTP Control Protocol (RTCP) Canonical Name (CNAME) is a
persistent transport-level identifier for an RTP endpoint.  While the
Synchronization Source (SSRC) identifier of an RTP endpoint may
change if a collision is detected, or when the RTP application is
restarted, its RTCP CNAME is meant to stay unchanged, so that RTP
endpoints can be uniquely identified and associated with their RTP
media streams.  For proper functionality, RTCP CNAMEs should be
unique within the participants of an RTP session.  However, the
existing guidelines for choosing the RTCP CNAME provided in the RTP
standard are insufficient to achieve this uniqueness.  This memo
updates those guidelines to allow endpoints to choose unique RTCP
CNAMEs.

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

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ftp://ftp.ietf.org/internet-drafts/

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From frans.de.bont@philips.com  Tue Jan 11 01:47:36 2011
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From: "Bont, Frans de" <frans.de.bont@philips.com>
To: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>, "avt@ietf.org" <avt@ietf.org>
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Hi Keith,

Thanks for your extensive review.
Please find our comments below in-line.

Best regards,
Frans

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of DRA=
GE,
> Keith (Keith)
> Sent: 2010 Dec 06 12:59 AM
> To: avt@ietf.org
> Cc: draft-ietf-avt-rfc3016bis.all@tools.ietf.org
> Subject: Re: [AVT] WGLC on draft-ietf-avt-rfc3016bis
>
> I've reviewed the document, comments as follows
>
> 1)      This document has inherited a considerable problem from the previ=
ous
> RFC 3016, in that much of that document is written with strange RFC 2119
> usage, and pretty much entirely in the passive case. How readers assess
> conformance to this document I don't know. However, given that anyone who=
 is
> going to implement these parts of RFC 3016 will presumably already have d=
one
> so, I have limited my comments on this part to areas where I think the
> requirements are totally unclear, and specifically tagged them as on the
> existing text. If anyone thinks it would be useful to go further, then I =
can
> certainly submit a large number of further comments on the existing text,=
 but
> they are not included here.

The purpose of this document is mainly to correct some errors in RFC 3016, =
and to provide an extension to signal the presence of MPEG Surround audio d=
ata. Indeed existing users will use this, so we see no need to go much furt=
her with the level of the comments.

> 2)      (Editorial). Abstract.
>
>    Comments are solicited and should be addressed to the working group's
>    mailing list at avt@ietf.org and/or the author(s).
>
> Delete this text. I don't know why people include this in an internet-dra=
ft in
> the first place, given that it states the obvious, and it is not needed i=
n the
> RFC.

Will be deleted, Roni made the same comment.

> 3)      (Existing text). Section 1.1
>
>    The fragmentation rule recommends not to map more than one VOP in an
>    RTP packet so that the RTP timestamp uniquely indicates the VOP time
>    framing.  On the other hand, MPEG-4 video may generate VOPs of very
>    small size, in cases with an empty VOP (vop_coded=3D0) containing only
>    VOP header or an arbitrary shaped VOP with a small number of coding
>    blocks.  To reduce the overhead for such cases, the fragmentation
>    rule permits concatenating multiple VOPs in an RTP packet.  (See
>    fragmentation rule (4) in section 3.2 and marker bit and timestamp in
>    section 3.1.)
>
> The use of recommends here is a pseudo quote from later in the document. =
It
> would be better if it was a real quote as in the previous paragraph, thus=
 it
> should be modified as follows,
>
>    The fragmentation rule "Different VOPs SHOULD be fragmented into diffe=
rent
> RTP packets" is made
>    so that the RTP timestamp uniquely indicates the VOP time
>    framing.  On the other hand, MPEG-4 video may generate VOPs of very
>    small size, in cases with an empty VOP (vop_coded=3D0) containing only
>    VOP header or an arbitrary shaped VOP with a small number of coding
>    blocks.  To reduce the overhead for such cases, the fragmentation
>    rule permits concatenating multiple VOPs in an RTP packet.  (See
>    fragmentation rule (4) in section 3.2 and marker bit and timestamp in
>    section 3.1.)

Agreed.

> 4)      (Existing text). Section 1.2
>
>    in RTP transmission there is no need for the last two features.
>    Therefore, these two features MUST NOT be used in applications based
>    on RTP packetization specified by this document.  Since LATM has been
>    developed for only natural audio coding tools, i.e., not for
>    synthesis tools, it seems difficult to transmit Structured Audio (SA)
>    data and Text to Speech Interface (TTSI) data by LATM.  Therefore, SA
>    data and TTSI data MUST NOT be transported by the RTP packetization
>    in this document.
>
> Section 1.2 is an introductory section. Therefore it is inappropriate to =
start
> introducing RFC 2119 language at this stage. As far as I can see, these
> requirements are not elsewhere in the document, so we will need to create=
 a
> place for them. I suggest a new section 3 immediate before the existing
> section 3, and renumber accordingly (given that the numbering of the exis=
ting
> RFC 3016 has not been preserved.

A new section 3 will be created.

> Similar considerations apply to the following two paragraphs:
>
>    For transmission of scalable streams, audio data of each layer SHOULD
>    be packetized onto different RTP streams allowing for the different
>    layers to be treated differently at the IP level, for example via
>    some means of differentiated service.  On the other hand, all
>    configuration data of the scalable streams are contained in one LATM
>    configuration data "StreamMuxConfig" and every scalable layer shares
>    the StreamMuxConfig.  The mapping between each layer and its
>    configuration data is achieved by LATM header information attached to
>    the audio data.  In order to indicate the dependency information of
>    the scalable streams, the signaling mechanism as specified in
>    [RFC5583] SHOULD be used (see section 4.2).

This paragraph will be added to the new section 3.

>    For MPEG-4 Audio coding tools, as is true for other audio coders, if
>    the payload is a single audio frame, packet loss will not impair the
>    decodability of adjacent packets.  Therefore, the additional media
>    specific header for recovering errors will not be required for MPEG-4
>    Audio.  Existing RTP protection mechanisms, such as Generic Forward
>    Error Correction (RFC 5109 [RFC5109]) and Redundant Audio Data (RFC
>    2198 [RFC2198]), MAY be applied to improve error resiliency.

As this paragraph doesn't contain RFC 2119 language we feel that this parag=
raph can stay in the introduction.

> 5)      Section 1.3.
>
> I first looked for this at the end of the document, in some kind of appen=
dix,
> because that is where I expected to find it, and I wanted to check this f=
irst.
> So my first comment is that while there is no rule on this, I would prefe=
r it
> to appear at the end of the document.

Will be moved to the end of the document.

> 6)      Section 1.3.
>
> I do not think the list as currently drafted gives a very good indication=
 to
> existing implementors at what they need to look at in order to update the=
ir
> implementations. This should be redrafted to identify key things that are
> revised implementation will need to look at. For example:
>
>    o  The audio parameter "SBR-enabled" is not defined within RFC 3016
>       but used by 3GPP
>
> should become something like:
>
>    o  The use of an audio parameter "SBR-enabled" is now defined in this
> document, which is used by 3GPP implementations [informative reference to=
 3GPP
> specification here].

Agreed.

> 6)      Section 1.3. For the text:
>
>    Furthermore some comments have been addressed and signaling support
>    for MPEG surround [23003-1] was added.  It should be noted that the
>    audio payload format described here has some known limitations.  For
>    new system designs RFC 3640 [RFC3640] is recommended.
>
> Remove the contruct "It should be noted that the". It is mixing material =
that
> could be confused as normative with definitely informative material.

Agreed.

> It is also unclear how RFC 3640 is recommended, as the reference is
> informative, and this is the only place it is referenced. I assume you me=
an
> that new implementations would not be using this document at all. But tha=
t
> surely should be called out in the Abstract of this document.

The last two sentences of this paragraph will be moved to the Abstract.

> 7)      Section 2 (Editorial)
>
>    This memo makes use of terms, specified in [14496-2], [14496-3], and
>    [23003-1].  In addition, the following terms are used in this
>    document and have specific meaning within the context of this
>    document.
>
> Change to:
>
>    This document makes use of terms, specified in [14496-2], [14496-3], a=
nd
>    [23003-1].  In addition, the following terms are used in this
>    document and have specific meaning within the context of this
>    document.

Agreed.

> 8)      Section 3 (Existing text)
>
>    This section specifies RTP packetization rules for MPEG-4 Visual
>    content.  An MPEG-4 Visual bitstream is mapped directly onto RTP
>    packets without the addition of extra header fields or any removal of
>    Visual syntax elements.  The Combined Configuration/Elementary stream
>    mode MUST be used so that configuration information will be carried
>    to the same RTP port as the elementary stream. (see 6.2.1 "Start
>    codes" of ISO/IEC 14496-2 [14496-2]) The configuration information
>    MAY additionally be specified by some out-of-band means.  If needed
>    for an H.323 terminal, H.245 codepoint
>    "decoderConfigurationInformation" MUST be used for this purpose.  If
>    needed by systems using Media Type parameters and SDP parameters,
>    e.g., SIP and RTSP, the optional parameter "config" MUST be used to
>    specify the configuration information (see 5.1 and 5.2).
>
> The abbreviation "e.g." means for example, and "MUST" can never be and
> example. I supect the example here is only the string "e.g., SIP and RTSP=
" so
> I suggest that this bit is placed in parenthesis, as follows: "(e.g., SIP=
 and
> RTSP)"

You are correct, the draft will be updated accordingly.

> 9)      Section 3.2 (Existing text). In the following text:
>
>   (5) It is RECOMMENDED that a single video packet is sent as a single
>    RTP packet.  The size of a video packet SHOULD be adjusted in such a
>    way that the resulting RTP packet is not larger than the path-MTU.
>    Note: Rule (5) does not apply when the video packet is disabled by
>    the coder configuration (by setting resync_marker_disable in the VOL
>    header to 1), or in coding tools where the video packet is not
>    supported.  In this case, a VOP MAY be split at arbitrary byte-
>    positions.
>
> remove the mix of the word "note" with normative text. I suggest:
>
>   (5) It is RECOMMENDED that a single video packet is sent as a single
>    RTP packet.  The size of a video packet SHOULD be adjusted in such a
>    way that the resulting RTP packet is not larger than the path-MTU.
>    If the video packet is disabled by
>    the coder configuration (by setting resync_marker_disable in the VOL
>    header to 1), or in coding tools where the video packet is not
>    supported, a VOP MAY be split at arbitrary byte-
>    positions.

Agreed.

> 10)     Section 3.3 (Existing text. In the following text:
>
>    When concatenating more than one video packets into an RTP packet,
>    VOP header or video_packet_header() shall not be placed in the middle
>    of the RTP payload.  The packetization as in (b) is not allowed by
>    criterion (2) due to the aspect of the error resiliency.  Comparing
>    this example with Figure 2(d), although two video packets are mapped
>    onto two RTP packets in both cases, the packet-loss resiliency is not
>    identical.  Namely, if the second RTP packet is lost, both video
>    packets 1 and 2 are lost in the case of Figure 3(b) whereas only
>    video packet 2 is lost in the case of Figure 2(d).
>
> is the "shall" here denoting a requirements (in which case it should be
> capitalised and replaced with "MUST", and moved to a section that is not
> titled "Example") or is it denoting something else, e.g. an impossibility=
, in
> which case change the modal auxiliary to something else.

The requirements used here are already listed in (old) section 3.2 ".. is p=
rohibited by criterion of 3.2".
The "shall not be placed" will be replaced by "are not allowed to be placed=
".

> 11)     Section 4.2, 1st paragraph:
>
>    Payload Type (PT): The assignment of an RTP payload type for this new
>    packet format is outside the scope of this document, and will not be
>    specified here.  It is expected that the RTP profile for a particular
>    class of applications will assign a payload type for this encoding,
>    or if that is not done then a payload type in the dynamic range shall
>    be chosen by means of an out-of-band signaling protocol (e.g., H.245,
>    SIP, etc).  In the dynamic assignment of RTP payload types for
>    scalable streams, a different value SHOULD be assigned to each layer.
>    The dependency relationships between the enhance layer and the base
>    layer SHOULD be signaled as specified in [RFC5583].  An example of
>    the use of such signaling for scalable audio streams can be found in
>    [RFC5691].
>
> Write the above new text in the active, i.e. "In the dynamic assignment o=
f RTP
> payload types for scalable streams, the server SHOULD assign a different =
value
> to each layer. The server SHOULD signal ...".

Agreed.

> Under what circumstances does the "SHOULD" not apply. It is good practice=
 when
> using "SHOULD" to give guidance tot he reader of the conditions under whi=
ch it
> is safe to ignore the requirement. Otherwise may implementors treat the w=
ord
> as a straight option and ignore it.

Will be changed into "SHALL".

> It is also noted that a requirement specifying the assignment of the payl=
oad
> type is inconsistent with the first sentence, which says "assignment of a=
n RTP
> payload type for this new packet format is outside the scope of this
> document". This sentence will therefore need to be altered.

Will be changed into "... is outside the scope of this document, and will o=
nly be restricted here.".

> 12)     Section 4.2 (Existing text). Is the following meant to be normati=
ve:
>
>    Timestamp: The timestamp indicates the sampling instance of the first
>    audio frame contained in the RTP packet.  Timestamps are recommended
>    to start at a random value for security reasons.
>
> It is currently drafted as such, but "recommended" is lower case.

Will be made upper case.

> 13)     Section 4.3 (Existing text). What is the meaning of the final
> sentence:
>
>    If it cannot, the audioMuxElement MAY
>    be fragmented and spread across multiple packets.
>
> As written it is telling me I have an option to fragment it and spread it
> across multiple packets. Is that what is meant.

Indeed, there is no option. The "MAY" will be changed into "SHALL".

> 14)     Section 5.1. The following text:
>
>    Note, any unspecified parameter MUST be ignored by the receiver to
>    ensure that additional parameters can be added in any future revision
>    of this specification.
>
> has the word "MUST" mandatory requirement, following the word "note" whic=
h
> many people interpret as informative. Assuming a mandatory requirement is
> meant, delete the word "Note". The text should also be redrafted in the
> active, e.g.:
>
>    The receiver MUST ignore any unspecified parameter, to
>    ensure that additional parameters can be added in any future revision
>    of this specification.

Agreed.

> This is also a registration section. Surely normative requirements on the
> implementation should be elsewhere in the document, so I suggest this tex=
t is
> moved to another more appropriate section.

We checked RFC 3640 and it has this text also in the registration section. =
Not sure where else to place this text.

> 15)     Section 5.1.
>
>    Published specification:
>
>       The specifications for MPEG-4 Visual streams are presented in ISO/
>       IEC 14469-2 [14496-2].  The RTP payload format is described in RFC
>       XXXX.
>
> Why not just say: "The RTP payload format is described in this document."
>
> Same change for the following two entries, and also in section 5.3.

Agreed.

> 16)     Section 5.3, 5.4.1.3 and 5.4.1.4. This document does not define t=
he
> coding of the SBR-enabled parameter. I get the impression from reading th=
e
> document that it is defined elsewhere, but no reference is made to the po=
int
> where it is defined. That would surely need to be a normative reference.

The "SBR-enabled" parameter is defined as a boolean parameter in 5.3. Do we=
 miss your point?

> 17)     Section 5.3.
>
>    Note, any unspecified parameter MUST be ignored by the receiver to
>    ensure that additional parameters can be added in any future revision
>    of this specification.
>
> See comments on similar text at the start of section 4.1.

And our response at 14)

> 18)     Section 5.3 (Existing text):
>
>   Required parameters:
>
>       rate: the rate parameter indicates the RTP time stamp clock rate.
>       The default value is 90000.  Other rates MAY be specified only if
>       they are set to the same value as the audio sampling rate (number
>       of samples per second).
>
> Change "specified" to "indicated". The AVT working group is the specifier=
.

Agreed.

> 19)     Section 5.3 (Existing text). There are a number of lower case "sh=
all"
> here. Do these constitute requirements. How is the reader of this
> specification expected to understand and interpret them.
>
> If they are not requirements, then change them.

Will be made upper case.

> 19)     Section 5.3 (existing text)
>
>       MPS-profile-level-id: a decimal representation of the MPEG
>       Surround Profile Level indication as defined in ISO/IEC 14496-3
>       [14496-3].  This parameter indicates the MPEG Surround profile and
>       level that the decoder must be capable in order to decode the
>       stream.
>
> What is the meaning of the final sentence?

This sentence will be modified into.
"This parameter indicates the support of the MPEG Surround profile and leve=
l by the decoder to be capable to decode the stream."

> 20)     Section 5.3 (existing text)
>
>      ptime: RECOMMENDED duration of each packet in milliseconds.
>
> Why is this a requirement. Hows does an implementation conform to it.

"RECOMMENDED" will be removed.

> 21)     Section 5.3
>
>      If this parameter is set to 0, a decoder SHALL expect that SBR is
>       not used.  If this parameter is set to 1, a decoder SHOULD
>       upsample the audio data with the SBR tool, regardless whether SBR
>       data is present in the stream or not.
>
> First sentence is not written as a conformable requirement. What does the
> decoder need to do to conform? For the second sentence, see comments earl=
ier
> on use of SHOULD.

The SHALL and SHOULD will be replaced by MAY and CAN, respectively.

> 22)     Section 5.3
>
>       If the presence of SBR can not be detected from out-of-band
>       configuration and the SBR-enabled parameter is not present, the
>       parameter defaults to 1 for an SBR-capable decoder.  If the
>       resulting output sampling rate or the computational complexity is
>       not supported, the SBR tool may be disabled or run in downsampled
>       mode.
>
> Change "may" to "can" to indicate that it is definitely not a conformance
> requirement.

Agreed.

> 23)     Section 5.4.1.3
>
>    In this example, the presence of SBR can not be determined by the SDP
>    parameter set.  The clock rate represents the core codec sampling
>    rate.  An SBR enabled decoder SHOULD use the SBR tool to upsample the
>    audio data if complexity and resulting output sampling rate permits.
>
> This section is entitled "Example". Such sections ideally should not incl=
ude
> conformable requirements. So if this is a requirement then move it somewh=
ere
> else. Also comments elsewhere about the use of SHOULD also apply here.

The "SHOULD" will be replaced by "can", as it is not a requirement.

> 24)     Section 5.4.1.4
>
>    In this example, the clock rate is still 24000 and this information
>    should be used for RTP timestamp calculation.  The value of 24000 is
>    used to support old AAC decoders.  This makes the decoder supporting
>    only AAC understand the HE AAC coded data, although only plain AAC is
>    supported.  A HE AAC decoder is able to generate output data with the
>    SBR sampling rate.
>
> Change "should be" to "is" to avoid confusion with a conformable requirem=
ent.

Agreed.

> 25)     Section 6.
>
>    This memo defines additional optional format parameters to the Media
>    Type "audio" and its subtype "MP4A-LATM", as defined in RFC XXXX.
>    The Media Type parameters are defined in sections 5.1 and 5.3 of RFC
>    XXXX.
>
> Just say "sections 5.1 and 5.3 of this document".

A similar comment was made by Roni.
The proposal is to change it into.

   This document updates the media subtypes "MP4A-LATM" and "MP4V-ES"
   from RFC 3016.  The new registrations are in Section 6.1 and
   Section 6.3 of this document.

> 26)     Section 6.1.
>
>    This memo defines the following additional optional parameters which
>    SHOULD be used if SBR or MPEG Surround data is present inside the
>    payload of an AAC elementary stream.
>
> We do not need a SHOULD (or any other RFC 2119 language in relation to an=
 IANA
> considerations section.
>
> I suspect this is just superfluous, and not a real conformable requiremen=
t on
> implementations.

(old) Sections 6.1 and 6.2 will be removed, see a comment made by Roni.

> 27)     Section 6
>
> An IANA considerations section should basically consists of a set of
> instructions to IANA as to what they need to include in their tables. See
> http://www.iana.org/protocols/
>
> I cannot ascertain from this section whether they need to do work.
>
> Are there any existing RFC 3013 registrations where the reference needs t=
o be
> updated?
>
> Are there any of these parameters that actually need registration?

See our remark at 25)

>
>
> regards
>
> Keith
>
>
> ________________________________
>
>         From: Roni Even [mailto:Even.roni@huawei.com]
>         Sent: Tuesday, November 23, 2010 2:00 PM
>         To: avt@ietf.org
>         Cc: DRAGE, Keith (Keith); draft-ietf-avt-rfc3016bis.all@tools.iet=
f.org
>         Subject: WGLC on draft-ietf-avt-rfc3016bis
>
>
>
>         Hi,
>
>         I would like to start a working group last call on
> http://tools.ietf.org/html/draft-ietf-avt-rfc3016bis-01
>
>
>
>         RTP Payload Format for MPEG-4 Audio/Visual Streams
>
>
>
>         The WGLC will end on December 13th , 2010
>
>         Please review the draft and send comments to the list
>
>
>
>         Note that section 1.3 and 1.4 discuss the changes from RFC3016
>
>
>
>         Roni Even
>
>         AVT co- ass=3DMsoNormal>
>
>
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt

The information contained in this message may be confidential and legally p=
rotected under applicable law. The message is intended solely for the addre=
ssee(s). If you are not the intended recipient, you are hereby notified tha=
t any use, forwarding, dissemination, or reproduction of this message is st=
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.


	Title           : RTP Payload Format for MPEG-4 Audio/Visual Streams
	Author(s)       : M. Schmidt, et al.
	Filename        : draft-ietf-avt-rfc3016bis-02.txt
	Pages           : 32
	Date            : 2011-01-11

This document describes Real-Time Transport Protocol (RTP) payload
formats for carrying each of MPEG-4 Audio and MPEG-4 Visual
bitstreams without using MPEG-4 Systems.  For the purpose of directly
mapping MPEG-4 Audio/Visual bitstreams onto RTP packets, it provides
specifications for the use of RTP header fields and also specifies
fragmentation rules.  It also provides specifications for Media Type
registration and the use of Session Description Protocol (SDP).  The
audio payload format described in this document has some limitations.
for new system designs [RFC3640] is preferred.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avt-rfc3016bis-02.txt

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From frans.de.bont@philips.com  Tue Jan 11 02:57:40 2011
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From: "Bont, Frans de" <frans.de.bont@philips.com>
To: "avt@ietf.org" <avt@ietf.org>
Date: Tue, 11 Jan 2011 11:59:30 +0100
Thread-Topic: [AVT] I-D Action:draft-ietf-avt-rfc3016bis-02.txt
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Hi all,

The WGLC ended last month and a number of reviews have been received,
thanks for those. We have responded to the individual reviews on the
list and have updated the draft accordingly. The new draft has been
submitted and can be found using the link in the announcement below.

Below a summary of the major changes.

1. The paragraphs including RFC 2119 keywords have been moved from the
   Introduction to a new section entitled "LATM Restrictions for RTP
   Packetization of MPEG-4 Audio Bitstreams".

2. The restrictions on the RTP packet format for MPEG-4 Audio bitstreams
   has been updated to make them better readable and unambiguous.

3. The IANA considerations section has been reduced to what is really
   needed.

4. The Security Considerations section been updated with text from
   recent payload format RFC's.

5. The differences with respect to RFC 3016 have been moved to a new
   section at the end of the draft.

6. An ackowledgement section has been added.

Your comments are welcome!

Best regards,
Frans


> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
> Internet-Drafts@ietf.org
> Sent: 2011 Jan 11 11:00 AM
> To: i-d-announce@ietf.org
> Cc: avt@ietf.org
> Subject: [AVT] I-D Action:draft-ietf-avt-rfc3016bis-02.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 Working Group of t=
he
> IETF.
>
>
>       Title           : RTP Payload Format for MPEG-4 Audio/Visual Stream=
s
>       Author(s)       : M. Schmidt, et al.
>       Filename        : draft-ietf-avt-rfc3016bis-02.txt
>       Pages           : 32
>       Date            : 2011-01-11
>
> This document describes Real-Time Transport Protocol (RTP) payload format=
s for
> carrying each of MPEG-4 Audio and MPEG-4 Visual bitstreams without using =
MPEG-
> 4 Systems.  For the purpose of directly mapping MPEG-4 Audio/Visual bitst=
reams
> onto RTP packets, it provides specifications for the use of RTP header fi=
elds
> and also specifies fragmentation rules.  It also provides specifications =
for
> Media Type registration and the use of Session Description Protocol (SDP)=
.
> The audio payload format described in this document has some limitations.
> for new system designs [RFC3640] is preferred.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-avt-rfc3016bis-02.txt
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> Below is the data which will enable a MIME compliant mail reader
> implementation to automatically retrieve the ASCII version of the Interne=
t-
> Draft.

The information contained in this message may be confidential and legally p=
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Subject: [AVT] I-D Action:draft-ietf-avt-rtp-ipmr-15.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.


	Title           : RTP Payload Format for IP-MR Speech Codec
	Author(s)       : D. Yudin
	Filename        : draft-ietf-avt-rtp-ipmr-15.txt
	Pages           : 20
	Date            : 2011-01-11

This document specifies the payload format for packetization of
SPIRIT IP-MR encoded speech signals into the real-time transport
protocol (RTP). The payload format supports transmission of multiple
frames per packet and introduced redundancy for robustness against
packet loss and bit errors.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-ipmr-15.txt

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From albrecht.schwarz@alcatel-lucent.com  Tue Jan 11 07:17:21 2011
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From: "Schwarz, Albrecht (Albrecht)" <albrecht.schwarz@alcatel-lucent.com>
To: "Belling, Thomas (NSN - DE/Munich)" <thomas.belling@nsn.com>, IETF AVT WG <avt@ietf.org>, Magnus Westerlund <magnus.westerlund@ericsson.com>
Date: Tue, 11 Jan 2011 16:19:26 +0100
Thread-Topic: [AVT] draft-ietf-avt-ecn-for-rtp: Is AVPF required for leap-of-faith initialisation?
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Subject: Re: [AVT] draft-ietf-avt-ecn-for-rtp: Is AVPF required for	leap-of-faith initialisation?
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>It would be quite helpful to have it available for the upcoming SA4 meetin=
g.

Thomas, there wasn't any update nor any ECN contribution to SA4#62 this wee=
k, right?

Thanks,
Albrecht

________________________________
From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Belli=
ng, Thomas (NSN - DE/Munich)
Sent: Dienstag, 21. Dezember 2010 18:26
To: IETF AVT WG; Magnus Westerlund
Subject: [AVT] draft-ietf-avt-ecn-for-rtp: Is AVPF required for leap-of-fai=
th initialisation?


Dear Magnus,

allow me to continue a discussion we had about a month ago.

You were suggesting that the leap of faith initialisation method requires A=
VPF because this allows XR ECN summary reports can then be sent quicker and=
 shorter reporting interval than the average 5secs can be negotiated. Corre=
ct?

The draft contains the following related procedure in 6.2.3:

   If the sender marks all packets as ECT while transmitting on a path

   that contains an ECN-blocking middlebox, then receivers downstream of

   that middlebox will not receive any RTP data packets from the sender,

   and hence will not consider it to be an active RTP SSRC.  The sender

   can detect this and revert to sending packets without ECT marks,

   since RTCP SR/RR packets from such receivers will either not include

   a report for sender's SSRC, or will report that no packets have been

   received, but this takes at least one RTCP reporting interval.  It

   should be noted that a receiver might generate its first RTCP packet

   immediately on joining a unicast session, or very shortly after

   joining a RTP/AVPF session, before it has had chance to receive any

   data packets.  A sender that receives RTCP SR/RR packet indicating

   lack of reception by a receiver SHOULD therefore wait for a second

   RTCP report from that receiver to be sure that the lack of reception

   is due to ECT-marking.  Since this recovery process can take several

   tens of seconds, during which time the RTP session is unusable for

   media, it is NOT RECOMMENDED that the leap-of-faith ECT initiation

   method be used in environments where ECN-blocking middleboxes are

   likely to be present.

So there is no particular trigger for the receiver to sent SR/RR reports th=
at would allow it to benefit from the more flexible timing rules of AVPF.

One could think about using the AVPF early RTCP feedback if the sender does=
 not receive any packets for a while at call setup, but the duration is qui=
te open and it might also be part of normal operation that no packets are r=
eceived - consider early media that are usually only transmitted in backwar=
d direction as an example.

You suggested to negotiate a higher RTCP transmission rate as a remedy. In =
my understanding this would work both with AVP and AVPF using SDP RTCP band=
width modifiers.

But I am not convinced this is a good idea given the extra required bandwid=
th. There is a clear wording already that the leap of faith method is not  =
recommended in environments where ECN-blocking middleboxes are likely to be=
 present. It is really meant for operator controlled environments where suc=
h problems will hardly occur. Is it really worth spending more bandwidth to=
 improve the handling of a very unlikely event? If there is a real risk tha=
t ECN-marked packets are blocked in a network, other initialisation methods=
 are preferable anyway.

As said, I also still doubt that you would have any benefit from AVPF for t=
hat initialisation method. But even I only fail to understand the advantage=
s you had in mind (do you assume some particular receiver behaviour describ=
ed nowhere up to now?),  I doubt this would be worth the extra complexity, =
only to improve  the handling of a very unlikely error. Again, in an networ=
k where there is a real risk that ECN-marked packets are blocked, other ini=
tialisation methods are preferable anyway.



As an unrelated question, are you working on an update of the draft?

It would be quite helpful to have it available for the upcoming SA4 meeting=
. I would also welcome if you could share the draft with me for a quick rev=
iew before distributing it further, at least if time allows.

Regards, Thomas

--_000_5F7BCCF5541B7444830A2288ABBEBC961D4FED0874FRMRSSXCHMBSD_
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>draft-ietf-avt-ecn-for-rtp: Is AVPF required for leap-of=
-faith initialisation?</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2900.6036" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><FONT face=3D"Trebuchet MS" color=3D#800000 siz=
e=3D2><SPAN=20
class=3D433081615-11012011>&gt;<FONT size=3D3><FONT color=3D#000000><SPAN=20
lang=3Den-gb><FONT face=3DCalibri>It would be quite</FONT></SPAN><SPAN=20
lang=3Dde></SPAN><SPAN lang=3Den-gb><FONT face=3D"Times New Roman"> </FONT>=
<FONT=20
face=3DCalibri>helpful</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-=
gb><FONT=20
face=3DCalibri></FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb><FON=
T=20
face=3D"Times New Roman"> </FONT><FONT face=3DCalibri>to have it available =
for the=20
upcoming SA4 meeting. </FONT></SPAN></FONT></FONT></SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3D"Trebuchet MS" color=3D#800000=20
size=3D2></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D433081615-11012011><FONT face=3D"=
Trebuchet MS"=20
color=3D#800000 size=3D2>Thomas, there wasn't any update nor any ECN contri=
bution to=20
SA4#62 this week, right?</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D433081615-11012011><FONT face=3D"=
Trebuchet MS"=20
color=3D#800000 size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D433081615-11012011><FONT face=3D"=
Trebuchet MS"=20
color=3D#800000 size=3D2>Thanks,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D433081615-11012011><FONT face=3D"=
Trebuchet MS"=20
color=3D#800000 size=3D2>Albrecht</FONT></SPAN></DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #800000 2px soli=
d; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> avt-bounces@ietf.org=20
  [mailto:avt-bounces@ietf.org] <B>On Behalf Of </B>Belling, Thomas (NSN -=
=20
  DE/Munich)<BR><B>Sent:</B> Dienstag, 21. Dezember 2010 18:26<BR><B>To:</B=
>=20
  IETF AVT WG; Magnus Westerlund<BR><B>Subject:</B> [AVT]=20
  draft-ietf-avt-ecn-for-rtp: Is AVPF required for leap-of-faith=20
  initialisation?<BR></FONT><BR></DIV>
  <DIV></DIV><!-- Converted from text/rtf format -->
  <P dir=3Dltr><SPAN lang=3Dde><FONT face=3DCalibri>Dear Magnus,</FONT></SP=
AN></P>
  <P dir=3Dltr><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb><FONT face=3DCalib=
ri>allow me=20
  to continue a discussion we ha</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN=
=20
  lang=3Den-gb><FONT face=3DCalibri>d</FONT></SPAN><SPAN lang=3Dde></SPAN><=
SPAN=20
  lang=3Den-gb><FONT face=3DCalibri> about a month ago.</FONT></SPAN></P>
  <P dir=3Dltr><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb><FONT=20
  face=3DCalibri>Yo</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb>=
<FONT=20
  face=3DCalibri>u</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb><=
FONT=20
  face=3DCalibri> were suggesting that the leap of faith initialisation=20
  m</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb><FONT=20
  face=3DCalibri>e</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb><=
FONT=20
  face=3DCalibri>thod requires AVPF because</FONT></SPAN><SPAN=20
  lang=3Dde></SPAN><SPAN lang=3Den-gb> <FONT face=3DCalibri>this allows XR =
ECN summary=20
  reports can then be sen</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN=20
  lang=3Den-gb><FONT face=3DCalibri>t quicker and short</FONT><FONT face=3D=
Calibri>er=20
  reporting</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb> <FONT=20
  face=3DCalibri>interval</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3D=
en-gb><FONT=20
  face=3DCalibri></FONT> <FONT face=3DCalibri>than</FONT> <FONT face=3DCali=
bri>the=20
  average 5secs can</FONT><FONT face=3DCalibri> be negotiated.</FONT></SPAN=
><SPAN=20
  lang=3Dde></SPAN><SPAN lang=3Den-gb><FONT face=3DCalibri>=20
  Correct?</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb></SPAN></=
P>
  <P dir=3Dltr><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb></SPAN></P>
  <P dir=3Dltr><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb><FONT face=3DCalib=
ri>The draft=20
  contains the following related</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN=
=20
  lang=3Den-gb> <FONT face=3DCalibri>procedure</FONT></SPAN><SPAN=20
  lang=3Dde></SPAN><SPAN lang=3Den-gb><FONT face=3DCalibri> in</FONT></SPAN=
><SPAN=20
  lang=3Dde></SPAN><SPAN lang=3Den-gb> <FONT face=3DCalibri>6.2.3:</FONT></=
SPAN></P>
  <P dir=3Dltr><SPAN lang=3Dde></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3De=
n-gb><FONT=20
  face=3D"Courier New" size=3D2>&nbsp;&nbsp;</FONT></SPAN><SPAN lang=3Dde><=
/SPAN><SPAN=20
  lang=3Dde></SPAN><SPAN lang=3Den-gb> <FONT face=3D"Courier New" size=3D2>=
If the sender=20
  marks all packets as ECT while transmitting on a path</FONT></SPAN></P>
  <P dir=3Dltr><SPAN lang=3Den-gb><FONT face=3D"Courier New" size=3D2>&nbsp=
;&nbsp; that=20
  contains an ECN-blocking middlebox, then receivers downstream=20
  of</FONT></SPAN></P>
  <P dir=3Dltr><SPAN lang=3Den-gb><FONT face=3D"Courier New" size=3D2>&nbsp=
;&nbsp; that=20
  middlebox will not receive any RTP data packets from the=20
  sender,</FONT></SPAN></P>
  <P dir=3Dltr><SPAN lang=3Den-gb><FONT face=3D"Courier New" size=3D2>&nbsp=
;&nbsp; and=20
  hence will not consider it to be an active RTP SSRC.&nbsp; The=20
  sender</FONT></SPAN></P>
  <P dir=3Dltr><SPAN lang=3Den-gb><FONT face=3D"Courier New" size=3D2>&nbsp=
;&nbsp; can=20
  detect this and revert to sending packets without ECT marks,</FONT></SPAN=
></P>
  <P dir=3Dltr><SPAN lang=3Den-gb><FONT face=3D"Courier New" size=3D2>&nbsp=
;&nbsp; since=20
  RTCP SR/RR packets from such receivers will either not=20
  include</FONT></SPAN></P>
  <P dir=3Dltr><SPAN lang=3Den-gb><FONT face=3D"Courier New" size=3D2>&nbsp=
;&nbsp; a=20
  report for sender's SSRC, or will report that no packets have=20
  been</FONT></SPAN></P>
  <P dir=3Dltr><SPAN lang=3Den-gb><FONT face=3D"Courier New" size=3D2>&nbsp=
;&nbsp;=20
  received, but this takes at least one RTCP reporting interval.&nbsp;=20
  It</FONT></SPAN></P>
  <P dir=3Dltr><SPAN lang=3Den-gb><FONT face=3D"Courier New" size=3D2>&nbsp=
;&nbsp;=20
  should be noted that a receiver might generate its first RTCP=20
  packet</FONT></SPAN></P>
  <P dir=3Dltr><SPAN lang=3Den-gb><FONT face=3D"Courier New" size=3D2>&nbsp=
;&nbsp;=20
  immediately on joining a unicast session, or very shortly=20
  after</FONT></SPAN></P>
  <P dir=3Dltr><SPAN lang=3Den-gb><FONT face=3D"Courier New" size=3D2>&nbsp=
;&nbsp;=20
  joining a RTP/AVPF session, before it has had chance to receive=20
  any</FONT></SPAN></P>
  <P dir=3Dltr><SPAN lang=3Den-gb><FONT face=3D"Courier New" size=3D2>&nbsp=
;&nbsp; data=20
  packets.&nbsp; A sender that receives RTCP SR/RR packet=20
  indicating</FONT></SPAN></P>
  <P dir=3Dltr><SPAN lang=3Den-gb><FONT face=3D"Courier New" size=3D2>&nbsp=
;&nbsp; lack=20
  of reception by a receiver SHOULD therefore wait for a=20
second</FONT></SPAN></P>
  <P dir=3Dltr><SPAN lang=3Den-gb><FONT face=3D"Courier New" size=3D2>&nbsp=
;&nbsp; RTCP=20
  report from that receiver to be sure that the lack of=20
  reception</FONT></SPAN></P>
  <P dir=3Dltr><SPAN lang=3Den-gb><FONT face=3D"Courier New" size=3D2>&nbsp=
;&nbsp; is=20
  due to ECT-marking.&nbsp; Since this recovery process can take=20
  several</FONT></SPAN></P>
  <P dir=3Dltr><SPAN lang=3Den-gb><FONT face=3D"Courier New" size=3D2>&nbsp=
;&nbsp; tens=20
  of seconds, during which time the RTP session is unusable=20
for</FONT></SPAN></P>
  <P dir=3Dltr><SPAN lang=3Den-gb><FONT face=3D"Courier New" size=3D2>&nbsp=
;&nbsp;=20
  media, it is NOT RECOMMENDED that the leap-of-faith ECT=20
  initiation</FONT></SPAN></P>
  <P dir=3Dltr><SPAN lang=3Den-gb><FONT face=3D"Courier New" size=3D2>&nbsp=
;&nbsp;=20
  method be used in environments where ECN-blocking middleboxes=20
  are</FONT></SPAN></P>
  <P dir=3Dltr><SPAN lang=3Den-gb><FONT face=3D"Courier New"=20
  size=3D2>&nbsp;&nbsp;</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Dde=
> <FONT=20
  face=3D"Courier New" size=3D2>likely to be present.</FONT></SPAN></P>
  <P dir=3Dltr><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb></SPAN></P>
  <P dir=3Dltr><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb></SPAN></P>
  <P dir=3Dltr><SPAN lang=3Den-gb><FONT face=3DCalibri>So there is=20
  no</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb> <FONT=20
  face=3DCalibri>particular</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN=20
  lang=3Den-gb><FONT face=3DCalibri> trigger for the receiver to=20
  sent</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb> <FONT=20
  face=3DCalibri>SR/RR reports</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN=20
  lang=3Den-gb><FONT face=3DCalibri> that would allow i</FONT></SPAN><SPAN=
=20
  lang=3Dde></SPAN><SPAN lang=3Den-gb><FONT face=3DCalibri>t to=20
  benefit</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb><FONT face=
=3DCalibri>=20
  from the more flexible timing rules of AVPF.</FONT></SPAN><SPAN=20
  lang=3Dde></SPAN><SPAN lang=3Den-gb></SPAN></P>
  <P dir=3Dltr><SPAN lang=3Den-gb><FONT face=3DCalibri>One could think abou=
t=20
  using</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb> <FONT=20
  face=3DCalibri>the AVPF</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3D=
en-gb>=20
  <FONT face=3DCalibri>early RTCP feedback if the sender does not=20
  receive</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb> <FONT=20
  face=3DCalibri>any packets for a while</FONT></SPAN><SPAN lang=3Dde></SPA=
N><SPAN=20
  lang=3Den-gb><FONT face=3DCalibri> at call setup</FONT></SPAN><SPAN=20
  lang=3Dde></SPAN><SPAN lang=3Den-gb><FONT face=3DCalibri>, but the durati=
on is quite=20
  open an</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb><FONT=20
  face=3DCalibri>d it</FONT> <FONT face=3DCalibri>might also be</FONT><FONT=
=20
  face=3DCalibri> part of normal operation</FONT><FONT face=3DCalibri> that=
</FONT>=20
  <FONT face=3DCalibri>no packets are received</FONT></SPAN><SPAN=20
  lang=3Dde></SPAN><SPAN lang=3Den-gb> <FONT face=3DCalibri>&#8211;</FONT><=
/SPAN><SPAN=20
  lang=3Dde></SPAN><SPAN lang=3Den-gb><FONT face=3DCalibri>=20
  consider</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb> <FONT=20
  face=3DCalibri>early media that are usually only transmitted in backward=
=20
  direction as an example.</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN=20
  lang=3Den-gb></SPAN></P>
  <P dir=3Dltr><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb></SPAN></P>
  <P dir=3Dltr><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb><FONT face=3DCalib=
ri>You=20
  suggested to negotiat</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den=
-gb><FONT=20
  face=3DCalibri>e a higher RTCP transmission r</FONT><FONT face=3DCalibri>=
ate as=20
  a</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb> <FONT=20
  face=3DCalibri>remedy</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den=
-gb>.<FONT=20
  face=3DCalibri> In my understanding this would work both with AVP and AVP=
F=20
  usin</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb><FONT=20
  face=3DCalibri>g</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb><=
FONT=20
  face=3DCalibri> SDP RTCP bandwidth mo</FONT></SPAN><SPAN lang=3Dde></SPAN=
><SPAN=20
  lang=3Den-gb><FONT face=3DCalibri>difiers.</FONT></SPAN><SPAN lang=3Dde><=
/SPAN><SPAN=20
  lang=3Den-gb></SPAN></P>
  <P dir=3Dltr><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb><FONT face=3DCalib=
ri>But I am=20
  not convinced this is a go</FONT><FONT face=3DCalibri>o</FONT><FONT=20
  face=3DCalibri>d idea</FONT> <FONT face=3DCalibri>given the extra require=
d=20
  bandwidth</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb>.</SPAN>=
<SPAN=20
  lang=3Dde></SPAN><SPAN lang=3Den-gb><FONT face=3DCalibri></FONT></SPAN><S=
PAN=20
  lang=3Dde></SPAN><SPAN lang=3Den-gb> <FONT face=3DCalibri>There is a clea=
r=20
  wording</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb> <FONT=20
  face=3DCalibri>already</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3De=
n-gb> <FONT=20
  face=3DCalibri>that the</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3D=
en-gb><FONT=20
  face=3DCalibri></FONT> <FONT face=3DCalibri>leap of faith method=20
  is</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb> <FONT=20
  face=3DCalibri>not</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN=20
  lang=3Den-gb>&nbsp;<FONT face=3DCalibri> recommended</FONT></SPAN><SPAN=20
  lang=3Dde></SPAN><SPAN lang=3Den-gb> <FONT face=3DCalibri>in</FONT> <FONT=
=20
  face=3DCalibri>environments where</FONT></SPAN><SPAN lang=3Dde></SPAN><SP=
AN=20
  lang=3Den-gb> <FONT face=3DCalibri>ECN-blocking middleboxes are</FONT><FO=
NT=20
  face=3DCalibri> likely to be present</FONT></SPAN><SPAN lang=3Dde></SPAN>=
<SPAN=20
  lang=3Den-gb><FONT face=3DCalibri>. It is really me</FONT><FONT=20
  face=3DCalibri>a</FONT><FONT face=3DCalibri>nt for operator controlled</F=
ONT>=20
  <FONT face=3DCalibri>environments where such problems</FONT></SPAN><SPAN=
=20
  lang=3Dde></SPAN><SPAN lang=3Den-gb> <FONT face=3DCalibri>will</FONT> <FO=
NT=20
  face=3DCalibri>hardly occur.</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN la=
ng=3Den-gb>=20
  <FONT face=3DCalibri>Is</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3D=
en-gb><FONT=20
  face=3DCalibri> it really wo</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN=20
  lang=3Den-gb><FONT face=3DCalibri>r</FONT></SPAN><SPAN lang=3Dde></SPAN><=
SPAN=20
  lang=3Den-gb><FONT face=3DCalibri>th spending more bandwidth</FONT></SPAN=
><SPAN=20
  lang=3Dde></SPAN><SPAN lang=3Den-gb> <FONT face=3DCalibri>to improve the =
handling of=20
  a very unlikely</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb> <=
FONT=20
  face=3DCalibri>event</FONT><FONT face=3DCalibri>?</FONT></SPAN><SPAN=20
  lang=3Dde></SPAN><SPAN lang=3Den-gb><FONT face=3DCalibri> If there is a r=
eal risk=20
  that ECN-marked packets are blocked i</FONT></SPAN><SPAN lang=3Dde></SPAN=
><SPAN=20
  lang=3Den-gb><FONT face=3DCalibri>n</FONT></SPAN><SPAN lang=3Dde></SPAN><=
SPAN=20
  lang=3Den-gb><FONT face=3DCalibri> a network</FONT></SPAN><SPAN=20
  lang=3Dde></SPAN><SPAN lang=3Den-gb><FONT face=3DCalibri>, other initiali=
sation=20
  methods are preferable anyway.</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN=
=20
  lang=3Den-gb></SPAN></P>
  <P dir=3Dltr><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb></SPAN></P>
  <P dir=3Dltr><SPAN lang=3Den-gb><FONT face=3DCalibri>As said, I=20
  also</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb> <FONT=20
  face=3DCalibri>still</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-=
gb> <FONT=20
  face=3DCalibri>doubt that you would have any benefit from=20
  AVPF</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb> <FONT=20
  face=3DCalibri>for that</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3D=
en-gb>=20
  <FONT face=3DCalibri>initialisation</FONT><FONT face=3DCalibri></FONT></S=
PAN><SPAN=20
  lang=3Dde></SPAN><SPAN lang=3Den-gb> <FONT face=3DCalibri>method</FONT></=
SPAN><SPAN=20
  lang=3Dde></SPAN><SPAN lang=3Den-gb><FONT face=3DCalibri>.</FONT></SPAN><=
SPAN=20
  lang=3Dde></SPAN><SPAN lang=3Den-gb> <FONT face=3DCalibri>But</FONT></SPA=
N><SPAN=20
  lang=3Dde></SPAN><SPAN lang=3Den-gb><FONT face=3DCalibri> even</FONT></SP=
AN><SPAN=20
  lang=3Dde></SPAN><SPAN lang=3Den-gb><FONT face=3DCalibri> I</FONT></SPAN>=
<SPAN=20
  lang=3Dde></SPAN><SPAN lang=3Den-gb> <FONT face=3DCalibri>only fail</FONT=
><FONT=20
  face=3DCalibri> to unders</FONT><FONT face=3DCalibri>tand the advantages =
you had=20
  in mind (</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb><FONT=20
  face=3DCalibri>d</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb><=
FONT=20
  face=3DCalibri>o you</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-=
gb><FONT=20
  face=3DCalibri> assume some particul</FONT></SPAN><SPAN lang=3Dde></SPAN>=
<SPAN=20
  lang=3Den-gb><FONT face=3DCalibri>ar receiver</FONT></SPAN><SPAN=20
  lang=3Dde></SPAN><SPAN lang=3Den-gb> <FONT=20
  face=3DCalibri>behaviour</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN=20
  lang=3Den-gb><FONT face=3DCalibri></FONT> <FONT face=3DCalibri>described =
nowhere up=20
  to now</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb><FONT=20
  face=3DCalibri>?</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb><=
FONT=20
  face=3DCalibri>),</FONT><FONT face=3DCalibri></FONT></SPAN><SPAN=20
  lang=3Dde></SPAN><SPAN lang=3Den-gb>&nbsp;<FONT face=3DCalibri></FONT> <F=
ONT=20
  face=3DCalibri>I doubt thi</FONT><FONT face=3DCalibri>s would=20
  be</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb> <FONT=20
  face=3DCalibri>wo</FONT><FONT face=3DCalibri>r</FONT><FONT face=3DCalibri=
>th the=20
  extra complexity</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb><=
FONT=20
  face=3DCalibri>, only to improve</FONT></SPAN><SPAN lang=3Dde></SPAN><SPA=
N=20
  lang=3Den-gb>&nbsp;<FONT face=3DCalibri> the handling of a very unlikely=
=20
  error</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb><FONT face=
=3DCalibri>.=20
  Again, in an</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb> <FON=
T=20
  face=3DCalibri>network</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3De=
n-gb><FONT=20
  face=3DCalibri></FONT> <FONT face=3DCalibri>where</FONT></SPAN><SPAN=20
  lang=3Dde></SPAN><SPAN lang=3Den-gb> <FONT face=3DCalibri>there is a real=
 risk that=20
  ECN-marked packets are blocked</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN=
=20
  lang=3Den-gb><FONT face=3DCalibri>, other initialisation methods are pref=
erable=20
  anyway.</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb></SPAN></P=
><BR><BR>
  <P dir=3Dltr><SPAN lang=3Den-gb><FONT face=3DCalibri>As an unrelated ques=
tion, are=20
  you working on an update of the draft?</FONT></SPAN><SPAN lang=3Dde></SPA=
N><SPAN=20
  lang=3Den-gb></SPAN></P>
  <P dir=3Dltr><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb><FONT face=3DCalib=
ri>It would=20
  be quite</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb> <FONT=20
  face=3DCalibri>helpful</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3De=
n-gb><FONT=20
  face=3DCalibri></FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb> <=
FONT=20
  face=3DCalibri>to have it available for the upcoming SA4 meeting. I=20
  woul</FONT><FONT face=3DCalibri>d also welcome if you could share the dra=
ft with=20
  me for a quick revi</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-g=
b><FONT=20
  face=3DCalibri>ew before distributing it further</FONT><FONT=20
  face=3DCalibri>,</FONT> <FONT face=3DCalibri>at least if time=20
  allows.</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb></SPAN></P=
>
  <P dir=3Dltr><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb></SPAN></P>
  <P dir=3Dltr><SPAN lang=3Den-gb><FONT face=3DCalibri>Regards,=20
  Thomas</FONT></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Den-gb></SPAN></P>
  <P dir=3Dltr><SPAN lang=3Dde></SPAN><SPAN=20
lang=3Den-gb></SPAN></P></BLOCKQUOTE></BODY></HTML>

--_000_5F7BCCF5541B7444830A2288ABBEBC961D4FED0874FRMRSSXCHMBSD_--

From keith.drage@alcatel-lucent.com  Tue Jan 11 07:46:18 2011
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To: "avt@ietf.org" <avt@ietf.org>
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Subject: [AVT] Conference announcements
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(As WG co-chair)

In general we try and discourage blanket conference announcements on IETF l=
ists. Generally when this occurs, anyone on an IETF list ends up receiving =
multiple copies, amongst their already overloaded inboxes.

Generally the criteria for any such announcements should be that they progr=
ess work on one of our deliverables, because that is what the scope of our =
IETF mailing list is. In general, if you believe such an announcement is ju=
stified, then tell us (the list) how it does that as part of the announceme=
nt. It will make us all happier that we understand why the conference is im=
portant, and you will probably get a better result from this list by doing =
so.

If you feel the need to apologise because we are receiving a message multip=
le times, then in general you are probably outside this criteria, and certa=
inly have not explained the relevance to AVT.

If you are in doubt, consult with the WG chairs.

regards

Keith=

From magnus.westerlund@ericsson.com  Tue Jan 11 09:01:42 2011
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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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To: "Belling, Thomas (NSN - DE/Munich)" <thomas.belling@nsn.com>
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Cc: IETF AVT WG <avt@ietf.org>
Subject: Re: [AVT] draft-ietf-avt-ecn-for-rtp: Is AVPF required for leap-of-faith initialisation?
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Please see some response at the bottom.

Magnus

Belling, Thomas (NSN - DE/Munich) skrev 2010-12-21 18:25:
> Dear Magnus,
> 
> allow me to continue a discussion we hadabout a month ago.
> 
> Youwere suggesting that the leap of faith initialisation method requires
> AVPF becausethis allows XR ECN summary reports can then be sent quicker
> and shorter reportingintervalthan the average 5secs canbe
> negotiated.Correct?
> 
> The draft contains the following relatedprocedurein6.2.3:
> 
>   If the sender marks all packets as ECT while transmitting on a path
> 
>    that contains an ECN-blocking middlebox, then receivers downstream of
> 
>    that middlebox will not receive any RTP data packets from the sender,
> 
>    and hence will not consider it to be an active RTP SSRC.  The sender
> 
>    can detect this and revert to sending packets without ECT marks,
> 
>    since RTCP SR/RR packets from such receivers will either not include
> 
>    a report for sender's SSRC, or will report that no packets have been
> 
>    received, but this takes at least one RTCP reporting interval.  It
> 
>    should be noted that a receiver might generate its first RTCP packet
> 
>    immediately on joining a unicast session, or very shortly after
> 
>    joining a RTP/AVPF session, before it has had chance to receive any
> 
>    data packets.  A sender that receives RTCP SR/RR packet indicating
> 
>    lack of reception by a receiver SHOULD therefore wait for a second
> 
>    RTCP report from that receiver to be sure that the lack of reception
> 
>    is due to ECT-marking.  Since this recovery process can take several
> 
>    tens of seconds, during which time the RTP session is unusable for
> 
>    media, it is NOT RECOMMENDED that the leap-of-faith ECT initiation
> 
>    method be used in environments where ECN-blocking middleboxes are
> 
>   likely to be present.
> 
> So there is noparticulartrigger for the receiver to sentSR/RR
> reportsthat would allow it to benefitfrom the more flexible timing rules
> of AVPF.
> 
> One could think about usingthe AVPFearly RTCP feedback if the sender
> does not receiveany packets for a whileat call setup, but the duration
> is quite open and it might also bepart of normal operationthat no
> packets are received–considerearly media that are usually only
> transmitted in backward direction as an example.
> 
> You suggested to negotiate a higher RTCP transmission rate as aremedy.In
> my understanding this would work both with AVP and AVPF usingSDP RTCP
> bandwidth modifiers.
> 
> But I am not convinced this is a good idea given the extra required
> bandwidth.There is a clear wordingalreadythat theleap of faith method
> isnot recommendedin environments whereECN-blocking middleboxes arelikely
> to be present. It is really meant for operator controlled environments
> where such problemswill hardly occur.Isit really worth spending more
> bandwidthto improve the handling of a very unlikelyevent?If there is a
> real risk that ECN-marked packets are blocked ina network, other
> initialisation methods are preferable anyway.
> 
> As said, I alsostilldoubt that you would have any benefit from AVPFfor
> thatinitialisationmethod.ButevenIonly failto understand the advantages
> you had in mind (do youassume some particular receiverbehaviourdescribed
> nowhere up to now?), I doubt this would beworth the extra complexity,
> only to improve the handling of a very unlikely error. Again, in
> annetworkwherethere is a real risk that ECN-marked packets are blocked,
> other initialisation methods are preferable anyway.
> 

On the above I will have to review the discussion and refresh myself on
our document. I will come back to you.

> 
> 
> As an unrelated question, are you working on an update of the draft?

The authors plan was to provide an update in December. That has clearly
failed. It is on my to do list and I do hope that we can provide one
within a few weeks.

> 
> It would be quitehelpfulto have it available for the upcoming SA4
> meeting. I would also welcome if you could share the draft with me for a
> quick review before distributing it further, at least if time allows.
> 

Sorry about that.


-- 

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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We revised the CNAME draft. For diff and the full text, refer to:
https://datatracker.ietf.org/doc/draft-ietf-avt-rtp-cnames/

There are some minor fixes. In addition, we were also asked to make some =
clarifications around the uniqueness requirements. Basically, we want =
uniqueness with a high probability. Collision will not result in RTP =
functionality failure anyway. We also changed hashing from sha-1 to =
sha-256 as the former is slowly being obsoleted. More importantly, we =
extended the security section and made it clear that the goal behind =
hashing was not security but to achieve a reasonable uniqueness.=20

We hope these will address the IESG comments. If you have questions, =
please email.

-acbegen

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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.


	Title           : The use of AES-192 and AES-256 in Secure RTP
	Author(s)       : D. McGrew
	Filename        : draft-ietf-avt-srtp-big-aes-06.txt
	Pages           : 21
	Date            : 2011-01-12

This memo describes the use of the Advanced Encryption Standard (AES)
with 192 and 256 bit keys within the Secure RTP protocol.  It details
Counter Mode encryption for SRTP and SRTCP and a new SRTP Key
Derivation Function (KDF) for AES-192 and AES-256.

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FYI, Please review and send comments to iesg@ietf.org

Roni

 

A new IETF working group has been proposed in the Real-Time Applications

and Infrastructure Area.  The IESG has not made any determination as yet.

The following draft charter was submitted, and is provided for

informational purposes only. Please send your comments to the IESG mailing

list (iesg at ietf.org) by Tuesday, January 18, 2011.               

 

Audio/Video Transport Core Maintenence (avtcore)

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

Current Status: Proposed Working Group

Last modified: 2010-12-03

 

Chairs: TBD

 

Real-Time Applications and Infrastructure Area Directors:

  Gonzalo Camarillo <gonzalo.camarillo at ericsson.com>

  Robert Sparks <rjsparks at nostrum.com>

 

Real-Time Applications and Infrastructure Area Advisor:

  Robert Sparks <rjsparks at nostrum.com>

 

Mailing Lists: TBD

 

Description of Working Group:

 

The Real-time Transport Protocol, RTP, along with its associated 

profiles and payload formats provides for real-time transmission of 

audio and video over unicast and multicast transports.  RTP itself has 

been shepherded to Full Standard. Its associated profiles, extensions, 

and payload formats are currently at various levels of standards 

maturity. 

 

The AVTCORE working group is chartered to maintain the core RTP/RTCP 

protocols and the AVP, SAVP, AVPF, and SAVPF profiles. The group will 

provide architectural guidance for extending the protocols and 

guidelines for their proper use. While other working groups may be 

chartered to work on application-specific extensions to the protocols, 

extensions that are generally applicable will be developed in AVTCORE.

 

The AVTCORE working group will coordinate closely with the Security Area

while working on maintenance and enhancements to the SRTP Profile.

 

In addition to the milestones called out below, the AVTCORE working 

group's initial tasks will include completing any remaining work 

identified in those drafts from the AVT working group already in IESG 

Evaluation, with the exception of the Rapid Acquisition of Multicast RTP 

sessions, which will complete in the AVTEXT working group.

 

Goals and Milestones:

Dec 2010  Submit Guideline for Choosing RTCP CNAME as Proposed Standard

Dec 2010  Submit use of AES-192 and AES-256 in Secure RTP for Proposed 

          Standard

Dec 2010  Submit Port Mapping Between Unicast and Multicast RTP Sessions 

          for Proposed Standard

Dec 2010  Submit Real-Time Transport Control Protocol (RTCP) in Overlay 

          Multicast for Proposed Standard

Feb 2011  Submit Monitoring Architecture for RTP for Informational

Feb 2011  Guidelines for the use of Variable Bit Rate Audio with Secure 

          RTP as informational (or possibly BCP)

Apr 2011  Submit in band keying mechanism for SRTP draft for Proposed 

          Standard

Apr 2011  Submit Explicit Congestion Notification (ECN) for RTP over UDP 

          for Proposed Standard

May 2011  RTCP indication for retransmission suppression as proposed 

          standard

Sep 2011  Encryption of Header Extensions in the Secure Real-Time 

          Transport Protocol (SRTP) for Proposed Standard

 

 

 


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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>FYI, Please review and send comments to <a href="mailto:iesg@ietf.org">iesg@ietf.org</a><o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Roni<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>A new IETF working group has been proposed in the Real-Time Applications<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>and Infrastructure Area.&nbsp; The IESG has not made any determination as yet.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>The following draft charter was submitted, and is provided for<o:p></o:
 p></span
span style='font-size:10.0pt;font-family:"Courier New"'>informational purposes only. Please send your comments to the IESG mailing<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>list (iesg at ietf.org) by Tuesday, January 18, 2011.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Audio/Video Transport Core Maintenence (avtcore)<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>-------------------------------------------<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Current Status: Proposed Working Group<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'
 >Last mo
o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Chairs: TBD<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Real-Time Applications and Infrastructure Area Directors:<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp; Gonzalo Camarillo &lt;gonzalo.camarillo at ericsson.com&gt;<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp; Robert Sparks &lt;rjsparks at nostrum.com&gt;<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Real-Time
  Applica
Area Advisor:<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp; Robert Sparks &lt;rjsparks at nostrum.com&gt;<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Mailing Lists: TBD<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Description of Working Group:<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>The Real-time Transport Protocol, RTP, along with its associated <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>profiles an
 d payloa
al-time transmission of <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>audio and video over unicast and multicast transports.&nbsp; RTP itself has <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>been shepherded to Full Standard. Its associated profiles, extensions, <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>and payload formats are currently at various levels of standards <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>maturity. <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>The AVTCORE working group is chartered to maintain the core RTP/RTCP <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-famil
 y:"Couri
e AVP, SAVP, AVPF, and SAVPF profiles. The group will <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>provide architectural guidance for extending the protocols and <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>guidelines for their proper use. While other working groups may be <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>chartered to work on application-specific extensions to the protocols, <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>extensions that are generally applicable will be developed in AVTCORE.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>The AVTCORE working group will coordinate closely with the Security Are
 a<o:p></
soNormal><span style='font-size:10.0pt;font-family:"Courier New"'>while working on maintenance and enhancements to the SRTP Profile.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>In addition to the milestones called out below, the AVTCORE working <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>group's initial tasks will include completing any remaining work <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>identified in those drafts from the AVT working group already in IESG <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Evaluation, with the exception of the Rapid Acquisition of Multicast RTP <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Co
 urier Ne
complete in the AVTEXT working group.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Goals and Milestones:<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Dec 2010&nbsp; Submit Guideline for Choosing RTCP CNAME as Proposed Standard<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Dec 2010&nbsp; Submit use of AES-192 and AES-256 in Secure RTP for Proposed <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Standard<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Dec 2010&nbsp; Submit Port Mapping Between Unicast and Multicast RTP Sessions <o:p></o:p></span></p><p class=MsoNormal
 ><span s
nt-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;for Proposed Standard<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Dec 2010&nbsp; Submit Real-Time Transport Control Protocol (RTCP) in Overlay <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Multicast for Proposed Standard<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Feb 2011&nbsp; Submit Monitoring Architecture for RTP for Informational<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Feb 2011&nbsp; Guidelines for the use of Variable Bit Rate Audio with Secure <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;RTP as information
 al (or p
span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Apr 2011&nbsp; Submit in band keying mechanism for SRTP draft for Proposed <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Standard<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Apr 2011&nbsp; Submit Explicit Congestion Notification (ECN) for RTP over UDP <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;for Proposed Standard<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>May 2011&nbsp; RTCP indication for retransmission suppression as proposed <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb
 sp;&nbsp
></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Sep 2011&nbsp; Encryption of Header Extensions in the Secure Real-Time <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Transport Protocol (SRTP) for Proposed Standard<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><o:p>&nbsp;</o:p></p><p class=MsoNormal><o:p>&nbsp;</o:p></p></div> <BR><BR>__________ Information from ESET NOD32 Antivirus, version of virus signature database 5781 (20110112) __________<BR><BR>The message was checked by ESET NOD32 Antivirus.<BR><BR><A HREF="http://www.eset.com">http://www.eset.com</A><BR> </body></html>

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Date: Wed, 12 Jan 2011 20:30:07 +0200
From: Roni Even <Even.roni@huawei.com>
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Subject: [AVT] WG Review: Audio/Video Transport Extensions (avtext)
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FYI, please review and send any comments to iesg@ietf.org

Roni Even

 

A new IETF working group has been proposed in the Real-Time Applications

and Infrastructure Area.  The IESG has not made any determination as yet.

The following draft charter was submitted, and is provided for

informational purposes only. Please send your comments to the IESG mailing

list (iesg at ietf.org) by Tuesday, January 18, 2011.               

 

Audio/Video Transport Extensions (avtext)

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

Current Status: Proposed Working Group

Last modified: 2010-12-03

 

Chairs: TBD

 

Real-Time Applications and Infrastructure Area Directors:

  Gonzalo Camarillo <gonzalo.camarillo at ericsson.com>

  Robert Sparks <rjsparks at nostrum.com>

 

Real-Time Applications and Infrastructure Area Advisor:

  Robert Sparks <rjsparks at nostrum.com>

 

Mailing Lists: TBD

 

Description of Working Group:

 

  The Full-Standard Real-time Transport Protocol, RTP [RFC 3550], along 

  with its associated profiles and payload formats provides for real-

  time transmission of audio and video over unicast and multicast 

  transports. RTP is widely implemented, and deployed for a wide range 

  of applications, ranging from telephony to television over IP. As new 

  applications have emerged, the need for guidelines for the use of the 

  RTP/RTCP protocols and extensions specific to those applications 

  has been identified.

 

  The AVTEXT working group is charted to develop application-specific 

  guidelines for the use of RTP/RTCP protocols with the AVP, SAVP, 

  AVPF, and SAVPF profiles, and extensions to those protocols that are 

  driven by application-specific needs. Proposals for extensions with 

  general applicability to many different RTP/RTCP usages should be 

  taken to the AVTCORE working group. 

 

  The AVTEXT working group is constrained to use the protocol extension 

  mechanisms defined in the core protocols (such as RTP Header 

  extensions [RFC5285], AVPF Feedback Messages [RFC4585], and 

  SDES Items [RFC3550]). If new ways to extend the core protocols are 

  needed, they will be developed in the AVTCORE working group.

 

  In addition to the milestones called out below, the AVTEXT working 

  group's initial tasks will include completing any new work identified 

  during IESG evaluation for the Rapid Acquisition of Multicast RTP 

  Sessions.

 

Goals and Milestones:

Dec 2010  Submit Considerations for RAMS Scenarios for Informational

Oct 2011  Submit RTP Header extension for mixer to client audio level 

          indication as Proposed Standard

Oct 2011  Submit RTP Header extension for client to mixer audio level 

          indication as proposed standard

Dec 2011  Submit Support for multiple clock rates in an RTP session for 

          Proposed Standard

Dec 2011  Submit SRTP Store-and-Forward Use Cases and Requirements for 

          Informational

Dec 2011  Submit Use of the Secure Real-time Transport Protocol (SRTP) 

          in Store-and-Forward Applications for Proposed Standard

 

 

 


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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>FYI, please review and send any comments to <a href="mailto:iesg@ietf.org">iesg@ietf.org</a><o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Roni Even<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>A new IETF working group has been proposed in the Real-Time Applications<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>and Infrastructure Area.&nbsp; The IESG has not made any determination as yet.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>The following draft charter was submitted, and is provided for
 <o:p></o
oNormal><span style='font-size:10.0pt;font-family:"Courier New"'>informational purposes only. Please send your comments to the IESG mailing<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>list (iesg at ietf.org) by Tuesday, January 18, 2011.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Audio/Video Transport Extensions (avtext)<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>-------------------------------------------<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Current Status: Proposed Working Group<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New
 "'>Last 
</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Chairs: TBD<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Real-Time Applications and Infrastructure Area Directors:<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp; Gonzalo Camarillo &lt;gonzalo.camarillo at ericsson.com&gt;<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp; Robert Sparks &lt;rjsparks at nostrum.com&gt;<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Real-Ti
 me Appli
e Area Advisor:<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp; Robert Sparks &lt;rjsparks at nostrum.com&gt;<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Mailing Lists: TBD<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Description of Working Group:<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp; The Full-Standard Real-time Transport Protocol, RTP [RFC 3550], along <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New
 "'>&nbsp
d profiles and payload formats provides for real-<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp; time transmission of audio and video over unicast and multicast <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;transports. RTP is widely implemented, and deployed for a wide range <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;of applications, ranging from telephony to television over IP. As new <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;applications have emerged, the need for guidelines for the use of the <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;RTP/RTCP protocols and extensions specific to those applications <o:p></o:p></span></p><p class=MsoNormal><span style='font
 -size:10
 New"'>&nbsp;&nbsp;has been identified.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp; The AVTEXT working group is charted to develop application-specific <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;guidelines for the use of RTP/RTCP protocols with the AVP, SAVP, <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;AVPF, and SAVPF profiles, and extensions to those protocols that are <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;driven by application-specific needs. Proposals for extensions with <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;general applicability to many di
 fferent 
 <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;taken to the AVTCORE working group. <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp; The AVTEXT working group is constrained to use the protocol extension <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;mechanisms defined in the core protocols (such as RTP Header <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;extensions [RFC5285], AVPF Feedback Messages [RFC4585], and <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;SDES Items [RFC3550]). If new ways to extend the core protocols are <o:p></o:p></span></p><p class=MsoNormal><spa
 n style=
mily:"Courier New"'>&nbsp;&nbsp;needed, they will be developed in the AVTCORE working group.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp; In addition to the milestones called out below, the AVTEXT working <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;group's initial tasks will include completing any new work identified <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;during IESG evaluation for the Rapid Acquisition of Multicast RTP <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;Sessions.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><s
 pan styl
family:"Courier New"'>Goals and Milestones:<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Dec 2010&nbsp; Submit Considerations for RAMS Scenarios for Informational<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Oct 2011&nbsp; Submit RTP Header extension for mixer to client audio level <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;indication as Proposed Standard<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Oct 2011&nbsp; Submit RTP Header extension for client to mixer audio level <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;indication as proposed standard<o:p></o:p></span></p><p class=MsoNormal><span style
 ='font-s
ourier New"'>Dec 2011&nbsp; Submit Support for multiple clock rates in an RTP session for <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Proposed Standard<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Dec 2011&nbsp; Submit SRTP Store-and-Forward Use Cases and Requirements for <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Informational<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Dec 2011&nbsp; Submit Use of the Secure Real-time Transport Protocol (SRTP) <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;in Store-and-Forward Applications for Proposed S
 tandard<
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Date: Wed, 12 Jan 2011 20:31:07 +0200
From: Roni Even <Even.roni@huawei.com>
To: avt@ietf.org
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Subject: [AVT] WG Review: Metric Blocks for use with RTCP's Extended Report Framework (xrblock)
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FYI,

Please review and send any comments to iesg@ietf.org

Roni Even

 

A new IETF working group has been proposed in the Real-Time Applications

and Infrastructure Area.  The IESG has not made any determination as yet.

The following draft charter was submitted, and is provided for

informational purposes only. Please send your comments to the IESG mailing

list (iesg at ietf.org) by Tuesday, January 18, 2011.               

 

Metric Blocks for use with RTCP's Extended Report Framework (xrblock)

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

Current Status: Proposed Working Group

Last modified: 2010-12-03

 

Chairs: TBD

 

Real-Time Applications and Infrastructure Area Directors:

  Gonzalo Camarillo <gonzalo.camarillo at ericsson.com>

  Robert Sparks <rjsparks at nostrum.com>

 

Real-Time Applications and Infrastructure Area Advisor:

  Robert Sparks <rjsparks at nostrum.com>

 

Mailing Lists: TBD

 

Description of Working Group:

 

RFC3611, "RTP Control Protocol Extended Reports (RTCP XR)" established

a framework to allow new information to be conveyed in RTCP, 

supplementing the original report blocks defined in RFC 3550, "RTP: A 

Transport Protocol for Real-Time Applications".

 

The XRBLOCK working group is chartered to work within this framework, 

evaluating proposals for report block definitions containing new 

metrics.  The group will follow the guidelines established in RFC5968, 

"Guidelines for Extending the RTP Control Protocol (RTCP)" and RTP 

Monitoring Architecture being developed in the RTPCORE working group.

 

Goals and Milestones:

 

Mar 2011  Submit RTCP XR Report Block for Measurement Identity

Mar 2011  Submit RTCP XR Report Block for Burst/Gap Discard metric 

          Reporting

Mar 2011  Submit RTCP XR Report Block for Burst/Gap Loss metric 

          Reporting

Jun 2011  Submit RTCP XR Report Block for Concealed Seconds metric 

          Reporting

Jun 2011  Submit RTCP XR Report Block for Delay metric Reporting

Jun 2011  Submit RTCP XR Report Block for Discard metric Reporting

Sep 2011  Submit RTCP XR Report Block for Jitter Buffer Metric Reporting

Sep 2011  Submit RTCP XR Report Block for Loss Concealment metric 

          Reporting

Sep 2011  Submit RTCP XR Report Block for Packet Delay Variation Metric 

          Reporting

Dec 2011  Submit RTCP XR Report Block for Run Length Encodings of 

          Discarded Packets

Dec 2011  Submit RTCP XR Report Block for QoE Metrics Reporting    

 

 


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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><span style='font-size:10.0pt;font-family:"Courier New"'>FYI,<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Please review and send any comments to <a href="mailto:iesg@ietf.org">iesg@ietf.org</a><o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Roni Even<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>A new IETF working group has been proposed in the Real-Time Applications<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>and Infrastructure Area.&nbsp; The IESG has not made any determination as yet.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size
 :10.0pt;
'>The following draft charter was submitted, and is provided for<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>informational purposes only. Please send your comments to the IESG mailing<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>list (iesg at ietf.org) by Tuesday, January 18, 2011.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Metric Blocks for use with RTCP's Extended Report Framework (xrblock)<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>-------------------------------------------<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Current 
 Status: 
p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Last modified: 2010-12-03<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Chairs: TBD<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Real-Time Applications and Infrastructure Area Directors:<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp; Gonzalo Camarillo &lt;gonzalo.camarillo at ericsson.com&gt;<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp; Robert Sparks &lt;rjsparks at nostrum.com&gt;<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"
 Courier 
span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Real-Time Applications and Infrastructure Area Advisor:<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp; Robert Sparks &lt;rjsparks at nostrum.com&gt;<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Mailing Lists: TBD<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Description of Working Group:<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>RFC3611, &quot;RTP Control Protocol Extended Re
 ports (R
<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>a framework to allow new information to be conveyed in RTCP, <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>supplementing the original report blocks defined in RFC 3550, &quot;RTP: A <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Transport Protocol for Real-Time Applications&quot;.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>The XRBLOCK working group is chartered to work within this framework, <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>evaluating proposals for report block definitions containing new <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;fo
 nt-famil
&nbsp; The group will follow the guidelines established in RFC5968, <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&quot;Guidelines for Extending the RTP Control Protocol (RTCP)&quot; and RTP <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Monitoring Architecture being developed in the RTPCORE working group.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Goals and Milestones:<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Mar 2011&nbsp; Submit RTCP XR Report Block for Measurement Identity<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Cour
 ier New"
TCP XR Report Block for Burst/Gap Discard metric <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Reporting<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Mar 2011&nbsp; Submit RTCP XR Report Block for Burst/Gap Loss metric <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Reporting<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Jun 2011&nbsp; Submit RTCP XR Report Block for Concealed Seconds metric <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Reporting<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Jun 2
 011&nbsp
lock for Delay metric Reporting<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Jun 2011&nbsp; Submit RTCP XR Report Block for Discard metric Reporting<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Sep 2011&nbsp; Submit RTCP XR Report Block for Jitter Buffer Metric Reporting<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Sep 2011&nbsp; Submit RTCP XR Report Block for Loss Concealment metric <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Reporting<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Sep 2011&nbsp; Submit RTCP XR Report Block for Packet Delay Variation Metric <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;
 &nbsp;&n
sp;&nbsp;&nbsp;&nbsp;Reporting<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Dec 2011&nbsp; Submit RTCP XR Report Block for Run Length Encodings of <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Discarded Packets<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Dec 2011&nbsp; Submit RTCP XR Report Block for QoE Metrics Reporting&nbsp;&nbsp;&nbsp; <o:p></o:p></span></p><p class=MsoNormal><o:p>&nbsp;</o:p></p><p class=MsoNormal><o:p>&nbsp;</o:p></p></div> <BR><BR>__________ Information from ESET NOD32 Antivirus, version of virus signature database 5781 (20110112) __________<BR><BR>The message was checked by ESET NOD32 Antivirus.<BR><BR><A HREF="http://www.eset.com">http://www.eset.com</A><BR> </body></html>

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Date: Wed, 12 Jan 2011 20:32:13 +0200
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Subject: [AVT] WG Review: Audio/Video Transport Payloads (payload)
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FYI, please review and send any comments to iesg@ietf.org

Roni Even

 

A new IETF working group has been proposed in the Real-Time Applications

and Infrastructure Area.  The IESG has not made any determination as yet.

The following draft charter was submitted, and is provided for

informational purposes only. Please send your comments to the IESG mailing

list (iesg at ietf.org) by Tuesday, January 18, 2011.               

 

Audio/Video Transport Payloads (payload)

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

Current Status: Proposed Working Group

Last modified: 2010-12-03

 

Chairs: TBD

 

Real-Time Applications and Infrastructure Area Directors:

  Gonzalo Camarillo <gonzalo.camarillo at ericsson.com>

  Robert Sparks <rjsparks at nostrum.com>

 

Real-Time Applications and Infrastructure Area Advisor:

  Robert Sparks <rjsparks at nostrum.com>

 

Mailing Lists: TBD

 

Description of Working Group:

 

  The PAYLOAD working group is tasked with the specification and 

  maintenance of payload formats for use with the Real-Time Transport 

  Protocol, RTP [RFC3550]. The group will follow the guidelines 

  established in "Guidelines for Writers of RTP Payload Format 

  Specifications" [BCP 36], the "How to Write an RTP Payload Format" 

  (under development) and is responsible for maintaining those 

  guidelines.

 

  This working group will develop RTP payload formats for new media 

  codecs, review and revise existing payload formats to advance those 

  which are useful to Draft Standard or Standard, and declare others 

  Historic.

 

Goals and Milestones:

Dec 2010  Submit RTP Payload Format for MIDI for Proposed Standard

Feb 2011  Submit How to Write an RTP Payload Format for Informational

Feb 2011  Submit RTP Payload Format for MPEG-4 Audio/Visual Streams for 

          Proposed Standard

Mar 2011  Submit RTP Payload Format for EVBR/G.718 for Proposed Standard

Mar 2011  Submit RTP Payload Format for Enhanced Variable Rate 

          Narrowband-Wideband Codec (EVRC-NW) for Proposed Standard

Mar 2011  Submit RTP Payload Format for Bluetooth's SBC audio codec for 

          Proposed Standard

Apr 2011  Submit RTP Payload Format for MPEG2-TS preamble for Proposed 

          Standard

Apr 2011  Submit RTP Payload Format for DV (IEC 61834) Video for 

          Proposed Standard

Apr 2011  Submit RTP Payload Format for the iSAC codec for Proposed 

          Standard

Apr 2011  Submit RTP profile for the carriage of SMPTE 336M data for 

          Proposed Standard

Jun 2011  Submit RTP Payload Format for MVC Video for Proposed Standard

 


--Boundary_(ID_OHPDpYzEJgDh+qojFKlfVA)
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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>FYI, please review and send any comments to <a href="mailto:iesg@ietf.org">iesg@ietf.org</a><o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Roni Even<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>A new IETF working group has been proposed in the Real-Time Applications<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>and Infrastructure Area.&nbsp; The IESG has not made any determination as yet.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>The following draft charter was submitted, and is provided for
 <o:p></o
oNormal><span style='font-size:10.0pt;font-family:"Courier New"'>informational purposes only. Please send your comments to the IESG mailing<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>list (iesg at ietf.org) by Tuesday, January 18, 2011.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Audio/Video Transport Payloads (payload)<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>-------------------------------------------<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Current Status: Proposed Working Group<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"
 '>Last m
/o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Chairs: TBD<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Real-Time Applications and Infrastructure Area Directors:<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp; Gonzalo Camarillo &lt;gonzalo.camarillo at ericsson.com&gt;<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp; Robert Sparks &lt;rjsparks at nostrum.com&gt;<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Real-Tim
 e Applic
 Area Advisor:<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp; Robert Sparks &lt;rjsparks at nostrum.com&gt;<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Mailing Lists: TBD<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Description of Working Group:<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp; The PAYLOAD working group is tasked with the specification and <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp
 ;&nbsp;m
mats for use with the Real-Time Transport <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;Protocol, RTP [RFC3550]. The group will follow the guidelines <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;established in &quot;Guidelines for Writers of RTP Payload Format <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;Specifications&quot; [BCP 36], the &quot;How to Write an RTP Payload Format&quot; <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;(under development) and is responsible for maintaining those <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;guidelines.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p><
 /span></
n style='font-size:10.0pt;font-family:"Courier New"'>&nbsp; This working group will develop RTP payload formats for new media <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;codecs, review and revise existing payload formats to advance those <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;which are useful to Draft Standard or Standard, and declare others <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;Historic.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Goals and Milestones:<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Dec 2010&nbsp; Submit RTP Payload Format for MIDI for Proposed Standa
 rd<o:p><
MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Feb 2011&nbsp; Submit How to Write an RTP Payload Format for Informational<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Feb 2011&nbsp; Submit RTP Payload Format for MPEG-4 Audio/Visual Streams for <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Proposed Standard<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Mar 2011 &nbsp;Submit RTP Payload Format for EVBR/G.718 for Proposed Standard<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Mar 2011&nbsp; Submit RTP Payload Format for Enhanced Variable Rate <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
 Narrowba
W) for Proposed Standard<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Mar 2011&nbsp; Submit RTP Payload Format for Bluetooth's SBC audio codec for <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Proposed Standard<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Apr 2011&nbsp; Submit RTP Payload Format for MPEG2-TS preamble for Proposed <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Standard<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Apr 2011&nbsp; Submit RTP Payload Format for DV (IEC 61834) Video for <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbs
 p;&nbsp;
nbsp;&nbsp;&nbsp;Proposed Standard<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Apr 2011&nbsp; Submit RTP Payload Format for the iSAC codec for Proposed <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Standard<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Apr 2011&nbsp; Submit RTP profile for the carriage of SMPTE 336M data for <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Proposed Standard<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>Jun 2011&nbsp; Submit RTP Payload Format for MVC Video for Proposed Standard<o:p></o:p></span></p><p class=MsoNormal><o:p>&nbsp;</o:p></p></div> <BR><BR>__________ Informatio
 n from E
ion of virus signature database 5781 (20110112) __________<BR><BR>The message was checked by ESET NOD32 Antivirus.<BR><BR><A HREF="http://www.eset.com">http://www.eset.com</A><BR> </body></html>

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From kpfleming@digium.com  Wed Jan 12 12:47:54 2011
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Date: Wed, 12 Jan 2011 14:50:12 -0600
From: "Kevin P. Fleming" <kpfleming@digium.com>
Organization: Digium, Inc.
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Subject: Re: [AVT] WG Review: Audio/Video Transport Payloads (payload)
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On 01/12/2011 12:32 PM, Roni Even wrote:
> FYI, please review and send any comments to iesg@ietf.org
> <mailto:iesg@ietf.org>
>
> Roni Even
>
> A new IETF working group has been proposed in the Real-Time Applications
>
> and Infrastructure Area. The IESG has not made any determination as yet.
>
> The following draft charter was submitted, and is provided for
> informational purposes only. Please send your comments to the IESG mailing
>
> list (iesg at ietf.org) by Tuesday, January 18, 2011.
>
> Audio/Video Transport Payloads (payload)
>
> -------------------------------------------
>
> Current Status: Proposed Working Group
>
> Last m /o:p>
>
> Chairs: TBD
>
> Real-Time Applications and Infrastructure Area Directors:
>
> Gonzalo Camarillo <gonzalo.camarillo at ericsson.com>
>
> Robert Sparks <rjsparks at nostrum.com>
>
> Real-Tim e Applic Area Advisor:
>
> Robert Sparks <rjsparks at nostrum.com>
>
> Mailing Lists: TBD
>
> Description of Working Group:
>
> The PAYLOAD working group is tasked with the specification and
>
> ; m mats for use with the Real-Time Transport
>
> Protocol, RTP [RFC3550]. The group will follow the guidelines
>
> established in "Guidelines for Writers of RTP Payload Format
>
> Specifications" [BCP 36], the "How to Write an RTP Payload Format"
>
> (under development) and is responsible for maintaining those
>
> guidelines.
>
> This working group will develop RTP payload formats for new media
>
> codecs, review and revise existing payload formats to advance those
>
> which are useful to Draft Standard or Standard, and declare others
>
> Historic.

This description is badly broken up, and it appears that other parts of 
the message were corrupted as well.

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

From Even.roni@huawei.com  Wed Jan 12 15:21:04 2011
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Date: Thu, 13 Jan 2011 01:23:09 +0200
From: Roni Even <Even.roni@huawei.com>
In-reply-to: <4D2E1404.1060004@digium.com>
To: "'Kevin P. Fleming'" <kpfleming@digium.com>, avt@ietf.org
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Subject: Re: [AVT] WG Review: Audio/Video Transport Payloads (payload)
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Hi,
I am sorry, but I did not get the original messages from the announce list.
I took it from the archive which can be seen at
http://www.ietf.org/mail-archive/web/ietf-announce/current/maillist.html on
January 11th.
So you can see the original message there.
Thanks
Roni

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
> Kevin P. Fleming
> Sent: Wednesday, January 12, 2011 10:50 PM
> To: avt@ietf.org
> Subject: Re: [AVT] WG Review: Audio/Video Transport Payloads (payload)
> 
> On 01/12/2011 12:32 PM, Roni Even wrote:
> > FYI, please review and send any comments to iesg@ietf.org
> > <mailto:iesg@ietf.org>
> >
> > Roni Even
> >
> > A new IETF working group has been proposed in the Real-Time
> Applications
> >
> > and Infrastructure Area. The IESG has not made any determination as
> yet.
> >
> > The following draft charter was submitted, and is provided for
> > informational purposes only. Please send your comments to the IESG
> mailing
> >
> > list (iesg at ietf.org) by Tuesday, January 18, 2011.
> >
> > Audio/Video Transport Payloads (payload)
> >
> > -------------------------------------------
> >
> > Current Status: Proposed Working Group
> >
> > Last m /o:p>
> >
> > Chairs: TBD
> >
> > Real-Time Applications and Infrastructure Area Directors:
> >
> > Gonzalo Camarillo <gonzalo.camarillo at ericsson.com>
> >
> > Robert Sparks <rjsparks at nostrum.com>
> >
> > Real-Tim e Applic Area Advisor:
> >
> > Robert Sparks <rjsparks at nostrum.com>
> >
> > Mailing Lists: TBD
> >
> > Description of Working Group:
> >
> > The PAYLOAD working group is tasked with the specification and
> >
> > ; m mats for use with the Real-Time Transport
> >
> > Protocol, RTP [RFC3550]. The group will follow the guidelines
> >
> > established in "Guidelines for Writers of RTP Payload Format
> >
> > Specifications" [BCP 36], the "How to Write an RTP Payload Format"
> >
> > (under development) and is responsible for maintaining those
> >
> > guidelines.
> >
> > This working group will develop RTP payload formats for new media
> >
> > codecs, review and revise existing payload formats to advance those
> >
> > which are useful to Draft Standard or Standard, and declare others
> >
> > Historic.
> 
> This description is badly broken up, and it appears that other parts of
> the message were corrupted as well.
> 
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> skype: kpfleming | jabber: kfleming@digium.com
> Check us out at www.digium.com & www.asterisk.org
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
> 
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From stewe@stewe.org  Wed Jan 12 16:09:18 2011
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From: Stephan Wenger <stewe@stewe.org>
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Thread-Topic: proposed AVT WG split
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Hi IESG,

As already communicated in Maastricht, I continue to believe that this step
is a bad idea.  I didn't attend Beijing, and don't know whether there was
any discussion about the subject.  I didn't see anything mentioned in the
AVT minutes, though, nor did I see any discussion since Beijing on
ietf@ietf.  If there was, some of the points below may be redundant, and I
apologize in advance.

Essentially, you guys are suggesting to split one "eternal" WG into four.
I'm writing "eternal", because I don't see that any of the WGs can fulfill
its charter before other work items will likely get chartered with a simila=
r
scope, and in the same WG.  For RTP core, RTP extensions, and RTP payloads
that is probably self-evident.  For the XR stuff: metrics pop up all the
time, like payload formats.  Metrics are also known to make people quick
bucks on patent licensing.  There will be more metric proposals.  If the
RTCP-XR metrics ever get momentum in the marketplace, that is.  Somehow, I
have my doubts.  But that's a different discussion.

The positive effects of the split could be
-chair workload is split among more people, and
-there are more meeting seconds for RTP related work at any given IETF
meeting. =20

Assuming unlimited resources, both are good things.  Unfortunately,
resources are not unlimited.

The negative effects are IMO:
-the key knowledgeable people in AVT ideally would have to sit in all four
meetings so that all four groups can tap their experience.  But they may no=
t
want to do so, because it leaves less time for cross-area involvement,
coordination, and draft-writing outside the AVT sessions, and whatnot.
Also, the burden for remote participants in far-away time zones goes up.  I=
n
other words, the resource meeting-seconds for those people is not unlimited=
,
and they will have to prioritize differently.    The result could be that
what I called "key knowledgable people" would perhaps not attend the mere
RTCP-XR or Payload session; and the guys there may be wasting their time
doing work that does not make sense architecturally, just to see the hammer
coming down at IETF last call.
-more people get chair positions and are burdened with administrative stuff
-> less time to do real work
-WG scheduling gets more difficult
-AD load potentially goes up (someone needs to coordinate between those
groups, and the AD would seem to me the natural choice)
-there is no reason to believe that the overall manpower availability goes
up through this change, so all the inefficiencies introduced go straight
against the available cycles.

Some of these issues can be solved by joint WG meetings and/or careful chai=
r
selection.  However, both options have drawbacks as well.  A joint session
would be counter-productive to the allocation of more meeting minutes.  And
when you have, say, 8 chairs for four WG, all of which know the RTP
architecture well enough, then you have pretty much exhausted, if not
overdrawn already, the pool of "key knowledgable people", and burdened them
with (in many cases: additional) chair duties.

No matter how I see it, splitting the group is not a good idea.

Other options to alleviate problems could come to mind:

If the problem to be solved is chair-workload: add more chairs (but not 6 o=
f
them).
If the problem to be solved is meeting-second shortages: have interim
meetings, conf calls, etc.  In other words, stretch out the AVT meeting in
the interval between IETF meetings.  Require a web-ex presentation of every
draft to be discussed in the AVT session at least 4 weeks before the
meeting, with a two hour timeslot for Q/A.  Should I continue with this
brainstorm?

If the problem is something else, I would like to know=8A

Thanks for your consideration, and apologies for the wordcount,

Stephan






--B_3377693488_23135744
Content-type: text/html;
	charset="ISO-8859-1"
Content-transfer-encoding: quoted-printable

<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>Hi IESG,</div><div><br></div=
><div>As already communicated in Maastricht, I continue to believe that this=
 step is a bad idea. &nbsp;I didn't attend Beijing, and don't know whether t=
here was any discussion about the subject. &nbsp;I didn't see anything menti=
oned in the AVT minutes, though, nor did I see any discussion since Beijing =
on ietf@ietf. &nbsp;If there was, some of the points below may be redundant,=
 and I apologize in advance.</div><div><br></div><div>Essentially, you guys =
are suggesting to split one "eternal" WG into four. &nbsp;I'm writing "etern=
al", because I don't see that any of the WGs can fulfill its charter before =
other work items will likely get chartered with a similar scope, and in the =
same WG. &nbsp;For RTP core, RTP extensions, and RTP payloads that is probab=
ly self-evident. &nbsp;For the XR stuff: metrics pop up all the time, like p=
ayload formats. &nbsp;Metrics are also known to make people quick bucks on p=
atent licensing. &nbsp;There will be more metric proposals. &nbsp;If the RTC=
P-XR metrics ever get momentum in the marketplace, that is. &nbsp;Somehow, I=
 have my doubts. &nbsp;But that's a different discussion. &nbsp;</div><div><=
br></div><div>The positive effects of the split could be&nbsp;</div><div>-ch=
air workload is split among more people, and&nbsp;</div><div>-there are more=
 meeting seconds for RTP related work at any given IETF meeting. &nbsp;</div=
><div><br></div><div>Assuming unlimited resources, both are good things. &nb=
sp;Unfortunately, resources are not unlimited.</div><div><br></div><div>The =
negative effects are IMO:</div><div>-the key knowledgeable people in AVT ide=
ally would have to sit in all four meetings so that all four groups can tap =
their experience. &nbsp;But they may not want to do so, because it leaves le=
ss time for cross-area involvement, coordination, and draft-writing outside =
the AVT sessions, and whatnot. &nbsp;Also, the burden for remote participant=
s in far-away time zones goes up. &nbsp;In other words, the resource meeting=
-seconds for those people is not unlimited, and they will have to prioritize=
 differently. &nbsp; &nbsp;The result could be that what I called "key knowl=
edgable people" would perhaps not attend the mere RTCP-XR or Payload session=
; and the guys there may be wasting their time doing work that does not make=
 sense architecturally, just to see the hammer coming down at IETF last call=
.</div><div>-more people get chair positions and are burdened with administr=
ative stuff -&gt; less time to do real work</div><div>-WG scheduling gets mo=
re difficult</div><div>-AD load potentially goes up (someone needs to coordi=
nate between those groups, and the AD would seem to me the natural choice)</=
div><div>-there is no reason to believe that the overall manpower availabili=
ty goes up through this change, so all the inefficiencies introduced go stra=
ight against the available cycles.</div><div><br></div><div>Some of these is=
sues can be solved by joint WG meetings and/or careful chair selection. &nbs=
p;However, both options have drawbacks as well. &nbsp;A joint session would =
be counter-productive to the allocation of more meeting minutes. &nbsp;And w=
hen you have, say, 8 chairs for four WG, all of which know the RTP architect=
ure well enough, then you have pretty much exhausted, if not overdrawn alrea=
dy, the pool of "key knowledgable people", and burdened them with (in many c=
ases: additional) chair duties.</div><div><br></div><div>No matter how I see=
 it, splitting the group is not a good idea.</div><div><br></div><div>Other =
options to alleviate problems could come to mind:</div><div><br></div><div>I=
f the problem to be solved is chair-workload: add more chairs (but not 6 of =
them).</div><div>If the problem to be solved is meeting-second shortages: ha=
ve interim meetings, conf calls, etc. &nbsp;In other words, stretch out the =
AVT meeting in the interval between IETF meetings. &nbsp;Require a web-ex pr=
esentation of every draft to be discussed in the AVT session at least 4 week=
s before the meeting, with a two hour timeslot for Q/A. &nbsp;Should I conti=
nue with this brainstorm?</div><div><br></div><div>If the problem is somethi=
ng else, I would like to know&#8230;</div><div><br></div><div>Thanks for you=
r consideration, and apologies for the wordcount,</div><div><br></div><div>S=
tephan</div><div><br></div><div><br></div><div><br></div></body></html>

--B_3377693488_23135744--



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Date: Thu, 13 Jan 2011 10:17:42 +0800
From: Qin Wu <sunseawq@huawei.com>
To: avt@ietf.org
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Subject: [AVT] Fw: WG Review: Metric Blocks for use with RTCP's Extended Report Framework (xrblock)
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FYI.
----- Original Message ----- 
From: Qin Wu 
To: iesg@ietf.org 
Sent: Thursday, January 13, 2011 10:07 AM
Subject: Fw: [AVT] WG Review: Metric Blocks for use with RTCP's Extended Report Framework (xrblock)


Hi, IESG:
Since the proposed XRBLOCK new working group is chartered based on RTP Monitoring Architecture
and RFC5968, rather than develop RTP Monitoring Architecture in AVTCore,
why not move the milestone for RTP Monitoring Architecture 
from AVTCore to  XRBlock and build with other Report Block work items together?

Regards!
-Qin
----- Original Message ----- 
From: Roni Even 
To: avt@ietf.org 
Sent: Thursday, January 13, 2011 2:31 AM
Subject: [AVT] WG Review: Metric Blocks for use with RTCP's Extended Report Framework (xrblock)


FYI,

Please review and send any comments to iesg@ietf.org

Roni Even

 

A new IETF working group has been proposed in the Real-Time Applications

and Infrastructure Area.  The IESG has not made any determination as yet.

The following draft charter was submitted, and is provided for

informational purposes only. Please send your comments to the IESG mailing

list (iesg at ietf.org) by Tuesday, January 18, 2011.               

 

Metric Blocks for use with RTCP's Extended Report Framework (xrblock)

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

Current Status: p>

Last modified: 2010-12-03

 

Chairs: TBD

 

Real-Time Applications and Infrastructure Area Directors:

  Gonzalo Camarillo <gonzalo.camarillo at ericsson.com>

  Robert Sparks <rjsparks at nostrum.com>

Real-Time Applications and Infrastructure Area Advisor:

  Robert Sparks <rjsparks at nostrum.com>

 

Mailing Lists: TBD

 

Description of Working Group:

 

RFC3611, "RTP Control Protocol Extended Re ports (R 

a framework to allow new information to be conveyed in RTCP, 

supplementing the original report blocks defined in RFC 3550, "RTP: A 

Transport Protocol for Real-Time Applications".

 

The XRBLOCK working group is chartered to work within this framework, 

evaluating proposals for report block definitions containing new 

"Guidelines for Extending the RTP Control Protocol (RTCP)" and RTP 

Monitoring Architecture being developed in the RTPCORE working group.

 

Goals and Milestones:

 

Mar 2011  Submit RTCP XR Report Block for Measurement Identity

          Reporting

Mar 2011  Submit RTCP XR Report Block for Burst/Gap Loss metric 

          Reporting

Jun 2011  Submit RTCP XR Report Block for Concealed Seconds metric 

          Reporting

Jun 2 011  lock for Delay metric Reporting

Jun 2011  Submit RTCP XR Report Block for Discard metric Reporting

Sep 2011  Submit RTCP XR Report Block for Jitter Buffer Metric Reporting

Sep 2011  Submit RTCP XR Report Block for Loss Concealment metric 

          Reporting

Sep 2011  Submit RTCP XR Report Block for Packet Delay Variation Metric 

   &n sp;   Reporting

Dec 2011  Submit RTCP XR Report Block for Run Length Encodings of 

          Discarded Packets

Dec 2011  Submit RTCP XR Report Block for QoE Metrics Reporting    

 

 



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


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

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<DIV>FYI.</DIV>
<DIV style="FONT: 9pt &#23435;&#20307;">----- Original Message ----- 
<DIV style="BACKGROUND: #e4e4e4; font-color: black"><B>From:</B> <A 
title=sunseawq@huawei.com href="mailto:sunseawq@huawei.com">Qin Wu</A> </DIV>
<DIV><B>To:</B> <A title=iesg@ietf.org 
href="mailto:iesg@ietf.org">iesg@ietf.org</A> </DIV>
<DIV><B>Sent:</B> Thursday, January 13, 2011 10:07 AM</DIV>
<DIV><B>Subject:</B> Fw: [AVT] WG Review: Metric Blocks for use with RTCP's 
Extended Report Framework (xrblock)</DIV></DIV>
<DIV><BR></DIV>
<DIV><SPAN style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt"><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt"><FONT size=3 
face="Times New Roman">Hi, IESG:<BR>Since the proposed XRBLOCK new working group 
is chartered based on RTP Monitoring Architecture<BR>and RFC5968, rather than 
develop RTP Monitoring Architecture in AVTCore,<BR>why not move the milestone 
for RTP Monitoring Architecture <BR>from AVTCore to&nbsp; XRBlock and build with 
other Report Block work items together?</FONT></SPAN></SPAN></DIV>
<DIV>&nbsp;</DIV>
<DIV><SPAN style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt"><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt"><FONT size=3 
face="Times New Roman">Regards!<BR>-Qin</FONT></SPAN></SPAN></DIV>
<DIV style="FONT: 9pt &#23435;&#20307;">----- Original Message ----- 
<DIV style="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><B>To:</B> <A title=avt@ietf.org 
href="mailto:avt@ietf.org">avt@ietf.org</A> </DIV>
<DIV><B>Sent:</B> Thursday, January 13, 2011 2:31 AM</DIV>
<DIV><B>Subject:</B> [AVT] WG Review: Metric Blocks for use with RTCP's Extended 
Report Framework (xrblock)</DIV></DIV>
<DIV><BR></DIV>
<DIV class=WordSection1>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">FYI,<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Please review and send any 
comments to <A 
href="mailto:iesg@ietf.org">iesg@ietf.org</A><o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Roni 
Even<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt"><o:p>&nbsp;</o:p></SPAN></P>
<P class=MsoNormal><SPAN style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">A 
new IETF working group has been proposed in the Real-Time 
Applications<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">and 
Infrastructure Area.&nbsp; The IESG has not made any determination as 
yet.<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN style="FONT-SIZE: 10pt">The following draft charter was 
submitted, and is provided for<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">informational purposes only. 
Please send your comments to the IESG mailing<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">list (iesg at ietf.org) by 
Tuesday, January 18, 
2011.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt"><o:p>&nbsp;</o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Metric Blocks for use with 
RTCP's Extended Report Framework (xrblock)<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">-------------------------------------------<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Current Status: 
p&gt;</SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Last modified: 
2010-12-03<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt"><o:p>&nbsp;</o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Chairs: 
TBD<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt"><o:p>&nbsp;</o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Real-Time Applications and 
Infrastructure Area Directors:<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">&nbsp; Gonzalo Camarillo 
&lt;gonzalo.camarillo at ericsson.com&gt;<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">&nbsp; Robert Sparks 
&lt;rjsparks at nostrum.com&gt;<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN style="FONT-FAMILY: ' Courier 
span></p><p class=MsoNormal><span style=; FONT-SIZE: 10pt" 
font-size:10.0pt;font-family:?Courier New??>Real-Time Applications and 
Infrastructure Area Advisor:<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">&nbsp; Robert Sparks 
&lt;rjsparks at nostrum.com&gt;<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt"><o:p>&nbsp;</o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Mailing Lists: 
TBD<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt"><o:p>&nbsp;</o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Description of Working 
Group:<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt"><o:p>&nbsp;</o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">RFC3611, "RTP Control 
Protocol Extended Re ports (R <o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">a 
framework to allow new information to be conveyed in RTCP, 
<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">supplementing the original 
report blocks defined in RFC 3550, "RTP: A <o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Transport Protocol for 
Real-Time Applications".<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt"><o:p>&nbsp;</o:p></SPAN></P>
<P class=MsoNormal><SPAN style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">The 
XRBLOCK working group is chartered to work within this framework, 
<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">evaluating proposals for 
report block definitions containing new <o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-SIZE: 10pt; fo: p></o:p></span></p><p class=MsoNormal><span style=" 
font-size:10.0pt;font-family:?Courier New??>"Guidelines for Extending the RTP 
Control Protocol (RTCP)" and RTP <o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Monitoring Architecture 
being developed in the RTPCORE working group.<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt"><o:p>&nbsp;</o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Goals and 
Milestones:<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt"><o:p>&nbsp;</o:p></SPAN></P>
<P class=MsoNormal><SPAN style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Mar 
2011&nbsp; Submit RTCP XR Report Block for Measurement 
Identity<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN style="FONT-FAMILY: 'Cour ier New'
TCP XR Report Block for Burst/Gap Discard metric <o:p></o:p></span></p><p class=MsoNormal><span style=; FONT-SIZE: 10pt" 
font-size:10.0pt;font-family:?Courier 
New??>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Reporting<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Mar 
2011&nbsp; Submit RTCP XR Report Block for Burst/Gap Loss metric 
<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Reporting<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Jun 
2011&nbsp; Submit RTCP XR Report Block for Concealed Seconds metric 
<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Reporting<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Jun 
2 011&nbsp; lock for Delay metric Reporting<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Jun 
2011&nbsp; Submit RTCP XR Report Block for Discard metric 
Reporting<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Sep 
2011&nbsp; Submit RTCP XR Report Block for Jitter Buffer Metric 
Reporting<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Sep 
2011&nbsp; Submit RTCP XR Report Block for Loss Concealment metric 
<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Reporting<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Sep 
2011&nbsp; Submit RTCP XR Report Block for Packet Delay Variation Metric 
<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">&nbsp; &nbsp;&amp;n 
sp;&nbsp;&nbsp;&nbsp;Reporting<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Dec 
2011&nbsp; Submit RTCP XR Report Block for Run Length Encodings of 
<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Discarded 
Packets<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Dec 
2011&nbsp; Submit RTCP XR Report Block for QoE Metrics 
Reporting&nbsp;&nbsp;&nbsp; <o:p></o:p></SPAN></P>
<P class=MsoNormal><o:p>&nbsp;</o:p></P>
<P class=MsoNormal><o:p>&nbsp;</o:p></P></DIV><BR><BR>__________ Information 
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<P></P>_______________________________________________<BR>Audio/Video Transport 
Working 
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Date: Thu, 13 Jan 2011 11:13:43 +0800
From: Qin Wu <sunseawq@huawei.com>
To: iesg@ietf.org, avt@ietf.org
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Subject: [AVT] Fw: WG Review: Metric Blocks for use with RTCP's Extended Report Framework (xrblock)
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Hi, iesg:
As for milestone, I remembered some new XR works have been identified and discussed in the last two or three meetings. 
And based on the results of meetings, these work are waiting for being taken on after monitoring architecture work is complete. 
So isn't better to allow XRblock identify or take on some potential new works?

Regards!
-Qin
----- Original Message ----- 
From: Roni Even 
To: avt@ietf.org 
Sent: Thursday, January 13, 2011 2:31 AM
Subject: [AVT] WG Review: Metric Blocks for use with RTCP's Extended Report Framework (xrblock)


FYI,

Please review and send any comments to iesg@ietf.org

Roni Even

 

A new IETF working group has been proposed in the Real-Time Applications

and Infrastructure Area.  The IESG has not made any determination as yet.

The following draft charter was submitted, and is provided for

informational purposes only. Please send your comments to the IESG mailing

list (iesg at ietf.org) by Tuesday, January 18, 2011.               

 

Metric Blocks for use with RTCP's Extended Report Framework (xrblock)

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

Current Status: p>

Last modified: 2010-12-03

 

Chairs: TBD

 

Real-Time Applications and Infrastructure Area Directors:

  Gonzalo Camarillo <gonzalo.camarillo at ericsson.com>

  Robert Sparks <rjsparks at nostrum.com>

Real-Time Applications and Infrastructure Area Advisor:

  Robert Sparks <rjsparks at nostrum.com>

 

Mailing Lists: TBD

 

Description of Working Group:

 

RFC3611, "RTP Control Protocol Extended Re ports (R 

a framework to allow new information to be conveyed in RTCP, 

supplementing the original report blocks defined in RFC 3550, "RTP: A 

Transport Protocol for Real-Time Applications".

 

The XRBLOCK working group is chartered to work within this framework, 

evaluating proposals for report block definitions containing new 

"Guidelines for Extending the RTP Control Protocol (RTCP)" and RTP 

Monitoring Architecture being developed in the RTPCORE working group.

 

Goals and Milestones:

 

Mar 2011  Submit RTCP XR Report Block for Measurement Identity

          Reporting

Mar 2011  Submit RTCP XR Report Block for Burst/Gap Loss metric 

          Reporting

Jun 2011  Submit RTCP XR Report Block for Concealed Seconds metric 

          Reporting

Jun 2 011  lock for Delay metric Reporting

Jun 2011  Submit RTCP XR Report Block for Discard metric Reporting

Sep 2011  Submit RTCP XR Report Block for Jitter Buffer Metric Reporting

Sep 2011  Submit RTCP XR Report Block for Loss Concealment metric 

          Reporting

Sep 2011  Submit RTCP XR Report Block for Packet Delay Variation Metric 

   &n sp;   Reporting

Dec 2011  Submit RTCP XR Report Block for Run Length Encodings of 

          Discarded Packets

Dec 2011  Submit RTCP XR Report Block for QoE Metrics Reporting    

 

 



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

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<DIV>Hi, iesg:</DIV>
<DIV>As for milestone, I remembered some new XR works have been identified and 
discussed in the last two or three&nbsp;meetings. </DIV>
<DIV>And based on the results of meetings, these work are waiting for being 
taken on after monitoring architecture work is complete. </DIV>
<DIV>So isn't better to&nbsp;allow&nbsp;XRblock identify or take on some 
potential new works?</DIV>
<DIV><FONT size=2 face=&#23435;&#20307;></FONT>&nbsp;</DIV>
<DIV>Regards!</DIV>
<DIV>-Qin</DIV>
<DIV style="FONT: 9pt &#23435;&#20307;">----- Original Message ----- 
<DIV style="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><B>To:</B> <A title=avt@ietf.org 
href="mailto:avt@ietf.org">avt@ietf.org</A> </DIV>
<DIV><B>Sent:</B> Thursday, January 13, 2011 2:31 AM</DIV>
<DIV><B>Subject:</B> [AVT] WG Review: Metric Blocks for use with RTCP's Extended 
Report Framework (xrblock)</DIV></DIV>
<DIV><BR></DIV>
<DIV class=WordSection1>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">FYI,<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Please review and send any 
comments to <A 
href="mailto:iesg@ietf.org">iesg@ietf.org</A><o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Roni 
Even<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt"><o:p>&nbsp;</o:p></SPAN></P>
<P class=MsoNormal><SPAN style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">A 
new IETF working group has been proposed in the Real-Time 
Applications<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">and 
Infrastructure Area.&nbsp; The IESG has not made any determination as 
yet.<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN style="FONT-SIZE: 10pt">The following draft charter was 
submitted, and is provided for<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">informational purposes only. 
Please send your comments to the IESG mailing<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">list (iesg at ietf.org) by 
Tuesday, January 18, 
2011.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt"><o:p>&nbsp;</o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Metric Blocks for use with 
RTCP's Extended Report Framework (xrblock)<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">-------------------------------------------<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Current Status: 
p&gt;</SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Last modified: 
2010-12-03<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt"><o:p>&nbsp;</o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Chairs: 
TBD<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt"><o:p>&nbsp;</o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Real-Time Applications and 
Infrastructure Area Directors:<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">&nbsp; Gonzalo Camarillo 
&lt;gonzalo.camarillo at ericsson.com&gt;<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">&nbsp; Robert Sparks 
&lt;rjsparks at nostrum.com&gt;<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN style="FONT-FAMILY: ' Courier 
span></p><p class=MsoNormal><span style=; FONT-SIZE: 10pt" New?? 
font-size:10.0pt;font-family:?Courier>Real-Time Applications and Infrastructure 
Area Advisor:<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">&nbsp; Robert Sparks 
&lt;rjsparks at nostrum.com&gt;<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt"><o:p>&nbsp;</o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Mailing Lists: 
TBD<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt"><o:p>&nbsp;</o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Description of Working 
Group:<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt"><o:p>&nbsp;</o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">RFC3611, "RTP Control 
Protocol Extended Re ports (R <o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">a 
framework to allow new information to be conveyed in RTCP, 
<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">supplementing the original 
report blocks defined in RFC 3550, "RTP: A <o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Transport Protocol for 
Real-Time Applications".<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt"><o:p>&nbsp;</o:p></SPAN></P>
<P class=MsoNormal><SPAN style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">The 
XRBLOCK working group is chartered to work within this framework, 
<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">evaluating proposals for 
report block definitions containing new <o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-SIZE: 10pt; fo: p></o:p></span></p><p class=MsoNormal><span style=" 
New?? font-size:10.0pt;font-family:?Courier>"Guidelines for Extending the RTP 
Control Protocol (RTCP)" and RTP <o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Monitoring Architecture 
being developed in the RTPCORE working group.<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt"><o:p>&nbsp;</o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Goals and 
Milestones:<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt"><o:p>&nbsp;</o:p></SPAN></P>
<P class=MsoNormal><SPAN style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Mar 
2011&nbsp; Submit RTCP XR Report Block for Measurement 
Identity<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN style="FONT-FAMILY: 'Cour ier New'
TCP XR Report Block for Burst/Gap Discard metric <o:p></o:p></span></p><p class=MsoNormal><span style=; FONT-SIZE: 10pt" 
New?? 
font-size:10.0pt;font-family:?Courier>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Reporting<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Mar 
2011&nbsp; Submit RTCP XR Report Block for Burst/Gap Loss metric 
<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Reporting<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Jun 
2011&nbsp; Submit RTCP XR Report Block for Concealed Seconds metric 
<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Reporting<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Jun 
2 011&nbsp; lock for Delay metric Reporting<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Jun 
2011&nbsp; Submit RTCP XR Report Block for Discard metric 
Reporting<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Sep 
2011&nbsp; Submit RTCP XR Report Block for Jitter Buffer Metric 
Reporting<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Sep 
2011&nbsp; Submit RTCP XR Report Block for Loss Concealment metric 
<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Reporting<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Sep 
2011&nbsp; Submit RTCP XR Report Block for Packet Delay Variation Metric 
<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">&nbsp; &nbsp;&amp;n 
sp;&nbsp;&nbsp;&nbsp;Reporting<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Dec 
2011&nbsp; Submit RTCP XR Report Block for Run Length Encodings of 
<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN 
style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Discarded 
Packets<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN style="FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Dec 
2011&nbsp; Submit RTCP XR Report Block for QoE Metrics 
Reporting&nbsp;&nbsp;&nbsp; <o:p></o:p></SPAN></P>
<P class=MsoNormal><o:p>&nbsp;</o:p></P>
<P class=MsoNormal><o:p>&nbsp;</o:p></P></DIV><BR><BR>__________ Information 
from ESET NOD32 Antivirus, version of virus signature database 5781 (20110112) 
__________<BR><BR>The message was checked by ESET NOD32 Antivirus.<BR><BR><A 
href="http://www.eset.com">http://www.eset.com</A><BR>
<P>
<HR>

<P></P>_______________________________________________<BR>Audio/Video Transport 
Working 
Group<BR>avt@ietf.org<BR>https://www.ietf.org/mailman/listinfo/avt<BR></BODY></HTML>

--Boundary_(ID_n4XmMuBRhfdBHVBsaSD9aA)--

From casner@acm.org  Wed Jan 12 20:30:37 2011
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Date: Wed, 12 Jan 2011 20:32:46 -0800 (PST)
From: Stephen Casner <casner@acm.org>
To: Stephan Wenger <stewe@stewe.org>
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+1, definitely.

When I was chair, I once thought that AVT should try to draw to a
close.  Later I realized that this was not only unlikely but also not
required.

                                                        -- Steve

On Wed, 12 Jan 2011, Stephan Wenger wrote:

> Hi IESG,
>
> As already communicated in Maastricht, I continue to believe that this step
> is a bad idea.  I didn't attend Beijing, and don't know whether there was
> any discussion about the subject.  I didn't see anything mentioned in the
> AVT minutes, though, nor did I see any discussion since Beijing on
> ietf@ietf.  If there was, some of the points below may be redundant, and I
> apologize in advance.
>
> Essentially, you guys are suggesting to split one "eternal" WG into four.
> I'm writing "eternal", because I don't see that any of the WGs can fulfill
> its charter before other work items will likely get chartered with a similar
> scope, and in the same WG.  For RTP core, RTP extensions, and RTP payloads
> that is probably self-evident.  For the XR stuff: metrics pop up all the
> time, like payload formats.  Metrics are also known to make people quick
> bucks on patent licensing.  There will be more metric proposals.  If the
> RTCP-XR metrics ever get momentum in the marketplace, that is.  Somehow, I
> have my doubts.  But that's a different discussion.
>
> The positive effects of the split could be
> -chair workload is split among more people, and
> -there are more meeting seconds for RTP related work at any given IETF
> meeting.
>
> Assuming unlimited resources, both are good things.  Unfortunately,
> resources are not unlimited.
>
> The negative effects are IMO:
> -the key knowledgeable people in AVT ideally would have to sit in all four
> meetings so that all four groups can tap their experience.  But they may not
> want to do so, because it leaves less time for cross-area involvement,
> coordination, and draft-writing outside the AVT sessions, and whatnot.
> Also, the burden for remote participants in far-away time zones goes up.  In
> other words, the resource meeting-seconds for those people is not unlimited,
> and they will have to prioritize differently.    The result could be that
> what I called "key knowledgable people" would perhaps not attend the mere
> RTCP-XR or Payload session; and the guys there may be wasting their time
> doing work that does not make sense architecturally, just to see the hammer
> coming down at IETF last call.
> -more people get chair positions and are burdened with administrative stuff
> -> less time to do real work
> -WG scheduling gets more difficult
> -AD load potentially goes up (someone needs to coordinate between those
> groups, and the AD would seem to me the natural choice)
> -there is no reason to believe that the overall manpower availability goes
> up through this change, so all the inefficiencies introduced go straight
> against the available cycles.
>
> Some of these issues can be solved by joint WG meetings and/or careful chair
> selection.  However, both options have drawbacks as well.  A joint session
> would be counter-productive to the allocation of more meeting minutes.  And
> when you have, say, 8 chairs for four WG, all of which know the RTP
> architecture well enough, then you have pretty much exhausted, if not
> overdrawn already, the pool of "key knowledgable people", and burdened them
> with (in many cases: additional) chair duties.
>
> No matter how I see it, splitting the group is not a good idea.
>
> Other options to alleviate problems could come to mind:
>
> If the problem to be solved is chair-workload: add more chairs (but not 6 of
> them).
> If the problem to be solved is meeting-second shortages: have interim
> meetings, conf calls, etc.  In other words, stretch out the AVT meeting in
> the interval between IETF meetings.  Require a web-ex presentation of every
> draft to be discussed in the AVT session at least 4 weeks before the
> meeting, with a two hour timeslot for Q/A.  Should I continue with this
> brainstorm?
>
> If the problem is something else, I would like to know?
>
> Thanks for your consideration, and apologies for the wordcount,
>
> Stephan

From michael.speer@oracle.com  Wed Jan 12 22:28:34 2011
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+1, go for it.

On 1/12/11 8:32 PM, Stephen Casner wrote:
> +1, definitely.
>
> When I was chair, I once thought that AVT should try to draw to a
> close.  Later I realized that this was not only unlikely but also not
> required.
>
>                                                          -- Steve
>
> On Wed, 12 Jan 2011, Stephan Wenger wrote:
>
>> Hi IESG,
>>
>> As already communicated in Maastricht, I continue to believe that this step
>> is a bad idea.  I didn't attend Beijing, and don't know whether there was
>> any discussion about the subject.  I didn't see anything mentioned in the
>> AVT minutes, though, nor did I see any discussion since Beijing on
>> ietf@ietf.  If there was, some of the points below may be redundant, and I
>> apologize in advance.
>>
>> Essentially, you guys are suggesting to split one "eternal" WG into four.
>> I'm writing "eternal", because I don't see that any of the WGs can fulfill
>> its charter before other work items will likely get chartered with a similar
>> scope, and in the same WG.  For RTP core, RTP extensions, and RTP payloads
>> that is probably self-evident.  For the XR stuff: metrics pop up all the
>> time, like payload formats.  Metrics are also known to make people quick
>> bucks on patent licensing.  There will be more metric proposals.  If the
>> RTCP-XR metrics ever get momentum in the marketplace, that is.  Somehow, I
>> have my doubts.  But that's a different discussion.
>>
>> The positive effects of the split could be
>> -chair workload is split among more people, and
>> -there are more meeting seconds for RTP related work at any given IETF
>> meeting.
>>
>> Assuming unlimited resources, both are good things.  Unfortunately,
>> resources are not unlimited.
>>
>> The negative effects are IMO:
>> -the key knowledgeable people in AVT ideally would have to sit in all four
>> meetings so that all four groups can tap their experience.  But they may not
>> want to do so, because it leaves less time for cross-area involvement,
>> coordination, and draft-writing outside the AVT sessions, and whatnot.
>> Also, the burden for remote participants in far-away time zones goes up.  In
>> other words, the resource meeting-seconds for those people is not unlimited,
>> and they will have to prioritize differently.    The result could be that
>> what I called "key knowledgable people" would perhaps not attend the mere
>> RTCP-XR or Payload session; and the guys there may be wasting their time
>> doing work that does not make sense architecturally, just to see the hammer
>> coming down at IETF last call.
>> -more people get chair positions and are burdened with administrative stuff
>> ->  less time to do real work
>> -WG scheduling gets more difficult
>> -AD load potentially goes up (someone needs to coordinate between those
>> groups, and the AD would seem to me the natural choice)
>> -there is no reason to believe that the overall manpower availability goes
>> up through this change, so all the inefficiencies introduced go straight
>> against the available cycles.
>>
>> Some of these issues can be solved by joint WG meetings and/or careful chair
>> selection.  However, both options have drawbacks as well.  A joint session
>> would be counter-productive to the allocation of more meeting minutes.  And
>> when you have, say, 8 chairs for four WG, all of which know the RTP
>> architecture well enough, then you have pretty much exhausted, if not
>> overdrawn already, the pool of "key knowledgable people", and burdened them
>> with (in many cases: additional) chair duties.
>>
>> No matter how I see it, splitting the group is not a good idea.
>>
>> Other options to alleviate problems could come to mind:
>>
>> If the problem to be solved is chair-workload: add more chairs (but not 6 of
>> them).
>> If the problem to be solved is meeting-second shortages: have interim
>> meetings, conf calls, etc.  In other words, stretch out the AVT meeting in
>> the interval between IETF meetings.  Require a web-ex presentation of every
>> draft to be discussed in the AVT session at least 4 weeks before the
>> meeting, with a two hour timeslot for Q/A.  Should I continue with this
>> brainstorm?
>>
>> If the problem is something else, I would like to know?
>>
>> Thanks for your consideration, and apologies for the wordcount,
>>
>> Stephan
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt


-- 
Michael Speer
Principal Software Engineer, Solaris Core OS
Oracle
e-mail: michael.speer@oracle.com
phone: +1 408 276-6943


From casner@acm.org  Wed Jan 12 23:28:21 2011
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From: Stephen Casner <casner@acm.org>
To: Michael Speer <michael.speer@oracle.com>
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Hmmm, Michael's message makes me think mine may have not been clear.
By +1, I meant that I agree with Stephan that splitting AVT into four
WGs is a bad idea.

                                                        -- Steve

On Wed, 12 Jan 2011, Michael Speer wrote:

> +1, go for it.
>
> On 1/12/11 8:32 PM, Stephen Casner wrote:
> > +1, definitely.
> >
> > When I was chair, I once thought that AVT should try to draw to a
> > close.  Later I realized that this was not only unlikely but also not
> > required.
> >
> >                                                          -- Steve
> >
> > On Wed, 12 Jan 2011, Stephan Wenger wrote:
> >
> > > Hi IESG,
> > >
> > > As already communicated in Maastricht, I continue to believe that this
> > > step
> > > is a bad idea.  I didn't attend Beijing, and don't know whether there was
> > > any discussion about the subject.  I didn't see anything mentioned in the
> > > AVT minutes, though, nor did I see any discussion since Beijing on
> > > ietf@ietf.  If there was, some of the points below may be redundant, and I
> > > apologize in advance.
> > >
> > > Essentially, you guys are suggesting to split one "eternal" WG into four.
> > > I'm writing "eternal", because I don't see that any of the WGs can fulfill
> > > its charter before other work items will likely get chartered with a
> > > similar
> > > scope, and in the same WG.  For RTP core, RTP extensions, and RTP payloads
> > > that is probably self-evident.  For the XR stuff: metrics pop up all the
> > > time, like payload formats.  Metrics are also known to make people quick
> > > bucks on patent licensing.  There will be more metric proposals.  If the
> > > RTCP-XR metrics ever get momentum in the marketplace, that is.  Somehow, I
> > > have my doubts.  But that's a different discussion.
> > >
> > > The positive effects of the split could be
> > > -chair workload is split among more people, and
> > > -there are more meeting seconds for RTP related work at any given IETF
> > > meeting.
> > >
> > > Assuming unlimited resources, both are good things.  Unfortunately,
> > > resources are not unlimited.
> > >
> > > The negative effects are IMO:
> > > -the key knowledgeable people in AVT ideally would have to sit in all four
> > > meetings so that all four groups can tap their experience.  But they may
> > > not
> > > want to do so, because it leaves less time for cross-area involvement,
> > > coordination, and draft-writing outside the AVT sessions, and whatnot.
> > > Also, the burden for remote participants in far-away time zones goes up.
> > > In
> > > other words, the resource meeting-seconds for those people is not
> > > unlimited,
> > > and they will have to prioritize differently.    The result could be that
> > > what I called "key knowledgable people" would perhaps not attend the mere
> > > RTCP-XR or Payload session; and the guys there may be wasting their time
> > > doing work that does not make sense architecturally, just to see the
> > > hammer
> > > coming down at IETF last call.
> > > -more people get chair positions and are burdened with administrative
> > > stuff
> > > ->  less time to do real work
> > > -WG scheduling gets more difficult
> > > -AD load potentially goes up (someone needs to coordinate between those
> > > groups, and the AD would seem to me the natural choice)
> > > -there is no reason to believe that the overall manpower availability goes
> > > up through this change, so all the inefficiencies introduced go straight
> > > against the available cycles.
> > >
> > > Some of these issues can be solved by joint WG meetings and/or careful
> > > chair
> > > selection.  However, both options have drawbacks as well.  A joint session
> > > would be counter-productive to the allocation of more meeting minutes.
> > > And
> > > when you have, say, 8 chairs for four WG, all of which know the RTP
> > > architecture well enough, then you have pretty much exhausted, if not
> > > overdrawn already, the pool of "key knowledgable people", and burdened
> > > them
> > > with (in many cases: additional) chair duties.
> > >
> > > No matter how I see it, splitting the group is not a good idea.
> > >
> > > Other options to alleviate problems could come to mind:
> > >
> > > If the problem to be solved is chair-workload: add more chairs (but not 6
> > > of
> > > them).
> > > If the problem to be solved is meeting-second shortages: have interim
> > > meetings, conf calls, etc.  In other words, stretch out the AVT meeting in
> > > the interval between IETF meetings.  Require a web-ex presentation of
> > > every
> > > draft to be discussed in the AVT session at least 4 weeks before the
> > > meeting, with a two hour timeslot for Q/A.  Should I continue with this
> > > brainstorm?
> > >
> > > If the problem is something else, I would like to know?
> > >
> > > Thanks for your consideration, and apologies for the wordcount,
> > >
> > > Stephan
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www.ietf.org/mailman/listinfo/avt
>
>
> --
> Michael Speer
> Principal Software Engineer, Solaris Core OS
> Oracle
> e-mail: michael.speer@oracle.com
> phone: +1 408 276-6943
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt

From john.elwell@siemens-enterprise.com  Thu Jan 13 00:53:14 2011
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Stephan Wenger <stewe@stewe.org>, "iesg@ietf.org" <iesg@ietf.org>
Date: Thu, 13 Jan 2011 09:55:22 +0100
Thread-Topic: proposed AVT WG split
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Subject: Re: [AVT] proposed AVT WG split
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I have to agree with much, if not all, of what Stephan says. It would help,=
 of course, if those proposing the change could give a clear statement as t=
o what they consider to be broken with the current arrangement.=20

I tend to have an interest in some, but not all, of the activities AVT is e=
ngaged in. Therefore, I tend to pay attention to only parts of AVT face-to-=
face sessions. It is not clear that the proposed split would make my partic=
ipation easier - I suspect I would still be in a position of selecting part=
icular topics from each of the WG sessions. If anything it would probably m=
ake things harder - 4 WGs would typically involve 4 face-to-face sessions, =
rather than the 2 that AVT typically holds, making conflicts with other WGs=
 in the RAI area (or elsewhere, for that matter) more likely.

John

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On=20
> Behalf Of Stephan Wenger
> Sent: 13 January 2011 00:11
> To: iesg@ietf.org
> Cc: avt@ietf.org
> Subject: [AVT] proposed AVT WG split
>=20
> Hi IESG,
>=20
> As already communicated in Maastricht, I continue to believe=20
> that this step is a bad idea.  I didn't attend Beijing, and=20
> don't know whether there was any discussion about the=20
> subject.  I didn't see anything mentioned in the AVT minutes,=20
> though, nor did I see any discussion since Beijing on=20
> ietf@ietf.  If there was, some of the points below may be=20
> redundant, and I apologize in advance.
>=20
> Essentially, you guys are suggesting to split one "eternal"=20
> WG into four.  I'm writing "eternal", because I don't see=20
> that any of the WGs can fulfill its charter before other work=20
> items will likely get chartered with a similar scope, and in=20
> the same WG.  For RTP core, RTP extensions, and RTP payloads=20
> that is probably self-evident.  For the XR stuff: metrics pop=20
> up all the time, like payload formats.  Metrics are also=20
> known to make people quick bucks on patent licensing.  There=20
> will be more metric proposals.  If the RTCP-XR metrics ever=20
> get momentum in the marketplace, that is.  Somehow, I have my=20
> doubts.  But that's a different discussion. =20
>=20
> The positive effects of the split could be=20
> -chair workload is split among more people, and=20
> -there are more meeting seconds for RTP related work at any=20
> given IETF meeting. =20
>=20
> Assuming unlimited resources, both are good things. =20
> Unfortunately, resources are not unlimited.
>=20
> The negative effects are IMO:
> -the key knowledgeable people in AVT ideally would have to=20
> sit in all four meetings so that all four groups can tap=20
> their experience.  But they may not want to do so, because it=20
> leaves less time for cross-area involvement, coordination,=20
> and draft-writing outside the AVT sessions, and whatnot. =20
> Also, the burden for remote participants in far-away time=20
> zones goes up.  In other words, the resource meeting-seconds=20
> for those people is not unlimited, and they will have to=20
> prioritize differently.    The result could be that what I=20
> called "key knowledgable people" would perhaps not attend the=20
> mere RTCP-XR or Payload session; and the guys there may be=20
> wasting their time doing work that does not make sense=20
> architecturally, just to see the hammer coming down at IETF last call.
> -more people get chair positions and are burdened with=20
> administrative stuff -> less time to do real work
> -WG scheduling gets more difficult
> -AD load potentially goes up (someone needs to coordinate=20
> between those groups, and the AD would seem to me the natural choice)
> -there is no reason to believe that the overall manpower=20
> availability goes up through this change, so all the=20
> inefficiencies introduced go straight against the available cycles.
>=20
> Some of these issues can be solved by joint WG meetings=20
> and/or careful chair selection.  However, both options have=20
> drawbacks as well.  A joint session would be=20
> counter-productive to the allocation of more meeting minutes.=20
>  And when you have, say, 8 chairs for four WG, all of which=20
> know the RTP architecture well enough, then you have pretty=20
> much exhausted, if not overdrawn already, the pool of "key=20
> knowledgable people", and burdened them with (in many cases:=20
> additional) chair duties.
>=20
> No matter how I see it, splitting the group is not a good idea.
>=20
> Other options to alleviate problems could come to mind:
>=20
> If the problem to be solved is chair-workload: add more=20
> chairs (but not 6 of them).
> If the problem to be solved is meeting-second shortages: have=20
> interim meetings, conf calls, etc.  In other words, stretch=20
> out the AVT meeting in the interval between IETF meetings. =20
> Require a web-ex presentation of every draft to be discussed=20
> in the AVT session at least 4 weeks before the meeting, with=20
> a two hour timeslot for Q/A.  Should I continue with this brainstorm?
>=20
> If the problem is something else, I would like to know...
>=20
> Thanks for your consideration, and apologies for the wordcount,
>=20
> Stephan
>=20
>=20
>=20
> =

From keith.drage@alcatel-lucent.com  Thu Jan 13 10:57:02 2011
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, Stephan Wenger <stewe@stewe.org>, "iesg@ietf.org" <iesg@ietf.org>
Date: Thu, 13 Jan 2011 19:59:20 +0100
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Cc: "avt@ietf.org" <avt@ietf.org>
Subject: Re: [AVT] proposed AVT WG split
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This is AD initiated rather than WG chair initiated, although the ADs have =
been discussing it with the WG chairs, and indeed also with the other RAI c=
hairs.

One of the issues we do have is WG chair load. Every time I do a review of =
the current status of our working group items, it takes me over 3 hours. Th=
at is just the sheer quantity of milestones we have. It therefore doesn't h=
appen as often as it ought to because that is 3 hours away from doing other=
 tasks. =20

Currently in each meeting cycle we are starting the publication request pro=
cess for on avaerage two documents each - i.e. AVT is forwarding 12 documen=
ts a year to IESG. That is also a heavy chair load.

The meeting load issues has already been discussed. I believe that the payl=
oad group would meet rarely. In formal IETF terms it is really doing an exp=
ert review process, but apparently needs a WG to do it, because we haven't =
been able to codify all the good practice that an expert review should take=
 into account. We are not spending a lot of our current meeting time as AVT=
 discussing new or revised payload types.

So your average 2 + 1 AVT sessions will probably turn into 1 + 1 + 1 workin=
g group sessions of the new groups if this goes ahead.

Keith

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On=20
> Behalf Of Elwell, John
> Sent: Thursday, January 13, 2011 8:55 AM
> To: Stephan Wenger; iesg@ietf.org
> Cc: avt@ietf.org
> Subject: Re: [AVT] proposed AVT WG split
>=20
> I have to agree with much, if not all, of what Stephan says.=20
> It would help, of course, if those proposing the change could=20
> give a clear statement as to what they consider to be broken=20
> with the current arrangement.=20
>=20
> I tend to have an interest in some, but not all, of the=20
> activities AVT is engaged in. Therefore, I tend to pay=20
> attention to only parts of AVT face-to-face sessions. It is=20
> not clear that the proposed split would make my participation=20
> easier - I suspect I would still be in a position of=20
> selecting particular topics from each of the WG sessions. If=20
> anything it would probably make things harder - 4 WGs would=20
> typically involve 4 face-to-face sessions, rather than the 2=20
> that AVT typically holds, making conflicts with other WGs in=20
> the RAI area (or elsewhere, for that matter) more likely.
>=20
> John
>=20
> > -----Original Message-----
> > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On=20
> Behalf Of=20
> > Stephan Wenger
> > Sent: 13 January 2011 00:11
> > To: iesg@ietf.org
> > Cc: avt@ietf.org
> > Subject: [AVT] proposed AVT WG split
> >=20
> > Hi IESG,
> >=20
> > As already communicated in Maastricht, I continue to=20
> believe that this=20
> > step is a bad idea.  I didn't attend Beijing, and don't=20
> know whether=20
> > there was any discussion about the subject.  I didn't see anything=20
> > mentioned in the AVT minutes, though, nor did I see any discussion=20
> > since Beijing on ietf@ietf.  If there was, some of the points below=20
> > may be redundant, and I apologize in advance.
> >=20
> > Essentially, you guys are suggesting to split one "eternal"=20
> > WG into four.  I'm writing "eternal", because I don't see=20
> that any of=20
> > the WGs can fulfill its charter before other work items will likely=20
> > get chartered with a similar scope, and in the same WG. =20
> For RTP core,=20
> > RTP extensions, and RTP payloads that is probably=20
> self-evident.  For=20
> > the XR stuff: metrics pop up all the time, like payload formats. =20
> > Metrics are also known to make people quick bucks on patent=20
> licensing. =20
> > There will be more metric proposals.  If the RTCP-XR=20
> metrics ever get=20
> > momentum in the marketplace, that is.  Somehow, I have my=20
> doubts.  But=20
> > that's a different discussion.
> >=20
> > The positive effects of the split could be -chair workload is split=20
> > among more people, and -there are more meeting seconds for=20
> RTP related=20
> > work at any given IETF meeting.
> >=20
> > Assuming unlimited resources, both are good things. =20
> > Unfortunately, resources are not unlimited.
> >=20
> > The negative effects are IMO:
> > -the key knowledgeable people in AVT ideally would have to=20
> sit in all=20
> > four meetings so that all four groups can tap their=20
> experience.  But=20
> > they may not want to do so, because it leaves less time for=20
> cross-area=20
> > involvement, coordination, and draft-writing outside the=20
> AVT sessions,=20
> > and whatnot.
> > Also, the burden for remote participants in far-away time=20
> zones goes=20
> > up.  In other words, the resource meeting-seconds for those=20
> people is=20
> > not unlimited, and they will have to
> > prioritize differently.    The result could be that what I=20
> > called "key knowledgable people" would perhaps not attend the mere=20
> > RTCP-XR or Payload session; and the guys there may be wasting their=20
> > time doing work that does not make sense architecturally,=20
> just to see=20
> > the hammer coming down at IETF last call.
> > -more people get chair positions and are burdened with=20
> administrative=20
> > stuff -> less time to do real work -WG scheduling gets more=20
> difficult=20
> > -AD load potentially goes up (someone needs to coordinate between=20
> > those groups, and the AD would seem to me the natural=20
> choice) -there=20
> > is no reason to believe that the overall manpower=20
> availability goes up=20
> > through this change, so all the inefficiencies introduced=20
> go straight=20
> > against the available cycles.
> >=20
> > Some of these issues can be solved by joint WG meetings=20
> and/or careful=20
> > chair selection.  However, both options have drawbacks as well.  A=20
> > joint session would be counter-productive to the allocation of more=20
> > meeting minutes.
> >  And when you have, say, 8 chairs for four WG, all of which=20
> know the=20
> > RTP architecture well enough, then you have pretty much=20
> exhausted, if=20
> > not overdrawn already, the pool of "key knowledgable people", and=20
> > burdened them with (in many cases:
> > additional) chair duties.
> >=20
> > No matter how I see it, splitting the group is not a good idea.
> >=20
> > Other options to alleviate problems could come to mind:
> >=20
> > If the problem to be solved is chair-workload: add more chairs (but=20
> > not 6 of them).
> > If the problem to be solved is meeting-second shortages:=20
> have interim=20
> > meetings, conf calls, etc.  In other words, stretch out the AVT=20
> > meeting in the interval between IETF meetings.
> > Require a web-ex presentation of every draft to be discussed in the=20
> > AVT session at least 4 weeks before the meeting, with a two hour=20
> > timeslot for Q/A.  Should I continue with this brainstorm?
> >=20
> > If the problem is something else, I would like to know...
> >=20
> > Thanks for your consideration, and apologies for the wordcount,
> >=20
> > Stephan
> >=20
> >=20
> >=20
> >=20
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
> =

From Even.roni@huawei.com  Thu Jan 13 11:56:39 2011
Return-Path: <Even.roni@huawei.com>
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Date: Thu, 13 Jan 2011 21:59:19 +0200
From: Roni Even <Even.roni@huawei.com>
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To: 'Stephan Wenger' <stewe@stewe.org>, iesg@ietf.org
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Subject: Re: [AVT] proposed AVT WG split
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Hi,

I agree with Stephan view. My experience in AVT is that the WG review
process depends on a number of individual that has a general interest in RTP
and since payloads and RTCP XR are discussed in the same WG are contributing
to this effort. Breaking the WG to 3 or 4 will not be helpful for these
persons. If the issue is load on chairs we can add one more chair to help
with document shepherding

As for the current proposal if there is a strong support for it I think that
the XR WG make sense bit should include the monitoring architecture document
to it.

As for RTP core and text this is an artificial division. We need one WG and
if there is work that is not in the charter it  should go to DISPATCH who
will decide if to have a new group or assign it to AVT.

The payload work should be part of the main AVT group since we want the
people who are interested in AVT but are following and helping the payload
work just because it is part of AVT to continue and not go away.

To summarize the only division I can see if for RTCP XR for monitoring WG.

 

Thanks

Roni Even as individual

 

From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
Stephan Wenger
Sent: Thursday, January 13, 2011 2:11 AM
To: iesg@ietf.org
Cc: avt@ietf.org
Subject: [AVT] proposed AVT WG split

 

Hi IESG,

 

As already communicated in Maastricht, I continue to believe that this step
is a bad idea.  I didn't attend Beijing, and don't know whether there was
any discussion about the subject.  I didn't see anything mentioned in the
AVT minutes, though, nor did I see any discussion since Beijing on
ietf@ietf.  If there was, some of the points below may be redundant, and I
apologize in advance.

 

Essentially, you guys are suggesting to split one "eternal" WG into four.
I'm writing "eternal", because I don't see that any of the WGs can fulfill
its charter before other work items will likely get chartered with a similar
scope, and in the same WG.  For RTP core, RTP extensions, and RTP payloads
that is probably self-evident.  For the XR stuff: metrics pop up all the
time, like payload formats.  Metrics are also known to make people quick
bucks on patent licensing.  There will be more metric proposals.  If the
RTCP-XR metrics ever get momentum in the marketplace, that is.  Somehow, I
have my doubts.  But that's a different discussion.  

 

The positive effects of the split could be 

-chair workload is split among more people, and 

-there are more meeting seconds for RTP related work at any given IETF
meeting.  

 

Assuming unlimited resources, both are good things.  Unfortunately,
resources are not unlimited.

 

The negative effects are IMO:

-the key knowledgeable people in AVT ideally would have to sit in all four
meetings so that all four groups can tap their experience.  But they may not
want to do so, because it leaves less time for cross-area involvement,
coordination, and draft-writing outside the AVT sessions, and whatnot.
Also, the burden for remote participants in far-away time zones goes up.  In
other words, the resource meeting-seconds for those people is not unlimited,
and they will have to prioritize differently.    The result could be that
what I called "key knowledgable people" would perhaps not attend the mere
RTCP-XR or Payload session; and the guys there may be wasting their time
doing work that does not make sense architecturally, just to see the hammer
coming down at IETF last call.

-more people get chair positions and are burdened with administrative stuff
-> less time to do real work

-WG scheduling gets more difficult

-AD load potentially goes up (someone needs to coordinate between those
groups, and the AD would seem to me the natural choice)

-there is no reason to believe that the overall manpower availability goes
up through this change, so all the inefficiencies introduced go straight
against the available cycles.

 

Some of these issues can be solved by joint WG meetings and/or careful chair
selection.  However, both options have drawbacks as well.  A joint session
would be counter-productive to the allocation of more meeting minutes.  And
when you have, say, 8 chairs for four WG, all of which know the RTP
architecture well enough, then you have pretty much exhausted, if not
overdrawn already, the pool of "key knowledgable people", and burdened them
with (in many cases: additional) chair duties.

 

No matter how I see it, splitting the group is not a good idea.

 

Other options to alleviate problems could come to mind:

 

If the problem to be solved is chair-workload: add more chairs (but not 6 of
them).

If the problem to be solved is meeting-second shortages: have interim
meetings, conf calls, etc.  In other words, stretch out the AVT meeting in
the interval between IETF meetings.  Require a web-ex presentation of every
draft to be discussed in the AVT session at least 4 weeks before the
meeting, with a two hour timeslot for Q/A.  Should I continue with this
brainstorm?

 

If the problem is something else, I would like to know.

 

Thanks for your consideration, and apologies for the wordcount,

 

Stephan

 

 

 



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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple style='word-wrap: break-word;-webkit-nbsp-mode: space;-webkit-line-break: after-white-space'><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Hi,<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>I agree with Stephan view. My experience in AVT is that the WG review process depends on a number of individual that has a general interest in RTP and since payloads and RTCP XR are discussed in the same WG are contributing to this effort. Breaking the WG to 3 or 4 will not be helpful for these persons. If the issue is load on chairs we can add one more chair to help with document shepherding<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>As for the current proposal if there is a strong supp
 ort for 
 make sense bit should include the monitoring architecture document to it.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>As for RTP core and text this is an artificial division. We need one WG and if there is work that is not in the charter it&nbsp; should go to DISPATCH who will decide if to have a new group or assign it to AVT.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>The payload work should be part of the main AVT group since we want the people who are interested in AVT but are following and helping the payload work just because it is part of AVT to continue and not go away.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>To summarize the only division I can see if for RTCP XR for monitoring WG.<o:p></o:p></span></p><p class=MsoNormal><span style='font
 -size:11
","sans-serif";color:#1F497D'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Thanks<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Roni Even as individual<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p>&nbsp;</o:p></span></p><div style='border:none;border-right:solid blue 1.5pt;padding:0cm 0cm 0cm 4.0pt'><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] <b>On Behalf Of </b>Stephan Wenger<br><b>Sent:</b> Thursday, January 13, 2011 2:11 AM<br><b>To:</b> iesg@ietf.org<br><b>Cc:</b> avt@ietf.org<br><b
 >Subject
WG split<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p>&nbsp;</o:p></p><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>Hi IESG,<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'><o:p>&nbsp;</o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>As already communicated in Maastricht, I continue to believe that this step is a bad idea. &nbsp;I didn't attend Beijing, and don't know whether there was any discussion about the subject. &nbsp;I didn't see anything mentioned in the AVT minutes, though, nor did I see any discussion since Beijing on ietf@ietf. &nbsp;If there was, some of the points below may be redundant, and I apologize in advance.<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>
 <o:p>&nb
><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>Essentially, you guys are suggesting to split one &quot;eternal&quot; WG into four. &nbsp;I'm writing &quot;eternal&quot;, because I don't see that any of the WGs can fulfill its charter before other work items will likely get chartered with a similar scope, and in the same WG. &nbsp;For RTP core, RTP extensions, and RTP payloads that is probably self-evident. &nbsp;For the XR stuff: metrics pop up all the time, like payload formats. &nbsp;Metrics are also known to make people quick bucks on patent licensing. &nbsp;There will be more metric proposals. &nbsp;If the RTCP-XR metrics ever get momentum in the marketplace, that is. &nbsp;Somehow, I have my doubts. &nbsp;But that's a different discussion. &nbsp;<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'><o:p>&nbsp;</o:p></span></p></div><div><p c
 lass=Mso
size:10.5pt;font-family:"Calibri","sans-serif";color:black'>The positive effects of the split could be&nbsp;<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>-chair workload is split among more people, and&nbsp;<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>-there are more meeting seconds for RTP related work at any given IETF meeting. &nbsp;<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'><o:p>&nbsp;</o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>Assuming unlimited resources, both are good things. &nbsp;Unfortunately, resources are not unlimited.<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif"
 ;color:b
span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>The negative effects are IMO:<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>-the key knowledgeable people in AVT ideally would have to sit in all four meetings so that all four groups can tap their experience. &nbsp;But they may not want to do so, because it leaves less time for cross-area involvement, coordination, and draft-writing outside the AVT sessions, and whatnot. &nbsp;Also, the burden for remote participants in far-away time zones goes up. &nbsp;In other words, the resource meeting-seconds for those people is not unlimited, and they will have to prioritize differently. &nbsp; &nbsp;The result could be that what I called &quot;key knowledgable people&quot; would perhaps not attend the mere RTCP-XR or Payload session; and the guys there may be wasting their time doing wor
 k that d
ecturally, just to see the hammer coming down at IETF last call.<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>-more people get chair positions and are burdened with administrative stuff -&gt; less time to do real work<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>-WG scheduling gets more difficult<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>-AD load potentially goes up (someone needs to coordinate between those groups, and the AD would seem to me the natural choice)<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>-there is no reason to believe that the overall manpower availability goes up through this change, so all the inefficiencies introduced go
  straigh
ycles.<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'><o:p>&nbsp;</o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>Some of these issues can be solved by joint WG meetings and/or careful chair selection. &nbsp;However, both options have drawbacks as well. &nbsp;A joint session would be counter-productive to the allocation of more meeting minutes. &nbsp;And when you have, say, 8 chairs for four WG, all of which know the RTP architecture well enough, then you have pretty much exhausted, if not overdrawn already, the pool of &quot;key knowledgable people&quot;, and burdened them with (in many cases: additional) chair duties.<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'><o:p>&nbsp;</o:p></span></p></div><div><p class=MsoNormal><span style='font
 -size:10
","sans-serif";color:black'>No matter how I see it, splitting the group is not a good idea.<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'><o:p>&nbsp;</o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>Other options to alleviate problems could come to mind:<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'><o:p>&nbsp;</o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>If the problem to be solved is chair-workload: add more chairs (but not 6 of them).<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>If the problem to be solved is meeting-second shortages: have interim meetings, conf calls, etc. &n
 bsp;In o
he AVT meeting in the interval between IETF meetings. &nbsp;Require a web-ex presentation of every draft to be discussed in the AVT session at least 4 weeks before the meeting, with a two hour timeslot for Q/A. &nbsp;Should I continue with this brainstorm?<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'><o:p>&nbsp;</o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>If the problem is something else, I would like to know&#8230;<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'><o:p>&nbsp;</o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>Thanks for your consideration, and apologies for the wordcount,<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.
 5pt;font
erif";color:black'><o:p>&nbsp;</o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>Stephan<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'><o:p>&nbsp;</o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'><o:p>&nbsp;</o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'><o:p>&nbsp;</o:p></span></p></div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'><br><br>__________ Information from ESET NOD32 Antivirus, version of virus signature database 5782 (20110112) __________<br><br>The message was checked by ESET NOD32 Antivirus.<br><br><a href="http://www.eset.com">http://www.eset.com</a><br><br><br>__________ Information from ESET NOD32 Antivir
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Subject: Re: [AVT] proposed AVT WG split
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Stephan -

As I noted in response to your comments in Maastricht, I disagree.

There are separable streams of work in AVT - it is really multiple =
working groups in one shell at the moment.

While there are key people currently in AVT that would overlap these =
groups, they are not the majority of the contributors to the different =
types of activities.
The intended positive effect of this explicit division is to allow these =
different streams to better parallelize, on list between meetings, not =
just during the IETF
in-person meeting time.=20

We'd like to try to channel these streams separately for a while. That =
will require some additional coordination work from the area directors =
and chairs, but
it will bring significant benefits, especially in being able to better =
prioritize work items within the streams.  I realize that for folks like =
you, who will actively participate=20
in all of the efforts, the benefits are more limited.

I have had several participants express frustration, for example, with =
the way the RAMs work eclipsed basic architectural work in AVT.=20
While that could be  managed in place  (we could have only one working =
group per area in the extreme), it has shown itself to be difficult=20
with the set of things this group is working on.

I also expect  the separation will make it easier to evaluate when the =
energy for a particular effort (such as the XR metrics you point to =
below) exists,=20
and when that level of energy changes.

Yes, conflicts between these working groups when they do meet in person =
will have to be managed carefully, but we have mechanisms for that. As =
Keith noted
separately, we do not anticipate this significantly increasing the =
number-of-minutes footprint on any given in-person IETF meeting. Also, =
your comments assume all=20
four of these groups would have a session at each in-person meeting. I =
don't expect that to be the case.


RjS


On Jan 12, 2011, at 6:11 PM, Stephan Wenger wrote:

> Hi IESG,
>=20
> As already communicated in Maastricht, I continue to believe that this =
step is a bad idea.  I didn't attend Beijing, and don't know whether =
there was any discussion about the subject.  I didn't see anything =
mentioned in the AVT minutes, though, nor did I see any discussion since =
Beijing on ietf@ietf.  If there was, some of the points below may be =
redundant, and I apologize in advance.
>=20
> Essentially, you guys are suggesting to split one "eternal" WG into =
four.  I'm writing "eternal", because I don't see that any of the WGs =
can fulfill its charter before other work items will likely get =
chartered with a similar scope, and in the same WG.  For RTP core, RTP =
extensions, and RTP payloads that is probably self-evident.  For the XR =
stuff: metrics pop up all the time, like payload formats.  Metrics are =
also known to make people quick bucks on patent licensing.  There will =
be more metric proposals.  If the RTCP-XR metrics ever get momentum in =
the marketplace, that is.  Somehow, I have my doubts.  But that's a =
different discussion. =20
>=20
> The positive effects of the split could be=20
> -chair workload is split among more people, and=20
> -there are more meeting seconds for RTP related work at any given IETF =
meeting. =20
>=20
> Assuming unlimited resources, both are good things.  Unfortunately, =
resources are not unlimited.
>=20
> The negative effects are IMO:
> -the key knowledgeable people in AVT ideally would have to sit in all =
four meetings so that all four groups can tap their experience.  But =
they may not want to do so, because it leaves less time for cross-area =
involvement, coordination, and draft-writing outside the AVT sessions, =
and whatnot.  Also, the burden for remote participants in far-away time =
zones goes up.  In other words, the resource meeting-seconds for those =
people is not unlimited, and they will have to prioritize differently.   =
 The result could be that what I called "key knowledgable people" would =
perhaps not attend the mere RTCP-XR or Payload session; and the guys =
there may be wasting their time doing work that does not make sense =
architecturally, just to see the hammer coming down at IETF last call.
> -more people get chair positions and are burdened with administrative =
stuff -> less time to do real work
> -WG scheduling gets more difficult
> -AD load potentially goes up (someone needs to coordinate between =
those groups, and the AD would seem to me the natural choice)
> -there is no reason to believe that the overall manpower availability =
goes up through this change, so all the inefficiencies introduced go =
straight against the available cycles.
>=20
> Some of these issues can be solved by joint WG meetings and/or careful =
chair selection.  However, both options have drawbacks as well.  A joint =
session would be counter-productive to the allocation of more meeting =
minutes.  And when you have, say, 8 chairs for four WG, all of which =
know the RTP architecture well enough, then you have pretty much =
exhausted, if not overdrawn already, the pool of "key knowledgable =
people", and burdened them with (in many cases: additional) chair =
duties.
>=20
> No matter how I see it, splitting the group is not a good idea.
>=20
> Other options to alleviate problems could come to mind:
>=20
> If the problem to be solved is chair-workload: add more chairs (but =
not 6 of them).
> If the problem to be solved is meeting-second shortages: have interim =
meetings, conf calls, etc.  In other words, stretch out the AVT meeting =
in the interval between IETF meetings.  Require a web-ex presentation of =
every draft to be discussed in the AVT session at least 4 weeks before =
the meeting, with a two hour timeslot for Q/A.  Should I continue with =
this brainstorm?
>=20
> If the problem is something else, I would like to know=85
>=20
> Thanks for your consideration, and apologies for the wordcount,
>=20
> Stephan
>=20
>=20
>=20


--Apple-Mail-98-966779175
Content-Transfer-Encoding: quoted-printable
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	charset=windows-1252

<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
"><div>Stephan -</div><div><br></div><div>As I noted in response to your =
comments in Maastricht, I disagree.</div><div><br></div><div>There are =
separable streams of work in AVT - it is really multiple working groups =
in one shell at the moment.</div><div><br></div><div>While there are key =
people currently in AVT that would overlap these groups, they are not =
the majority of the contributors to the different types of =
activities.</div><div>The intended positive effect of this explicit =
division is to allow these different streams to better parallelize, on =
list between meetings, not just during the IETF</div><div>in-person =
meeting time.&nbsp;</div><div><br></div><div>We'd like to try to channel =
these streams separately for a while. That will require some additional =
coordination work from the area directors and chairs, but</div><div>it =
will bring significant benefits,&nbsp;especially in being able to better =
prioritize work items within the streams. &nbsp;I realize that for folks =
like you, who will actively participate&nbsp;</div><div>in all of =
the&nbsp;efforts, the benefits are more =
limited.</div><div><br></div><div>I have had&nbsp;several participants =
express frustration, for example, with the way the&nbsp;RAMs work =
eclipsed basic architectural work in AVT.&nbsp;</div><div>While that =
could be &nbsp;managed&nbsp;in place &nbsp;(we could have only one =
working group per area in the extreme), it has&nbsp;shown itself to be =
difficult&nbsp;</div><div>with the set of things this group is working =
on.</div><div><br></div><div>I also expect &nbsp;the separation will =
make it easier to evaluate when the energy for a particular&nbsp;effort =
(such as the XR metrics you point to below) exists,&nbsp;</div><div>and =
when that level&nbsp;of energy =
changes.</div><div><div><br></div><div>Yes, conflicts between these =
working groups when they do meet in person will have to be managed =
carefully, but we have mechanisms for that. As Keith =
noted</div></div><div>separately, we do not anticipate this =
significantly increasing the number-of-minutes footprint on any given =
in-person IETF meeting. Also, your comments assume =
all&nbsp;</div><div>four of these groups would have a session at each =
in-person meeting. I don't expect that to be the =
case.</div><div><br></div><div><br></div><div>RjS</div><div><br></div><br>=
<div><div>On Jan 12, 2011, at 6:11 PM, Stephan Wenger wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><div =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; color: rgb(0, 0, 0); font-size: =
14px; font-family: Calibri, sans-serif; "><div>Hi =
IESG,</div><div><br></div><div>As already communicated in Maastricht, I =
continue to believe that this step is a bad idea. &nbsp;I didn't attend =
Beijing, and don't know whether there was any discussion about the =
subject. &nbsp;I didn't see anything mentioned in the AVT minutes, =
though, nor did I see any discussion since Beijing on ietf@ietf. =
&nbsp;If there was, some of the points below may be redundant, and I =
apologize in advance.</div><div><br></div><div>Essentially, you guys are =
suggesting to split one "eternal" WG into four. &nbsp;I'm writing =
"eternal", because I don't see that any of the WGs can fulfill its =
charter before other work items will likely get chartered with a similar =
scope, and in the same WG. &nbsp;For RTP core, RTP extensions, and RTP =
payloads that is probably self-evident. &nbsp;For the XR stuff: metrics =
pop up all the time, like payload formats. &nbsp;Metrics are also known =
to make people quick bucks on patent licensing. &nbsp;There will be more =
metric proposals. &nbsp;If the RTCP-XR metrics ever get momentum in the =
marketplace, that is. &nbsp;Somehow, I have my doubts. &nbsp;But that's =
a different discussion. &nbsp;</div><div><br></div><div>The positive =
effects of the split could be&nbsp;</div><div>-chair workload is split =
among more people, and&nbsp;</div><div>-there are more meeting seconds =
for RTP related work at any given IETF meeting. =
&nbsp;</div><div><br></div><div>Assuming unlimited resources, both are =
good things. &nbsp;Unfortunately, resources are not =
unlimited.</div><div><br></div><div>The negative effects are =
IMO:</div><div>-the key knowledgeable people in AVT ideally would have =
to sit in all four meetings so that all four groups can tap their =
experience. &nbsp;But they may not want to do so, because it leaves less =
time for cross-area involvement, coordination, and draft-writing outside =
the AVT sessions, and whatnot. &nbsp;Also, the burden for remote =
participants in far-away time zones goes up. &nbsp;In other words, the =
resource meeting-seconds for those people is not unlimited, and they =
will have to prioritize differently. &nbsp; &nbsp;The result could be =
that what I called "key knowledgable people" would perhaps not attend =
the mere RTCP-XR or Payload session; and the guys there may be wasting =
their time doing work that does not make sense architecturally, just to =
see the hammer coming down at IETF last call.</div><div>-more people get =
chair positions and are burdened with administrative stuff -&gt; less =
time to do real work</div><div>-WG scheduling gets more =
difficult</div><div>-AD load potentially goes up (someone needs to =
coordinate between those groups, and the AD would seem to me the natural =
choice)</div><div>-there is no reason to believe that the overall =
manpower availability goes up through this change, so all the =
inefficiencies introduced go straight against the available =
cycles.</div><div><br></div><div>Some of these issues can be solved by =
joint WG meetings and/or careful chair selection. &nbsp;However, both =
options have drawbacks as well. &nbsp;A joint session would be =
counter-productive to the allocation of more meeting minutes. &nbsp;And =
when you have, say, 8 chairs for four WG, all of which know the RTP =
architecture well enough, then you have pretty much exhausted, if not =
overdrawn already, the pool of "key knowledgable people", and burdened =
them with (in many cases: additional) chair =
duties.</div><div><br></div><div>No matter how I see it, splitting the =
group is not a good idea.</div><div><br></div><div>Other options to =
alleviate problems could come to mind:</div><div><br></div><div>If the =
problem to be solved is chair-workload: add more chairs (but not 6 of =
them).</div><div>If the problem to be solved is meeting-second =
shortages: have interim meetings, conf calls, etc. &nbsp;In other words, =
stretch out the AVT meeting in the interval between IETF meetings. =
&nbsp;Require a web-ex presentation of every draft to be discussed in =
the AVT session at least 4 weeks before the meeting, with a two hour =
timeslot for Q/A. &nbsp;Should I continue with this =
brainstorm?</div><div><br></div><div>If the problem is something else, I =
would like to know=85</div><div><br></div><div>Thanks for your =
consideration, and apologies for the =
wordcount,</div><div><br></div><div>Stephan</div><div><br></div><div><br><=
/div><div><br></div></div>
</blockquote></div><br></body></html>=

--Apple-Mail-98-966779175--

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To: Stephan Wenger <stewe@stewe.org>
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Subject: Re: [AVT] proposed AVT WG split
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Stephan, I see this differently than you. I think you are assuming that =
four WG will take on more work in net than just one. First I don't think =
this is true, and even if it is, then I would see that as a good thing =
as it would be proof that the 1 WG was blocking useful work that the =
community wanted to do. Let me address your cons inline.


On Jan 12, 2011, at 5:11 PM, Stephan Wenger wrote:

>=20
>=20
> The negative effects are IMO:
> -the key knowledgeable people in AVT ideally would have to sit in all =
four meetings so that all four groups can tap their experience.

I view that as the current situation. Right now, given the times on the =
agenda move around in a meeting, everyone has to be at all the AVT =
related talks regardless if they want to see just some parts of them or =
all of them. It seem like the split would improve this. It can't make it =
worse.=20

>  But they may not want to do so, because it leaves less time for =
cross-area involvement, coordination, and draft-writing outside the AVT =
sessions, and whatnot.  Also, the burden for remote participants in =
far-away time zones goes up.  In other words, the resource =
meeting-seconds for those people is not unlimited, and they will have to =
prioritize differently.    The result could be that what I called "key =
knowledgable people" would perhaps not attend the mere RTCP-XR or =
Payload session; and the guys there may be wasting their time doing work =
that does not make sense architecturally, just to see the hammer coming =
down at IETF last call.
> -more people get chair positions and are burdened with administrative =
stuff -> less time to do real work

Given the current chair bottlenecks, and the fact we have people that =
would like to do this, I view the bunch more chairs a pro not a con

> -WG scheduling gets more difficult

Having been very involved with scheduling RAI for several years, I =
strongly disagree. Right now you can't conflict AVT with anything, yet I =
bet you could conflict the XR metrics with say BLISS without too many =
problems.=20

> -AD load potentially goes up (someone needs to coordinate between =
those groups, and the AD would seem to me the natural choice)

The AD is going to have to be aware of all the work related to RTP =
regardless of if it is split in one group or four. I know when I was AD =
for AVT I felt this split would make things easier for me.=20

> -there is no reason to believe that the overall manpower availability =
goes up through this change, so all the inefficiencies introduced go =
straight against the available cycles.

I agree the manpower to do AVT technical work does not improve. However, =
having people who care less about 1 of the four stream be able to ignore =
theses meetings and go elsewhere would improve the manpower available =
available to other topics. The administrative work would go from two =
chairs and split over more. I don't think it significantly increases the =
total amount of admin-a-triva and having more people doing that is =
likely to reduce administrative delays. Now you might think the =
administrative delays have no impact on how many person hours are =
required to do the technical work but if an admin delay causes something =
to sit for 3 months, then all the technical people need to spend a bunch =
of time reloading the context in their brain and getting back up to =
speed on  where the technical work is. This decreases the efficiently of =
the the technical work.=20

>=20
> Some of these issues can be solved by joint WG meetings and/or careful =
chair selection.  However, both options have drawbacks as well.  A joint =
session would be counter-productive to the allocation of more meeting =
minutes.  And when you have, say, 8 chairs for four WG, all of which =
know the RTP architecture well enough, then you have pretty much =
exhausted, if not overdrawn already, the pool of "key knowledgable =
people", and burdened them with (in many cases: additional) chair =
duties.
>=20
> No matter how I see it, splitting the group is not a good idea.
>=20
> Other options to alleviate problems could come to mind:
>=20
> If the problem to be solved is chair-workload: add more chairs (but =
not 6 of them).

The more chairs you have for a single WG, the more all chairs think some =
other chair will do it and nothing happens. See track record of sipping =
chairs for a concrete example of this and those were all chairs that had =
proven track record of being effect chairs in other WGs. I agree with =
your point that if all the top RTP architecture people were suddenly =
spending all their time doing admin work this would be a disaster. I'm =
sure the ADs won't do that and that theses people would not accept. But =
I do think we need to ask, is our pool of really top RTP people that get =
the architecture growing or shrinking? What can we do to grow it? =
Dropping in some keen and smart person as chair is a not bad way of =
growing deep expertise in an area.=20

> If the problem to be solved is meeting-second shortages: have interim =
meetings, conf calls, etc.  In other words, stretch out the AVT meeting =
in the interval between IETF meetings.  Require a web-ex presentation of =
every draft to be discussed in the AVT session at least 4 weeks before =
the meeting, with a two hour timeslot for Q/A.  Should I continue with =
this brainstorm?
( ahhgg I see all kinds of problems with that but lets ignore theses, =
this is all based on the idea that we want to do more work which I think =
is an orthogonal issue to the split proposal)

>=20
> If the problem is something else, I would like to know=85
>=20
> Thanks for your consideration, and apologies for the wordcount,

When I read what you are saying, I think much of it can be tracked back =
to if we decide to try and do more work in the 4 groups than we were =
doing in one group, that may not be a good thing. I agree with that and =
think we need to manage that careful but that is equally true across the =
area as a whole as it is in any single WG. The IETF as a whole is =
familiar with managing that trade off.=20

>=20
> Stephan
>=20
>=20
>=20
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt


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Date: Thu, 13 Jan 2011 23:56:07 +0200
From: Roni Even <Even.roni@huawei.com>
In-reply-to: <E8393051-32DD-4DDE-8E92-E64ABCFF02E0@cisco.com>
To: 'Cullen Jennings' <fluffy@cisco.com>, 'Stephan Wenger' <stewe@stewe.org>
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Cc: iesg@ietf.org, avt@ietf.org
Subject: Re: [AVT] proposed AVT WG split
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Hi Cullen,
I would like to point out that the AVT work was split between the two Ads so
maybe we need a third AD to help with the work.
I would also to state that I did not complain about work load. Some of the
work is limited by the editors and reviewers. 
As for you other comments, see inline.
Roni Even

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
> Cullen Jennings
> Sent: Thursday, January 13, 2011 11:36 PM
> To: Stephan Wenger
> Cc: iesg@ietf.org; avt@ietf.org
> Subject: Re: [AVT] proposed AVT WG split
> 
> 
> Stephan, I see this differently than you. I think you are assuming that
> four WG will take on more work in net than just one. First I don't
> think this is true, and even if it is, then I would see that as a good
> thing as it would be proof that the 1 WG was blocking useful work that
> the community wanted to do. Let me address your cons inline.
> 
> 
> On Jan 12, 2011, at 5:11 PM, Stephan Wenger wrote:
> 
> >
> >
> > The negative effects are IMO:
> > -the key knowledgeable people in AVT ideally would have to sit in all
> four meetings so that all four groups can tap their experience.
> 
> I view that as the current situation. Right now, given the times on the
> agenda move around in a meeting, everyone has to be at all the AVT
> related talks regardless if they want to see just some parts of them or
> all of them. It seem like the split would improve this. It can't make
> it worse.
[Roni Even] I see this as a pro since we get the relevant people in the
meeting while if we divide I believe that some of them will not follow the
payload and RTCP XR work.
> 
> >  But they may not want to do so, because it leaves less time for
> cross-area involvement, coordination, and draft-writing outside the AVT
> sessions, and whatnot.  Also, the burden for remote participants in
> far-away time zones goes up.  In other words, the resource meeting-
> seconds for those people is not unlimited, and they will have to
> prioritize differently.    The result could be that what I called "key
> knowledgable people" would perhaps not attend the mere RTCP-XR or
> Payload session; and the guys there may be wasting their time doing
> work that does not make sense architecturally, just to see the hammer
> coming down at IETF last call.
> > -more people get chair positions and are burdened with administrative
> stuff -> less time to do real work
> 
> Given the current chair bottlenecks, and the fact we have people that
> would like to do this, I view the bunch more chairs a pro not a con
> 
> > -WG scheduling gets more difficult
> 
> Having been very involved with scheduling RAI for several years, I
> strongly disagree. Right now you can't conflict AVT with anything, yet
> I bet you could conflict the XR metrics with say BLISS without too many
> problems.
[Roni Even] The conflicts will continue wand will be bigger since more
chairs add conflicts (each chair has different preferences).
> 
> > -AD load potentially goes up (someone needs to coordinate between
> those groups, and the AD would seem to me the natural choice)
> 
> The AD is going to have to be aware of all the work related to RTP
> regardless of if it is split in one group or four. I know when I was AD
> for AVT I felt this split would make things easier for me.
> 
> > -there is no reason to believe that the overall manpower availability
> goes up through this change, so all the inefficiencies introduced go
> straight against the available cycles.
> 
> I agree the manpower to do AVT technical work does not improve.
> However, having people who care less about 1 of the four stream be able
> to ignore theses meetings and go elsewhere would improve the manpower
> available available to other topics. The administrative work would go
> from two chairs and split over more. I don't think it significantly
> increases the total amount of admin-a-triva and having more people
> doing that is likely to reduce administrative delays. Now you might
> think the administrative delays have no impact on how many person hours
> are required to do the technical work but if an admin delay causes
> something to sit for 3 months, then all the technical people need to
> spend a bunch of time reloading the context in their brain and getting
> back up to speed on  where the technical work is. This decreases the
> efficiently of the technical work.
[Roni Even] Some of the delays is not due to the chairs but to the editors
of the drafts. This is mostly true for payload. It also depends on how many
items the Ads allow. We always have the DISPATCH WG for non chartered work.
> 
> >
> > Some of these issues can be solved by joint WG meetings and/or
> careful chair selection.  However, both options have drawbacks as well.
> A joint session would be counter-productive to the allocation of more
> meeting minutes.  And when you have, say, 8 chairs for four WG, all of
> which know the RTP architecture well enough, then you have pretty much
> exhausted, if not overdrawn already, the pool of "key knowledgable
> people", and burdened them with (in many cases: additional) chair
> duties.
> >
> > No matter how I see it, splitting the group is not a good idea.
> >
> > Other options to alleviate problems could come to mind:
> >
> > If the problem to be solved is chair-workload: add more chairs (but
> not 6 of them).
> 
> The more chairs you have for a single WG, the more all chairs think
> some other chair will do it and nothing happens. See track record of
> sipping chairs for a concrete example of this and those were all chairs
> that had proven track record of being effect chairs in other WGs. I
> agree with your point that if all the top RTP architecture people were
> suddenly spending all their time doing admin work this would be a
> disaster. I'm sure the ADs won't do that and that theses people would
> not accept. But I do think we need to ask, is our pool of really top
> RTP people that get the architecture growing or shrinking? What can we
> do to grow it? Dropping in some keen and smart person as chair is a not
> bad way of growing deep expertise in an area.
> 
> > If the problem to be solved is meeting-second shortages: have interim
> meetings, conf calls, etc.  In other words, stretch out the AVT meeting
> in the interval between IETF meetings.  Require a web-ex presentation
> of every draft to be discussed in the AVT session at least 4 weeks
> before the meeting, with a two hour timeslot for Q/A.  Should I
> continue with this brainstorm?
> ( ahhgg I see all kinds of problems with that but lets ignore theses,
> this is all based on the idea that we want to do more work which I
> think is an orthogonal issue to the split proposal)
> 
> >
> > If the problem is something else, I would like to know.
> >
> > Thanks for your consideration, and apologies for the wordcount,
> 
> When I read what you are saying, I think much of it can be tracked back
> to if we decide to try and do more work in the 4 groups than we were
> doing in one group, that may not be a good thing. I agree with that and
> think we need to manage that careful but that is equally true across
> the area as a whole as it is in any single WG. The IETF as a whole is
> familiar with managing that trade off.
> 
> >
> > Stephan
> >
> >
> >
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www.ietf.org/mailman/listinfo/avt
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
> 
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Date: Fri, 14 Jan 2011 00:04:24 +0200
From: Roni Even <Even.roni@huawei.com>
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To: 'Robert Sparks' <rjsparks@nostrum.com>, 'Stephan Wenger' <stewe@stewe.org>
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Subject: Re: [AVT] proposed AVT WG split
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Roberts,

I do not think that the RAMS work should be the motivation to split AVT.

1.       RAMS in today's structure should have gone to DISPATCH and not to
AVT

2.       The time it took was due to discussions on the correctness of the
general solution and not because of AVT load. It always was the high
priority work in AVT.

Roni Even

 

From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Robert
Sparks
Sent: Thursday, January 13, 2011 11:28 PM
To: Stephan Wenger
Cc: IETF AVT WG; IESG IESG
Subject: Re: [AVT] proposed AVT WG split

 

Stephan -

 

As I noted in response to your comments in Maastricht, I disagree.

 

There are separable streams of work in AVT - it is really multiple working
groups in one shell at the moment.

 

While there are key people currently in AVT that would overlap these groups,
they are not the majority of the contributors to the different types of
activities.

The intended positive effect of this explicit division is to allow these
different streams to better parallelize, on list between meetings, not just
during the IETF

in-person meeting time. 

 

We'd like to try to channel these streams separately for a while. That will
require some additional coordination work from the area directors and
chairs, but

it will bring significant benefits, especially in being able to better
prioritize work items within the streams.  I realize that for folks like
you, who will actively participate 

in all of the efforts, the benefits are more limited.

 

I have had several participants express frustration, for example, with the
way the RAMs work eclipsed basic architectural work in AVT. 

While that could be  managed in place  (we could have only one working group
per area in the extreme), it has shown itself to be difficult 

with the set of things this group is working on.

 

I also expect  the separation will make it easier to evaluate when the
energy for a particular effort (such as the XR metrics you point to below)
exists, 

and when that level of energy changes.

 

Yes, conflicts between these working groups when they do meet in person will
have to be managed carefully, but we have mechanisms for that. As Keith
noted

separately, we do not anticipate this significantly increasing the
number-of-minutes footprint on any given in-person IETF meeting. Also, your
comments assume all 

four of these groups would have a session at each in-person meeting. I don't
expect that to be the case.

 

 

RjS

 

 

On Jan 12, 2011, at 6:11 PM, Stephan Wenger wrote:





Hi IESG,

 

As already communicated in Maastricht, I continue to believe that this step
is a bad idea.  I didn't attend Beijing, and don't know whether there was
any discussion about the subject.  I didn't see anything mentioned in the
AVT minutes, though, nor did I see any discussion since Beijing on
ietf@ietf.  If there was, some of the points below may be redundant, and I
apologize in advance.

 

Essentially, you guys are suggesting to split one "eternal" WG into four.
I'm writing "eternal", because I don't see that any of the WGs can fulfill
its charter before other work items will likely get chartered with a similar
scope, and in the same WG.  For RTP core, RTP extensions, and RTP payloads
that is probably self-evident.  For the XR stuff: metrics pop up all the
time, like payload formats.  Metrics are also known to make people quick
bucks on patent licensing.  There will be more metric proposals.  If the
RTCP-XR metrics ever get momentum in the marketplace, that is.  Somehow, I
have my doubts.  But that's a different discussion.  

 

The positive effects of the split could be 

-chair workload is split among more people, and 

-there are more meeting seconds for RTP related work at any given IETF
meeting.  

 

Assuming unlimited resources, both are good things.  Unfortunately,
resources are not unlimited.

 

The negative effects are IMO:

-the key knowledgeable people in AVT ideally would have to sit in all four
meetings so that all four groups can tap their experience.  But they may not
want to do so, because it leaves less time for cross-area involvement,
coordination, and draft-writing outside the AVT sessions, and whatnot.
Also, the burden for remote participants in far-away time zones goes up.  In
other words, the resource meeting-seconds for those people is not unlimited,
and they will have to prioritize differently.    The result could be that
what I called "key knowledgable people" would perhaps not attend the mere
RTCP-XR or Payload session; and the guys there may be wasting their time
doing work that does not make sense architecturally, just to see the hammer
coming down at IETF last call.

-more people get chair positions and are burdened with administrative stuff
-> less time to do real work

-WG scheduling gets more difficult

-AD load potentially goes up (someone needs to coordinate between those
groups, and the AD would seem to me the natural choice)

-there is no reason to believe that the overall manpower availability goes
up through this change, so all the inefficiencies introduced go straight
against the available cycles.

 

Some of these issues can be solved by joint WG meetings and/or careful chair
selection.  However, both options have drawbacks as well.  A joint session
would be counter-productive to the allocation of more meeting minutes.  And
when you have, say, 8 chairs for four WG, all of which know the RTP
architecture well enough, then you have pretty much exhausted, if not
overdrawn already, the pool of "key knowledgable people", and burdened them
with (in many cases: additional) chair duties.

 

No matter how I see it, splitting the group is not a good idea.

 

Other options to alleviate problems could come to mind:

 

If the problem to be solved is chair-workload: add more chairs (but not 6 of
them).

If the problem to be solved is meeting-second shortages: have interim
meetings, conf calls, etc.  In other words, stretch out the AVT meeting in
the interval between IETF meetings.  Require a web-ex presentation of every
draft to be discussed in the AVT session at least 4 weeks before the
meeting, with a two hour timeslot for Q/A.  Should I continue with this
brainstorm?

 

If the problem is something else, I would like to know.

 

Thanks for your consideration, and apologies for the wordcount,

 

Stephan

 

 

 




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 ISPATCH 
</span></p><p class=MsoListParagraph style='text-indent:-18.0pt;mso-list:l0 level1 lfo1'><![if !supportLists]><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><span style='mso-list:Ignore'>2.<span style='font:7.0pt "Times New Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span></span><![endif]><span dir=LTR></span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>The time it took was due to discussions on the correctness of the general solution and not because of AVT load. It always was the high priority work in AVT.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Roni Even<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p>&nbsp;</o:p></span></p><div style='border:none;border-right:solid blue 1.5pt;padding:0cm 0cm 0cm 4.0pt'><div><div style='border:none;border-top:
 solid #B
 0cm 0cm 0cm'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] <b>On Behalf Of </b>Robert Sparks<br><b>Sent:</b> Thursday, January 13, 2011 11:28 PM<br><b>To:</b> Stephan Wenger<br><b>Cc:</b> IETF AVT WG; IESG IESG<br><b>Subject:</b> Re: [AVT] proposed AVT WG split<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p>&nbsp;</o:p></p><div><p class=MsoNormal>Stephan -<o:p></o:p></p></div><div><p class=MsoNormal><o:p>&nbsp;</o:p></p></div><div><p class=MsoNormal>As I noted in response to your comments in Maastricht, I disagree.<o:p></o:p></p></div><div><p class=MsoNormal><o:p>&nbsp;</o:p></p></div><div><p class=MsoNormal>There are separable streams of work in AVT - it is really multiple working groups in one shell at the moment.<o:p></o:p></p></div><div><p class=MsoNormal><o:p>&nbsp;</o:p></p></div><div><p 
 class=Ms
ey people currently in AVT that would overlap these groups, they are not the majority of the contributors to the different types of activities.<o:p></o:p></p></div><div><p class=MsoNormal>The intended positive effect of this explicit division is to allow these different streams to better parallelize, on list between meetings, not just during the IETF<o:p></o:p></p></div><div><p class=MsoNormal>in-person meeting time.&nbsp;<o:p></o:p></p></div><div><p class=MsoNormal><o:p>&nbsp;</o:p></p></div><div><p class=MsoNormal>We'd like to try to channel these streams separately for a while. That will require some additional coordination work from the area directors and chairs, but<o:p></o:p></p></div><div><p class=MsoNormal>it will bring significant benefits,&nbsp;especially in being able to better prioritize work items within the streams. &nbsp;I realize that for folks like you, who will actively participate&nbsp;<o:p></o:p></p></div><div><p class=MsoNormal>in all of the&nbsp;efforts,
  the ben
:p></o:p></p></div><div><p class=MsoNormal><o:p>&nbsp;</o:p></p></div><div><p class=MsoNormal>I have had&nbsp;several participants express frustration, for example, with the way the&nbsp;RAMs work eclipsed basic architectural work in AVT.&nbsp;<o:p></o:p></p></div><div><p class=MsoNormal>While that could be &nbsp;managed&nbsp;in place &nbsp;(we could have only one working group per area in the extreme), it has&nbsp;shown itself to be difficult&nbsp;<o:p></o:p></p></div><div><p class=MsoNormal>with the set of things this group is working on.<o:p></o:p></p></div><div><p class=MsoNormal><o:p>&nbsp;</o:p></p></div><div><p class=MsoNormal>I also expect &nbsp;the separation will make it easier to evaluate when the energy for a particular&nbsp;effort (such as the XR metrics you point to below) exists,&nbsp;<o:p></o:p></p></div><div><p class=MsoNormal>and when that level&nbsp;of energy changes.<o:p></o:p></p></div><div><div><p class=MsoNormal><o:p>&nbsp;</o:p></p></div><div><p class=
 MsoNorma
these working groups when they do meet in person will have to be managed carefully, but we have mechanisms for that. As Keith noted<o:p></o:p></p></div></div><div><p class=MsoNormal>separately, we do not anticipate this significantly increasing the number-of-minutes footprint on any given in-person IETF meeting. Also, your comments assume all&nbsp;<o:p></o:p></p></div><div><p class=MsoNormal>four of these groups would have a session at each in-person meeting. I don't expect that to be the case.<o:p></o:p></p></div><div><p class=MsoNormal><o:p>&nbsp;</o:p></p></div><div><p class=MsoNormal><o:p>&nbsp;</o:p></p></div><div><p class=MsoNormal>RjS<o:p></o:p></p></div><div><p class=MsoNormal><o:p>&nbsp;</o:p></p></div><p class=MsoNormal><o:p>&nbsp;</o:p></p><div><div><p class=MsoNormal>On Jan 12, 2011, at 6:11 PM, Stephan Wenger wrote:<o:p></o:p></p></div><p class=MsoNormal><br><br><o:p></o:p></p><div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-
 serif";c
</o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'><o:p>&nbsp;</o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>As already communicated in Maastricht, I continue to believe that this step is a bad idea. &nbsp;I didn't attend Beijing, and don't know whether there was any discussion about the subject. &nbsp;I didn't see anything mentioned in the AVT minutes, though, nor did I see any discussion since Beijing on ietf@ietf. &nbsp;If there was, some of the points below may be redundant, and I apologize in advance.<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'><o:p>&nbsp;</o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>Essentially, you guys are suggesting to split one &quot;etern
 al&quot;
writing &quot;eternal&quot;, because I don't see that any of the WGs can fulfill its charter before other work items will likely get chartered with a similar scope, and in the same WG. &nbsp;For RTP core, RTP extensions, and RTP payloads that is probably self-evident. &nbsp;For the XR stuff: metrics pop up all the time, like payload formats. &nbsp;Metrics are also known to make people quick bucks on patent licensing. &nbsp;There will be more metric proposals. &nbsp;If the RTCP-XR metrics ever get momentum in the marketplace, that is. &nbsp;Somehow, I have my doubts. &nbsp;But that's a different discussion. &nbsp;<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'><o:p>&nbsp;</o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>The positive effects of the split could be&nbsp;<o:p></o:p></span></p></div><div><p class=MsoNormal><spa
 n style=
mily:"Calibri","sans-serif";color:black'>-chair workload is split among more people, and&nbsp;<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>-there are more meeting seconds for RTP related work at any given IETF meeting. &nbsp;<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'><o:p>&nbsp;</o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>Assuming unlimited resources, both are good things. &nbsp;Unfortunately, resources are not unlimited.<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'><o:p>&nbsp;</o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>The negative effects are IMO:<o:p></o:p></span>
 </p></di
><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>-the key knowledgeable people in AVT ideally would have to sit in all four meetings so that all four groups can tap their experience. &nbsp;But they may not want to do so, because it leaves less time for cross-area involvement, coordination, and draft-writing outside the AVT sessions, and whatnot. &nbsp;Also, the burden for remote participants in far-away time zones goes up. &nbsp;In other words, the resource meeting-seconds for those people is not unlimited, and they will have to prioritize differently. &nbsp; &nbsp;The result could be that what I called &quot;key knowledgable people&quot; would perhaps not attend the mere RTCP-XR or Payload session; and the guys there may be wasting their time doing work that does not make sense architecturally, just to see the hammer coming down at IETF last call.<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calib
 ri","san
re people get chair positions and are burdened with administrative stuff -&gt; less time to do real work<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>-WG scheduling gets more difficult<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>-AD load potentially goes up (someone needs to coordinate between those groups, and the AD would seem to me the natural choice)<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>-there is no reason to believe that the overall manpower availability goes up through this change, so all the inefficiencies introduced go straight against the available cycles.<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'><o:p>&nbsp;</o:p></span></p>
 </div><d
an style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>Some of these issues can be solved by joint WG meetings and/or careful chair selection. &nbsp;However, both options have drawbacks as well. &nbsp;A joint session would be counter-productive to the allocation of more meeting minutes. &nbsp;And when you have, say, 8 chairs for four WG, all of which know the RTP architecture well enough, then you have pretty much exhausted, if not overdrawn already, the pool of &quot;key knowledgable people&quot;, and burdened them with (in many cases: additional) chair duties.<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'><o:p>&nbsp;</o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>No matter how I see it, splitting the group is not a good idea.<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-siz
 e:10.5pt
ans-serif";color:black'><o:p>&nbsp;</o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>Other options to alleviate problems could come to mind:<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'><o:p>&nbsp;</o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>If the problem to be solved is chair-workload: add more chairs (but not 6 of them).<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>If the problem to be solved is meeting-second shortages: have interim meetings, conf calls, etc. &nbsp;In other words, stretch out the AVT meeting in the interval between IETF meetings. &nbsp;Require a web-ex presentation of every draft to be discussed in the AVT session at least 4 weeks before 
 the meet
slot for Q/A. &nbsp;Should I continue with this brainstorm?<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'><o:p>&nbsp;</o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>If the problem is something else, I would like to know&#8230;<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'><o:p>&nbsp;</o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>Thanks for your consideration, and apologies for the wordcount,<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'><o:p>&nbsp;</o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>Stephan<o
 :p></o:p
 class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'><o:p>&nbsp;</o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'><o:p>&nbsp;</o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'><o:p>&nbsp;</o:p></span></p></div></div></div><p class=MsoNormal><br><br><br>__________ Information from ESET NOD32 Antivirus, version of virus signature database 5785 (20110113) __________<br><br>The message was checked by ESET NOD32 Antivirus.<br><br><a href="http://www.eset.com">http://www.eset.com</a><br><br><br>__________ Information from ESET NOD32 Antivirus, version of virus signature database 5785 (20110113) __________<br><br>The message was checked by ESET NOD32 Antivirus.<br><br><a href="http://www.eset.com">http://www.eset.com</a><o:p></o:p></p></div></div> <BR><BR>__________ Information from ESET NOD32 A
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From stewe@stewe.org  Thu Jan 13 17:16:47 2011
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Date: Thu, 13 Jan 2011 17:18:59 -0800
From: Stephan Wenger <stewe@stewe.org>
To: Cullen Jennings <fluffy@cisco.com>
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Subject: Re: [AVT] proposed AVT WG split
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On 1.13.2011 13:35 , "Cullen Jennings" <fluffy@cisco.com> wrote:

>
>Stephan, I see this differently than you. I think you are assuming that
>four WG will take on more work in net than just one. First I don't think
>this is true, and even if it is, then I would see that as a good thing as
>it would be proof that the 1 WG was blocking useful work that the
>community wanted to do. Let me address your cons inline.

Actually, that's not my assumption.  Quite the opposite.  My main argument
is that handling the same number of docs in a larger number of groups
naturally leads to less efficiency.  Also a few additional replies inline.

>
>
>On Jan 12, 2011, at 5:11 PM, Stephan Wenger wrote:
>
>>=20
>>=20
>> The negative effects are IMO:
>> -the key knowledgeable people in AVT ideally would have to sit in all
>>four meetings so that all four groups can tap their experience.
>
>I view that as the current situation. Right now, given the times on the
>agenda move around in a meeting, everyone has to be at all the AVT
>related talks regardless if they want to see just some parts of them or
>all of them. It seem like the split would improve this. It can't make it
>worse.

Your analysis would be correct if the number of meeting seconds for the
(currently two) AVT sessions is similar to the proposed four successor
WG's sessions.  Somehow I doubt that would be the case in the long term,
even if you guys would succeed to enforce it in the short term...

>=20
>
>>  But they may not want to do so, because it leaves less time for
>>cross-area involvement, coordination, and draft-writing outside the AVT
>>sessions, and whatnot.  Also, the burden for remote participants in
>>far-away time zones goes up.  In other words, the resource
>>meeting-seconds for those people is not unlimited, and they will have to
>>prioritize differently.    The result could be that what I called "key
>>knowledgable people" would perhaps not attend the mere RTCP-XR or
>>Payload session; and the guys there may be wasting their time doing work
>>that does not make sense architecturally, just to see the hammer coming
>>down at IETF last call.
>> -more people get chair positions and are burdened with administrative
>>stuff -> less time to do real work
>
>Given the current chair bottlenecks, and the fact we have people that
>would like to do this, I view the bunch more chairs a pro not a con

Really?  At almost every plenary I hear folks whining about how difficult
it is to get *qualified* chairs.  (Some would say that AVT, in its present
incarnation, clearly shows that difficulty)

>
>> -WG scheduling gets more difficult
>
>Having been very involved with scheduling RAI for several years, I
>strongly disagree. Right now you can't conflict AVT with anything, yet I
>bet you could conflict the XR metrics with say BLISS without too many
>problems.

Perhaps.  You are way more experienced than I am on these matters.

>=20
>
>> -AD load potentially goes up (someone needs to coordinate between those
>>groups, and the AD would seem to me the natural choice)
>
>The AD is going to have to be aware of all the work related to RTP
>regardless of if it is split in one group or four. I know when I was AD
>for AVT I felt this split would make things easier for me.

Interesting.  I didn't know you like multiparty coordination so much :-)
Your insight is not intuitive, though, at least not to me.

>=20
>
>> -there is no reason to believe that the overall manpower availability
>>goes up through this change, so all the inefficiencies introduced go
>>straight against the available cycles.
>
>I agree the manpower to do AVT technical work does not improve. However,
>having people who care less about 1 of the four stream be able to ignore
>theses meetings and go elsewhere would improve the manpower available
>available to other topics. The administrative work would go from two
>chairs and split over more. I don't think it significantly increases the
>total amount of admin-a-triva and having more people doing that is likely
>to reduce administrative delays. Now you might think the administrative
>delays have no impact on how many person hours are required to do the
>technical work but if an admin delay causes something to sit for 3
>months, then all the technical people need to spend a bunch of time
>reloading the context in their brain and getting back up to speed on
>where the technical work is. This decreases the efficiently of the the
>technical work.

Roni has answered this point already, and I concur: things are not slow
because of admin delays.  Things are slow because parts of AVT's work is
highly contentious.  That's natural, to me at least.

>=20
>
>>=20
>> Some of these issues can be solved by joint WG meetings and/or careful
>>chair selection.  However, both options have drawbacks as well.  A joint
>>session would be counter-productive to the allocation of more meeting
>>minutes.  And when you have, say, 8 chairs for four WG, all of which
>>know the RTP architecture well enough, then you have pretty much
>>exhausted, if not overdrawn already, the pool of "key knowledgable
>>people", and burdened them with (in many cases: additional) chair duties.
>>=20
>> No matter how I see it, splitting the group is not a good idea.
>>=20
>> Other options to alleviate problems could come to mind:
>>=20
>> If the problem to be solved is chair-workload: add more chairs (but not
>>6 of them).
>
>The more chairs you have for a single WG, the more all chairs think some
>other chair will do it and nothing happens. See track record of sipping
>chairs for a concrete example of this and those were all chairs that had
>proven track record of being effect chairs in other WGs. I agree with
>your point that if all the top RTP architecture people were suddenly
>spending all their time doing admin work this would be a disaster. I'm
>sure the ADs won't do that and that theses people would not accept. But I
>do think we need to ask, is our pool of really top RTP people that get
>the architecture growing or shrinking? What can we do to grow it?
>Dropping in some keen and smart person as chair is a not bad way of
>growing deep expertise in an area.
>
>> If the problem to be solved is meeting-second shortages: have interim
>>meetings, conf calls, etc.  In other words, stretch out the AVT meeting
>>in the interval between IETF meetings.  Require a web-ex presentation of
>>every draft to be discussed in the AVT session at least 4 weeks before
>>the meeting, with a two hour timeslot for Q/A.  Should I continue with
>>this brainstorm?
>( ahhgg I see all kinds of problems with that but lets ignore theses,
>this is all based on the idea that we want to do more work which I think
>is an orthogonal issue to the split proposal)
>
>>=20
>> If the problem is something else, I would like to know=8A
>>=20
>> Thanks for your consideration, and apologies for the wordcount,
>
>When I read what you are saying, I think much of it can be tracked back
>to if we decide to try and do more work in the 4 groups than we were
>doing in one group, that may not be a good thing. I agree with that and
>think we need to manage that careful but that is equally true across the
>area as a whole as it is in any single WG. The IETF as a whole is
>familiar with managing that trade off.
>
>>=20
>> Stephan
>>=20
>>=20
>>=20
>> _______________________________________________
>> Audio/Video Transport Working Group
>> avt@ietf.org
>> https://www.ietf.org/mailman/listinfo/avt
>
>_______________________________________________
>Audio/Video Transport Working Group
>avt@ietf.org
>https://www.ietf.org/mailman/listinfo/avt



From stewe@stewe.org  Thu Jan 13 17:25:05 2011
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Date: Thu, 13 Jan 2011 17:27:08 -0800
From: Stephan Wenger <stewe@stewe.org>
To: Robert Sparks <rjsparks@nostrum.com>
Message-ID: <C954E495.263C7%stewe@stewe.org>
Thread-Topic: [AVT] proposed AVT WG split
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Cc: IETF AVT WG <avt@ietf.org>, IESG IESG <iesg@ietf.org>
Subject: Re: [AVT] proposed AVT WG split
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Hi Robert,

We have exchanged these arguments before.  I guess it's most efficient for
both of us to agree to disagree, and continue with our day job=8A (which in
your case may require making a decision I don't like, but I still hope for
what I consider common sense :-).

Just one word on the RAMS example: if you have two major contributing
companies at each other's throat, combined with cultural differences, and a
couple of difficult individuals to spice up the broth, it does not matter a
bit how many groups you have: things are going to be slow in this group.
More importantly for this discussion, slowdown will also occur in related
groups.  Overspill happens.  Horse-trading happens.  Retaliation by blockin=
g
happens.  It's IMO na=EFve to assume a good set of chair can prevent any of
these deplorable events.

Stephan


From:  "rjsparks@nostrum.com" <rjsparks@nostrum.com>
Date:  Thu, 13 Jan 2011 15:27:55 -0600
To:  Stephan Wenger <stewe@stewe.org>
Cc:  IETF AVT WG <avt@ietf.org>, IESG IESG <iesg@ietf.org>
Subject:  Re: [AVT] proposed AVT WG split

Stephan -

As I noted in response to your comments in Maastricht, I disagree.

There are separable streams of work in AVT - it is really multiple working
groups in one shell at the moment.

While there are key people currently in AVT that would overlap these groups=
,
they are not the majority of the contributors to the different types of
activities.
The intended positive effect of this explicit division is to allow these
different streams to better parallelize, on list between meetings, not just
during the IETF
in-person meeting time.

We'd like to try to channel these streams separately for a while. That will
require some additional coordination work from the area directors and
chairs, but
it will bring significant benefits, especially in being able to better
prioritize work items within the streams.  I realize that for folks like
you, who will actively participate
in all of the efforts, the benefits are more limited.

I have had several participants express frustration, for example, with the
way the RAMs work eclipsed basic architectural work in AVT.
While that could be  managed in place  (we could have only one working grou=
p
per area in the extreme), it has shown itself to be difficult
with the set of things this group is working on.

I also expect  the separation will make it easier to evaluate when the
energy for a particular effort (such as the XR metrics you point to below)
exists,=20
and when that level of energy changes.

Yes, conflicts between these working groups when they do meet in person wil=
l
have to be managed carefully, but we have mechanisms for that. As Keith
noted
separately, we do not anticipate this significantly increasing the
number-of-minutes footprint on any given in-person IETF meeting. Also, your
comments assume all
four of these groups would have a session at each in-person meeting. I don'=
t
expect that to be the case.


RjS


On Jan 12, 2011, at 6:11 PM, Stephan Wenger wrote:

> Hi IESG,
>=20
> As already communicated in Maastricht, I continue to believe that this st=
ep is
> a bad idea.  I didn't attend Beijing, and don't know whether there was an=
y
> discussion about the subject.  I didn't see anything mentioned in the AVT
> minutes, though, nor did I see any discussion since Beijing on ietf@ietf.=
  If
> there was, some of the points below may be redundant, and I apologize in
> advance.
>=20
> Essentially, you guys are suggesting to split one "eternal" WG into four.=
  I'm
> writing "eternal", because I don't see that any of the WGs can fulfill it=
s
> charter before other work items will likely get chartered with a similar
> scope, and in the same WG.  For RTP core, RTP extensions, and RTP payload=
s
> that is probably self-evident.  For the XR stuff: metrics pop up all the =
time,
> like payload formats.  Metrics are also known to make people quick bucks =
on
> patent licensing.  There will be more metric proposals.  If the RTCP-XR
> metrics ever get momentum in the marketplace, that is.  Somehow, I have m=
y
> doubts.  But that's a different discussion.
>=20
> The positive effects of the split could be
> -chair workload is split among more people, and
> -there are more meeting seconds for RTP related work at any given IETF
> meeting. =20
>=20
> Assuming unlimited resources, both are good things.  Unfortunately, resou=
rces
> are not unlimited.
>=20
> The negative effects are IMO:
> -the key knowledgeable people in AVT ideally would have to sit in all fou=
r
> meetings so that all four groups can tap their experience.  But they may =
not
> want to do so, because it leaves less time for cross-area involvement,
> coordination, and draft-writing outside the AVT sessions, and whatnot.  A=
lso,
> the burden for remote participants in far-away time zones goes up.  In ot=
her
> words, the resource meeting-seconds for those people is not unlimited, an=
d
> they will have to prioritize differently.    The result could be that wha=
t I
> called "key knowledgable people" would perhaps not attend the mere RTCP-X=
R or
> Payload session; and the guys there may be wasting their time doing work =
that
> does not make sense architecturally, just to see the hammer coming down a=
t
> IETF last call.
> -more people get chair positions and are burdened with administrative stu=
ff ->
> less time to do real work
> -WG scheduling gets more difficult
> -AD load potentially goes up (someone needs to coordinate between those
> groups, and the AD would seem to me the natural choice)
> -there is no reason to believe that the overall manpower availability goe=
s up
> through this change, so all the inefficiencies introduced go straight aga=
inst
> the available cycles.
>=20
> Some of these issues can be solved by joint WG meetings and/or careful ch=
air
> selection.  However, both options have drawbacks as well.  A joint sessio=
n
> would be counter-productive to the allocation of more meeting minutes.  A=
nd
> when you have, say, 8 chairs for four WG, all of which know the RTP
> architecture well enough, then you have pretty much exhausted, if not
> overdrawn already, the pool of "key knowledgable people", and burdened th=
em
> with (in many cases: additional) chair duties.
>=20
> No matter how I see it, splitting the group is not a good idea.
>=20
> Other options to alleviate problems could come to mind:
>=20
> If the problem to be solved is chair-workload: add more chairs (but not 6=
 of
> them).
> If the problem to be solved is meeting-second shortages: have interim
> meetings, conf calls, etc.  In other words, stretch out the AVT meeting i=
n the
> interval between IETF meetings.  Require a web-ex presentation of every d=
raft
> to be discussed in the AVT session at least 4 weeks before the meeting, w=
ith a
> two hour timeslot for Q/A.  Should I continue with this brainstorm?
>=20
> If the problem is something else, I would like to know=8A
>=20
> Thanks for your consideration, and apologies for the wordcount,
>=20
> Stephan
>=20
>=20
>=20

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


--B_3377784433_28584930
Content-type: text/html;
	charset="ISO-8859-1"
Content-transfer-encoding: quoted-printable

<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>Hi Robert,</div><div><br></d=
iv><div>We have exchanged these arguments before. &nbsp;I guess it's most ef=
ficient for both of us to agree to disagree, and continue with our day job&#=
8230; (which in your case may require making a decision I don't like, but I =
still hope for what I consider common sense :-).</div><div><br></div><div>Ju=
st one word on the RAMS example: if you have two major contributing companie=
s at each other's throat, combined with cultural differences, and a couple o=
f difficult individuals to spice up the broth, it does not matter a bit how =
many groups you have: things are going to be slow in this group. &nbsp;More =
importantly for this discussion, slowdown will also occur in related groups.=
 &nbsp;Overspill happens. &nbsp;Horse-trading happens. &nbsp;Retaliation by =
blocking happens. &nbsp;It's IMO na=EFve to assume a good set of chair can pre=
vent any of these deplorable events.</div><div><br></div><div>Stephan</div><=
div><br></div><div><br></div><span id=3D"OLK_SRC_BODY_SECTION"><div style=3D"fon=
t-family:Calibri; font-size:11pt; text-align:left; color:black; BORDER-BOTTO=
M: medium none; BORDER-LEFT: medium none; PADDING-BOTTOM: 0in; PADDING-LEFT:=
 0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt solid; BORDER-RIGHT: mediu=
m none; PADDING-TOP: 3pt"><span style=3D"font-weight:bold">From: </span> "<a h=
ref=3D"mailto:rjsparks@nostrum.com">rjsparks@nostrum.com</a>" &lt;<a href=3D"mai=
lto:rjsparks@nostrum.com">rjsparks@nostrum.com</a>&gt;<br><span style=3D"font-=
weight:bold">Date: </span> Thu, 13 Jan 2011 15:27:55 -0600<br><span style=3D"f=
ont-weight:bold">To: </span> Stephan Wenger &lt;<a href=3D"mailto:stewe@stewe.=
org">stewe@stewe.org</a>&gt;<br><span style=3D"font-weight:bold">Cc: </span> I=
ETF AVT WG &lt;<a href=3D"mailto:avt@ietf.org">avt@ietf.org</a>&gt;, IESG IESG=
 &lt;<a href=3D"mailto:iesg@ietf.org">iesg@ietf.org</a>&gt;<br><span style=3D"fo=
nt-weight:bold">Subject: </span> Re: [AVT] proposed AVT WG split<br></div><d=
iv><br></div><div><div style=3D"word-wrap: break-word; -webkit-nbsp-mode: spac=
e; -webkit-line-break: after-white-space; "><div>Stephan -</div><div><br></d=
iv><div>As I noted in response to your comments in Maastricht, I disagree.</=
div><div><br></div><div>There are separable streams of work in AVT - it is r=
eally multiple working groups in one shell at the moment.</div><div><br></di=
v><div>While there are key people currently in AVT that would overlap these =
groups, they are not the majority of the contributors to the different types=
 of activities.</div><div>The intended positive effect of this explicit divi=
sion is to allow these different streams to better parallelize, on list betw=
een meetings, not just during the IETF</div><div>in-person meeting time.&nbs=
p;</div><div><br></div><div>We'd like to try to channel these streams separa=
tely for a while. That will require some additional coordination work from t=
he area directors and chairs, but</div><div>it will bring significant benefi=
ts,&nbsp;especially in being able to better prioritize work items within the=
 streams. &nbsp;I realize that for folks like you, who will actively partici=
pate&nbsp;</div><div>in all of the&nbsp;efforts, the benefits are more limit=
ed.</div><div><br></div><div>I have had&nbsp;several participants express fr=
ustration, for example, with the way the&nbsp;RAMs work eclipsed basic archi=
tectural work in AVT.&nbsp;</div><div>While that could be &nbsp;managed&nbsp=
;in place &nbsp;(we could have only one working group per area in the extrem=
e), it has&nbsp;shown itself to be difficult&nbsp;</div><div>with the set of=
 things this group is working on.</div><div><br></div><div>I also expect &nb=
sp;the separation will make it easier to evaluate when the energy for a part=
icular&nbsp;effort (such as the XR metrics you point to below) exists,&nbsp;=
</div><div>and when that level&nbsp;of energy changes.</div><div><div><br></=
div><div>Yes, conflicts between these working groups when they do meet in pe=
rson will have to be managed carefully, but we have mechanisms for that. As =
Keith noted</div></div><div>separately, we do not anticipate this significan=
tly increasing the number-of-minutes footprint on any given in-person IETF m=
eeting. Also, your comments assume all&nbsp;</div><div>four of these groups =
would have a session at each in-person meeting. I don't expect that to be th=
e case.</div><div><br></div><div><br></div><div>RjS</div><div><br></div><br>=
<div><div>On Jan 12, 2011, at 6:11 PM, Stephan Wenger wrote:</div><br class=3D=
"Apple-interchange-newline"><blockquote type=3D"cite"><div style=3D"word-wrap: b=
reak-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
color: rgb(0, 0, 0); font-size: 14px; font-family: Calibri, sans-serif; "><d=
iv>Hi IESG,</div><div><br></div><div>As already communicated in Maastricht, =
I continue to believe that this step is a bad idea. &nbsp;I didn't attend Be=
ijing, and don't know whether there was any discussion about the subject. &n=
bsp;I didn't see anything mentioned in the AVT minutes, though, nor did I se=
e any discussion since Beijing on ietf@ietf. &nbsp;If there was, some of the=
 points below may be redundant, and I apologize in advance.</div><div><br></=
div><div>Essentially, you guys are suggesting to split one "eternal" WG into=
 four. &nbsp;I'm writing "eternal", because I don't see that any of the WGs =
can fulfill its charter before other work items will likely get chartered wi=
th a similar scope, and in the same WG. &nbsp;For RTP core, RTP extensions, =
and RTP payloads that is probably self-evident. &nbsp;For the XR stuff: metr=
ics pop up all the time, like payload formats. &nbsp;Metrics are also known =
to make people quick bucks on patent licensing. &nbsp;There will be more met=
ric proposals. &nbsp;If the RTCP-XR metrics ever get momentum in the marketp=
lace, that is. &nbsp;Somehow, I have my doubts. &nbsp;But that's a different=
 discussion. &nbsp;</div><div><br></div><div>The positive effects of the spl=
it could be&nbsp;</div><div>-chair workload is split among more people, and&=
nbsp;</div><div>-there are more meeting seconds for RTP related work at any =
given IETF meeting. &nbsp;</div><div><br></div><div>Assuming unlimited resou=
rces, both are good things. &nbsp;Unfortunately, resources are not unlimited=
.</div><div><br></div><div>The negative effects are IMO:</div><div>-the key =
knowledgeable people in AVT ideally would have to sit in all four meetings s=
o that all four groups can tap their experience. &nbsp;But they may not want=
 to do so, because it leaves less time for cross-area involvement, coordinat=
ion, and draft-writing outside the AVT sessions, and whatnot. &nbsp;Also, th=
e burden for remote participants in far-away time zones goes up. &nbsp;In ot=
her words, the resource meeting-seconds for those people is not unlimited, a=
nd they will have to prioritize differently. &nbsp; &nbsp;The result could b=
e that what I called "key knowledgable people" would perhaps not attend the =
mere RTCP-XR or Payload session; and the guys there may be wasting their tim=
e doing work that does not make sense architecturally, just to see the hamme=
r coming down at IETF last call.</div><div>-more people get chair positions =
and are burdened with administrative stuff -&gt; less time to do real work</=
div><div>-WG scheduling gets more difficult</div><div>-AD load potentially g=
oes up (someone needs to coordinate between those groups, and the AD would s=
eem to me the natural choice)</div><div>-there is no reason to believe that =
the overall manpower availability goes up through this change, so all the in=
efficiencies introduced go straight against the available cycles.</div><div>=
<br></div><div>Some of these issues can be solved by joint WG meetings and/o=
r careful chair selection. &nbsp;However, both options have drawbacks as wel=
l. &nbsp;A joint session would be counter-productive to the allocation of mo=
re meeting minutes. &nbsp;And when you have, say, 8 chairs for four WG, all =
of which know the RTP architecture well enough, then you have pretty much ex=
hausted, if not overdrawn already, the pool of "key knowledgable people", an=
d burdened them with (in many cases: additional) chair duties.</div><div><br=
></div><div>No matter how I see it, splitting the group is not a good idea.<=
/div><div><br></div><div>Other options to alleviate problems could come to m=
ind:</div><div><br></div><div>If the problem to be solved is chair-workload:=
 add more chairs (but not 6 of them).</div><div>If the problem to be solved =
is meeting-second shortages: have interim meetings, conf calls, etc. &nbsp;I=
n other words, stretch out the AVT meeting in the interval between IETF meet=
ings. &nbsp;Require a web-ex presentation of every draft to be discussed in =
the AVT session at least 4 weeks before the meeting, with a two hour timeslo=
t for Q/A. &nbsp;Should I continue with this brainstorm?</div><div><br></div=
><div>If the problem is something else, I would like to know&#8230;</div><di=
v><br></div><div>Thanks for your consideration, and apologies for the wordco=
unt,</div><div><br></div><div>Stephan</div><div><br></div><div><br></div><di=
v><br></div></div></blockquote></div><br></div></div>_______________________=
________________________
Audio/Video Transport Working Group
<a href=3D"mailto:avt@ietf.org">avt@ietf.org</a>
<a href=3D"https://www.ietf.org/mailman/listinfo/avt">https://www.ietf.org/ma=
ilman/listinfo/avt</a>
</span></body></html>

--B_3377784433_28584930--



From abegen@cisco.com  Thu Jan 13 18:17:46 2011
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Robert Sparks" <rjsparks@nostrum.com>, "Stephan Wenger" <stewe@stewe.org>
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Subject: Re: [AVT] proposed AVT WG split
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I do believe that some re-organization along the lines proposed by the =
ADs might help with the unnecessarily long delays for important specs =
like RAMS.=20

If you ask my opinion, RAMS draft was finished rather slowly: my =
personal target was to finish it in a year. While we had urgency since =
other organizations were waiting for us, starting the wglc process and =
completing it took almost a year. We did more than 20 revisions btw. So, =
editors were not the blocking factor here.=20

Yet, many avt-seasoned people told me that 2 years was actually quite a =
big success for a spec like RAMS. They are probably right, but often a =
late spec is almost as good as no spec, if not worse, especially if =
another organization does something that would break something like RTP. =
It is often impossible to clean up the mess once the damage is done.
=20
I don't think the process can get much slower if some form of splitting =
happens. I am sure you know this better than I do but many other =
organizations are afraid to bring their work to the IETF in general or =
seek approval because they are concerned about the delays. If AVT wants =
to stay relevant in related standards, I think we need to address this =
problem.

-acbegen

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of =
Robert Sparks
> Sent: Thursday, January 13, 2011 10:28 PM
> To: Stephan Wenger
> Cc: IETF AVT WG; IESG IESG
> Subject: Re: [AVT] proposed AVT WG split
>=20
> Stephan -
>=20
> As I noted in response to your comments in Maastricht, I disagree.
>=20
> There are separable streams of work in AVT - it is really multiple =
working groups in one shell at the moment.
>=20
> While there are key people currently in AVT that would overlap these =
groups, they are not the majority of the contributors to
> the different types of activities.
> The intended positive effect of this explicit division is to allow =
these different streams to better parallelize, on list between
> meetings, not just during the IETF
> in-person meeting time.
>=20
> We'd like to try to channel these streams separately for a while. That =
will require some additional coordination work from
> the area directors and chairs, but
> it will bring significant benefits, especially in being able to better =
prioritize work items within the streams.  I realize that for
> folks like you, who will actively participate
> in all of the efforts, the benefits are more limited.
>=20
> I have had several participants express frustration, for example, with =
the way the RAMs work eclipsed basic architectural
> work in AVT.
> While that could be  managed in place  (we could have only one working =
group per area in the extreme), it has shown itself
> to be difficult
> with the set of things this group is working on.
>=20
> I also expect  the separation will make it easier to evaluate when the =
energy for a particular effort (such as the XR metrics
> you point to below) exists,
> and when that level of energy changes.
>=20
> Yes, conflicts between these working groups when they do meet in =
person will have to be managed carefully, but we have
> mechanisms for that. As Keith noted
> separately, we do not anticipate this significantly increasing the =
number-of-minutes footprint on any given in-person IETF
> meeting. Also, your comments assume all
> four of these groups would have a session at each in-person meeting. I =
don't expect that to be the case.
>=20
>=20
> RjS
>=20
>=20
> On Jan 12, 2011, at 6:11 PM, Stephan Wenger wrote:
>=20
>=20
> 	Hi IESG,
>=20
> 	As already communicated in Maastricht, I continue to believe that =
this step is a bad idea.  I didn't attend Beijing, and
> don't know whether there was any discussion about the subject.  I =
didn't see anything mentioned in the AVT minutes, though,
> nor did I see any discussion since Beijing on ietf@ietf.  If there =
was, some of the points below may be redundant, and I
> apologize in advance.
>=20
> 	Essentially, you guys are suggesting to split one "eternal" WG into =
four.  I'm writing "eternal", because I don't see that
> any of the WGs can fulfill its charter before other work items will =
likely get chartered with a similar scope, and in the same
> WG.  For RTP core, RTP extensions, and RTP payloads that is probably =
self-evident.  For the XR stuff: metrics pop up all the
> time, like payload formats.  Metrics are also known to make people =
quick bucks on patent licensing.  There will be more
> metric proposals.  If the RTCP-XR metrics ever get momentum in the =
marketplace, that is.  Somehow, I have my doubts.  But
> that's a different discussion.
>=20
> 	The positive effects of the split could be
> 	-chair workload is split among more people, and
> 	-there are more meeting seconds for RTP related work at any given =
IETF meeting.
>=20
> 	Assuming unlimited resources, both are good things.  Unfortunately, =
resources are not unlimited.
>=20
> 	The negative effects are IMO:
> 	-the key knowledgeable people in AVT ideally would have to sit in all =
four meetings so that all four groups can tap
> their experience.  But they may not want to do so, because it leaves =
less time for cross-area involvement, coordination, and
> draft-writing outside the AVT sessions, and whatnot.  Also, the burden =
for remote participants in far-away time zones goes
> up.  In other words, the resource meeting-seconds for those people is =
not unlimited, and they will have to prioritize
> differently.    The result could be that what I called "key =
knowledgable people" would perhaps not attend the mere RTCP-XR
> or Payload session; and the guys there may be wasting their time doing =
work that does not make sense architecturally, just to
> see the hammer coming down at IETF last call.
> 	-more people get chair positions and are burdened with administrative =
stuff -> less time to do real work
> 	-WG scheduling gets more difficult
> 	-AD load potentially goes up (someone needs to coordinate between =
those groups, and the AD would seem to me the
> natural choice)
> 	-there is no reason to believe that the overall manpower availability =
goes up through this change, so all the
> inefficiencies introduced go straight against the available cycles.
>=20
> 	Some of these issues can be solved by joint WG meetings and/or =
careful chair selection.  However, both options have
> drawbacks as well.  A joint session would be counter-productive to the =
allocation of more meeting minutes.  And when you
> have, say, 8 chairs for four WG, all of which know the RTP =
architecture well enough, then you have pretty much exhausted, if
> not overdrawn already, the pool of "key knowledgable people", and =
burdened them with (in many cases: additional) chair
> duties.
>=20
> 	No matter how I see it, splitting the group is not a good idea.
>=20
> 	Other options to alleviate problems could come to mind:
>=20
> 	If the problem to be solved is chair-workload: add more chairs (but =
not 6 of them).
> 	If the problem to be solved is meeting-second shortages: have interim =
meetings, conf calls, etc.  In other words,
> stretch out the AVT meeting in the interval between IETF meetings.  =
Require a web-ex presentation of every draft to be
> discussed in the AVT session at least 4 weeks before the meeting, with =
a two hour timeslot for Q/A.  Should I continue with
> this brainstorm?
>=20
> 	If the problem is something else, I would like to know=85
>=20
> 	Thanks for your consideration, and apologies for the wordcount,
>=20
> 	Stephan
>=20
>=20
>=20
>=20


From Even.roni@huawei.com  Thu Jan 13 19:30:34 2011
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Date: Fri, 14 Jan 2011 05:30:52 +0200
From: Roni Even <Even.roni@huawei.com>
In-reply-to: <04CAD96D4C5A3D48B1919248A8FE0D540E16D87A@xmb-sjc-215.amer.cisco.com>
To: "'Ali C. Begen (abegen)'" <abegen@cisco.com>, 'Robert Sparks' <rjsparks@nostrum.com>, 'Stephan Wenger' <stewe@stewe.org>
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Cc: 'IETF AVT WG' <avt@ietf.org>, 'IESG IESG' <iesg@ietf.org>
Subject: Re: [AVT] proposed AVT WG split
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Hi Ali,
I think that some work can be spawned as a separate group using the dispatch
WG to do so.
As for the time it took I suggest you look at the port mapping draft, when
it started and how the discussion led to changes even late in the process.
This draft and the cname and ssrc ports were the ones delaying the RAMS work
and not the main body
Roni Even

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
> Ali C. Begen (abegen)
> Sent: Friday, January 14, 2011 4:20 AM
> To: Robert Sparks; Stephan Wenger
> Cc: IETF AVT WG; IESG IESG
> Subject: Re: [AVT] proposed AVT WG split
> 
> I do believe that some re-organization along the lines proposed by the
> ADs might help with the unnecessarily long delays for important specs
> like RAMS.
> 
> If you ask my opinion, RAMS draft was finished rather slowly: my
> personal target was to finish it in a year. While we had urgency since
> other organizations were waiting for us, starting the wglc process and
> completing it took almost a year. We did more than 20 revisions btw.
> So, editors were not the blocking factor here.
> 
> Yet, many avt-seasoned people told me that 2 years was actually quite a
> big success for a spec like RAMS. They are probably right, but often a
> late spec is almost as good as no spec, if not worse, especially if
> another organization does something that would break something like
> RTP. It is often impossible to clean up the mess once the damage is
> done.
> 
> I don't think the process can get much slower if some form of splitting
> happens. I am sure you know this better than I do but many other
> organizations are afraid to bring their work to the IETF in general or
> seek approval because they are concerned about the delays. If AVT wants
> to stay relevant in related standards, I think we need to address this
> problem.
> 
> -acbegen
> 
> > -----Original Message-----
> > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
> Robert Sparks
> > Sent: Thursday, January 13, 2011 10:28 PM
> > To: Stephan Wenger
> > Cc: IETF AVT WG; IESG IESG
> > Subject: Re: [AVT] proposed AVT WG split
> >
> > Stephan -
> >
> > As I noted in response to your comments in Maastricht, I disagree.
> >
> > There are separable streams of work in AVT - it is really multiple
> working groups in one shell at the moment.
> >
> > While there are key people currently in AVT that would overlap these
> groups, they are not the majority of the contributors to
> > the different types of activities.
> > The intended positive effect of this explicit division is to allow
> these different streams to better parallelize, on list between
> > meetings, not just during the IETF
> > in-person meeting time.
> >
> > We'd like to try to channel these streams separately for a while.
> That will require some additional coordination work from
> > the area directors and chairs, but
> > it will bring significant benefits, especially in being able to
> better prioritize work items within the streams.  I realize that for
> > folks like you, who will actively participate
> > in all of the efforts, the benefits are more limited.
> >
> > I have had several participants express frustration, for example,
> with the way the RAMs work eclipsed basic architectural
> > work in AVT.
> > While that could be  managed in place  (we could have only one
> working group per area in the extreme), it has shown itself
> > to be difficult
> > with the set of things this group is working on.
> >
> > I also expect  the separation will make it easier to evaluate when
> the energy for a particular effort (such as the XR metrics
> > you point to below) exists,
> > and when that level of energy changes.
> >
> > Yes, conflicts between these working groups when they do meet in
> person will have to be managed carefully, but we have
> > mechanisms for that. As Keith noted
> > separately, we do not anticipate this significantly increasing the
> number-of-minutes footprint on any given in-person IETF
> > meeting. Also, your comments assume all
> > four of these groups would have a session at each in-person meeting.
> I don't expect that to be the case.
> >
> >
> > RjS
> >
> >
> > On Jan 12, 2011, at 6:11 PM, Stephan Wenger wrote:
> >
> >
> > 	Hi IESG,
> >
> > 	As already communicated in Maastricht, I continue to believe that
> this step is a bad idea.  I didn't attend Beijing, and
> > don't know whether there was any discussion about the subject.  I
> didn't see anything mentioned in the AVT minutes, though,
> > nor did I see any discussion since Beijing on ietf@ietf.  If there
> was, some of the points below may be redundant, and I
> > apologize in advance.
> >
> > 	Essentially, you guys are suggesting to split one "eternal" WG
> into four.  I'm writing "eternal", because I don't see that
> > any of the WGs can fulfill its charter before other work items will
> likely get chartered with a similar scope, and in the same
> > WG.  For RTP core, RTP extensions, and RTP payloads that is probably
> self-evident.  For the XR stuff: metrics pop up all the
> > time, like payload formats.  Metrics are also known to make people
> quick bucks on patent licensing.  There will be more
> > metric proposals.  If the RTCP-XR metrics ever get momentum in the
> marketplace, that is.  Somehow, I have my doubts.  But
> > that's a different discussion.
> >
> > 	The positive effects of the split could be
> > 	-chair workload is split among more people, and
> > 	-there are more meeting seconds for RTP related work at any given
> IETF meeting.
> >
> > 	Assuming unlimited resources, both are good things.
> Unfortunately, resources are not unlimited.
> >
> > 	The negative effects are IMO:
> > 	-the key knowledgeable people in AVT ideally would have to sit in
> all four meetings so that all four groups can tap
> > their experience.  But they may not want to do so, because it leaves
> less time for cross-area involvement, coordination, and
> > draft-writing outside the AVT sessions, and whatnot.  Also, the
> burden for remote participants in far-away time zones goes
> > up.  In other words, the resource meeting-seconds for those people is
> not unlimited, and they will have to prioritize
> > differently.    The result could be that what I called "key
> knowledgable people" would perhaps not attend the mere RTCP-XR
> > or Payload session; and the guys there may be wasting their time
> doing work that does not make sense architecturally, just to
> > see the hammer coming down at IETF last call.
> > 	-more people get chair positions and are burdened with
> administrative stuff -> less time to do real work
> > 	-WG scheduling gets more difficult
> > 	-AD load potentially goes up (someone needs to coordinate between
> those groups, and the AD would seem to me the
> > natural choice)
> > 	-there is no reason to believe that the overall manpower
> availability goes up through this change, so all the
> > inefficiencies introduced go straight against the available cycles.
> >
> > 	Some of these issues can be solved by joint WG meetings and/or
> careful chair selection.  However, both options have
> > drawbacks as well.  A joint session would be counter-productive to
> the allocation of more meeting minutes.  And when you
> > have, say, 8 chairs for four WG, all of which know the RTP
> architecture well enough, then you have pretty much exhausted, if
> > not overdrawn already, the pool of "key knowledgable people", and
> burdened them with (in many cases: additional) chair
> > duties.
> >
> > 	No matter how I see it, splitting the group is not a good idea.
> >
> > 	Other options to alleviate problems could come to mind:
> >
> > 	If the problem to be solved is chair-workload: add more chairs
> (but not 6 of them).
> > 	If the problem to be solved is meeting-second shortages: have
> interim meetings, conf calls, etc.  In other words,
> > stretch out the AVT meeting in the interval between IETF meetings.
> Require a web-ex presentation of every draft to be
> > discussed in the AVT session at least 4 weeks before the meeting,
> with a two hour timeslot for Q/A.  Should I continue with
> > this brainstorm?
> >
> > 	If the problem is something else, I would like to know.
> >
> > 	Thanks for your consideration, and apologies for the wordcount,
> >
> > 	Stephan
> >
> >
> >
> >
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
> 
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From magnus.westerlund@ericsson.com  Fri Jan 14 01:24:50 2011
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Cc: Cullen Jennings <fluffy@cisco.com>, "iesg@ietf.org" <iesg@ietf.org>, "avt@ietf.org" <avt@ietf.org>
Subject: Re: [AVT] proposed AVT WG split
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Stephan Wenger skrev 2011-01-14 02:18:
>>
>>
>>>  But they may not want to do so, because it leaves less time for
>>> cross-area involvement, coordination, and draft-writing outside the AVT
>>> sessions, and whatnot.  Also, the burden for remote participants in
>>> far-away time zones goes up.  In other words, the resource
>>> meeting-seconds for those people is not unlimited, and they will have to
>>> prioritize differently.    The result could be that what I called "key
>>> knowledgable people" would perhaps not attend the mere RTCP-XR or
>>> Payload session; and the guys there may be wasting their time doing work
>>> that does not make sense architecturally, just to see the hammer coming
>>> down at IETF last call.
>>> -more people get chair positions and are burdened with administrative
>>> stuff -> less time to do real work
>>
>> Given the current chair bottlenecks, and the fact we have people that
>> would like to do this, I view the bunch more chairs a pro not a con
> 
> Really?  At almost every plenary I hear folks whining about how difficult
> it is to get *qualified* chairs.  (Some would say that AVT, in its present
> incarnation, clearly shows that difficulty)

In this regard I do think splitting it does help. I will publicly state
that I have volunteered to be a candidate to chair AVTCORE and AVTEXT.
For me this is easier to take on as I at least don't need to handle also
the documents of PAYLOAD and XR. I do still remember my 3 years as AVT
chair. When we for a number of meeting had around 600000 words of AVT
documents. I would be much more hesitant to take on the non-split AVT.


In general I think this split is clearly worth trying out. We might
realize down the road that it is over splitting things, and that some
groups can merge or simply close. But lets gather some experience on this.

Yes, it will have downsides, it also will have positive effects. Lets
try this somewhat different trade-off.

I do think smaller groups with less content might allow for a more
focused work in the groups. But it will require the chairs to use the
smaller scope of each group to speed up things. I also agree that when
there is contention on how to do things, things take longer time. And
that needs to be taken into account, and for example do more virtual
interim meetings etc to give the subject more focus.

In conclusion, I think this proposal is well worth trying out. And if it
becomes misery and pain with no gain, then the AD can change things again.

Cheers

Magnus Westerlund

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

From magnus.westerlund@ericsson.com  Fri Jan 14 05:13:43 2011
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Hi,

I haven't re-reveiew the latest version of document yet. But I do want
to respond to a few issues here. Or try to clarify what I meant.


Ali C. Begen (abegen) skrev 2010-12-14 15:32:
>> 1. Section 3.1: "By default, Tokens are server specific.  However, the
>> client
>>    can use the same Token to communicate with different servers if these
>>    servers are provided with the same secret key and algorithm used to
>>    generate the Token and are at least loosely clock-synchronized."
>>
>> How a client determine this is the case still doesn't appear to be defined.
> 
> The client does not need to know anything besides it is the same address (explicit or implicit from the c line) and the port.
> 
>> My suggestion is that a client determines if it has a valid token based
>> on the port and address of where to send the token request. But this
>> needs to be spelled out somewhere.
> 
> I think this is pretty obvious.

Well, it does affect client implementation and behavior for this. And I
think there is a point to make it clear what is expected to be done. So
that if we find a reason to need to make some adjustments around this,
we can be fairly certain about behavior.

> 
>> 2. Section 3.1:
>> "The
>>    Token becomes invalid if client's public IP address changes or when
>>    the server expires the Token.  In these cases, the client has to
>>    request a new Token."
>>
>> This might be misleading as the client in a number of cases can't
>> determine if an address change happens to them. Thus only determining
>> this when failure happens.
> 
> If you read carefully, the text does not imply that the client knows when its address changes. It simply says "when client's address changes".

Well, it is implied through "In these cases, the client has to request a
new Token." Instead of writing. "If this occurs, then the token will be
invalid and after receiving an error message on a request the client
needs to request a new token".

> 
>> 5. Section 3.2.1: "Ports *c0, *c1 and *c2 " bullet:
>>
>> There is no discussion of if C2 should be equal to C1 or not. I think
> 
> They could be the same as the text says.
> 
>> there might be actually considerations here. This have to do with the
>> binding of the RTCP reports to the unicast session transmission
>> happening P3 -> C1. If that isn't possible then the server doesn't know
> 
> C2 is the port client uses to send the RTCP reports for the unicast session. They go to port P4, which is different than P3. Since P4 cannot be the same as P3, it does not matter what you choose for c2.
> 
>> for which outgoing transmission the reports are for. I think also C3 has
>> to be equal to C2, or the server can't match on anything on transport
>> level. However, if the NAT is not address and port independent mapping
>> also that breaks down due the NAT properties.
> 
> We don’t have a c3 in the figure or text.

Sorry, C2 = C1, and C3 = C2. The issue as I say is that you can't bind
the "burst" or RTX transmission to the RTCP feedback message on those.


> 
>>
>> If one are going to enable to mapp on heuristics then I think the only
>> possibility for the server is to mapp on reported SSRC's and ensure to
>> use unique SSRC identifiers across all unicast RTP session going out
>> from P3.
>>
>> Or have I forgotten something here?
> 
> See my response above and let me know whether it addresses your concerns (which I probably did not get).

Yes, unless you have done something about this I think it something that
needs clarification.

> 
>> 6. Section 3.2.2:
>>
>> I am missing an RTP session definitions. I would like to have something
>> along the following lines included. This I think is the main part of how
>> a unicast session is established.
>>
>> Port mapping starts with a primary RTP session, where the server acting
>> on a port mapping request has an unidirectional flow of RTCP messages
>> from the client port (C1) to the server on a designated port (P3). There
>> MUST be no RTCP traffic related to the primary RTP session being sent
>> from P3 on the server to the client on port (C1).
> 
> This agrees with figure 2.
> 
>>
>> Upon a port mapping request RTCP message in the primary session from the
>> client on local port (C1) to the server port (P3) a new RTP point to
>> point unicast RTP session is created, unless active RTP session within
>> this context already exist. This new session consist of one leg of
>> multiplex RTP and RTCP messages from the Server (P3) to the client (C1).
> 
> Correct.
> 
>> The client to server leg in this unicast RTP session is going from
>> client local port (C2 perhaps C1 if mandated to be equal) to the Server
>> port (P4). The server port for the unicast needs to be signaled to the
>> client.
> 
> Correct.
> 
>>
>> The point I am trying to get across in the above is the need to make it
>> clear what flows of messages are considered to be different RTP sessions.
> 
> This is what we convey with Figure 2. If you look at it, the sessions are properly separated and each session has a number of flows with directions and port numbers clearly identified. It is concrete and clear.
> 
> If further text is needed, we can put additional wording.

Yes, the problem is that you don't spell out which legs are which RTP
sessions. As you do have two, and you don't have the regular symmetric
session. This is important when it comes to using other functionality on
these sessions.

> 
>> Another thing that I think is unclear is when an existing RTP session is
>> reused correctly. Like that a sequence of NACK messages for the same
>> primary RTP session shall of course continue to use the same RTP session.
> 
> Indeed, here we introduce nothing new. The regular RTP rules are still valid. If the session has timed out or was ended by a BYE, it ended. A subsequent NACK will open a new session. Ow, the existing session will continue.

This is far from obvious when you have this functionality. This is new
functionality not previously existing. Before a session was existing and
any request was within that sessions context. Or in certain cases where
multiple sessions exist, the relation between them is clearly stated.

> 
>> A third thing is the connection of the SSRC spaces that happens between
>> these two sessions. That as RTCP response to requests in the primary are
>> going into the unicast sessions packet flow.
>>
>> I have to say that this is starting more and more to trouble me as I
>> peel away the layers of implications. Can I please have a time machine
>> and go back 1.5 years and change things?
> 
> Probably you cannot. However, IMO, you are overthinking things in the sense some of the issues you raise above do not even belong to this spec. The SSRC behavior for example will be defined with the unicast application you are running. It could be RAMS, retransmission, multi-layer video, etc. They define their behavior individually.

Maybe you are correct that this is part of the RAMS, etc specification.
But is an question that needs to be considered. Getting all the quirks
of RTP is not easy, despite now having 10 years of experience with it.
Thus, you need to think through a lot of cases to be certain things
work. When doing that you should document how it is intended to work.
Otherwise we will have issues down the line. But you might be right that
this is an issue that belongs in RAMS and any other mechanism using this
mechanism. But this port mapping do mandate this relation, and thus
should point out this detail.

> 
>>
>> 7. Section 3.2.2, bullet 4:
>>
>> Until a unicast
>>        session is established, neither the server nor the client needs
>>        to send RTCP reports for the unicast session.
>>
>> Well, my question based on this, is when it is considered to not any
>> longer be established. I see two issues around that needs to be made clear.
> 
> Requesting and receiving Tokens does not create the unicast session. That is it.

Correct, but it is this spec that defines how the different legs of the
different RTP sessions exist.

> 
>> 1. How long are the client and server going to send RTCP receiver
>> reports related to the RTP transmissions happening in the session. Are
>> this 5 reporting interval since last RTP traffic in the session, or
>> something else?
> 
> Why define new rules here? This is just another RTP session and is supposed to follow the RTP rules.

Actually this is something new for RAMS/RTX with port mapping. We never
had RTP sessions that was dynamically created from one RTP session. We
always had external signalling that defined when an RTP session could
exist.

> 
>> 2. Secondly, what about RTCP BYE signaling? If either ends has sent
>> this, must the client select a new local port and never send any new
>> requests in the primary that ends up in this unicast session?
> 
> No, why would it mean that? BYE does not have such semantics.
> 

The issue is what is creating this session. It is the combination of a
request needing a unicast session, and the port mapping parts. But we
have completely failed to go into details when a session is terminated.

Yes, BYE's normal semantics is that this SSRC leaves the RTP session
now. So in a context where a request with a certain SSRC leaves a
session, does that imply that the created RTP session goes away?

It is a question, and as I see it a hole in the spec.

>>
>>
>> 8. Section 3.2.2: RTP session question.
>>
>> Are RTP packets in the direction C2->P4 forbidden? Currently there
>> doesn't appear to be any wording on this. Secondly, should it be
>> forbidden? Because I can see there is potential applications that needs
>> a client to server RTP stream, rather than the opposite. And as this is
>> a general RTP tool I think we need to work these questions out.
> 
> Yes, they are forbidden. This was the choice since the beginning. If you need a full bi-directional RTP session on the unicast side, you can do a full offer/answer exchange to set up that session. It does not fit into this context anyway.
> 
> To make this crystal clear, we can say that no rtp traffic must flow from client to server.

Hope this has made clear.

> 
>> 9. Section 4:
>> When a client sends a
>>    Port Mapping Request or Token Verification Request message but it
>>    does not receive a response back from the server (either a Port
>>    Mapping Response or Token Verification Failure message), it MAY
>>    resend its request by following the timer rules defined for RTCP
>>    feedback messages in Section 3.5 of [RFC4585] as a good practice.
>>
>> What are the criteria the client shall use to determine when a packet it
>> sent is lost?
> 
> Seriously? The client is ruled by the timer rules. It has a min waiting time. Then, it can resend or give up. What is there to define here?

Ok, this is serious nitpicking. And in most cases this isn't a problem
becuase the RTCP sending intervall allowed will be longer than the RTT.
But what happens if the one has "unlimited" RTCP bandwidth and can
always send immediately. You still need to wait to request until a time
that is long enough for the RTT + server response time.

> 
>> 15. Section 6:
>> A
>>    client that received a Token Verification Failure message can request
>>    a new Token from the server.
>>
>> I am missing discussion one when there is persistent errors. I think
>> there should be some recommendation about exponentially backing off ones
>> tries to retrieve a new token and then use it.
> 
> Would not 4585 scheduling algorithms take care of this? This is not different than sending broken NACKs for example.

Not really. The 4585 scheduling algorithm allows one to reschedule the
same NACK for every transmission and never give up. For knowing expected
behavior there is a point of making clear what the behavior here is.

> 
>> One may also need to consider if one after a period should try to
>> retrieve fresh declaration information for the service one tries to
>> access to see if the error is there?
> 
> If a server has given me a token valid for the next 12 hours, I can use it for the next 12 hours. If the service gets disabled in the mean time say after 6 hours, I will eventually get an error. So what? I can try to get a new token. Or, as you suggest I can be conservative and check with the server every hour to see whether I can get a valid token (because if the service got disabled, I won't get a valid token). This behavior is something up to the implementation. Either is fine and has pros/cons for different scenarios.
> 

That is not what I am saying. I asking for the service declartion, like
SDP or XML. Basically saying that is you have a persistent error at this
part, you better pop up to the application layer and have them check
their information.

> 
>> 15. Section 7.1:
>>
>> If used at a media level, the attribute MUST be used in a unicast
>>    media block.
>>
>> I think this sentence should go. Because this is in conflict with the
>> current semantics. I think it is better to include the a=portmapping-req
>> attribute at the places where one needs to send token verification
>> requests in RTCP. Rather than consider this to be derived implicitly.
> 
> It is already gone since we currently switched to media-level attributes only. And the text simply says "The 'portmapping-req' attribute SHALL be used as a media-level attribute."
> 
> 
>> This would make the SDP example in Figure 7 look like this:
>>
>>        v=0
>>         o=ali 1122334455 1122334466 IN IP4 nack.example.com
>>         s=Local Retransmissions
>>         t=0 0
>>         a=group:FID 1 2
>>         a=rtcp-unicast:rsi
>>         m=video 41000 RTP/AVPF 98
>>         i=Multicast Stream
>>         c=IN IP4 233.252.0.2/255
>>         a=source-filter:incl IN IP4 233.252.0.2 198.51.100.1   ; Note 1
>>         a=rtpmap:98 MP2T/90000
>>         a=multicast-rtcp:41500                                 ; Note 1
>>         a=rtcp:42000 IN IP4 192.0.2.1                          ; Note 2
>>         a=rtcp-fb:98 nack                                      ; Note 2
>>         a=portmapping-req:30000                                ; Note 6
> 
> As I explained a few times before, putting this attribute here is a mistake. It needs to be in the unicast session associated with the multicast session.
> 

I still strongly disagree on this.

>>         a=mid:1
>>         m=video 42000 RTP/AVPF 99                              ; Note 3
>>         i=Unicast Retransmission Stream
>>         c=IN IP4 192.0.2.1
>>         a=sendonly
>>         a=rtpmap:99 rtx/90000
>>         a=rtcp-mux                                             ; Note 4
>>         a=rtcp:42500                                           ; Note 5
>>         a=fmtp:99 apt=98; rtx-time=5000
>>         a=portmapping-req:30000                                ; Note 7
>>         a=mid:2
>>
>> The line with Note 7 would only be needed if we actually use RTCP bye in
>> the nack context to kill of the unicast session. Otherwise I don't think
>> it is needed here in Nack. But if the unicast session was for RAMS, then
>> it would be needed.
> 
> I respectfully disagree that the attribute will be repeated in the multicast block. It does not make sense.

No, I think it makes total sense to indicate the RTP sessions and there
corresponding RTCP flows where you will need a token for submitting some
requests. Rather than placing it in the m= for an RTP session where you
don't needed, only need a token to create this session.

I guess this is different views on how one should signal this. I will
review draft-11 now and see if you then made the definition distinct
enough that it is clear when the attribute should be used.

Cheers

Magnus Westerlund

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Färögatan 6                | Mobile +46 73 0949079
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From magnus.westerlund@ericsson.com  Fri Jan 14 05:24:13 2011
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Subject: Re: [AVT] Comment on Section 5	ofdraft-ietf-avt-ports-for-ucast-mcast-rtp-09.txt
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Hi,

I would say that adding gives you nothing, even if detected. They you
get token failure, and anyway request a new token. Thus detecting it or
not the server will anyway eat the increased load.

I don't see that one can mitigate this attack without integrity checking
the token response message

Cheers

Magnus

Ali C. Begen (abegen) skrev 2010-12-19 15:39:
> 
> 
>> -----Original Message-----
>> From: Glen Zorn [mailto:gwz@net-zen.net]
>> Sent: Saturday, December 18, 2010 8:56 PM
>> To: Ali C. Begen (abegen)
>> Cc: avt@ietf.org
>> Subject: RE: [AVT] Comment on Section 5 ofdraft-ietf-avt-ports-for-ucast-mcast-rtp-09.txt
>>
>> Ali C. Begen (abegen) [mailto://abegen@cisco.com] writes:
>>
>>
>>>> SHA1 isn't free, or even particularly cheap; to me, this has DoS
>>> written all
>>>> over it.  Further, AFAICT, the attack is undetectable except through
>>> traffic
>>>> monitoring.
>>>
>>> If there is a concern that this may overload the server, then, yes,
>>> getting this into the token will fix it. However, if I were the
>>> attacker, I would rather mess with the token or other fields, so that
>>> both the service would fail and the client would try getting a new token
>>> right away (even without waiting for 2 mins).
>>
>> OK, but again, the difference between that and what I'm describing is that
>> the latter is AFAICT indistinguishable (by the parties involved) from the
>> normal operation of the protocol, while the former is easily recognizable as
>> an attack.
> 
> I think there is an interesting point here, which is the following:
> 
> If an attacker messes with the relative exp time value, this can still go unnoticed very easily if the attacker "fixes" its value on the reverse path. For example, the attacker reduces the value from 2 hours to 2 mins. The client will think that the token is valid only for two minutes, and will request a new token before the 2-min deadline expires. In the meantime, if the token is used by the client, the attacker can revert 2 minutes to 2 hours in the Token Verification Request messages and make everything look perfect from server's viewpoint.
> 
> Two seconds later, the client will request a new Token. This will go unnoticed by the server no matter whether we include this field in the token or not. Here, the cost, if any, will be on the server (for extra token generations). This attack AFAIK can only be avoided if the RTCP messages are protected or maybe the server is monitored for redundant token requests from a particular client.
> 
> Oppositely, if the attacker changes the value from 2 hours to 4 hours and the client uses the token after 2 hours, this will cause a failure message from the server. But, again the server may not necessarily notice this attack as the attacker can undo its changes on the reverse path. In this case, the server will assume the client is trying to use expired tokens. Here, the cost (which is delay) is to the client not the server. I think this also requires protected RTCP messages.
> 
> So, I think we can mention this and say that if there is such a risk (MITM), monitoring activity on the server or protecting RTCP messages may help.
> 
> -acbegen
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
> 


-- 

Magnus Westerlund

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Multimedia Technologies, Ericsson Research EAB/TVM
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Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
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From ray.vanbrandenburg@tno.nl  Fri Jan 14 08:12:55 2011
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From: "Brandenburg, R. (Ray) van" <ray.vanbrandenburg@tno.nl>
To: "avt@ietf.org" <avt@ietf.org>
Thread-Topic: New version of RTCP for IDMS
Thread-Index: Acu0BjkkNJ4VJH75TgKyYQ88CLubgw==
Date: Fri, 14 Jan 2011 16:15:13 +0000
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Subject: [AVT] New version of RTCP for IDMS
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Hi all,

We have just submitted a new version of our internet draft describing an RT=
CP XR block for Inter-Destination Media Synchronization (IDMS).

http://tools.ietf.org/html/draft-brandenburg-avt-rtcp-for-idms-03

We have tried to include all feedback received both from the mailing list a=
s well as from IETF 79 in Beijing.

Major changes include, apart from numerous clarifications and rewrites:

- an example/tutorial section that we hope will help make clear how IDMS ca=
n be used in practice.
- an 'Implementation Considerations' section that adresses a number of issu=
es that were raised on the mailing list, including how to deal with NTP syn=
chronization in different use cases, echo cancellation and the issue of rep=
orting versus controlling.
- the removal of the mixer functionality

In the past there have been some concerns about whether RTCP XR blocks are =
the best solution for IDMS and if a new RTCP block wouldn't be better. Also=
, some questions were raised about the advantage of having both client->ser=
ver and server->client communication with the same XR block. On these issue=
s we feel it is important to understand that the proposed IDMS system is no=
t a control protocol; client are not told how to synchronize their playback=
 to others. The proposed XR block only informs clients of the slowest playb=
ack within their synchronization group, how they deal with this information=
, if at all, is up to themselves. In this way, IDMS is not that different f=
rom how RTCP is used to deal with synchronization between audio and video s=
treams on the same terminal.

I look forward to your comments.

Kind regards,

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


From magnus.westerlund@ericsson.com  Fri Jan 14 08:19:52 2011
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Subject: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 1 SSRC spaces
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Hi,

This is one of several issues I still see remaining in the document. It
has clearly improved. I am sorry that I become sick before Christmas and
couldn't continue the discussion then.

There is an issue with SSRC usage in this document related to the
different RTP sessions. Please remember that the most fundamental part
of the RTP session definition is that each RTP session has its own SSRC
space.

So there is one SSRC number space for the multicast RTP session. An SSRC
from this space is used in the Token Verification Request to identify
the requesting client. In the Token Verification response message this
SSRC identifier is included in the Token response message. The problem
is that this message is conveyed in the unicast RTP session that has its
own SSRC space. Thus we have potentially create a coupling between the
two RTP sessions.

As I see it this can be fixed in two ways. Either one explicitly notes
that this RTCP Token Response message field carries a value that refers
to the SSRC space in another RTP session then one currently are in. This
is likely the simplest solution.

The alternative is to accept the coupling. But then one has to make
clear on how to handle SSRC collisions. This seems problematic as the
coupled session is a multicast session. A multicast session has bigger
difficulties with changes due to its number of receivers and also that
if one forces an active sender to change it may causes widespread
disruption.

I also observe that it is not clear what SSRC space the Port Mapping
request and response messages uses. Does this need to be synchronized
with any other space or can it be its own unique one. A unique one is
probably easiest. I don't see any direct coupling to the other sessions.

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 keith.drage@alcatel-lucent.com  Fri Jan 14 08:31:01 2011
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: "avt@ietf.org" <avt@ietf.org>
Date: Fri, 14 Jan 2011 17:33:19 +0100
Thread-Topic: WGLC for draft-ietf-avt-ports-for-ucast-mcast-rtp-11
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Subject: Re: [AVT] WGLC for draft-ietf-avt-ports-for-ucast-mcast-rtp-11
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(As WG co-chair)

A reminder of the WGLC on this document.

Please provide your comments by the 19th January 2011.

If you believe you need more time, then please mail the chairs, with an exp=
ected date you believe you can meet.

If you have read it, and believe the document addresses all issues appropri=
ately as it is, then please send the chairs a mail as well. This helps cons=
iderably in identifying the degree of review for the document.

regards

Keith

> -----Original Message-----
> From: DRAGE, Keith (Keith)=20
> Sent: Wednesday, January 05, 2011 5:11 PM
> To: DRAGE, Keith (Keith); avt@ietf.org
> Cc: draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
> Subject: RE: WGLC for draft-ietf-avt-ports-for-ucast-mcast-rtp-11
>=20
> (As WG co-chair)
>=20
> I should add that the document is intended as proposed=20
> standard, so please take that into account in your review.
>=20
> Keith=20
>=20
> > -----Original Message-----
> > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On=20
> Behalf Of=20
> > DRAGE, Keith (Keith)
> > Sent: Wednesday, January 05, 2011 4:56 PM
> > To: avt@ietf.org
> > Cc: draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
> > Subject: [AVT] WGLC for draft-ietf-avt-ports-for-ucast-mcast-rtp-11
> >=20
> > (As WG co-chair)
> >=20
> > The WG chairs believe this document now represents working group=20
> > consensus on the technical solution so...
> >=20
> > This is to announce a working group last call for
> > draft-ietf-avt-ports-for-ucast-mcast-rtp-11 with comments to be=20
> > provided by 19th January.
> >=20
> > http://www.ietf.org/internet-drafts/draft-ietf-avt-ports-for-u
> > cast-mcast-rtp-11.txt
> >=20
> > In providing comments, please identify the level you believe the=20
> > comment is (editorial, minor technical, major technical) in=20
> order to=20
> > give some guidance to prioritizing the show stoppers.
> >=20
> > Please provide your comments in reply to this message.
> >=20
> > If you read it and have no comments, and believe the document is=20
> > therefore OK, then please send a message to the WG chairs
> > - this is extremely helpful in determining level of review the=20
> > document has had.
> >=20
> > regards
> >=20
> > Keith
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www.ietf.org/mailman/listinfo/avt
> > =

From magnus.westerlund@ericsson.com  Fri Jan 14 08:31:19 2011
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Subject: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 2 Unicast RTP session Termination
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Hi,

The second issue is getting back to the unicast RTP session termination.

Having re-read this document it is very clear on when an Unicast RTP
session is created. Section 3.2, bullet 4 makes this clear. However,
there is no discussion on what the criterias are for when the session
can be considered terminated.

This has previously been discussed in my review comments for 08 version
but we didn't conclude on the discussion due to me being unable to
respond then.

To follow up on one thing. Yes, if we had an architectural and usage
requirements document for unicast sessions created in relation to
multicast then that document would make sense to discuss this issue. As
I see it now, that will not be created. This is the usage independent
component so I think it makes sense to discuss it here, even if not
perfectly fit. But I like to point out that creation and some other
treatment things are discussed. So lets include how it is terminated.

The reason i think it is important to discuss is that the end-points
need to know when the other side will consider it terminated and create
a new session in its eyes rather than continue using the current one.
Also, the other signaling mechanisms do have some way of terminating.
SIP Dialog termination kills the media RTP sessions, they can also
become terminated in SDP by re-invites. RTSP has its Teardown method.
SAP relies on SDP t= attribute to tell when the session is valid.

But for this solution it is unclear. That becomes a bit more problematic
as this is a protocol that is intended for highly dynamic cases, like
RAMS. So I would really like to know what the criteria are. Otherwise we
can have clients sending RTCP to a server for a session it considers
closed. It also become a point of resource handling. One should clean up
what one create.

I am fine making this timeout based, like 5 regular report interval (or
TRR-int) without RTP activity. But I think we need to consider what it
means if one peer sends BYE. Can you time out earlier then?


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 iesg-secretary@ietf.org  Fri Jan 14 08:39:23 2011
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Subject: [AVT] Protocol Action: 'The use of AES-192 and AES-256 in Secure RTP' to	Proposed Standard (draft-ietf-avt-srtp-big-aes-06.txt)
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The IESG has approved the following document:
- 'The use of AES-192 and AES-256 in Secure RTP'
  (draft-ietf-avt-srtp-big-aes-06.txt) as a Proposed Standard

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

The IESG contact persons are Robert Sparks and Gonzalo Camarillo.

A URL of this Internet Draft is:
http://datatracker.ietf.org/doc/draft-ietf-avt-srtp-big-aes/




Technical Summary

This document describes the use of the Advanced Encryption Standard
(AES) with 192 and 256 bit keys within the Secure RTP protocol.  It
details Counter Mode encryption for SRTP and SRTCP and a new SRTP Key
Derivation Function (KDF) for AES-192 and AES-256.

Working Group Summary

The document achieved consensus in the AVT working group.

Document Quality

David McGrew, Jonathan Lennox (in the the open-source libsrtp project) and
Philip Zimmermann (in libZRTP) have indicated implementations of this
internet-draft. John Mattsson independently verified the tests. 

Personnel

Keith Drage is the document shepherd. Robert Sparks is the responsible AD.

From magnus.westerlund@ericsson.com  Fri Jan 14 08:40:07 2011
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Subject: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 3 Binding of C2->P4 RTCP to correct RTP session
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Hi,

This third issue was also brought up before and also Tom Van Caenegem
comment on it.

The RTCP reports going from C2->P4 from a receiver is not obvious on how
to correlate with the Unicast RTP session it is reporting on that are
sent P3->C1.

The issues that may arise are the following. First there might be a NAT
with port and address dependent mapping between them. Thus, even if
C2=C1, the externally seen address+port can be different between traffic
going to P3 and P4. In addition we shall not forget that there might be
multiple clients behind the same NAT that participate in the same
multicast session. So several RTCP flow might go to P4 from this IP
address.

I think some details on how to correlate this RTCP flow needs to be
included in the document. What are the determining factors? SSRC can't
be it unless we couple the multicast and unicast session. Which I think
is a bad idea. CNAME is likely the only method that is directly
applicable and would match from the C1->P3 request to the C2->P4 one.
But this needs to be stated as a requirement.


Cheers

Magnus Westerlund

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From tom.van_caenegem@alcatel-lucent.com  Fri Jan 14 08:42:23 2011
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From: "Van Caenegem, Tom (Tom)" <tom.van_caenegem@alcatel-lucent.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>, IETF AVT WG <avt@ietf.org>, "draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org" <draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org>
Date: Fri, 14 Jan 2011 17:44:43 +0100
Thread-Topic: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 1 SSRC spaces
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Subject: Re: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 1 SSRC spaces
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Magnus,

Thx for the comment, see below=20
Tom
-----Original Message-----
From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Magnu=
s Westerlund
Sent: vrijdag 14 januari 2011 17:22
To: IETF AVT WG; draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
Subject: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11=
: Part 1 SSRC spaces

Hi,

This is one of several issues I still see remaining in the document. It has=
 clearly improved. I am sorry that I become sick before Christmas and could=
n't continue the discussion then.

There is an issue with SSRC usage in this document related to the different=
 RTP sessions. Please remember that the most fundamental part of the RTP se=
ssion definition is that each RTP session has its own SSRC space.

So there is one SSRC number space for the multicast RTP session. An SSRC fr=
om this space is used in the Token Verification Request to identify the req=
uesting client. In the Token Verification response message this SSRC identi=
fier is included in the Token response message. The problem is that this me=
ssage is conveyed in the unicast RTP session that has its own SSRC space. T=
hus we have potentially create a coupling between the two RTP sessions.

<Tom: well the text says for the "token verification failure message":

SSRC of Requesting Client (32 bits):  Field that contains the SSRC
      of the client who sent the verification request.

..but this does not mean it equals the SSRC in the token verification reque=
st message.  The port mapping "client" has its SSRC in the SSM session, its=
 SSRC in the port mapping exchange messages with the server and its SSRC in=
 the unicast sessioon. It needs to use the correct SSRC in the messages sen=
t in all these sessions. Of course, a relaxing implementation may use the s=
ame client SSRC across
these sessions.. (but in conflict with RFC 3550)>

As I see it this can be fixed in two ways. Either one explicitly notes that=
 this RTCP Token Response message field carries a value that refers to the =
SSRC space in another RTP session then one currently are in.=20

<Tom: this is what was implied by the text..but should be called out more e=
xplictly perhaps>

This is likely the simplest solution.


The alternative is to accept the coupling. But then one has to make clear o=
n how to handle SSRC collisions. This seems problematic as the coupled sess=
ion is a multicast session. A multicast session has bigger difficulties wit=
h changes due to its number of receivers and also that if one forces an act=
ive sender to change it may causes widespread disruption.

I also observe that it is not clear what SSRC space the Port Mapping reques=
t and response messages uses. Does this need to be synchronized with any ot=
her space or can it be its own unique one. A unique one is probably easiest=
. I don't see any direct coupling to the other sessions.

<Tom: has its own SSRC space....  See also the figure that shows the three =
sessions decoupled by means of the dashed lines..>

Cheers

Magnus Westerlund

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----------------------------------------------------------------------
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F=E4r=F6gatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
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_______________________________________________
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avt@ietf.org
https://www.ietf.org/mailman/listinfo/avt

From magnus.westerlund@ericsson.com  Fri Jan 14 08:54:50 2011
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Subject: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 4 SDP attribute semantics
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Hi,

I think the change for the SDP attributes "portmapping" and
"portmapping-req" was an improvement. However, I think the semantics and
usage of these SDP Attributes needs clarification.

Lets start with "portmapping", currently it seems vague and not match
the generalize requirements needed. The current semantics appears to be:
Include in the multicast m= block which in relation there might be need
for token verification request. For example if one looks at Section
4.3.1 where BYE messages related to RAMS bursts may need tokens. That is
within the unicast session created if I understand correctly. Thus there
is no indication in this media block. Only in the Multicast session
block. This appears to be rather fuzzy and unclear.

I would like to resuggest the semantics I was considering before:

The "portmapping" attribute SHALL be included in any SDP media block
(m=) that define an RTP session that require the usage of token
verification requests for one or more RTCP message.

Then I frankly have to say that I don't grasp what the semantics are
related to the placement of the "portmapping-req" attribute. Yes, it
conveys the port and possibly address for where to send the Mapping
request. But what is the meaning of its placement on a particular media
block? Why is it placed in unicast blocks? Even if no Token is required
in this RTP session?

To me it appears that this attribute would be better served by becoming
an Session + possibly override in media blocks. At session level it
would server as the port mapping port and possibly address, unless c= at
session level is unicast to the mapping server, that is used for all
tokens used in the RTP sessions indicated by the "portmapping"
attribute. Then it may be included at media level in an media block
containing a=portmapping to override this port and address within that
particular media block.

Cheers

Magnus Westerlund

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From magnus.westerlund@ericsson.com  Fri Jan 14 08:58:18 2011
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Subject: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 5 Page 11, minimal time recommendation for re-use
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Hi,

On page 11 last paragraph the following is written:

	After a sufficient
   time, a previously used receive port could be used again.

It doesn't define what a sufficiently time is. I would suggest that a
value that is extremely likely to be sufficent would be to use one TCP
Maximum Segment Lifetime which is 2 minutes. I think some recommendation
is in order.

Cheers

Magnus Westerlund

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From magnus.westerlund@ericsson.com  Fri Jan 14 09:06:16 2011
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Subject: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 6: Resend behavior
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Hi,

In Section 4, there is some statement about resending. I want to come to
some conclusion on this topic that was brought up before.

The basic issue is the question on how one detects packet loss for RTCP.
There is no other mechanism than detecting lack of response / action
from the corresponding peer.

The issue I have is that the RTT + server delay in responding can easily
be many times the minimal sending interval that RTCP configure to serve
even 1 mbit/s stream. Thus the formulation in the draft:

"When a client sends a
   Port Mapping Request or Token Verification Request message but it
   does not receive a response back from the server (either a Port
   Mapping Response or Token Verification Failure message), it MAY
   resend its request by following the timer rules defined for RTCP
   feedback messages in Section 3.5 of [RFC4585] as a good practice."

Might not be sufficient. Because the issue is how one detects the packet
loss and how quickly the packet is considered lost.

Cheers

Magnus Westerlund

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From magnus.westerlund@ericsson.com  Fri Jan 14 09:11:31 2011
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Subject: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 7 Persistent Errors
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Hi,

This is a continuation of the issue 15A from the 08 review.

The issue is when a client should consider an error to be persistent, or
at least not short time temporarily or resolvable. If a client continues
to get Mapping responses with errors after having tried several times.
Maybe it should do exponential back-off. Maybe it should update its
session description. Similar for Verification Requests.

Some recommendation about such behavior I think has clear benefit and
reduced uncertainty in behavior by clients.

This was my last issue I have found in this review. A clear improvement
from the previous review of the document.

Cheers

Magnus Westerlund

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From magnus.westerlund@ericsson.com  Fri Jan 14 09:15:32 2011
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Subject: Re: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 1 SSRC spaces
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Van Caenegem, Tom (Tom) skrev 2011-01-14 17:44:
> Magnus,
> 
> Thx for the comment, see below 
> Tom
> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Magnus Westerlund
> Sent: vrijdag 14 januari 2011 17:22
> To: IETF AVT WG; draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
> Subject: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 1 SSRC spaces
> 
> Hi,
> 
> This is one of several issues I still see remaining in the document. It has clearly improved. I am sorry that I become sick before Christmas and couldn't continue the discussion then.
> 
> There is an issue with SSRC usage in this document related to the different RTP sessions. Please remember that the most fundamental part of the RTP session definition is that each RTP session has its own SSRC space.
> 
> So there is one SSRC number space for the multicast RTP session. An SSRC from this space is used in the Token Verification Request to identify the requesting client. In the Token Verification response message this SSRC identifier is included in the Token response message. The problem is that this message is conveyed in the unicast RTP session that has its own SSRC space. Thus we have potentially create a coupling between the two RTP sessions.
> 
> <Tom: well the text says for the "token verification failure message":
> 
> SSRC of Requesting Client (32 bits):  Field that contains the SSRC
>       of the client who sent the verification request.
> 
> ..but this does not mean it equals the SSRC in the token verification request message.  The port mapping "client" has its SSRC in the SSM session, its SSRC in the port mapping exchange messages with the server and its SSRC in the unicast sessioon. It needs to use the correct SSRC in the messages sent in all these sessions. Of course, a relaxing implementation may use the same client SSRC across
> these sessions.. (but in conflict with RFC 3550)>
> 
> As I see it this can be fixed in two ways. Either one explicitly notes that this RTCP Token Response message field carries a value that refers to the SSRC space in another RTP session then one currently are in. 
> 
> <Tom: this is what was implied by the text..but should be called out more explictly perhaps>

Yes, just being a bit more explict about that this SSRC value are in
another RTP session SSRC space resolve this.

> 
> This is likely the simplest solution.
> 
> 
> The alternative is to accept the coupling. But then one has to make clear on how to handle SSRC collisions. This seems problematic as the coupled session is a multicast session. A multicast session has bigger difficulties with changes due to its number of receivers and also that if one forces an active sender to change it may causes widespread disruption.
> 
> I also observe that it is not clear what SSRC space the Port Mapping request and response messages uses. Does this need to be synchronized with any other space or can it be its own unique one. A unique one is probably easiest. I don't see any direct coupling to the other sessions.
> 
> <Tom: has its own SSRC space....  See also the figure that shows the three sessions decoupled by means of the dashed lines..>

Ok, it is not obvious. And I have to say that this session is far from a
common RTP session. It is only RTCP messages, and also in fact not sent
using normal timer rules. Only when needed, which might actually need to
be pointed out also.

Cheers

Magnus Westerlund

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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Roni Even" <Even.roni@huawei.com>, "Robert Sparks" <rjsparks@nostrum.com>, "Stephan Wenger" <stewe@stewe.org>
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Subject: Re: [AVT] proposed AVT WG split
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I guess you missed my point. Dependencies do cause delay and RAMS is =
only now being delayed due to the drafts you mentioned. I was not =
referring to those delays in my email below.

-acbegen

> -----Original Message-----
> From: Roni Even [mailto:Even.roni@huawei.com]
> Sent: Friday, January 14, 2011 4:31 AM
> To: Ali C. Begen (abegen); 'Robert Sparks'; 'Stephan Wenger'
> Cc: 'IETF AVT WG'; 'IESG IESG'
> Subject: RE: [AVT] proposed AVT WG split
>=20
> Hi Ali,
> I think that some work can be spawned as a separate group using the =
dispatch
> WG to do so.
> As for the time it took I suggest you look at the port mapping draft, =
when
> it started and how the discussion led to changes even late in the =
process.
> This draft and the cname and ssrc ports were the ones delaying the =
RAMS work
> and not the main body
> Roni Even
>=20
> > -----Original Message-----
> > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf =
Of
> > Ali C. Begen (abegen)
> > Sent: Friday, January 14, 2011 4:20 AM
> > To: Robert Sparks; Stephan Wenger
> > Cc: IETF AVT WG; IESG IESG
> > Subject: Re: [AVT] proposed AVT WG split
> >
> > I do believe that some re-organization along the lines proposed by =
the
> > ADs might help with the unnecessarily long delays for important =
specs
> > like RAMS.
> >
> > If you ask my opinion, RAMS draft was finished rather slowly: my
> > personal target was to finish it in a year. While we had urgency =
since
> > other organizations were waiting for us, starting the wglc process =
and
> > completing it took almost a year. We did more than 20 revisions btw.
> > So, editors were not the blocking factor here.
> >
> > Yet, many avt-seasoned people told me that 2 years was actually =
quite a
> > big success for a spec like RAMS. They are probably right, but often =
a
> > late spec is almost as good as no spec, if not worse, especially if
> > another organization does something that would break something like
> > RTP. It is often impossible to clean up the mess once the damage is
> > done.
> >
> > I don't think the process can get much slower if some form of =
splitting
> > happens. I am sure you know this better than I do but many other
> > organizations are afraid to bring their work to the IETF in general =
or
> > seek approval because they are concerned about the delays. If AVT =
wants
> > to stay relevant in related standards, I think we need to address =
this
> > problem.
> >
> > -acbegen
> >
> > > -----Original Message-----
> > > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf =
Of
> > Robert Sparks
> > > Sent: Thursday, January 13, 2011 10:28 PM
> > > To: Stephan Wenger
> > > Cc: IETF AVT WG; IESG IESG
> > > Subject: Re: [AVT] proposed AVT WG split
> > >
> > > Stephan -
> > >
> > > As I noted in response to your comments in Maastricht, I disagree.
> > >
> > > There are separable streams of work in AVT - it is really multiple
> > working groups in one shell at the moment.
> > >
> > > While there are key people currently in AVT that would overlap =
these
> > groups, they are not the majority of the contributors to
> > > the different types of activities.
> > > The intended positive effect of this explicit division is to allow
> > these different streams to better parallelize, on list between
> > > meetings, not just during the IETF
> > > in-person meeting time.
> > >
> > > We'd like to try to channel these streams separately for a while.
> > That will require some additional coordination work from
> > > the area directors and chairs, but
> > > it will bring significant benefits, especially in being able to
> > better prioritize work items within the streams.  I realize that for
> > > folks like you, who will actively participate
> > > in all of the efforts, the benefits are more limited.
> > >
> > > I have had several participants express frustration, for example,
> > with the way the RAMs work eclipsed basic architectural
> > > work in AVT.
> > > While that could be  managed in place  (we could have only one
> > working group per area in the extreme), it has shown itself
> > > to be difficult
> > > with the set of things this group is working on.
> > >
> > > I also expect  the separation will make it easier to evaluate when
> > the energy for a particular effort (such as the XR metrics
> > > you point to below) exists,
> > > and when that level of energy changes.
> > >
> > > Yes, conflicts between these working groups when they do meet in
> > person will have to be managed carefully, but we have
> > > mechanisms for that. As Keith noted
> > > separately, we do not anticipate this significantly increasing the
> > number-of-minutes footprint on any given in-person IETF
> > > meeting. Also, your comments assume all
> > > four of these groups would have a session at each in-person =
meeting.
> > I don't expect that to be the case.
> > >
> > >
> > > RjS
> > >
> > >
> > > On Jan 12, 2011, at 6:11 PM, Stephan Wenger wrote:
> > >
> > >
> > > 	Hi IESG,
> > >
> > > 	As already communicated in Maastricht, I continue to believe that
> > this step is a bad idea.  I didn't attend Beijing, and
> > > don't know whether there was any discussion about the subject.  I
> > didn't see anything mentioned in the AVT minutes, though,
> > > nor did I see any discussion since Beijing on ietf@ietf.  If there
> > was, some of the points below may be redundant, and I
> > > apologize in advance.
> > >
> > > 	Essentially, you guys are suggesting to split one "eternal" WG
> > into four.  I'm writing "eternal", because I don't see that
> > > any of the WGs can fulfill its charter before other work items =
will
> > likely get chartered with a similar scope, and in the same
> > > WG.  For RTP core, RTP extensions, and RTP payloads that is =
probably
> > self-evident.  For the XR stuff: metrics pop up all the
> > > time, like payload formats.  Metrics are also known to make people
> > quick bucks on patent licensing.  There will be more
> > > metric proposals.  If the RTCP-XR metrics ever get momentum in the
> > marketplace, that is.  Somehow, I have my doubts.  But
> > > that's a different discussion.
> > >
> > > 	The positive effects of the split could be
> > > 	-chair workload is split among more people, and
> > > 	-there are more meeting seconds for RTP related work at any given
> > IETF meeting.
> > >
> > > 	Assuming unlimited resources, both are good things.
> > Unfortunately, resources are not unlimited.
> > >
> > > 	The negative effects are IMO:
> > > 	-the key knowledgeable people in AVT ideally would have to sit in
> > all four meetings so that all four groups can tap
> > > their experience.  But they may not want to do so, because it =
leaves
> > less time for cross-area involvement, coordination, and
> > > draft-writing outside the AVT sessions, and whatnot.  Also, the
> > burden for remote participants in far-away time zones goes
> > > up.  In other words, the resource meeting-seconds for those people =
is
> > not unlimited, and they will have to prioritize
> > > differently.    The result could be that what I called "key
> > knowledgable people" would perhaps not attend the mere RTCP-XR
> > > or Payload session; and the guys there may be wasting their time
> > doing work that does not make sense architecturally, just to
> > > see the hammer coming down at IETF last call.
> > > 	-more people get chair positions and are burdened with
> > administrative stuff -> less time to do real work
> > > 	-WG scheduling gets more difficult
> > > 	-AD load potentially goes up (someone needs to coordinate between
> > those groups, and the AD would seem to me the
> > > natural choice)
> > > 	-there is no reason to believe that the overall manpower
> > availability goes up through this change, so all the
> > > inefficiencies introduced go straight against the available =
cycles.
> > >
> > > 	Some of these issues can be solved by joint WG meetings and/or
> > careful chair selection.  However, both options have
> > > drawbacks as well.  A joint session would be counter-productive to
> > the allocation of more meeting minutes.  And when you
> > > have, say, 8 chairs for four WG, all of which know the RTP
> > architecture well enough, then you have pretty much exhausted, if
> > > not overdrawn already, the pool of "key knowledgable people", and
> > burdened them with (in many cases: additional) chair
> > > duties.
> > >
> > > 	No matter how I see it, splitting the group is not a good idea.
> > >
> > > 	Other options to alleviate problems could come to mind:
> > >
> > > 	If the problem to be solved is chair-workload: add more chairs
> > (but not 6 of them).
> > > 	If the problem to be solved is meeting-second shortages: have
> > interim meetings, conf calls, etc.  In other words,
> > > stretch out the AVT meeting in the interval between IETF meetings.
> > Require a web-ex presentation of every draft to be
> > > discussed in the AVT session at least 4 weeks before the meeting,
> > with a two hour timeslot for Q/A.  Should I continue with
> > > this brainstorm?
> > >
> > > 	If the problem is something else, I would like to know.
> > >
> > > 	Thanks for your consideration, and apologies for the wordcount,
> > >
> > > 	Stephan
> > >
> > >
> > >
> > >
> >
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www.ietf.org/mailman/listinfo/avt
> >
> > __________ Information from ESET NOD32 Antivirus, version of virus
> > signature database 5785 (20110113) __________
> >
> > The message was checked by ESET NOD32 Antivirus.
> >
> > http://www.eset.com
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> >
> >
> >
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> > The message was checked by ESET NOD32 Antivirus.
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>=20
>=20
> __________ Information from ESET NOD32 Antivirus, version of virus =
signature
> database 5785 (20110113) __________
>=20
> The message was checked by ESET NOD32 Antivirus.
>=20
> http://www.eset.com
>=20


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Thread-Topic: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 1 SSRC spaces
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>, "IETF AVT WG" <avt@ietf.org>, <draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org>
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Subject: Re: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 1 SSRC spaces
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> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of =
Magnus Westerlund
> Sent: Friday, January 14, 2011 5:22 PM
> To: IETF AVT WG; =
draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
> Subject: [AVT] WGLC comments on =
draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 1 SSRC spaces
>=20
> Hi,
>=20
> This is one of several issues I still see remaining in the document. =
It
> has clearly improved. I am sorry that I become sick before Christmas =
and
> couldn't continue the discussion then.
>=20
> There is an issue with SSRC usage in this document related to the
> different RTP sessions. Please remember that the most fundamental part
> of the RTP session definition is that each RTP session has its own =
SSRC
> space.
>=20
> So there is one SSRC number space for the multicast RTP session. An =
SSRC
> from this space is used in the Token Verification Request to identify
> the requesting client. In the Token Verification response message this
> SSRC identifier is included in the Token response message. The problem

Are you referring to the port mapping request/response messages? In that =
request, client chooses an SSRC randomly. This exchange does not belong =
to either session (not necessarily) since the same token can be used in =
different unicast sessions.

In the token verification request message, the client uses whatever SSRC =
it needs to use for that particular unicast session.

> is that this message is conveyed in the unicast RTP session that has =
its
> own SSRC space. Thus we have potentially create a coupling between the
> two RTP sessions.

Nope, this is not correct.
=20
> As I see it this can be fixed in two ways. Either one explicitly notes
> that this RTCP Token Response message field carries a value that =
refers
> to the SSRC space in another RTP session then one currently are in. =
This
> is likely the simplest solution.

This is already mentioned in section 4.1 where we say the ssrc is chosen =
randomly.=20
=20
> The alternative is to accept the coupling. But then one has to make
> clear on how to handle SSRC collisions. This seems problematic as the
> coupled session is a multicast session. A multicast session has bigger
> difficulties with changes due to its number of receivers and also that
> if one forces an active sender to change it may causes widespread
> disruption.
>=20
> I also observe that it is not clear what SSRC space the Port Mapping
> request and response messages uses. Does this need to be synchronized
> with any other space or can it be its own unique one. A unique one is
> probably easiest. I don't see any direct coupling to the other =
sessions.

That SSRC is not used anywhere for a purpose, so that's why we decided =
to say "choose it randomly and forget about it".

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

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Thread-Topic: WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 2 Unicast RTP session Termination
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>, "IETF AVT WG" <avt@ietf.org>, <draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org>
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Subject: Re: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 2 Unicast RTP session Termination
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> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Friday, January 14, 2011 5:34 PM
> To: IETF AVT WG; =
draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
> Subject: WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: =
Part 2 Unicast RTP session Termination
>=20
> Hi,
>=20
> The second issue is getting back to the unicast RTP session =
termination.
>=20
> Having re-read this document it is very clear on when an Unicast RTP
> session is created. Section 3.2, bullet 4 makes this clear. However,
> there is no discussion on what the criterias are for when the session
> can be considered terminated.

Of course, this draft is about how to achieve port mapping and =
establish/control a unicast session. For termination, the 3550 rules =
apply as usual. It is because it is the unicast session that is the =
important here. This draft cannot say anything at all how the sessions =
established by using this draft should be terminated.
=20
> This has previously been discussed in my review comments for 08 =
version
> but we didn't conclude on the discussion due to me being unable to
> respond then.
>=20
> To follow up on one thing. Yes, if we had an architectural and usage
> requirements document for unicast sessions created in relation to
> multicast then that document would make sense to discuss this issue. =
As
> I see it now, that will not be created. This is the usage independent
> component so I think it makes sense to discuss it here, even if not
> perfectly fit. But I like to point out that creation and some other
> treatment things are discussed. So lets include how it is terminated.

I disagree. It is up to the unicast session how long it lasts or when it =
ends. E.g., if it is a retransmission session, it could be used for one =
rtx and then may never get used, eventually it will time out. Or, the =
client keeps asking for rtx and it will stay open. It has nothing to do =
with the token or port mapping.
=20
> The reason i think it is important to discuss is that the end-points
> need to know when the other side will consider it terminated and =
create
> a new session in its eyes rather than continue using the current one.

Server does not need to know anything other than usual RTP rules. When =
the session times out, it is done.=20

> Also, the other signaling mechanisms do have some way of terminating.
> SIP Dialog termination kills the media RTP sessions, they can also
> become terminated in SDP by re-invites. RTSP has its Teardown method.
> SAP relies on SDP t=3D attribute to tell when the session is valid.
>=20
> But for this solution it is unclear. That becomes a bit more =
problematic
> as this is a protocol that is intended for highly dynamic cases, like
> RAMS. So I would really like to know what the criteria are. Otherwise =
we
> can have clients sending RTCP to a server for a session it considers
> closed. It also become a point of resource handling. One should clean =
up
> what one create.
>=20
> I am fine making this timeout based, like 5 regular report interval =
(or
> TRR-int) without RTP activity. But I think we need to consider what it
> means if one peer sends BYE. Can you time out earlier then?

If there is a BYE in the unicast, of course, it will end without waiting =
for the 5xinterval timeout duration. I don't see why we need to say =
something about this on top of what is already in the text.

-acbegen

>=20
>=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
> ----------------------------------------------------------------------

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Thread-Topic: WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 3 Binding of C2->P4 RTCP to correct RTP session
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>, "IETF AVT WG" <avt@ietf.org>, <draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org>
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> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Friday, January 14, 2011 5:43 PM
> To: IETF AVT WG; =
draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
> Subject: WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: =
Part 3 Binding of C2->P4 RTCP to correct RTP
> session
>=20
> Hi,
>=20
> This third issue was also brought up before and also Tom Van Caenegem
> comment on it.
>=20
> The RTCP reports going from C2->P4 from a receiver is not obvious on =
how
> to correlate with the Unicast RTP session it is reporting on that are
> sent P3->C1.
>=20
> The issues that may arise are the following. First there might be a =
NAT
> with port and address dependent mapping between them. Thus, even if
> C2=3DC1, the externally seen address+port can be different between =
traffic
> going to P3 and P4. In addition we shall not forget that there might =
be
> multiple clients behind the same NAT that participate in the same
> multicast session. So several RTCP flow might go to P4 from this IP
> address.

Correct, there could exist multiple clients sending rtcp to p4.
=20
> I think some details on how to correlate this RTCP flow needs to be
> included in the document. What are the determining factors? SSRC can't
> be it unless we couple the multicast and unicast session. Which I =
think
> is a bad idea. CNAME is likely the only method that is directly
> applicable and would match from the C1->P3 request to the C2->P4 one.

Yes, via a CNAME. And since we now have a good CNAME guideline document, =
this should work just fine.

Report correlation is about the individual unicast sessions, but I agree =
it is a good idea to state this explicitly in this draft, too and also =
require the new CNAME guideline document so that cnames are unique.

> But this needs to be stated as a requirement.

+1.
-acbegen
=20
>=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
> ----------------------------------------------------------------------

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Thread-Topic: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 5 Page 11, minimal time recommendation for re-use
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>, "IETF AVT WG" <avt@ietf.org>, <draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org>
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Subject: Re: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 5 Page 11, minimal time recommendation for re-use
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> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of =
Magnus Westerlund
> Sent: Friday, January 14, 2011 6:01 PM
> To: IETF AVT WG; =
draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
> Subject: [AVT] WGLC comments on =
draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 5 Page 11, minimal =
time
> recommendation for re-use
>=20
> Hi,
>=20
> On page 11 last paragraph the following is written:
>=20
> 	After a sufficient
>    time, a previously used receive port could be used again.
>=20
> It doesn't define what a sufficiently time is. I would suggest that a
> value that is extremely likely to be sufficent would be to use one TCP
> Maximum Segment Lifetime which is 2 minutes. I think some =
recommendation
> is in order.

Sure, any reference we can use for this?

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

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Thread-Topic: WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 4 SDP attribute semantics
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>, "IETF AVT WG" <avt@ietf.org>, <draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org>
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Subject: Re: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 4 SDP attribute semantics
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> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Friday, January 14, 2011 5:57 PM
> To: IETF AVT WG; =
draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
> Subject: WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: =
Part 4 SDP attribute semantics
>=20
> Hi,
>=20
> I think the change for the SDP attributes "portmapping" and
> "portmapping-req" was an improvement. However, I think the semantics =
and
> usage of these SDP Attributes needs clarification.
>=20
> Lets start with "portmapping", currently it seems vague and not match
> the generalize requirements needed. The current semantics appears to =
be:
> Include in the multicast m=3D block which in relation there might be =
need
> for token verification request. For example if one looks at Section
> 4.3.1 where BYE messages related to RAMS bursts may need tokens. That =
is
> within the unicast session created if I understand correctly. Thus =
there
> is no indication in this media block. Only in the Multicast session

There is indication in the unicast block.

a=3Dportmapping-req:30000

> block. This appears to be rather fuzzy and unclear.
>=20
> I would like to resuggest the semantics I was considering before:
>=20
> The "portmapping" attribute SHALL be included in any SDP media block
> (m=3D) that define an RTP session that require the usage of token
> verification requests for one or more RTCP message.

We discussed this at length in the call and the group decided to propose =
what we have in the draft now. Your proposal, repeating the same =
attribute in two blocks, is actually worse, and even more confusing.
=20
> Then I frankly have to say that I don't grasp what the semantics are
> related to the placement of the "portmapping-req" attribute. Yes, it
> conveys the port and possibly address for where to send the Mapping
> request. But what is the meaning of its placement on a particular =
media
> block? Why is it placed in unicast blocks? Even if no Token is =
required
> in this RTP session?

I don=92t know what else to add on top of what I have written before.=20
=20
> To me it appears that this attribute would be better served by =
becoming
> an Session + possibly override in media blocks. At session level it
> would server as the port mapping port and possibly address, unless =
c=3D at
> session level is unicast to the mapping server, that is used for all
> tokens used in the RTP sessions indicated by the "portmapping"
> attribute. Then it may be included at media level in an media block
> containing a=3Dportmapping to override this port and address within =
that
> particular media block.

We decided to have both attributes media-level only. Session-level =
attributes created a lot of corner cases and confusion, some of which =
was raised by Roni.=20

Overall, I think the current SDP stuff works just fine.

-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
> ----------------------------------------------------------------------

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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>, "IETF AVT WG" <avt@ietf.org>, <draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org>
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Subject: Re: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 6: Resend behavior
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> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of =
Magnus Westerlund
> Sent: Friday, January 14, 2011 6:09 PM
> To: IETF AVT WG; =
draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
> Subject: [AVT] WGLC comments on =
draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 6: Resend behavior
>=20
> Hi,
>=20
> In Section 4, there is some statement about resending. I want to come =
to
> some conclusion on this topic that was brought up before.
>=20
> The basic issue is the question on how one detects packet loss for =
RTCP.
> There is no other mechanism than detecting lack of response / action
> from the corresponding peer.

Indeed and that is what we do.
=20
> The issue I have is that the RTT + server delay in responding can =
easily
> be many times the minimal sending interval that RTCP configure to =
serve

Hmm, why unless you have an awfully slow server? If the server is really =
that slow, then I guess the system has more major issues than just =
losing the token request messages.

> even 1 mbit/s stream. Thus the formulation in the draft:
>=20
> "When a client sends a
>    Port Mapping Request or Token Verification Request message but it
>    does not receive a response back from the server (either a Port
>    Mapping Response or Token Verification Failure message), it MAY
>    resend its request by following the timer rules defined for RTCP
>    feedback messages in Section 3.5 of [RFC4585] as a good practice."
>=20
> Might not be sufficient. Because the issue is how one detects the =
packet
> loss and how quickly the packet is considered lost.

Well, we cannot say "wait X seconds before you ask again" unless you =
wanna propose a value of X.

I understand this is not perfect but I don=92t think there exists =
anything better than the rules we have in place. Re-using them is easy =
for implementation as well.
=20
BTW, the same problem exists with NACKs or other RTCP packets that can =
actually create a lot of traffic. Ours is merely a corner case IMO.

-acbegen

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

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To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>, "IETF AVT WG" <avt@ietf.org>, <draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org>
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Subject: Re: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 7 Persistent Errors
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> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Friday, January 14, 2011 6:14 PM
> To: IETF AVT WG; =
draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
> Subject: WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: =
Part 7 Persistent Errors
>=20
> Hi,
>=20
> This is a continuation of the issue 15A from the 08 review.
>=20
> The issue is when a client should consider an error to be persistent, =
or
> at least not short time temporarily or resolvable. If a client =
continues
> to get Mapping responses with errors after having tried several times.
> Maybe it should do exponential back-off. Maybe it should update its
> session description. Similar for Verification Requests.
>=20
> Some recommendation about such behavior I think has clear benefit and
> reduced uncertainty in behavior by clients.

This sounds like a good suggestion. How about adding the following to =
the end of section 6:

If a client receives a Port Mapping Response message with an invalid =
Token (i.e., the relative expiration time is set to zero) two or more =
times for a particular Port Mapping Request message, or the client =
receives a Token Verification Failure message two or more times for the =
same Token Verification Request message, the client SHOULD do the =
following:
1- Check whether the session description has been updated or not. If it =
was updated, act according to the new session description.
2- Exponentially back-off for the third and subsequent attempts.

Exponential back-off does not apply when the client sends a Port Mapping =
Request or Token Verification Request message to a new address and/or =
port.

Let me know if this text is good or you wanna modify it.

-acbegen
=20
> This was my last issue I have found in this review. A clear =
improvement
> from the previous review of the document.
>=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 gwz@net-zen.net  Fri Jan 14 20:40:36 2011
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Subject: Re: [AVT] Comment on Section 5	ofdraft-ietf-avt-ports-for-ucast-mcast-rtp-09.txt
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Magnus Westerlund [mailto:magnus.westerlund@ericsson.com] writes:

> Hi,
> 
> I would say that adding gives you nothing, even if detected. They you
> get token failure, and anyway request a new token. Thus detecting it or
> not the server will anyway eat the increased load.
> 
> I don't see that one can mitigate this attack without integrity checking
> the token response message

That would depend upon your definition of "mitigate", I suppose.  If you
take it to mean "fully & completely foil" the attack then you are correct,
but if you take the word at its more usual meaning of "lessen or to try to
lessen the seriousness or extent of" the attack, then the attack can be
mitigated (if detected) at little cost.  Note that in my description I
didn't assume perfect behavior on the part of the attacker, just some
consistency, while Ali's reworking does assume that the attacker behaves
perfectly & is in the path for every packet; I would say that Ali's version
is not only much harder to defeat but also much more difficult to pull off.
"Perfect security or none at all" is a dangerous false dichotomy...

...


From Even.roni@huawei.com  Sat Jan 15 05:05:40 2011
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Date: Sat, 15 Jan 2011 15:08:04 +0200
From: Roni Even <Even.roni@huawei.com>
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Subject: Re: [AVT] proposed AVT WG split
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Magnus,
I understand your position but what you stated is reflecting my concern
about who will review payloads. You know that most payload drafts =
authors
are not IETF participants and they may attend only where their documents =
are
in a review stages to progress the work and afterwards they dropout
Roni

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
> Magnus Westerlund
> Sent: Friday, January 14, 2011 11:27 AM
> To: Stephan Wenger
> Cc: Cullen Jennings; iesg@ietf.org; avt@ietf.org
> Subject: Re: [AVT] proposed AVT WG split
>=20
> Stephan Wenger skrev 2011-01-14 02:18:
> >>
> >>
> >>>  But they may not want to do so, because it leaves less time for
> >>> cross-area involvement, coordination, and draft-writing outside =
the
> AVT
> >>> sessions, and whatnot.  Also, the burden for remote participants =
in
> >>> far-away time zones goes up.  In other words, the resource
> >>> meeting-seconds for those people is not unlimited, and they will
> have to
> >>> prioritize differently.    The result could be that what I called
> "key
> >>> knowledgable people" would perhaps not attend the mere RTCP-XR or
> >>> Payload session; and the guys there may be wasting their time =
doing
> work
> >>> that does not make sense architecturally, just to see the hammer
> coming
> >>> down at IETF last call.
> >>> -more people get chair positions and are burdened with
> administrative
> >>> stuff -> less time to do real work
> >>
> >> Given the current chair bottlenecks, and the fact we have people
> that
> >> would like to do this, I view the bunch more chairs a pro not a con
> >
> > Really?  At almost every plenary I hear folks whining about how
> difficult
> > it is to get *qualified* chairs.  (Some would say that AVT, in its
> present
> > incarnation, clearly shows that difficulty)
>=20
> In this regard I do think splitting it does help. I will publicly =
state
> that I have volunteered to be a candidate to chair AVTCORE and AVTEXT.
> For me this is easier to take on as I at least don't need to handle
> also
> the documents of PAYLOAD and XR. I do still remember my 3 years as AVT
> chair. When we for a number of meeting had around 600000 words of AVT
> documents. I would be much more hesitant to take on the non-split AVT.
>=20
>=20
> In general I think this split is clearly worth trying out. We might
> realize down the road that it is over splitting things, and that some
> groups can merge or simply close. But lets gather some experience on
> this.
>=20
> Yes, it will have downsides, it also will have positive effects. Lets
> try this somewhat different trade-off.
>=20
> I do think smaller groups with less content might allow for a more
> focused work in the groups. But it will require the chairs to use the
> smaller scope of each group to speed up things. I also agree that when
> there is contention on how to do things, things take longer time. And
> that needs to be taken into account, and for example do more virtual
> interim meetings etc to give the subject more focus.
>=20
> In conclusion, I think this proposal is well worth trying out. And if
> it
> becomes misery and pain with no gain, then the AD can change things
> again.
>=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
> ----------------------------------------------------------------------
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
>=20
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> signature database 5785 (20110113) __________
>=20
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>=20
> http://www.eset.com
>=20
>=20
>=20
>=20
> __________ Information from ESET NOD32 Antivirus, version of virus
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>=20
> The message was checked by ESET NOD32 Antivirus.
>=20
> http://www.eset.com
>=20
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From gonzalo.camarillo@ericsson.com  Mon Jan 17 01:02:49 2011
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Subject: Re: [AVT] proposed AVT WG split
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Hi,

> I would like to point out that the AVT work was split between the two Ads so
> maybe we need a third AD to help with the work.

yes, I have been taking care of a few AVT documents... but AD load is
not the main reason for this split. I think we have already discussed
all the main arguments and counter arguments before and again in this
thread. We probably need running code to understand all the actual prons
and cons of the split... and we will only get it by actually trying this
out.

Cheers,

Gonzalo


From magnus.westerlund@ericsson.com  Mon Jan 17 02:17:16 2011
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Subject: Re: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 7 Persistent Errors
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Ali C. Begen (abegen) skrev 2011-01-14 19:34:
> 
> 
>> -----Original Message-----
>> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
>> Sent: Friday, January 14, 2011 6:14 PM
>> To: IETF AVT WG; draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
>> Subject: WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 7 Persistent Errors
>>
>> Hi,
>>
>> This is a continuation of the issue 15A from the 08 review.
>>
>> The issue is when a client should consider an error to be persistent, or
>> at least not short time temporarily or resolvable. If a client continues
>> to get Mapping responses with errors after having tried several times.
>> Maybe it should do exponential back-off. Maybe it should update its
>> session description. Similar for Verification Requests.
>>
>> Some recommendation about such behavior I think has clear benefit and
>> reduced uncertainty in behavior by clients.
> 
> This sounds like a good suggestion. How about adding the following to the end of section 6:
> 
> If a client receives a Port Mapping Response message with an invalid Token (i.e., the relative expiration time is set to zero) two or more times for a particular Port Mapping Request message, or the client receives a Token Verification Failure message two or more times for the same Token Verification Request message, the client SHOULD do the following:
> 1- Check whether the session description has been updated or not. If it was updated, act according to the new session description.
> 2- Exponentially back-off for the third and subsequent attempts.
> 
> Exponential back-off does not apply when the client sends a Port Mapping Request or Token Verification Request message to a new address and/or port.
> 
> Let me know if this text is good or you wanna modify it.

Good enough I think.

Cheers

Magnus Westerlund

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

From magnus.westerlund@ericsson.com  Mon Jan 17 02:28:06 2011
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Subject: Re: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 6: Resend behavior
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Ali C. Begen (abegen) skrev 2011-01-14 19:33:
> 
> 
>> -----Original Message-----
>> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Magnus Westerlund
>> Sent: Friday, January 14, 2011 6:09 PM
>> To: IETF AVT WG; draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
>> Subject: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 6: Resend behavior
>>
>> Hi,
>>
>> In Section 4, there is some statement about resending. I want to come to
>> some conclusion on this topic that was brought up before.
>>
>> The basic issue is the question on how one detects packet loss for RTCP.
>> There is no other mechanism than detecting lack of response / action
>> from the corresponding peer.
> 
> Indeed and that is what we do.
>  
>> The issue I have is that the RTT + server delay in responding can easily
>> be many times the minimal sending interval that RTCP configure to serve
> 
> Hmm, why unless you have an awfully slow server? If the server is really that slow, then I guess the system has more major issues than just losing the token request messages.

The problem isn't really in that. The issue is that the minimum interval
period for sending RTCP can be configured to not exist, "immediate"
mode. Also even in early mode AVPF can you configure an RTCP bandwidth
that allows you to send packets very often and rapidly.


> 
>> even 1 mbit/s stream. Thus the formulation in the draft:
>>
>> "When a client sends a
>>    Port Mapping Request or Token Verification Request message but it
>>    does not receive a response back from the server (either a Port
>>    Mapping Response or Token Verification Failure message), it MAY
>>    resend its request by following the timer rules defined for RTCP
>>    feedback messages in Section 3.5 of [RFC4585] as a good practice."
>>
>> Might not be sufficient. Because the issue is how one detects the packet
>> loss and how quickly the packet is considered lost.
> 
> Well, we cannot say "wait X seconds before you ask again" unless you wanna propose a value of X.
> 

The algorithm I have in mind would be 1.5*RTT to the server in question
+ Y server processing delay. I think Y can be fairly small a few ms,
like 5 ms. Unless you know that the processing delay is going to be
bigger. Then add exponential backoff for detecting additional
consecutive packet losses.



If you don't know RTT, then at least 100 ms in a managed network
situation, otherwise I would go with TCP's 500 ms as initial backoff.
And for the initial mapping request, using 500 ms may not be a big issue.

However, it is problematic to measure the RTT using RTCP in this
setting. This as the client can't include an SR that the server can
respond to. One would need to use XR's extensions for RTT measurement to
achieve that.

> I understand this is not perfect but I don’t think there exists anything better than the rules we have in place. Re-using them is easy for implementation as well.

The issue is that with RTCP bandwidth values configured differently,
then you can have a lot of spurious retransmissions.

>  
> BTW, the same problem exists with NACKs or other RTCP packets that can actually create a lot of traffic. Ours is merely a corner case IMO.
> 

Agreed, and this is probably something that a future update should
strongly consider to discuss.


Cheers

Magnus Westerlund

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Subject: Re: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 1 SSRC spaces
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Ali C. Begen (abegen) skrev 2011-01-14 19:32:
> 
> 
>> -----Original Message-----
>> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Magnus Westerlund
>> Sent: Friday, January 14, 2011 5:22 PM
>> To: IETF AVT WG; draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
>> Subject: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 1 SSRC spaces
>>
>> Hi,
>>
>> This is one of several issues I still see remaining in the document. It
>> has clearly improved. I am sorry that I become sick before Christmas and
>> couldn't continue the discussion then.
>>
>> There is an issue with SSRC usage in this document related to the
>> different RTP sessions. Please remember that the most fundamental part
>> of the RTP session definition is that each RTP session has its own SSRC
>> space.
>>
>> So there is one SSRC number space for the multicast RTP session. An SSRC
>> from this space is used in the Token Verification Request to identify
>> the requesting client. In the Token Verification response message this
>> SSRC identifier is included in the Token response message. The problem
> 
> Are you referring to the port mapping request/response messages? In that request, client chooses an SSRC randomly. This exchange does not belong to either session (not necessarily) since the same token can be used in different unicast sessions.
> 
> In the token verification request message, the client uses whatever SSRC it needs to use for that particular unicast session.
> 
>> is that this message is conveyed in the unicast RTP session that has its
>> own SSRC space. Thus we have potentially create a coupling between the
>> two RTP sessions.
> 
> Nope, this is not correct.

Lets try again to describe the above paragraph, as we clearly
missunderstanding each other.

The client has an SSRC that is uses for regular RTCP feedback to the
Retransmission server. If is want to send a NACK, then it include a
Token Verification Request. In that message it include the SSRC of the
client. This SSRC comes from the Multicast RTP sessions SSRC space.

Then the Retransmission server may send an RTCP message back to the
client in a form of the Token Verification Failure message. This message
include the SSRC of the requesting client. As this Token Verification
Failure message is sent in the RS -> Client leg of the created unicast
session, but is in reality a response to the message in the multicast
RTP session. This creates a case where it is uncertain which SSRC space
the SSRCs in the failure message belong to. Based on network selectors,
e.g. UDP ports this message is in the unicast space. But semantically it
belongs to the Multicast channel.

Please make it clear in the description in 4.4 what space the SSRC
values belongs to.



>  
>> As I see it this can be fixed in two ways. Either one explicitly notes
>> that this RTCP Token Response message field carries a value that refers
>> to the SSRC space in another RTP session then one currently are in. This
>> is likely the simplest solution.
> 
> This is already mentioned in section 4.1 where we say the ssrc is chosen randomly. 
>  
>> The alternative is to accept the coupling. But then one has to make
>> clear on how to handle SSRC collisions. This seems problematic as the
>> coupled session is a multicast session. A multicast session has bigger
>> difficulties with changes due to its number of receivers and also that
>> if one forces an active sender to change it may causes widespread
>> disruption.
>>
>> I also observe that it is not clear what SSRC space the Port Mapping
>> request and response messages uses. Does this need to be synchronized
>> with any other space or can it be its own unique one. A unique one is
>> probably easiest. I don't see any direct coupling to the other sessions.
> 
> That SSRC is not used anywhere for a purpose, so that's why we decided to say "choose it randomly and forget about it".

Yes, for mapping message that is fine. The thing I can complain about is
why the SSRC fields aren't there own bullets and instead hidden in the
pre-facing text to the figures?

-- 

Magnus Westerlund

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Subject: Re: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 2 Unicast RTP session Termination
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Ali C. Begen (abegen) skrev 2011-01-14 19:33:
> 
> 
>> -----Original Message-----
>> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
>> Sent: Friday, January 14, 2011 5:34 PM
>> To: IETF AVT WG; draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
>> Subject: WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 2 Unicast RTP session Termination
>>
>> Hi,
>>
>> The second issue is getting back to the unicast RTP session termination.
>>
>> Having re-read this document it is very clear on when an Unicast RTP
>> session is created. Section 3.2, bullet 4 makes this clear. However,
>> there is no discussion on what the criterias are for when the session
>> can be considered terminated.
> 
> Of course, this draft is about how to achieve port mapping and establish/control a unicast session. For termination, the 3550 rules apply as usual. It is because it is the unicast session that is the important here. This draft cannot say anything at all how the sessions established by using this draft should be terminated.

The issue, I have is that you say RFC 3550 rules. I don't see how RFC
3550 applies to RTP session termination. Because that happening through
external mechanism. RFC 3550 discusses when to consider an SSRC timed
out. But that is not the same as terminating the session.

It might be that you need to tell any using specification to declare how
a unicast session is terminated. But currently this is a gapping whole
in the specification.

Based on the below comments it appears that you have decided that RTP
session termination is when all external SSRCs has timed out. But, I
wonder if the servers really are fine with that. If I as a client
continue to send RTCP RR to a server in a RAMS scenario for the unicast
session, then the server is not allowed to terminate the session? I
think they will consider the session terminated when the burst is
completed.

For NACK I think one can take two approaches. One more aggressive, but
both appear to work.

A. Lazy, server and client consider the session terminated when the far
side SSRC times out completely. This will not happen as long as both
end-points sees each other RTCP RR even if no data traffic is present.

B. More aggressive: The RTP session is timed out when no RTP data
traffic has been sent in the RTP session for 2 or 5 regular RTCP
reporting intervals.

Once more my summary of this topic is: This is not defined nor obvious
as this is an issue that has not existed prior to using RTP/RTCP for
creating new RTP sessions.


Cheers

Magnus Westerlund

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Subject: Re: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 3 Binding of C2->P4 RTCP to correct RTP session
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Ali C. Begen (abegen) skrev 2011-01-14 19:33:
> 
> 
>> -----Original Message-----
>> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
>> Sent: Friday, January 14, 2011 5:43 PM
>> To: IETF AVT WG; draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
>> Subject: WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 3 Binding of C2->P4 RTCP to correct RTP
>> session
>>
>> Hi,
>>
>> This third issue was also brought up before and also Tom Van Caenegem
>> comment on it.
>>
>> The RTCP reports going from C2->P4 from a receiver is not obvious on how
>> to correlate with the Unicast RTP session it is reporting on that are
>> sent P3->C1.
>>
>> The issues that may arise are the following. First there might be a NAT
>> with port and address dependent mapping between them. Thus, even if
>> C2=C1, the externally seen address+port can be different between traffic
>> going to P3 and P4. In addition we shall not forget that there might be
>> multiple clients behind the same NAT that participate in the same
>> multicast session. So several RTCP flow might go to P4 from this IP
>> address.
> 
> Correct, there could exist multiple clients sending rtcp to p4.
>  
>> I think some details on how to correlate this RTCP flow needs to be
>> included in the document. What are the determining factors? SSRC can't
>> be it unless we couple the multicast and unicast session. Which I think
>> is a bad idea. CNAME is likely the only method that is directly
>> applicable and would match from the C1->P3 request to the C2->P4 one.
> 
> Yes, via a CNAME. And since we now have a good CNAME guideline document, this should work just fine.
> 
> Report correlation is about the individual unicast sessions, but I agree it is a good idea to state this explicitly in this draft, too and also require the new CNAME guideline document so that cnames are unique.
> 
>> But this needs to be stated as a requirement.
> 
> +1.

Great, that seems to close on this issue.

-- 

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: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 4 SDP attribute semantics
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Ali C. Begen (abegen) skrev 2011-01-14 19:33:
> 
> 
>> -----Original Message-----
>> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
>> Sent: Friday, January 14, 2011 5:57 PM
>> To: IETF AVT WG; draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
>> Subject: WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 4 SDP attribute semantics
>>
>> Hi,
>>
>> I think the change for the SDP attributes "portmapping" and
>> "portmapping-req" was an improvement. However, I think the semantics and
>> usage of these SDP Attributes needs clarification.
>>
>> Lets start with "portmapping", currently it seems vague and not match
>> the generalize requirements needed. The current semantics appears to be:
>> Include in the multicast m= block which in relation there might be need
>> for token verification request. For example if one looks at Section
>> 4.3.1 where BYE messages related to RAMS bursts may need tokens. That is
>> within the unicast session created if I understand correctly. Thus there
>> is no indication in this media block. Only in the Multicast session
> 
> There is indication in the unicast block.
> 
> a=portmapping-req:30000
> 
>> block. This appears to be rather fuzzy and unclear.
>>
>> I would like to resuggest the semantics I was considering before:
>>
>> The "portmapping" attribute SHALL be included in any SDP media block
>> (m=) that define an RTP session that require the usage of token
>> verification requests for one or more RTCP message.
> 
> We discussed this at length in the call and the group decided to propose what we have in the draft now. Your proposal, repeating the same attribute in two blocks, is actually worse, and even more confusing.
>  

Ok, can you then write down the exact semantics of the portmapping-req
and portmapping attribute? Because to me their semantics aren't clear.
And without clear semantics this is going to cause confusion when used
the next time.


>> Then I frankly have to say that I don't grasp what the semantics are
>> related to the placement of the "portmapping-req" attribute. Yes, it
>> conveys the port and possibly address for where to send the Mapping
>> request. But what is the meaning of its placement on a particular media
>> block? Why is it placed in unicast blocks? Even if no Token is required
>> in this RTP session?
> 
> I don’t know what else to add on top of what I have written before. 
>  
>> To me it appears that this attribute would be better served by becoming
>> an Session + possibly override in media blocks. At session level it
>> would server as the port mapping port and possibly address, unless c= at
>> session level is unicast to the mapping server, that is used for all
>> tokens used in the RTP sessions indicated by the "portmapping"
>> attribute. Then it may be included at media level in an media block
>> containing a=portmapping to override this port and address within that
>> particular media block.
> 
> We decided to have both attributes media-level only. Session-level attributes created a lot of corner cases and confusion, some of which was raised by Roni. 

Fine, I can live with that.

> 
> Overall, I think the current SDP stuff works just fine.

I disagree, because in cases where you need token verification requests
also in the unicast client to server RTCP leg C2->P4, then I don't
understand how that is separated from when it is only needed in C1->P3.
This as I don't understand the semantics of the SDP attributes.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
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Färögatan 6                | Mobile +46 73 0949079
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Subject: Re: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 5 Page 11, minimal time recommendation for re-use
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Ali C. Begen (abegen) skrev 2011-01-14 19:33:
> 
> 
>> -----Original Message-----
>> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Magnus Westerlund
>> Sent: Friday, January 14, 2011 6:01 PM
>> To: IETF AVT WG; draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
>> Subject: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 5 Page 11, minimal time
>> recommendation for re-use
>>
>> Hi,
>>
>> On page 11 last paragraph the following is written:
>>
>> 	After a sufficient
>>    time, a previously used receive port could be used again.
>>
>> It doesn't define what a sufficiently time is. I would suggest that a
>> value that is extremely likely to be sufficent would be to use one TCP
>> Maximum Segment Lifetime which is 2 minutes. I think some recommendation
>> is in order.
> 
> Sure, any reference we can use for this?
> 

RFC 793 defines MSL to be 2 minutes for that spec. I think you should
specify the timer, and then informally reference the TCP spec.

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


-- 

Magnus Westerlund

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Multimedia Technologies, Ericsson Research EAB/TVM
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Subject: Re: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 2 Unicast RTP session Termination
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> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Monday, January 17, 2011 5:54 AM
> To: Ali C. Begen (abegen)
> Cc: IETF AVT WG; =
draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
> Subject: Re: WGLC comments on =
draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 2 Unicast RTP session =
Termination
>=20
> Ali C. Begen (abegen) skrev 2011-01-14 19:33:
> >
> >
> >> -----Original Message-----
> >> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> >> Sent: Friday, January 14, 2011 5:34 PM
> >> To: IETF AVT WG; =
draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
> >> Subject: WGLC comments on =
draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 2 Unicast RTP session =
Termination
> >>
> >> Hi,
> >>
> >> The second issue is getting back to the unicast RTP session =
termination.
> >>
> >> Having re-read this document it is very clear on when an Unicast =
RTP
> >> session is created. Section 3.2, bullet 4 makes this clear. =
However,
> >> there is no discussion on what the criterias are for when the =
session
> >> can be considered terminated.
> >
> > Of course, this draft is about how to achieve port mapping and =
establish/control a unicast session. For termination, the
> 3550 rules apply as usual. It is because it is the unicast session =
that is the important here. This draft cannot say anything at all
> how the sessions established by using this draft should be terminated.
>=20
> The issue, I have is that you say RFC 3550 rules. I don't see how RFC
> 3550 applies to RTP session termination. Because that happening =
through
> external mechanism. RFC 3550 discusses when to consider an SSRC timed
> out. But that is not the same as terminating the session.

As a good citizen, the client can send a BYE to end the unicast session. =
It is not just when SSRC times out.
=20
> It might be that you need to tell any using specification to declare =
how
> a unicast session is terminated. But currently this is a gapping whole
> in the specification.

Actually, we clearly specified how a RAMS session is terminated or when =
it ends. Repeating all that text here is not needed. So, maybe we should =
simply say the spec using the Token approach needs to discuss the =
session termination and ending sessions.=20
=20
> Based on the below comments it appears that you have decided that RTP
> session termination is when all external SSRCs has timed out. But, I
> wonder if the servers really are fine with that. If I as a client
> continue to send RTCP RR to a server in a RAMS scenario for the =
unicast
> session, then the server is not allowed to terminate the session? I

In RAMS, the client will continue sending RRs if and only if the client =
does retransmissions in that unicast session. If that is the case, of =
course, the server is fine since the session is still active.

If the client does not do any further retransmissions after the burst is =
completed, the RRs will not be sent so the session will end after the =
timeout.

> think they will consider the session terminated when the burst is
> completed.

Not necessarily. RAMS server is not just a burst server,  it is a =
retransmission server, too.
=20
> For NACK I think one can take two approaches. One more aggressive, but
> both appear to work.
>=20
> A. Lazy, server and client consider the session terminated when the =
far
> side SSRC times out completely. This will not happen as long as both
> end-points sees each other RTCP RR even if no data traffic is present.
>=20
> B. More aggressive: The RTP session is timed out when no RTP data
> traffic has been sent in the RTP session for 2 or 5 regular RTCP
> reporting intervals.
>=20
> Once more my summary of this topic is: This is not defined nor obvious
> as this is an issue that has not existed prior to using RTP/RTCP for
> creating new RTP sessions.

The difference is in creating the RTP session not ending it. Just =
because a session is dynamically created via RTCP does not mean that it =
must have a different termination method from the regularly established =
RTP sessions (3550). Once a session is up, it is up - no difference from =
any other regular RTP session. Following the KISS principle here makes =
sense IMO.

Cheers,
-acbegen
=20
>=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
> ----------------------------------------------------------------------

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Thread-Topic: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 1 SSRC spaces
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>
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Subject: Re: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 1 SSRC spaces
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> Lets try again to describe the above paragraph, as we clearly
> missunderstanding each other.
>=20
> The client has an SSRC that is uses for regular RTCP feedback to the
> Retransmission server. If is want to send a NACK, then it include a
> Token Verification Request. In that message it include the SSRC of the
> client. This SSRC comes from the Multicast RTP sessions SSRC space.

Right.
=20
> Then the Retransmission server may send an RTCP message back to the
> client in a form of the Token Verification Failure message. This =
message
> include the SSRC of the requesting client. As this Token Verification
> Failure message is sent in the RS -> Client leg of the created unicast

Yes, and the server simply gets this SSRC from the request message.=20

> session, but is in reality a response to the message in the multicast

Yes, maybe it was a NACK sent for the multicast session.

> RTP session. This creates a case where it is uncertain which SSRC =
space
> the SSRCs in the failure message belong to. Based on network =
selectors,
> e.g. UDP ports this message is in the unicast space. But semantically =
it
> belongs to the Multicast channel.
>=20
> Please make it clear in the description in 4.4 what space the SSRC
> values belongs to.

In the failure message:

SSRC of Packet Sender: This is server's SSRC, which equals the SSRC of =
the respective primary multicast stream.
SSRC of Requesting Client: This SSRC is directly copied from the Token =
Verification Request message (this text already exists in 4.4).
=20
Is the "packet sender SSRC" definition fine with you?
=20
> >> As I see it this can be fixed in two ways. Either one explicitly =
notes
> >> that this RTCP Token Response message field carries a value that =
refers
> >> to the SSRC space in another RTP session then one currently are in. =
This
> >> is likely the simplest solution.
> >
> > This is already mentioned in section 4.1 where we say the ssrc is =
chosen randomly.
> >
> >> The alternative is to accept the coupling. But then one has to make
> >> clear on how to handle SSRC collisions. This seems problematic as =
the
> >> coupled session is a multicast session. A multicast session has =
bigger
> >> difficulties with changes due to its number of receivers and also =
that
> >> if one forces an active sender to change it may causes widespread
> >> disruption.
> >>
> >> I also observe that it is not clear what SSRC space the Port =
Mapping
> >> request and response messages uses. Does this need to be =
synchronized
> >> with any other space or can it be its own unique one. A unique one =
is
> >> probably easiest. I don't see any direct coupling to the other =
sessions.
> >
> > That SSRC is not used anywhere for a purpose, so that's why we =
decided to say "choose it randomly and forget about it".
>=20
> Yes, for mapping message that is fine. The thing I can complain about =
is
> why the SSRC fields aren't there own bullets and instead hidden in the
> pre-facing text to the figures?

We can move them to the bullets. We probably did not so because, they =
are pretty much part of the baseline format.

-acbegen
=20
> --
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------

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Thread-Topic: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 6: Resend behavior
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>
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Subject: Re: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 6: Resend behavior
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OK, I see your concern, however, should not we still follow 4585 as it =
is and once it gets updated, it will be reflected here? If the immediate =
mode is chosen, the NACK/RAMS could result in many more spurious =
transmissions whereas the token stuff only results in one such =
transmission. So putting some extra rules here for the tokens only will =
not get us much.

Maybe, we can add some text to disallow the immediate mode?

-acbegen

> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Monday, January 17, 2011 5:31 AM
> To: Ali C. Begen (abegen)
> Cc: IETF AVT WG; =
draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
> Subject: Re: [AVT] WGLC comments on =
draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 6: Resend behavior
>=20
> Ali C. Begen (abegen) skrev 2011-01-14 19:33:
> >
> >
> >> -----Original Message-----
> >> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf =
Of Magnus Westerlund
> >> Sent: Friday, January 14, 2011 6:09 PM
> >> To: IETF AVT WG; =
draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
> >> Subject: [AVT] WGLC comments on =
draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 6: Resend behavior
> >>
> >> Hi,
> >>
> >> In Section 4, there is some statement about resending. I want to =
come to
> >> some conclusion on this topic that was brought up before.
> >>
> >> The basic issue is the question on how one detects packet loss for =
RTCP.
> >> There is no other mechanism than detecting lack of response / =
action
> >> from the corresponding peer.
> >
> > Indeed and that is what we do.
> >
> >> The issue I have is that the RTT + server delay in responding can =
easily
> >> be many times the minimal sending interval that RTCP configure to =
serve
> >
> > Hmm, why unless you have an awfully slow server? If the server is =
really that slow, then I guess the system has more major
> issues than just losing the token request messages.
>=20
> The problem isn't really in that. The issue is that the minimum =
interval
> period for sending RTCP can be configured to not exist, "immediate"
> mode. Also even in early mode AVPF can you configure an RTCP bandwidth
> that allows you to send packets very often and rapidly.
>=20
>=20
> >
> >> even 1 mbit/s stream. Thus the formulation in the draft:
> >>
> >> "When a client sends a
> >>    Port Mapping Request or Token Verification Request message but =
it
> >>    does not receive a response back from the server (either a Port
> >>    Mapping Response or Token Verification Failure message), it MAY
> >>    resend its request by following the timer rules defined for RTCP
> >>    feedback messages in Section 3.5 of [RFC4585] as a good =
practice."
> >>
> >> Might not be sufficient. Because the issue is how one detects the =
packet
> >> loss and how quickly the packet is considered lost.
> >
> > Well, we cannot say "wait X seconds before you ask again" unless you =
wanna propose a value of X.
> >
>=20
> The algorithm I have in mind would be 1.5*RTT to the server in =
question
> + Y server processing delay. I think Y can be fairly small a few ms,
> like 5 ms. Unless you know that the processing delay is going to be
> bigger. Then add exponential backoff for detecting additional
> consecutive packet losses.
>=20
>=20
>=20
> If you don't know RTT, then at least 100 ms in a managed network
> situation, otherwise I would go with TCP's 500 ms as initial backoff.
> And for the initial mapping request, using 500 ms may not be a big =
issue.
>=20
> However, it is problematic to measure the RTT using RTCP in this
> setting. This as the client can't include an SR that the server can
> respond to. One would need to use XR's extensions for RTT measurement =
to
> achieve that.
>=20
> > I understand this is not perfect but I don=92t think there exists =
anything better than the rules we have in place. Re-using them
> is easy for implementation as well.
>=20
> The issue is that with RTCP bandwidth values configured differently,
> then you can have a lot of spurious retransmissions.
>=20
> >
> > BTW, the same problem exists with NACKs or other RTCP packets that =
can actually create a lot of traffic. Ours is merely a
> corner case IMO.
> >
>=20
> Agreed, and this is probably something that a future update should
> strongly consider to discuss.
>=20
>=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
> ----------------------------------------------------------------------

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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
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> Ok, can you then write down the exact semantics of the portmapping-req
> and portmapping attribute? Because to me their semantics aren't clear.
> And without clear semantics this is going to cause confusion when used
> the next time.

I am not sure what you are looking for in addition to what section 7 =
says.  they are used in pairs. portmapping is used in the multicast =
block and portmapping-req is used in the unicast block. Together they =
mean, certain rtcp messages will require the token. The portmapping-req =
will tell you where to get the token from. Any confusion or question =
here?
=20
> >> Then I frankly have to say that I don't grasp what the semantics =
are
> >> related to the placement of the "portmapping-req" attribute. Yes, =
it
> >> conveys the port and possibly address for where to send the Mapping
> >> request. But what is the meaning of its placement on a particular =
media
> >> block? Why is it placed in unicast blocks? Even if no Token is =
required
> >> in this RTP session?
> >
> > I don=92t know what else to add on top of what I have written =
before.
> >
> >> To me it appears that this attribute would be better served by =
becoming
> >> an Session + possibly override in media blocks. At session level it
> >> would server as the port mapping port and possibly address, unless =
c=3D at
> >> session level is unicast to the mapping server, that is used for =
all
> >> tokens used in the RTP sessions indicated by the "portmapping"
> >> attribute. Then it may be included at media level in an media block
> >> containing a=3Dportmapping to override this port and address within =
that
> >> particular media block.
> >
> > We decided to have both attributes media-level only. Session-level =
attributes created a lot of corner cases and confusion,
> some of which was raised by Roni.
>=20
> Fine, I can live with that.
>=20
> >
> > Overall, I think the current SDP stuff works just fine.
>=20
> I disagree, because in cases where you need token verification =
requests
> also in the unicast client to server RTCP leg C2->P4, then I don't

The only thing that travels from c2->p4 is the receiver reports for the =
unicast session. How would you send a token verification request message =
on that path? The verification request message goes from c1 to p3.

-acbegen

> understand how that is separated from when it is only needed in =
C1->P3.
> This as I don't understand the semantics of the SDP attributes.
>=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
> ----------------------------------------------------------------------

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Subject: Re: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 4 SDP attribute semantics
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Ali C. Begen (abegen) skrev 2011-01-17 14:58:
>> Ok, can you then write down the exact semantics of the portmapping-req
>> and portmapping attribute? Because to me their semantics aren't clear.
>> And without clear semantics this is going to cause confusion when used
>> the next time.
> 
> I am not sure what you are looking for in addition to what section 7 says.  they are used in pairs. portmapping is used in the multicast block and portmapping-req is used in the unicast block. Together they mean, certain rtcp messages will require the token. The portmapping-req will tell you where to get the token from. Any confusion or question here?

I think the confusion is that I believe it will be cases where Token
Verification Requests are going to be need to be sent on C2->P4. If that
is the case one needs semantics to say that token may also be included
in RTCP message related to this session.

In addition the semantics for the portmapping-req that now is mandated
to be on the declaration of the unicast RTP session. If we take your
argument there is no need for token verification in that session. Thus,
can you please explain why it is part of that session declaration?

In my world there could be token verification messages in the RTCP for
the unicast session. However, the declaration of where the Token server
should then either be in all RTP sessions that uses Token Verification
Messages, or at session level and applies to all "portmapping".



>  
>>>> Then I frankly have to say that I don't grasp what the semantics are
>>>> related to the placement of the "portmapping-req" attribute. Yes, it
>>>> conveys the port and possibly address for where to send the Mapping
>>>> request. But what is the meaning of its placement on a particular media
>>>> block? Why is it placed in unicast blocks? Even if no Token is required
>>>> in this RTP session?
>>>
>>> I don’t know what else to add on top of what I have written before.
>>>
>>>> To me it appears that this attribute would be better served by becoming
>>>> an Session + possibly override in media blocks. At session level it
>>>> would server as the port mapping port and possibly address, unless c= at
>>>> session level is unicast to the mapping server, that is used for all
>>>> tokens used in the RTP sessions indicated by the "portmapping"
>>>> attribute. Then it may be included at media level in an media block
>>>> containing a=portmapping to override this port and address within that
>>>> particular media block.
>>>
>>> We decided to have both attributes media-level only. Session-level attributes created a lot of corner cases and confusion,
>> some of which was raised by Roni.
>>
>> Fine, I can live with that.
>>
>>>
>>> Overall, I think the current SDP stuff works just fine.
>>
>> I disagree, because in cases where you need token verification requests
>> also in the unicast client to server RTCP leg C2->P4, then I don't
> 
> The only thing that travels from c2->p4 is the receiver reports for the unicast session. How would you send a token verification request message on that path? The verification request message goes from c1 to p3.

Here we seem to disagree. If one looks at the list in section 4.3.1, i
think that this does contain messages where you would like to attach
token, such as BYE or CCM messages.

It might be me taking the generalized look on this specification. Then I
do see a need to make sure that it can be used in any RTP session where
indicated.

If the intention to not have any session level attribute, then I don't
understand why we have two SDP attributes, then only portmapping-req is
needed. And I think it should be defined as having the meaning. Shall be
included in the SDP media blocks that define an RTP session that needs
Token Verification messages. Which means that portmapping-req are needed
in the multicast m= block, not the unicast.



Cheers

Magnus Westerlund

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Ali C. Begen (abegen) skrev 2011-01-17 14:24:
> OK, I see your concern, however, should not we still follow 4585 as it is and once it gets updated, it will be reflected here? If the immediate mode is chosen, the NACK/RAMS could result in many more spurious transmissions whereas the token stuff only results in one such transmission. So putting some extra rules here for the tokens only will not get us much.
> 
> Maybe, we can add some text to disallow the immediate mode?

You seem to be punting on this issue. I think Immediate mode needs to be
allowed, it makes all the sense for these type of applications in
certain environments. However, what is needed is a separation of the
"packet loss" / "failure for server to act on event" detection mechanism
and the RTCP transmission rate limiting.

I think despite our discussion of these things in CCM (RFC 5104) we did
fail to take all the AVPF cases into account. I have to say that 3.5.1.1
do rely on repeating FIR messages until you get an result. It does fail
to say in which mode and what type of transmission should be used for
the repetitions. This as one in early mode, this is either classified as
an event that are allowed to send early, or as a regular transmission.

There need to be some sanity checking in this, and going forward being
aware that what we specify can be seriously broken is not the best
thing. In addition RAMS can be fixed. And the NACK still doesn't have a
spec for this use case.


Cheers

Magnus Westerlund

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Subject: Re: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 2 Unicast RTP session Termination
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Ali C. Begen (abegen) skrev 2011-01-17 14:23:
> 
> 
>> -----Original Message-----
>> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
>> Sent: Monday, January 17, 2011 5:54 AM
>> To: Ali C. Begen (abegen)
>> Cc: IETF AVT WG; draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
>> Subject: Re: WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 2 Unicast RTP session Termination
>>
>> Ali C. Begen (abegen) skrev 2011-01-14 19:33:
>>>
>>>
>>>> -----Original Message-----
>>>> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
>>>> Sent: Friday, January 14, 2011 5:34 PM
>>>> To: IETF AVT WG; draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
>>>> Subject: WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 2 Unicast RTP session Termination
>>>>
>>>> Hi,
>>>>
>>>> The second issue is getting back to the unicast RTP session termination.
>>>>
>>>> Having re-read this document it is very clear on when an Unicast RTP
>>>> session is created. Section 3.2, bullet 4 makes this clear. However,
>>>> there is no discussion on what the criterias are for when the session
>>>> can be considered terminated.
>>>
>>> Of course, this draft is about how to achieve port mapping and establish/control a unicast session. For termination, the
>> 3550 rules apply as usual. It is because it is the unicast session that is the important here. This draft cannot say anything at all
>> how the sessions established by using this draft should be terminated.
>>
>> The issue, I have is that you say RFC 3550 rules. I don't see how RFC
>> 3550 applies to RTP session termination. Because that happening through
>> external mechanism. RFC 3550 discusses when to consider an SSRC timed
>> out. But that is not the same as terminating the session.
> 
> As a good citizen, the client can send a BYE to end the unicast session. It is not just when SSRC times out.
Sure, but it is the criteria that is the question.

>  
>> It might be that you need to tell any using specification to declare how
>> a unicast session is terminated. But currently this is a gapping whole
>> in the specification.
> 
> Actually, we clearly specified how a RAMS session is terminated or when it ends. Repeating all that text here is not needed. So, maybe we should simply say the spec using the Token approach needs to discuss the session termination and ending sessions. 

No, if that requirement is put on application of this, then I am mostly
happy.

>  
>> Based on the below comments it appears that you have decided that RTP
>> session termination is when all external SSRCs has timed out. But, I
>> wonder if the servers really are fine with that. If I as a client
>> continue to send RTCP RR to a server in a RAMS scenario for the unicast
>> session, then the server is not allowed to terminate the session? I
> 
> In RAMS, the client will continue sending RRs if and only if the client does retransmissions in that unicast session. If that is the case, of course, the server is fine since the session is still active.
> 
> If the client does not do any further retransmissions after the burst is completed, the RRs will not be sent so the session will end after the timeout.

But, basically you are saying that unless the client keeps the session
alive, the server will terminate it. Which gives non symmetric behavior
between the client and the retransmission server.

> 
>> think they will consider the session terminated when the burst is
>> completed.
> 
> Not necessarily. RAMS server is not just a burst server,  it is a retransmission server, too.



>  
>> For NACK I think one can take two approaches. One more aggressive, but
>> both appear to work.
>>
>> A. Lazy, server and client consider the session terminated when the far
>> side SSRC times out completely. This will not happen as long as both
>> end-points sees each other RTCP RR even if no data traffic is present.
>>
>> B. More aggressive: The RTP session is timed out when no RTP data
>> traffic has been sent in the RTP session for 2 or 5 regular RTCP
>> reporting intervals.
>>
>> Once more my summary of this topic is: This is not defined nor obvious
>> as this is an issue that has not existed prior to using RTP/RTCP for
>> creating new RTP sessions.
> 
> The difference is in creating the RTP session not ending it. Just because a session is dynamically created via RTCP does not mean that it must have a different termination method from the regularly established RTP sessions (3550). Once a session is up, it is up - no difference from any other regular RTP session. Following the KISS principle here makes sense IMO.

Well, you have above described a system for retransmissions where there
is a difference between the client and the server. And where there is
cases for when the session is considered terminated. That is not the
same as saying that it isn't terminated. I haven't argued that they must
be an explicit message. Only that the rules are made clear.

I am not certain that the restransmission case is well described here.
And this document is the generalized form of when the unicasat session
is created. I think describing this general case of termiantion has its
point. And with the general case I mean:

If the client stops sending RTCP regularly and the client SSRC times out
from the unicast RTP session, the server is allowed to terminate/close
the session. New message for creating it is needed. If the client sends
BYE in the unicast session, the session can also be considered
terminated (Needs Token verification or there is a DOS angle).

Cheers

Magnus Westerlund

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Subject: Re: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 1 SSRC spaces
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Ali C. Begen (abegen) skrev 2011-01-17 14:23:
>> Lets try again to describe the above paragraph, as we clearly
>> missunderstanding each other.
>>
>> The client has an SSRC that is uses for regular RTCP feedback to the
>> Retransmission server. If is want to send a NACK, then it include a
>> Token Verification Request. In that message it include the SSRC of the
>> client. This SSRC comes from the Multicast RTP sessions SSRC space.
> 
> Right.
>  
>> Then the Retransmission server may send an RTCP message back to the
>> client in a form of the Token Verification Failure message. This message
>> include the SSRC of the requesting client. As this Token Verification
>> Failure message is sent in the RS -> Client leg of the created unicast
> 
> Yes, and the server simply gets this SSRC from the request message. 
> 
>> session, but is in reality a response to the message in the multicast
> 
> Yes, maybe it was a NACK sent for the multicast session.
> 
>> RTP session. This creates a case where it is uncertain which SSRC space
>> the SSRCs in the failure message belong to. Based on network selectors,
>> e.g. UDP ports this message is in the unicast space. But semantically it
>> belongs to the Multicast channel.
>>
>> Please make it clear in the description in 4.4 what space the SSRC
>> values belongs to.
> 
> In the failure message:
> 
> SSRC of Packet Sender: This is server's SSRC, which equals the SSRC of the respective primary multicast stream.
> SSRC of Requesting Client: This SSRC is directly copied from the Token Verification Request message (this text already exists in 4.4).
>  
> Is the "packet sender SSRC" definition fine with you?

I think I would like to have extra warning: "Note, theis SSRC value are
from a different SSRC space than the one used in the Unicast RTP session."

>> Yes, for mapping message that is fine. The thing I can complain about is
>> why the SSRC fields aren't there own bullets and instead hidden in the
>> pre-facing text to the figures?
> 
> We can move them to the bullets. We probably did not so because, they are pretty much part of the baseline format.

I think it has its point when we do have some irregularities here in
some cases.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
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From csp@csperkins.org  Tue Jan 18 01:43:50 2011
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To: "Brandenburg, R. (Ray) van" <ray.vanbrandenburg@tno.nl>
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Subject: Re: [AVT] New version of RTCP for IDMS
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On 14 Jan 2011, at 16:15, Brandenburg, R. (Ray) van wrote:
> We have just submitted a new version of our internet draft describing =
an RTCP XR block for Inter-Destination Media Synchronization (IDMS).
>=20
> http://tools.ietf.org/html/draft-brandenburg-avt-rtcp-for-idms-03
...
> In the past there have been some concerns about whether RTCP XR blocks =
are the best solution for IDMS and if a new RTCP block wouldn't be =
better. Also, some questions were raised about the advantage of having =
both client->server and server->client communication with the same XR =
block. On these issues we feel it is important to understand that the =
proposed IDMS system is not a control protocol; client are not told how =
to synchronize their playback to others. The proposed XR block only =
informs clients of the slowest playback within their synchronization =
group, how they deal with this information, if at all, is up to =
themselves. In this way, IDMS is not that different from how RTCP is =
used to deal with synchronization between audio and video streams on the =
same terminal.


RTCP uses a special-purpose packet type (SR) to information receivers of =
the synchronisation parameters, separate to the reception reports.=20

RTCP XR is appropriate for receivers to report on their synchronisation =
state. If IDMS were a peer-to-peer protocol, with each receiver directly =
determining its playout based on the reported state from receivers, then =
this would be sufficient for controlling playout too. However, if there =
is a server-to-client control message that informs receivers of their =
synchronisation point, that should, I believe, follow the previous RTCP =
approach, and be coded as a separate packet type.

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




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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>
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Cc: draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org, IETF AVT WG <avt@ietf.org>
Subject: Re: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 1 SSRC spaces
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> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Monday, January 17, 2011 10:15 AM
> To: Ali C. Begen (abegen)
> Cc: IETF AVT WG; =
draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
> Subject: Re: [AVT] WGLC comments on =
draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 1 SSRC spaces
>=20
> Ali C. Begen (abegen) skrev 2011-01-17 14:23:
> >> Lets try again to describe the above paragraph, as we clearly
> >> missunderstanding each other.
> >>
> >> The client has an SSRC that is uses for regular RTCP feedback to =
the
> >> Retransmission server. If is want to send a NACK, then it include a
> >> Token Verification Request. In that message it include the SSRC of =
the
> >> client. This SSRC comes from the Multicast RTP sessions SSRC space.
> >
> > Right.
> >
> >> Then the Retransmission server may send an RTCP message back to the
> >> client in a form of the Token Verification Failure message. This =
message
> >> include the SSRC of the requesting client. As this Token =
Verification
> >> Failure message is sent in the RS -> Client leg of the created =
unicast
> >
> > Yes, and the server simply gets this SSRC from the request message.
> >
> >> session, but is in reality a response to the message in the =
multicast
> >
> > Yes, maybe it was a NACK sent for the multicast session.
> >
> >> RTP session. This creates a case where it is uncertain which SSRC =
space
> >> the SSRCs in the failure message belong to. Based on network =
selectors,
> >> e.g. UDP ports this message is in the unicast space. But =
semantically it
> >> belongs to the Multicast channel.
> >>
> >> Please make it clear in the description in 4.4 what space the SSRC
> >> values belongs to.
> >
> > In the failure message:
> >
> > SSRC of Packet Sender: This is server's SSRC, which equals the SSRC =
of the respective primary multicast stream.
> > SSRC of Requesting Client: This SSRC is directly copied from the =
Token Verification Request message (this text already
> exists in 4.4).
> >
> > Is the "packet sender SSRC" definition fine with you?
>=20
> I think I would like to have extra warning: "Note, theis SSRC value =
are
> from a different SSRC space than the one used in the Unicast RTP =
session."

OK.

-acbegen
=20
> >> Yes, for mapping message that is fine. The thing I can complain =
about is
> >> why the SSRC fields aren't there own bullets and instead hidden in =
the
> >> pre-facing text to the figures?
> >
> > We can move them to the bullets. We probably did not so because, =
they are pretty much part of the baseline format.
>=20
> I think it has its point when we do have some irregularities here in
> some cases.
>=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
> ----------------------------------------------------------------------

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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>
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Subject: Re: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 2 Unicast RTP session Termination
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> >> It might be that you need to tell any using specification to =
declare how
> >> a unicast session is terminated. But currently this is a gapping =
whole
> >> in the specification.
> >
> > Actually, we clearly specified how a RAMS session is terminated or =
when it ends. Repeating all that text here is not needed.
> So, maybe we should simply say the spec using the Token approach needs =
to discuss the session termination and ending
> sessions.
>=20
> No, if that requirement is put on application of this, then I am =
mostly
> happy.

Just to be clear: are you satisfied if we say " the spec using the Token =
approach needs to discuss the session termination and ending
Sessions"? If so, do you think we need a 2119 language here?

> >
> >> Based on the below comments it appears that you have decided that =
RTP
> >> session termination is when all external SSRCs has timed out. But, =
I
> >> wonder if the servers really are fine with that. If I as a client
> >> continue to send RTCP RR to a server in a RAMS scenario for the =
unicast
> >> session, then the server is not allowed to terminate the session? I
> >
> > In RAMS, the client will continue sending RRs if and only if the =
client does retransmissions in that unicast session. If that is
> the case, of course, the server is fine since the session is still =
active.
> >
> > If the client does not do any further retransmissions after the =
burst is completed, the RRs will not be sent so the session will
> end after the timeout.
>=20
> But, basically you are saying that unless the client keeps the session
> alive, the server will terminate it. Which gives non symmetric =
behavior
> between the client and the retransmission server.

In practice, in an IPTV system, this is how it works to the best of my =
knowledge: The client once it completes a retransmission does not BYE =
the unicast ret session right after the ret packets were received. It =
will send the periodic RR/XR reports, if a new NACK is sent, the server =
is ready to go. When the client switches to another channel, it will BYE =
that unicast session so the session will end for both sides. Yet, some =
clients may stop sending reports, so the SSRC will time out on the =
server side.
=20
...
> >> Once more my summary of this topic is: This is not defined nor =
obvious
> >> as this is an issue that has not existed prior to using RTP/RTCP =
for
> >> creating new RTP sessions.
> >
> > The difference is in creating the RTP session not ending it. Just =
because a session is dynamically created via RTCP does not
> mean that it must have a different termination method from the =
regularly established RTP sessions (3550). Once a session is
> up, it is up - no difference from any other regular RTP session. =
Following the KISS principle here makes sense IMO.
>=20
> Well, you have above described a system for retransmissions where =
there
> is a difference between the client and the server. And where there is
> cases for when the session is considered terminated. That is not the
> same as saying that it isn't terminated. I haven't argued that they =
must
> be an explicit message. Only that the rules are made clear.
>=20
> I am not certain that the restransmission case is well described here.
> And this document is the generalized form of when the unicasat session
> is created. I think describing this general case of termiantion has =
its
> point. And with the general case I mean:
>=20
> If the client stops sending RTCP regularly and the client SSRC times =
out
> from the unicast RTP session, the server is allowed to terminate/close
> the session. New message for creating it is needed. If the client =
sends

Agreed.

> BYE in the unicast session, the session can also be considered
> terminated (Needs Token verification or there is a DOS angle).

Also agreed. More on this in an upcoming mail.
=20
-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
> ----------------------------------------------------------------------

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Thread-Topic: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 6: Resend behavior
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>
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Subject: Re: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 6: Resend behavior
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> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Monday, January 17, 2011 10:04 AM
> To: Ali C. Begen (abegen)
> Cc: IETF AVT WG; =
draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
> Subject: Re: [AVT] WGLC comments on =
draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 6: Resend behavior
>=20
> Ali C. Begen (abegen) skrev 2011-01-17 14:24:
> > OK, I see your concern, however, should not we still follow 4585 as =
it is and once it gets updated, it will be reflected here?
> If the immediate mode is chosen, the NACK/RAMS could result in many =
more spurious transmissions whereas the token stuff
> only results in one such transmission. So putting some extra rules =
here for the tokens only will not get us much.
> >
> > Maybe, we can add some text to disallow the immediate mode?
>=20
> You seem to be punting on this issue. I think Immediate mode needs to =
be

Did not even cross my mind :)

> allowed, it makes all the sense for these type of applications in
> certain environments. However, what is needed is a separation of the
> "packet loss" / "failure for server to act on event" detection =
mechanism
> and the RTCP transmission rate limiting.

Well said, but with RTCP one can do so much.
=20
> I think despite our discussion of these things in CCM (RFC 5104) we =
did
> fail to take all the AVPF cases into account. I have to say that =
3.5.1.1
> do rely on repeating FIR messages until you get an result. It does =
fail
> to say in which mode and what type of transmission should be used for
> the repetitions. This as one in early mode, this is either classified =
as
> an event that are allowed to send early, or as a regular transmission.
>=20
> There need to be some sanity checking in this, and going forward being
> aware that what we specify can be seriously broken is not the best
> thing. In addition RAMS can be fixed. And the NACK still doesn't have =
a
> spec for this use case.

OK, let me ask you this. Even if we come up with a miracle solution for =
the token stuff, that still does not fix the bigger issue that one might =
have with NACKs (I am omitting RAMS here since RAMS has some other =
natural limits, e.g., a client would not really have multiple =
outstanding RAMS requests unless it is attacking the system).

So, maybe we should actually talk about fixing this in general for 4585?

-acbegen
=20
>=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
> ----------------------------------------------------------------------

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Thread-Topic: WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 4 SDP attribute semantics
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>
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Cc: draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org, IETF AVT WG <avt@ietf.org>
Subject: Re: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 4 SDP attribute semantics
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> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Monday, January 17, 2011 9:55 AM
> To: Ali C. Begen (abegen)
> Cc: IETF AVT WG; =
draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
> Subject: Re: WGLC comments on =
draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 4 SDP attribute =
semantics
>=20
> Ali C. Begen (abegen) skrev 2011-01-17 14:58:
> >> Ok, can you then write down the exact semantics of the =
portmapping-req
> >> and portmapping attribute? Because to me their semantics aren't =
clear.
> >> And without clear semantics this is going to cause confusion when =
used
> >> the next time.
> >
> > I am not sure what you are looking for in addition to what section 7 =
says.  they are used in pairs. portmapping is used in the
> multicast block and portmapping-req is used in the unicast block. =
Together they mean, certain rtcp messages will require the
> token. The portmapping-req will tell you where to get the token from. =
Any confusion or question here?
>=20
> I think the confusion is that I believe it will be cases where Token
> Verification Requests are going to be need to be sent on C2->P4. If =
that
> is the case one needs semantics to say that token may also be included
> in RTCP message related to this session.

Right, the messages that control the unicast session and that are sent =
in the unicast session (like BYE and CCM) need to be authenticated (as =
written in 4.3.1).
=20
> In addition the semantics for the portmapping-req that now is mandated
> to be on the declaration of the unicast RTP session. If we take your
> argument there is no need for token verification in that session. =
Thus,
> can you please explain why it is part of that session declaration?
>=20
> In my world there could be token verification messages in the RTCP for
> the unicast session. However, the declaration of where the Token =
server
> should then either be in all RTP sessions that uses Token Verification
> Messages, or at session level and applies to all "portmapping".

Ok. I guess your concern is that the "portmapping" attribute is defined =
media level and used in multicast block, yet it means messages =
triggering or controlling the unicast session might need token =
authentication. Is this your concern?

I don't think we should define it as session level as I mentioned =
earlier, it opens other corner cases. So, let's see how we can address =
this while keeping it media level.=20

I guess we have two options here. We either repeat the "portmapping" =
attribute in both blocks and keep the "portmapping-req:port" in the =
unicast block only, or repeat "portmapping-req:port" in both blocks and =
remove the "portmapping" altogether. The latter saves one line from the =
SDP.
 =20
BTW, the last part of the 3rd paragraph in section 7.1 needs to be =
changed to:

  ... (ii) the client MUST receive the
   unicast session's RTP and RTCP packets from the server on the port
   from which it sent the RTCP message triggering the unicast session.

(Removed "or controlling" since that would mean port c2).

...
> > The only thing that travels from c2->p4 is the receiver reports for =
the unicast session. How would you send a token
> verification request message on that path? The verification request =
message goes from c1 to p3.
>=20
> Here we seem to disagree. If one looks at the list in section 4.3.1, i
> think that this does contain messages where you would like to attach
> token, such as BYE or CCM messages.

Correct.
=20
> It might be me taking the generalized look on this specification. Then =
I
> do see a need to make sure that it can be used in any RTP session =
where
> indicated.
>=20
> If the intention to not have any session level attribute, then I don't
> understand why we have two SDP attributes, then only portmapping-req =
is
> needed. And I think it should be defined as having the meaning. Shall =
be
> included in the SDP media blocks that define an RTP session that needs
> Token Verification messages. Which means that portmapping-req are =
needed
> in the multicast m=3D block, not the unicast.

Does putting the attribute only in the mcast block resolve your =
concerns? I don't think so as it will not cover the unicast session for =
which the rtcp messages are sent for.
=20
-acbegen

=20
>=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 ray.vanbrandenburg@tno.nl  Tue Jan 18 05:40:42 2011
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From: "Brandenburg, R. (Ray) van" <ray.vanbrandenburg@tno.nl>
To: 'Colin Perkins' <csp@csperkins.org>
Thread-Topic: [AVT] New version of RTCP for IDMS
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Subject: Re: [AVT] New version of RTCP for IDMS
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Hi Colin,

Thanks for your comments.

> RTCP XR is appropriate for receivers to report on their synchronisation s=
tate. If IDMS were a peer-to-peer protocol, with each receiver directly det=
ermining its playout based on the reported state from receivers, then this =
would be sufficient for controlling playout too. However, if there is a ser=
ver-to-client control message that informs receivers of their synchronisati=
on point, that should, I believe, follow the previous RTCP approach, and be=
 coded as a separate packet type.

What exactly do you mean by 'previous RTCP approach'?

The main reason we went with an RTCP-XR report block was that we believed i=
t would be less controversial to add an XR block type than a completely new=
 RTCP packet type. In RFC5968 (which I believe you authored) it says that t=
he only valid reason to create a new RTCP packet type is if the required fu=
nctionality would not be appropriate as part of one of the current packet t=
ypes (such as XR-). In the description of the XR- packet type in the same d=
ocument it says the XR blocks are appropriate for "for transporting new met=
rics regarding media transmission or reception quality". So I think we shou=
ld have a discussion about whether the IDMS block fits this description.=20

I believe the DLRR XR report block introduced in RFC3611, is somewhat simil=
ar in nature to the server->client IDMS block in that it is also a server-t=
o-client block that helps receivers get information on network delay.=20

Regards,

Ray van Brandenburg
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From Even.roni@huawei.com  Tue Jan 18 14:17:41 2011
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Date: Wed, 19 Jan 2011 00:16:26 +0200
From: Roni Even <Even.roni@huawei.com>
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Subject: [AVT] Review of evrc-nw, mvc and sbc payloads - plesae
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Hi,

The AVT chairs will like to do a WGLC very soon on the following three
payload specifications.

 

http://tools.ietf.org/html/draft-ietf-avt-rtp-evrc-nw-02

 

http://tools.ietf.org/html/draft-ietf-avt-rtp-mvc-01

 

http://tools.ietf.org/html/draft-ietf-avt-rtp-sbc-01

 

We will appreciate a preliminary review in order to see if the documents are
ready for last call. 

 

Please look at them and send comments to the list

 

Regards

Roni Even AVT co-chair 

 

 

 

 


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<o:shapedefaults v:ext="edit" spidmax="1026" />
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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>The AVT chairs will like to do a WGLC very soon on the following three payload specifications.<o:p></o:p></p><p class=MsoNormal><o:p>&nbsp;</o:p></p><p class=MsoNormal><a href="http://tools.ietf.org/html/draft-ietf-avt-rtp-evrc-nw-02">http://tools.ietf.org/html/draft-ietf-avt-rtp-evrc-nw-02</a><o:p></o:p></p><p class=MsoNormal><o:p>&nbsp;</o:p></p><p class=MsoNormal><a href="http://tools.ietf.org/html/draft-ietf-avt-rtp-mvc-01">http://tools.ietf.org/html/draft-ietf-avt-rtp-mvc-01</a><o:p></o:p></p><p class=MsoNormal><o:p>&nbsp;</o:p></p><p class=MsoNormal><a href="http://tools.ietf.org/html/draft-ietf-avt-rtp-sbc-01">http://tools.ietf.org/html/draft-ietf-avt-rtp-sbc-01</a><o:p></o:p></p><p class=MsoNormal><o:p>&nbsp;</o:p></p><p class=MsoNormal>We will appreciate a preliminary review in order to see if the documents a
 re ready
p></p><p class=MsoNormal><o:p>&nbsp;</o:p></p><p class=MsoNormal>Please look at them and send comments to the list<o:p></o:p></p><p class=MsoNormal><o:p>&nbsp;</o:p></p><p class=MsoNormal>Regards<o:p></o:p></p><p class=MsoNormal>Roni Even AVT co-chair <o:p></o:p></p><p class=MsoNormal><o:p>&nbsp;</o:p></p><p class=MsoNormal><o:p>&nbsp;</o:p></p><p class=MsoNormal><o:p>&nbsp;</o:p></p><p class=MsoNormal><o:p>&nbsp;</o:p></p></div></body></html>

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From peter.musgrave@magorcorp.com  Tue Jan 18 15:46:41 2011
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From: Peter Musgrave <peter.musgrave@magorcorp.com>
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To: "Brandenburg, R. (Ray) van" <ray.vanbrandenburg@tno.nl>
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Subject: Re: [AVT] New version of RTCP for IDMS
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Hi Ray,=20

I think the draft is in good shape and provides a lot of additional =
information.=20

I have one suggestion for adding some flexibility.=20

I continue to be concerned about using device NTP time as the basis for =
system playout sync. A device playout stack may not normally have a =
mechanism to force an NTP update prior to a session - and this draft =
recommends that it do an NTP update. Secondly the NTP local time is not =
in general a monotonic increasing clock. It will (after some amount of =
playout) drift in one direction or the other and shift when the next NTP =
update arrives.=20

One way to avoid this concern would be to use local free-running =
monotonically increasing clocks and use some other mechanism to =
determine the path delay from source to each player. NTP may work well =
enough most of the time - but in cases where that is not good enough - =
providing the local free running clock in the XR would provide the added =
information needed for a synch source which could determine path delays =
to make the necessary sync decisions.=20

I am therefore suggesting that an additional data element be added to =
the end of the report - a free running local device time.=20

Regards,=20

Peter Musgrave




On 2011-01-14, at 11:15 AM, Brandenburg, R. (Ray) van wrote:

> Hi all,
>=20
> We have just submitted a new version of our internet draft describing =
an RTCP XR block for Inter-Destination Media Synchronization (IDMS).
>=20
> http://tools.ietf.org/html/draft-brandenburg-avt-rtcp-for-idms-03
>=20
> We have tried to include all feedback received both from the mailing =
list as well as from IETF 79 in Beijing.
>=20
> Major changes include, apart from numerous clarifications and =
rewrites:
>=20
> - an example/tutorial section that we hope will help make clear how =
IDMS can be used in practice.
> - an 'Implementation Considerations' section that adresses a number of =
issues that were raised on the mailing list, including how to deal with =
NTP synchronization in different use cases, echo cancellation and the =
issue of reporting versus controlling.
> - the removal of the mixer functionality
>=20
> In the past there have been some concerns about whether RTCP XR blocks =
are the best solution for IDMS and if a new RTCP block wouldn't be =
better. Also, some questions were raised about the advantage of having =
both client->server and server->client communication with the same XR =
block. On these issues we feel it is important to understand that the =
proposed IDMS system is not a control protocol; client are not told how =
to synchronize their playback to others. The proposed XR block only =
informs clients of the slowest playback within their synchronization =
group, how they deal with this information, if at all, is up to =
themselves. In this way, IDMS is not that different from how RTCP is =
used to deal with synchronization between audio and video streams on the =
same terminal.
>=20
> I look forward to your comments.
>=20
> Kind regards,
>=20
> Ray van Brandenburg
> This e-mail and its contents are subject to the DISCLAIMER at =
http://www.tno.nl/disclaimer/email.html
>=20
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt


From magnus.westerlund@ericsson.com  Wed Jan 19 02:43:33 2011
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Subject: Re: [AVT] draft-ietf-avt-ecn-for-rtp: Is AVPF required forleap-of-faith initialisation?
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Belling, Thomas (NSN - DE/Munich) skrev 2011-01-18 11:37:
> Dear Magnus
> 
>  
> 
> I compiled the attached update proposal to the draft (with changemarks),
> adding support for a=rtcp-fb and a=rtcp-xr.
> 
> Could you please consider those changes?

Hi,

I have looked at the changes and I have the following comments:

SDP parameters. Yes, on the general level I agree this is what is
intended. On a more detailed level, I think there is a bit to much
respecification of things that are already defined.

When it comes to AVPF usage, I will go through that to try to generalize
it. I will respond separately on that.

We also have some conistency issues that needs to be fixed around
normative levels of mechanisms.

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 ray.vanbrandenburg@tno.nl  Wed Jan 19 03:27:30 2011
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From: "Brandenburg, R. (Ray) van" <ray.vanbrandenburg@tno.nl>
To: 'Peter Musgrave' <peter.musgrave@magorcorp.com>
Thread-Topic: [AVT] New version of RTCP for IDMS
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Subject: Re: [AVT] New version of RTCP for IDMS
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Hi Peter,

> I think the draft is in good shape and provides a lot of additional infor=
mation.=20

Thanks!

> I am therefore suggesting that an additional data element be added to the=
 end of the report - a free running local device time.=20

I understand your concern and agree that free-running clocks would be a pos=
sible solution. Would it be OK with you if we rename the 7th and 8th words =
of the XR block to '64bit timestamp' instead of 'NTP timestamp'? Applicatio=
ns can then decide whether this timestamp should be filled using NTP, local=
 device times or another timing solution such as GPS or PTP (precision time=
 protocol). This way the block becomes NTP-independent while not having to =
add additional elements making it unnecessarily big.=20

Regards,

Ray van Brandenburg

-----Original Message-----
From: Peter Musgrave [mailto:peter.musgrave@magorcorp.com]=20
Sent: woensdag 19 januari 2011 0:49
To: Brandenburg, R. (Ray) van
Cc: avt@ietf.org
Subject: Re: [AVT] New version of RTCP for IDMS

Hi Ray,=20

I think the draft is in good shape and provides a lot of additional informa=
tion.=20

I have one suggestion for adding some flexibility.=20

I continue to be concerned about using device NTP time as the basis for sys=
tem playout sync. A device playout stack may not normally have a mechanism =
to force an NTP update prior to a session - and this draft recommends that =
it do an NTP update. Secondly the NTP local time is not in general a monoto=
nic increasing clock. It will (after some amount of playout) drift in one d=
irection or the other and shift when the next NTP update arrives.=20

One way to avoid this concern would be to use local free-running monotonica=
lly increasing clocks and use some other mechanism to determine the path de=
lay from source to each player. NTP may work well enough most of the time -=
 but in cases where that is not good enough - providing the local free runn=
ing clock in the XR would provide the added information needed for a synch =
source which could determine path delays to make the necessary sync decisio=
ns.=20

I am therefore suggesting that an additional data element be added to the e=
nd of the report - a free running local device time.=20

Regards,=20

Peter Musgrave




On 2011-01-14, at 11:15 AM, Brandenburg, R. (Ray) van wrote:

> Hi all,
>=20
> We have just submitted a new version of our internet draft describing an =
RTCP XR block for Inter-Destination Media Synchronization (IDMS).
>=20
> http://tools.ietf.org/html/draft-brandenburg-avt-rtcp-for-idms-03
>=20
> We have tried to include all feedback received both from the mailing list=
 as well as from IETF 79 in Beijing.
>=20
> Major changes include, apart from numerous clarifications and rewrites:
>=20
> - an example/tutorial section that we hope will help make clear how IDMS =
can be used in practice.
> - an 'Implementation Considerations' section that adresses a number of is=
sues that were raised on the mailing list, including how to deal with NTP s=
ynchronization in different use cases, echo cancellation and the issue of r=
eporting versus controlling.
> - the removal of the mixer functionality
>=20
> In the past there have been some concerns about whether RTCP XR blocks ar=
e the best solution for IDMS and if a new RTCP block wouldn't be better. Al=
so, some questions were raised about the advantage of having both client->s=
erver and server->client communication with the same XR block. On these iss=
ues we feel it is important to understand that the proposed IDMS system is =
not a control protocol; client are not told how to synchronize their playba=
ck to others. The proposed XR block only informs clients of the slowest pla=
yback within their synchronization group, how they deal with this informati=
on, if at all, is up to themselves. In this way, IDMS is not that different=
 from how RTCP is used to deal with synchronization between audio and video=
 streams on the same terminal.
>=20
> I look forward to your comments.
>=20
> Kind regards,
>=20
> Ray van Brandenburg
> This e-mail and its contents are subject to the DISCLAIMER at=20
> http://www.tno.nl/disclaimer/email.html
>=20
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt

This e-mail and its contents are subject to the DISCLAIMER at http://www.tn=
o.nl/disclaimer/email.html


From peter.musgrave@magorcorp.com  Wed Jan 19 03:40:10 2011
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From: Peter Musgrave <peter.musgrave@magorcorp.com>
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To: "Brandenburg, R. (Ray) van" <ray.vanbrandenburg@tno.nl>
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Subject: Re: [AVT] New version of RTCP for IDMS
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Hi Ray,=20

I like that idea, but then I think having a flag field to indicate if it =
is e.g. NTP or "vendor specific" might be useful from an interop =
perspective.

Peter

On 2011-01-19, at 6:30 AM, Brandenburg, R. (Ray) van wrote:

> Hi Peter,
>=20
>> I think the draft is in good shape and provides a lot of additional =
information.=20
>=20
> Thanks!
>=20
>> I am therefore suggesting that an additional data element be added to =
the end of the report - a free running local device time.=20
>=20
> I understand your concern and agree that free-running clocks would be =
a possible solution. Would it be OK with you if we rename the 7th and =
8th words of the XR block to '64bit timestamp' instead of 'NTP =
timestamp'? Applications can then decide whether this timestamp should =
be filled using NTP, local device times or another timing solution such =
as GPS or PTP (precision time protocol). This way the block becomes =
NTP-independent while not having to add additional elements making it =
unnecessarily big.=20
>=20
> Regards,
>=20
> Ray van Brandenburg
>=20
> -----Original Message-----
> From: Peter Musgrave [mailto:peter.musgrave@magorcorp.com]=20
> Sent: woensdag 19 januari 2011 0:49
> To: Brandenburg, R. (Ray) van
> Cc: avt@ietf.org
> Subject: Re: [AVT] New version of RTCP for IDMS
>=20
> Hi Ray,=20
>=20
> I think the draft is in good shape and provides a lot of additional =
information.=20
>=20
> I have one suggestion for adding some flexibility.=20
>=20
> I continue to be concerned about using device NTP time as the basis =
for system playout sync. A device playout stack may not normally have a =
mechanism to force an NTP update prior to a session - and this draft =
recommends that it do an NTP update. Secondly the NTP local time is not =
in general a monotonic increasing clock. It will (after some amount of =
playout) drift in one direction or the other and shift when the next NTP =
update arrives.=20
>=20
> One way to avoid this concern would be to use local free-running =
monotonically increasing clocks and use some other mechanism to =
determine the path delay from source to each player. NTP may work well =
enough most of the time - but in cases where that is not good enough - =
providing the local free running clock in the XR would provide the added =
information needed for a synch source which could determine path delays =
to make the necessary sync decisions.=20
>=20
> I am therefore suggesting that an additional data element be added to =
the end of the report - a free running local device time.=20
>=20
> Regards,=20
>=20
> Peter Musgrave
>=20
>=20
>=20
>=20
> On 2011-01-14, at 11:15 AM, Brandenburg, R. (Ray) van wrote:
>=20
>> Hi all,
>>=20
>> We have just submitted a new version of our internet draft describing =
an RTCP XR block for Inter-Destination Media Synchronization (IDMS).
>>=20
>> http://tools.ietf.org/html/draft-brandenburg-avt-rtcp-for-idms-03
>>=20
>> We have tried to include all feedback received both from the mailing =
list as well as from IETF 79 in Beijing.
>>=20
>> Major changes include, apart from numerous clarifications and =
rewrites:
>>=20
>> - an example/tutorial section that we hope will help make clear how =
IDMS can be used in practice.
>> - an 'Implementation Considerations' section that adresses a number =
of issues that were raised on the mailing list, including how to deal =
with NTP synchronization in different use cases, echo cancellation and =
the issue of reporting versus controlling.
>> - the removal of the mixer functionality
>>=20
>> In the past there have been some concerns about whether RTCP XR =
blocks are the best solution for IDMS and if a new RTCP block wouldn't =
be better. Also, some questions were raised about the advantage of =
having both client->server and server->client communication with the =
same XR block. On these issues we feel it is important to understand =
that the proposed IDMS system is not a control protocol; client are not =
told how to synchronize their playback to others. The proposed XR block =
only informs clients of the slowest playback within their =
synchronization group, how they deal with this information, if at all, =
is up to themselves. In this way, IDMS is not that different from how =
RTCP is used to deal with synchronization between audio and video =
streams on the same terminal.
>>=20
>> I look forward to your comments.
>>=20
>> Kind regards,
>>=20
>> Ray van Brandenburg
>> This e-mail and its contents are subject to the DISCLAIMER at=20
>> http://www.tno.nl/disclaimer/email.html
>>=20
>> _______________________________________________
>> Audio/Video Transport Working Group
>> avt@ietf.org
>> https://www.ietf.org/mailman/listinfo/avt
>=20
> This e-mail and its contents are subject to the DISCLAIMER at =
http://www.tno.nl/disclaimer/email.html
>=20


From ray.vanbrandenburg@tno.nl  Wed Jan 19 03:54:58 2011
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From: "Brandenburg, R. (Ray) van" <ray.vanbrandenburg@tno.nl>
To: 'Peter Musgrave' <peter.musgrave@magorcorp.com>
Thread-Topic: [AVT] New version of RTCP for IDMS
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Subject: Re: [AVT] New version of RTCP for IDMS
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Hi Peter,

I agree there should be some way to indicate what type of timestamp is used=
. Apart from a flag field, for which there are some bits left, another opti=
on would be to have this information in an SDP parameter, since we also use=
 that to indicate the chosen group-sync identifier. =20

Ray

-----Original Message-----
From: Peter Musgrave [mailto:peter.musgrave@magorcorp.com]=20
Sent: woensdag 19 januari 2011 12:43
To: Brandenburg, R. (Ray) van
Cc: avt@ietf.org
Subject: Re: [AVT] New version of RTCP for IDMS

Hi Ray,=20

I like that idea, but then I think having a flag field to indicate if it is=
 e.g. NTP or "vendor specific" might be useful from an interop perspective.

Peter

On 2011-01-19, at 6:30 AM, Brandenburg, R. (Ray) van wrote:

> Hi Peter,
>=20
>> I think the draft is in good shape and provides a lot of additional info=
rmation.=20
>=20
> Thanks!
>=20
>> I am therefore suggesting that an additional data element be added to th=
e end of the report - a free running local device time.=20
>=20
> I understand your concern and agree that free-running clocks would be a p=
ossible solution. Would it be OK with you if we rename the 7th and 8th word=
s of the XR block to '64bit timestamp' instead of 'NTP timestamp'? Applicat=
ions can then decide whether this timestamp should be filled using NTP, loc=
al device times or another timing solution such as GPS or PTP (precision ti=
me protocol). This way the block becomes NTP-independent while not having t=
o add additional elements making it unnecessarily big.=20
>=20
> Regards,
>=20
> Ray van Brandenburg
>=20
> -----Original Message-----
> From: Peter Musgrave [mailto:peter.musgrave@magorcorp.com]
> Sent: woensdag 19 januari 2011 0:49
> To: Brandenburg, R. (Ray) van
> Cc: avt@ietf.org
> Subject: Re: [AVT] New version of RTCP for IDMS
>=20
> Hi Ray,
>=20
> I think the draft is in good shape and provides a lot of additional infor=
mation.=20
>=20
> I have one suggestion for adding some flexibility.=20
>=20
> I continue to be concerned about using device NTP time as the basis for s=
ystem playout sync. A device playout stack may not normally have a mechanis=
m to force an NTP update prior to a session - and this draft recommends tha=
t it do an NTP update. Secondly the NTP local time is not in general a mono=
tonic increasing clock. It will (after some amount of playout) drift in one=
 direction or the other and shift when the next NTP update arrives.=20
>=20
> One way to avoid this concern would be to use local free-running monotoni=
cally increasing clocks and use some other mechanism to determine the path =
delay from source to each player. NTP may work well enough most of the time=
 - but in cases where that is not good enough - providing the local free ru=
nning clock in the XR would provide the added information needed for a sync=
h source which could determine path delays to make the necessary sync decis=
ions.=20
>=20
> I am therefore suggesting that an additional data element be added to the=
 end of the report - a free running local device time.=20
>=20
> Regards,
>=20
> Peter Musgrave
>=20
>=20
>=20
>=20
> On 2011-01-14, at 11:15 AM, Brandenburg, R. (Ray) van wrote:
>=20
>> Hi all,
>>=20
>> We have just submitted a new version of our internet draft describing an=
 RTCP XR block for Inter-Destination Media Synchronization (IDMS).
>>=20
>> http://tools.ietf.org/html/draft-brandenburg-avt-rtcp-for-idms-03
>>=20
>> We have tried to include all feedback received both from the mailing lis=
t as well as from IETF 79 in Beijing.
>>=20
>> Major changes include, apart from numerous clarifications and rewrites:
>>=20
>> - an example/tutorial section that we hope will help make clear how IDMS=
 can be used in practice.
>> - an 'Implementation Considerations' section that adresses a number of i=
ssues that were raised on the mailing list, including how to deal with NTP =
synchronization in different use cases, echo cancellation and the issue of =
reporting versus controlling.
>> - the removal of the mixer functionality
>>=20
>> In the past there have been some concerns about whether RTCP XR blocks a=
re the best solution for IDMS and if a new RTCP block wouldn't be better. A=
lso, some questions were raised about the advantage of having both client->=
server and server->client communication with the same XR block. On these is=
sues we feel it is important to understand that the proposed IDMS system is=
 not a control protocol; client are not told how to synchronize their playb=
ack to others. The proposed XR block only informs clients of the slowest pl=
ayback within their synchronization group, how they deal with this informat=
ion, if at all, is up to themselves. In this way, IDMS is not that differen=
t from how RTCP is used to deal with synchronization between audio and vide=
o streams on the same terminal.
>>=20
>> I look forward to your comments.
>>=20
>> Kind regards,
>>=20
>> Ray van Brandenburg
>> This e-mail and its contents are subject to the DISCLAIMER at=20
>> http://www.tno.nl/disclaimer/email.html
>>=20
>> _______________________________________________
>> Audio/Video Transport Working Group
>> avt@ietf.org
>> https://www.ietf.org/mailman/listinfo/avt
>=20
> This e-mail and its contents are subject to the DISCLAIMER at=20
> http://www.tno.nl/disclaimer/email.html
>=20

This e-mail and its contents are subject to the DISCLAIMER at http://www.tn=
o.nl/disclaimer/email.html


From peter.musgrave@magorcorp.com  Wed Jan 19 04:00:29 2011
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From: Peter Musgrave <peter.musgrave@magorcorp.com>
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To: "Brandenburg, R. (Ray) van" <ray.vanbrandenburg@tno.nl>
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Cc: "avt@ietf.org" <avt@ietf.org>
Subject: Re: [AVT] New version of RTCP for IDMS
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You're right, an SDP parameter is way better....for now I think all we =
need are values ntp and local.

This does however open up a negotiation issue which will require =
clarification. If a device supports both - how does it say so, who =
decides etc.

Peter

On 2011-01-19, at 6:57 AM, Brandenburg, R. (Ray) van wrote:

> Hi Peter,
>=20
> I agree there should be some way to indicate what type of timestamp is =
used. Apart from a flag field, for which there are some bits left, =
another option would be to have this information in an SDP parameter, =
since we also use that to indicate the chosen group-sync identifier. =20
>=20
> Ray
>=20
> -----Original Message-----
> From: Peter Musgrave [mailto:peter.musgrave@magorcorp.com]=20
> Sent: woensdag 19 januari 2011 12:43
> To: Brandenburg, R. (Ray) van
> Cc: avt@ietf.org
> Subject: Re: [AVT] New version of RTCP for IDMS
>=20
> Hi Ray,=20
>=20
> I like that idea, but then I think having a flag field to indicate if =
it is e.g. NTP or "vendor specific" might be useful from an interop =
perspective.
>=20
> Peter
>=20
> On 2011-01-19, at 6:30 AM, Brandenburg, R. (Ray) van wrote:
>=20
>> Hi Peter,
>>=20
>>> I think the draft is in good shape and provides a lot of additional =
information.=20
>>=20
>> Thanks!
>>=20
>>> I am therefore suggesting that an additional data element be added =
to the end of the report - a free running local device time.=20
>>=20
>> I understand your concern and agree that free-running clocks would be =
a possible solution. Would it be OK with you if we rename the 7th and =
8th words of the XR block to '64bit timestamp' instead of 'NTP =
timestamp'? Applications can then decide whether this timestamp should =
be filled using NTP, local device times or another timing solution such =
as GPS or PTP (precision time protocol). This way the block becomes =
NTP-independent while not having to add additional elements making it =
unnecessarily big.=20
>>=20
>> Regards,
>>=20
>> Ray van Brandenburg
>>=20
>> -----Original Message-----
>> From: Peter Musgrave [mailto:peter.musgrave@magorcorp.com]
>> Sent: woensdag 19 januari 2011 0:49
>> To: Brandenburg, R. (Ray) van
>> Cc: avt@ietf.org
>> Subject: Re: [AVT] New version of RTCP for IDMS
>>=20
>> Hi Ray,
>>=20
>> I think the draft is in good shape and provides a lot of additional =
information.=20
>>=20
>> I have one suggestion for adding some flexibility.=20
>>=20
>> I continue to be concerned about using device NTP time as the basis =
for system playout sync. A device playout stack may not normally have a =
mechanism to force an NTP update prior to a session - and this draft =
recommends that it do an NTP update. Secondly the NTP local time is not =
in general a monotonic increasing clock. It will (after some amount of =
playout) drift in one direction or the other and shift when the next NTP =
update arrives.=20
>>=20
>> One way to avoid this concern would be to use local free-running =
monotonically increasing clocks and use some other mechanism to =
determine the path delay from source to each player. NTP may work well =
enough most of the time - but in cases where that is not good enough - =
providing the local free running clock in the XR would provide the added =
information needed for a synch source which could determine path delays =
to make the necessary sync decisions.=20
>>=20
>> I am therefore suggesting that an additional data element be added to =
the end of the report - a free running local device time.=20
>>=20
>> Regards,
>>=20
>> Peter Musgrave
>>=20
>>=20
>>=20
>>=20
>> On 2011-01-14, at 11:15 AM, Brandenburg, R. (Ray) van wrote:
>>=20
>>> Hi all,
>>>=20
>>> We have just submitted a new version of our internet draft =
describing an RTCP XR block for Inter-Destination Media Synchronization =
(IDMS).
>>>=20
>>> http://tools.ietf.org/html/draft-brandenburg-avt-rtcp-for-idms-03
>>>=20
>>> We have tried to include all feedback received both from the mailing =
list as well as from IETF 79 in Beijing.
>>>=20
>>> Major changes include, apart from numerous clarifications and =
rewrites:
>>>=20
>>> - an example/tutorial section that we hope will help make clear how =
IDMS can be used in practice.
>>> - an 'Implementation Considerations' section that adresses a number =
of issues that were raised on the mailing list, including how to deal =
with NTP synchronization in different use cases, echo cancellation and =
the issue of reporting versus controlling.
>>> - the removal of the mixer functionality
>>>=20
>>> In the past there have been some concerns about whether RTCP XR =
blocks are the best solution for IDMS and if a new RTCP block wouldn't =
be better. Also, some questions were raised about the advantage of =
having both client->server and server->client communication with the =
same XR block. On these issues we feel it is important to understand =
that the proposed IDMS system is not a control protocol; client are not =
told how to synchronize their playback to others. The proposed XR block =
only informs clients of the slowest playback within their =
synchronization group, how they deal with this information, if at all, =
is up to themselves. In this way, IDMS is not that different from how =
RTCP is used to deal with synchronization between audio and video =
streams on the same terminal.
>>>=20
>>> I look forward to your comments.
>>>=20
>>> Kind regards,
>>>=20
>>> Ray van Brandenburg
>>> This e-mail and its contents are subject to the DISCLAIMER at=20
>>> http://www.tno.nl/disclaimer/email.html
>>>=20
>>> _______________________________________________
>>> Audio/Video Transport Working Group
>>> avt@ietf.org
>>> https://www.ietf.org/mailman/listinfo/avt
>>=20
>> This e-mail and its contents are subject to the DISCLAIMER at=20
>> http://www.tno.nl/disclaimer/email.html
>>=20
>=20
> This e-mail and its contents are subject to the DISCLAIMER at =
http://www.tno.nl/disclaimer/email.html
>=20


From thomas.belling@nsn.com  Wed Jan 19 04:22:45 2011
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From: "Belling, Thomas (NSN - DE/Munich)" <thomas.belling@nsn.com>
To: "ext Magnus Westerlund" <magnus.westerlund@ericsson.com>
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Cc: "ext Jarvinen Kari.Ju \(Nokia-NRC/Tampere\)" <kari.ju.jarvinen@nokia.com>, Tomas Frankkila <tomas.frankkila@ericsson.com>, "Mutikainen Jari \(Nokia-MS/Helsinki\)" <Jari.Mutikainen@nokia.com>, "ext Leung, Nikolai" <nleung@QUALCOMM.COM>, IETF AVT WG <avt@ietf.org>
Subject: Re: [AVT] draft-ietf-avt-ecn-for-rtp: Is AVPF required forleap-of-faith initialisation?
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Thanks, Magnus.

My original email got blocked on the AVT exploder due to its size.
I thus resend it with a shortened attachment, showing only the parts of =
the draft where I propose changes. The changes are still exactly the =
same.

Thomas

-----Original Message-----
From: ext Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]=20
Sent: Wednesday, January 19, 2011 11:46 AM
To: Belling, Thomas (NSN - DE/Munich)
Cc: IETF AVT WG; ext Jarvinen Kari.Ju (Nokia-NRC/Tampere); Mutikainen =
Jari (Nokia-MS/Helsinki); ext Leung, Nikolai; Tomas Frankkila
Subject: Re: [AVT] draft-ietf-avt-ecn-for-rtp: Is AVPF required =
forleap-of-faith initialisation?

Belling, Thomas (NSN - DE/Munich) skrev 2011-01-18 11:37:
> Dear Magnus
>=20
> =20
>=20
> I compiled the attached update proposal to the draft (with=20
> changemarks), adding support for a=3Drtcp-fb and a=3Drtcp-xr.
>=20
> Could you please consider those changes?

Hi,

I have looked at the changes and I have the following comments:

SDP parameters. Yes, on the general level I agree this is what is =
intended. On a more detailed level, I think there is a bit to much =
respecification of things that are already defined.

When it comes to AVPF usage, I will go through that to try to generalize =
it. I will respond separately on that.

We also have some conistency issues that needs to be fixed around =
normative levels of mechanisms.

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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Content-Disposition: attachment;
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From magnus.westerlund@ericsson.com  Wed Jan 19 06:18:50 2011
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Cc: "draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org" <draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org>, IETF AVT WG <avt@ietf.org>
Subject: Re: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 2 Unicast RTP session Termination
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Ali C. Begen (abegen) skrev 2011-01-18 14:10:
>>>> It might be that you need to tell any using specification to declare how
>>>> a unicast session is terminated. But currently this is a gapping whole
>>>> in the specification.
>>>
>>> Actually, we clearly specified how a RAMS session is terminated or when it ends. Repeating all that text here is not needed.
>> So, maybe we should simply say the spec using the Token approach needs to discuss the session termination and ending
>> sessions.
>>
>> No, if that requirement is put on application of this, then I am mostly
>> happy.
> 
> Just to be clear: are you satisfied if we say " the spec using the Token approach needs to discuss the session termination and ending
> Sessions"? If so, do you think we need a 2119 language here?
> 

I have a preference for it, so that it is clear that it can't be skipped.

>>>
>>>> Based on the below comments it appears that you have decided that RTP
>>>> session termination is when all external SSRCs has timed out. But, I
>>>> wonder if the servers really are fine with that. If I as a client
>>>> continue to send RTCP RR to a server in a RAMS scenario for the unicast
>>>> session, then the server is not allowed to terminate the session? I
>>>
>>> In RAMS, the client will continue sending RRs if and only if the client does retransmissions in that unicast session. If that is
>> the case, of course, the server is fine since the session is still active.
>>>
>>> If the client does not do any further retransmissions after the burst is completed, the RRs will not be sent so the session will
>> end after the timeout.
>>
>> But, basically you are saying that unless the client keeps the session
>> alive, the server will terminate it. Which gives non symmetric behavior
>> between the client and the retransmission server.
> 
> In practice, in an IPTV system, this is how it works to the best of my knowledge: The client once it completes a retransmission does not BYE the unicast ret session right after the ret packets were received. It will send the periodic RR/XR reports, if a new NACK is sent, the server is ready to go. When the client switches to another channel, it will BYE that unicast session so the session will end for both sides. Yet, some clients may stop sending reports, so the SSRC will time out on the server side.

Yes, I agree that is the reasonable usage in that scenario. But, I don't
know if everyone will see it in the same light when implementing. Thus
resulting in interoperability issues.

That is why I do wonder if we shouldn't describe this reasonble base
case, and say that applications may modify or specify additional methods
of termination.

Cheers

Magnus Westerlund

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From magnus.westerlund@ericsson.com  Wed Jan 19 06:25:06 2011
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Subject: Re: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 6: Resend behavior
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Ali C. Begen (abegen) skrev 2011-01-18 14:10:
> 
> 
>> -----Original Message-----
>> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
>> Sent: Monday, January 17, 2011 10:04 AM
>> To: Ali C. Begen (abegen)
>> Cc: IETF AVT WG; draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
>> Subject: Re: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 6: Resend behavior
>>
>> Ali C. Begen (abegen) skrev 2011-01-17 14:24:
>>> OK, I see your concern, however, should not we still follow 4585 as it is and once it gets updated, it will be reflected here?
>> If the immediate mode is chosen, the NACK/RAMS could result in many more spurious transmissions whereas the token stuff
>> only results in one such transmission. So putting some extra rules here for the tokens only will not get us much.
>>>
>>> Maybe, we can add some text to disallow the immediate mode?
>>
>> You seem to be punting on this issue. I think Immediate mode needs to be
> 
> Did not even cross my mind :)
> 
>> allowed, it makes all the sense for these type of applications in
>> certain environments. However, what is needed is a separation of the
>> "packet loss" / "failure for server to act on event" detection mechanism
>> and the RTCP transmission rate limiting.
> 
> Well said, but with RTCP one can do so much.

Ok, can we at least point out that this issue exist and that
implementations needs to do something reasonable about detecting loss,
and not soley rely on sane configuration values of the RTCP bandwidth.

>  
>> I think despite our discussion of these things in CCM (RFC 5104) we did
>> fail to take all the AVPF cases into account. I have to say that 3.5.1.1
>> do rely on repeating FIR messages until you get an result. It does fail
>> to say in which mode and what type of transmission should be used for
>> the repetitions. This as one in early mode, this is either classified as
>> an event that are allowed to send early, or as a regular transmission.
>>
>> There need to be some sanity checking in this, and going forward being
>> aware that what we specify can be seriously broken is not the best
>> thing. In addition RAMS can be fixed. And the NACK still doesn't have a
>> spec for this use case.
> 
> OK, let me ask you this. Even if we come up with a miracle solution for the token stuff, that still does not fix the bigger issue that one might have with NACKs (I am omitting RAMS here since RAMS has some other natural limits, e.g., a client would not really have multiple outstanding RAMS requests unless it is attacking the system).
> 
> So, maybe we should actually talk about fixing this in general for 4585?

I do agree that RFC 4585 should be updated to take this issue into
consideration.

Cheers

Magnus Westerlund

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Subject: Re: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 4 SDP attribute semantics
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Ali C. Begen (abegen) skrev 2011-01-18 14:29:
> 
> 
>> -----Original Message-----
>> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
>> Sent: Monday, January 17, 2011 9:55 AM
>> To: Ali C. Begen (abegen)
>> Cc: IETF AVT WG; draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
>> Subject: Re: WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 4 SDP attribute semantics
>>
>> Ali C. Begen (abegen) skrev 2011-01-17 14:58:
>>>> Ok, can you then write down the exact semantics of the portmapping-req
>>>> and portmapping attribute? Because to me their semantics aren't clear.
>>>> And without clear semantics this is going to cause confusion when used
>>>> the next time.
>>>
>>> I am not sure what you are looking for in addition to what section 7 says.  they are used in pairs. portmapping is used in the
>> multicast block and portmapping-req is used in the unicast block. Together they mean, certain rtcp messages will require the
>> token. The portmapping-req will tell you where to get the token from. Any confusion or question here?
>>
>> I think the confusion is that I believe it will be cases where Token
>> Verification Requests are going to be need to be sent on C2->P4. If that
>> is the case one needs semantics to say that token may also be included
>> in RTCP message related to this session.
> 
> Right, the messages that control the unicast session and that are sent in the unicast session (like BYE and CCM) need to be authenticated (as written in 4.3.1).

Yes,

>  
>> In addition the semantics for the portmapping-req that now is mandated
>> to be on the declaration of the unicast RTP session. If we take your
>> argument there is no need for token verification in that session. Thus,
>> can you please explain why it is part of that session declaration?
>>
>> In my world there could be token verification messages in the RTCP for
>> the unicast session. However, the declaration of where the Token server
>> should then either be in all RTP sessions that uses Token Verification
>> Messages, or at session level and applies to all "portmapping".
> 
> Ok. I guess your concern is that the "portmapping" attribute is defined media level and used in multicast block, yet it means messages triggering or controlling the unicast session might need token authentication. Is this your concern?

Yes, lets indicate in which RTP session where token verification request
truly are needed.

> 
> I don't think we should define it as session level as I mentioned earlier, it opens other corner cases. So, let's see how we can address this while keeping it media level. 

Media level only is fine.

> 
> I guess we have two options here. We either repeat the "portmapping" attribute in both blocks and keep the "portmapping-req:port" in the unicast block only, or repeat "portmapping-req:port" in both blocks and remove the "portmapping" altogether. The latter saves one line from the SDP.

I would do the later. Keep portmapping-req and put it in all media
blocks that defines RTP session that uses Token Verification.

>   
> BTW, the last part of the 3rd paragraph in section 7.1 needs to be changed to:
> 
>   ... (ii) the client MUST receive the
>    unicast session's RTP and RTCP packets from the server on the port
>    from which it sent the RTCP message triggering the unicast session.
> 
> (Removed "or controlling" since that would mean port c2).
> 

Yes, I think you might have to explicit call out where the verification
failed message is sent from the server to the client for any
verification message sent from client at C2->P4.

> ...
>>> The only thing that travels from c2->p4 is the receiver reports for the unicast session. How would you send a token
>> verification request message on that path? The verification request message goes from c1 to p3.
>>
>> Here we seem to disagree. If one looks at the list in section 4.3.1, i
>> think that this does contain messages where you would like to attach
>> token, such as BYE or CCM messages.
> 
> Correct.
>  
>> It might be me taking the generalized look on this specification. Then I
>> do see a need to make sure that it can be used in any RTP session where
>> indicated.
>>
>> If the intention to not have any session level attribute, then I don't
>> understand why we have two SDP attributes, then only portmapping-req is
>> needed. And I think it should be defined as having the meaning. Shall be
>> included in the SDP media blocks that define an RTP session that needs
>> Token Verification messages. Which means that portmapping-req are needed
>> in the multicast m= block, not the unicast.
> 
> Does putting the attribute only in the mcast block resolve your concerns? I don't think so as it will not cover the unicast session for which the rtcp messages are sent for.

Correct, I want it the semantics of the attribute to be. In this RTP
session, there is need to use Token Verification for some RTCP messages.
And here is the server port and address where you can get a token.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
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From abegen@cisco.com  Wed Jan 19 07:20:13 2011
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Thread-Topic: WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 2 Unicast RTP session Termination
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>
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Subject: Re: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 2 Unicast RTP session Termination
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OK, let's not describe a specific method, and the following text:

A specification using the Token-based authentication described in this =
document is REQUIRED to discuss the methods and rules for =
terminating/ending the unicast session.

I propose to add the text above to the 5th parag. of section 3 as =
follows:

   Upon successful validation and once the unicast session is
   established, all the RTP and RTCP rules specified in [RFC3550] and
   other relevant specifications also apply in this session until it is
   terminated.  >>>


-acbegen

> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Wednesday, January 19, 2011 9:21 AM
> To: Ali C. Begen (abegen)
> Cc: IETF AVT WG; =
draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
> Subject: Re: WGLC comments on =
draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 2 Unicast RTP session =
Termination
>=20
> Ali C. Begen (abegen) skrev 2011-01-18 14:10:
> >>>> It might be that you need to tell any using specification to =
declare how
> >>>> a unicast session is terminated. But currently this is a gapping =
whole
> >>>> in the specification.
> >>>
> >>> Actually, we clearly specified how a RAMS session is terminated or =
when it ends. Repeating all that text here is not
> needed.
> >> So, maybe we should simply say the spec using the Token approach =
needs to discuss the session termination and ending
> >> sessions.
> >>
> >> No, if that requirement is put on application of this, then I am =
mostly
> >> happy.
> >
> > Just to be clear: are you satisfied if we say " the spec using the =
Token approach needs to discuss the session termination
> and ending
> > Sessions"? If so, do you think we need a 2119 language here?
> >
>=20
> I have a preference for it, so that it is clear that it can't be =
skipped.
>=20
> >>>
> >>>> Based on the below comments it appears that you have decided that =
RTP
> >>>> session termination is when all external SSRCs has timed out. =
But, I
> >>>> wonder if the servers really are fine with that. If I as a client
> >>>> continue to send RTCP RR to a server in a RAMS scenario for the =
unicast
> >>>> session, then the server is not allowed to terminate the session? =
I
> >>>
> >>> In RAMS, the client will continue sending RRs if and only if the =
client does retransmissions in that unicast session. If that
> is
> >> the case, of course, the server is fine since the session is still =
active.
> >>>
> >>> If the client does not do any further retransmissions after the =
burst is completed, the RRs will not be sent so the session
> will
> >> end after the timeout.
> >>
> >> But, basically you are saying that unless the client keeps the =
session
> >> alive, the server will terminate it. Which gives non symmetric =
behavior
> >> between the client and the retransmission server.
> >
> > In practice, in an IPTV system, this is how it works to the best of =
my knowledge: The client once it completes a
> retransmission does not BYE the unicast ret session right after the =
ret packets were received. It will send the periodic RR/XR
> reports, if a new NACK is sent, the server is ready to go. When the =
client switches to another channel, it will BYE that unicast
> session so the session will end for both sides. Yet, some clients may =
stop sending reports, so the SSRC will time out on the
> server side.
>=20
> Yes, I agree that is the reasonable usage in that scenario. But, I =
don't
> know if everyone will see it in the same light when implementing. Thus
> resulting in interoperability issues.
>=20
> That is why I do wonder if we shouldn't describe this reasonble base
> case, and say that applications may modify or specify additional =
methods
> of termination.
>=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
> ----------------------------------------------------------------------

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> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Wednesday, January 19, 2011 9:27 AM
> To: Ali C. Begen (abegen)
> Cc: IETF AVT WG; =
draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
> Subject: Re: [AVT] WGLC comments on =
draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 6: Resend behavior
>=20
> Ali C. Begen (abegen) skrev 2011-01-18 14:10:
> >
> >
> >> -----Original Message-----
> >> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> >> Sent: Monday, January 17, 2011 10:04 AM
> >> To: Ali C. Begen (abegen)
> >> Cc: IETF AVT WG; =
draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
> >> Subject: Re: [AVT] WGLC comments on =
draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 6: Resend behavior
> >>
> >> Ali C. Begen (abegen) skrev 2011-01-17 14:24:
> >>> OK, I see your concern, however, should not we still follow 4585 =
as it is and once it gets updated, it will be reflected
> here?
> >> If the immediate mode is chosen, the NACK/RAMS could result in many =
more spurious transmissions whereas the token
> stuff
> >> only results in one such transmission. So putting some extra rules =
here for the tokens only will not get us much.
> >>>
> >>> Maybe, we can add some text to disallow the immediate mode?
> >>
> >> You seem to be punting on this issue. I think Immediate mode needs =
to be
> >
> > Did not even cross my mind :)
> >
> >> allowed, it makes all the sense for these type of applications in
> >> certain environments. However, what is needed is a separation of =
the
> >> "packet loss" / "failure for server to act on event" detection =
mechanism
> >> and the RTCP transmission rate limiting.
> >
> > Well said, but with RTCP one can do so much.
>=20
> Ok, can we at least point out that this issue exist and that
> implementations needs to do something reasonable about detecting loss,
> and not soley rely on sane configuration values of the RTCP bandwidth.

OK, we will mention this and add a warning.
=20
> >
> >> I think despite our discussion of these things in CCM (RFC 5104) we =
did
> >> fail to take all the AVPF cases into account. I have to say that =
3.5.1.1
> >> do rely on repeating FIR messages until you get an result. It does =
fail
> >> to say in which mode and what type of transmission should be used =
for
> >> the repetitions. This as one in early mode, this is either =
classified as
> >> an event that are allowed to send early, or as a regular =
transmission.
> >>
> >> There need to be some sanity checking in this, and going forward =
being
> >> aware that what we specify can be seriously broken is not the best
> >> thing. In addition RAMS can be fixed. And the NACK still doesn't =
have a
> >> spec for this use case.
> >
> > OK, let me ask you this. Even if we come up with a miracle solution =
for the token stuff, that still does not fix the bigger issue
> that one might have with NACKs (I am omitting RAMS here since RAMS has =
some other natural limits, e.g., a client would not
> really have multiple outstanding RAMS requests unless it is attacking =
the system).
> >
> > So, maybe we should actually talk about fixing this in general for =
4585?
>=20
> I do agree that RFC 4585 should be updated to take this issue into
> consideration.

Cool. That will help us in the long term.
-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
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Thread-Topic: WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 4 SDP attribute semantics
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References: <4D308063.3020906@ericsson.com> <04CAD96D4C5A3D48B1919248A8FE0D540E16DA3B@xmb-sjc-215.amer.cisco.com> <4D3420EA.9040806@ericsson.com> <04CAD96D4C5A3D48B1919248A8FE0D540E16DCA7@xmb-sjc-215.amer.cisco.com> <4D345838.2050805@ericsson.com> <04CAD96D4C5A3D48B1919248A8FE0D540E16DD68@xmb-sjc-215.amer.cisco.com> <4D370160.6060002@ericsson.com>
From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>
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Cc: draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org, IETF AVT WG <avt@ietf.org>
Subject: Re: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 4 SDP attribute semantics
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OK, we will work on these changes and submit a revision after the =
chairs' go-ahead. I believe wglc ends today anyway.

Thanks again for the detailed review.
-acbegen

> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Wednesday, January 19, 2011 10:21 AM
> To: Ali C. Begen (abegen)
> Cc: IETF AVT WG; =
draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
> Subject: Re: WGLC comments on =
draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 4 SDP attribute =
semantics
>=20
> Ali C. Begen (abegen) skrev 2011-01-18 14:29:
> >
> >
> >> -----Original Message-----
> >> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> >> Sent: Monday, January 17, 2011 9:55 AM
> >> To: Ali C. Begen (abegen)
> >> Cc: IETF AVT WG; =
draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
> >> Subject: Re: WGLC comments on =
draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 4 SDP attribute =
semantics
> >>
> >> Ali C. Begen (abegen) skrev 2011-01-17 14:58:
> >>>> Ok, can you then write down the exact semantics of the =
portmapping-req
> >>>> and portmapping attribute? Because to me their semantics aren't =
clear.
> >>>> And without clear semantics this is going to cause confusion when =
used
> >>>> the next time.
> >>>
> >>> I am not sure what you are looking for in addition to what section =
7 says.  they are used in pairs. portmapping is used in
> the
> >> multicast block and portmapping-req is used in the unicast block. =
Together they mean, certain rtcp messages will require
> the
> >> token. The portmapping-req will tell you where to get the token =
from. Any confusion or question here?
> >>
> >> I think the confusion is that I believe it will be cases where =
Token
> >> Verification Requests are going to be need to be sent on C2->P4. If =
that
> >> is the case one needs semantics to say that token may also be =
included
> >> in RTCP message related to this session.
> >
> > Right, the messages that control the unicast session and that are =
sent in the unicast session (like BYE and CCM) need to be
> authenticated (as written in 4.3.1).
>=20
> Yes,
>=20
> >
> >> In addition the semantics for the portmapping-req that now is =
mandated
> >> to be on the declaration of the unicast RTP session. If we take =
your
> >> argument there is no need for token verification in that session. =
Thus,
> >> can you please explain why it is part of that session declaration?
> >>
> >> In my world there could be token verification messages in the RTCP =
for
> >> the unicast session. However, the declaration of where the Token =
server
> >> should then either be in all RTP sessions that uses Token =
Verification
> >> Messages, or at session level and applies to all "portmapping".
> >
> > Ok. I guess your concern is that the "portmapping" attribute is =
defined media level and used in multicast block, yet it means
> messages triggering or controlling the unicast session might need =
token authentication. Is this your concern?
>=20
> Yes, lets indicate in which RTP session where token verification =
request
> truly are needed.
>=20
> >
> > I don't think we should define it as session level as I mentioned =
earlier, it opens other corner cases. So, let's see how we can
> address this while keeping it media level.
>=20
> Media level only is fine.
>=20
> >
> > I guess we have two options here. We either repeat the "portmapping" =
attribute in both blocks and keep the "portmapping-
> req:port" in the unicast block only, or repeat "portmapping-req:port" =
in both blocks and remove the "portmapping"
> altogether. The latter saves one line from the SDP.
>=20
> I would do the later. Keep portmapping-req and put it in all media
> blocks that defines RTP session that uses Token Verification.
>=20
> >
> > BTW, the last part of the 3rd paragraph in section 7.1 needs to be =
changed to:
> >
> >   ... (ii) the client MUST receive the
> >    unicast session's RTP and RTCP packets from the server on the =
port
> >    from which it sent the RTCP message triggering the unicast =
session.
> >
> > (Removed "or controlling" since that would mean port c2).
> >
>=20
> Yes, I think you might have to explicit call out where the =
verification
> failed message is sent from the server to the client for any
> verification message sent from client at C2->P4.
>=20
> > ...
> >>> The only thing that travels from c2->p4 is the receiver reports =
for the unicast session. How would you send a token
> >> verification request message on that path? The verification request =
message goes from c1 to p3.
> >>
> >> Here we seem to disagree. If one looks at the list in section =
4.3.1, i
> >> think that this does contain messages where you would like to =
attach
> >> token, such as BYE or CCM messages.
> >
> > Correct.
> >
> >> It might be me taking the generalized look on this specification. =
Then I
> >> do see a need to make sure that it can be used in any RTP session =
where
> >> indicated.
> >>
> >> If the intention to not have any session level attribute, then I =
don't
> >> understand why we have two SDP attributes, then only =
portmapping-req is
> >> needed. And I think it should be defined as having the meaning. =
Shall be
> >> included in the SDP media blocks that define an RTP session that =
needs
> >> Token Verification messages. Which means that portmapping-req are =
needed
> >> in the multicast m=3D block, not the unicast.
> >
> > Does putting the attribute only in the mcast block resolve your =
concerns? I don't think so as it will not cover the unicast
> session for which the rtcp messages are sent for.
>=20
> Correct, I want it the semantics of the attribute to be. In this RTP
> session, there is need to use Token Verification for some RTCP =
messages.
> And here is the server port and address where you can get a token.
>=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  Wed Jan 19 07:51:20 2011
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Subject: Re: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 2 Unicast RTP session Termination
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Ali C. Begen (abegen) skrev 2011-01-19 16:22:
> OK, let's not describe a specific method, and the following text:
> 
> A specification using the Token-based authentication described in this document is REQUIRED to discuss the methods and rules for terminating/ending the unicast session.
> 
> I propose to add the text above to the 5th parag. of section 3 as follows:
> 
>    Upon successful validation and once the unicast session is
>    established, all the RTP and RTCP rules specified in [RFC3550] and
>    other relevant specifications also apply in this session until it is
>    terminated.  >>>
> 

Well based on this, I wouldn't come to the conclusion that one can stop
sending RTCP as server at all unless there has been some explicit
signalling for termination. this is Probably fine for the general
introduction. And it do leave the question of termination wide open.

So, I don't see other than that you have to discuss the termination in
the draft.

"The Unicast RTP session created using the port mapping method can be
explicitly terminated by procedures specified by the using application
or extension. Any unicast RTP session created using this method may be
terminated by a timeout procedure. If either client or Retransmission
Server has not seen any RTCP from any SSRC used by a peer for 5 regular
transmission intervals, the client or server SHOULD consider the RTP
session as terminated, and in that case not send any RTP or RTCP traffic
in the future. The client will have to establish a new unicast session
if required.

If no procedure for explicit termination exist, the client MAY stop
sending RTCP to the Retransmission Server to accomplish session
termination. The Server SHALL NOT stop sending RTCP until the unicast
RTP session is terminated. If Token Verification is signaled to be
allowed also on in the unicast RTP session, i.e. the RTCP messages sent
from c2 to P4, the client SHOULD prefer to terminate by sending RTCP BYE
on all SSRCs it has used in this unicast RTP session."

I think this works fine, and is way better than having no specification
at all.

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 abegen@cisco.com  Wed Jan 19 08:27:07 2011
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>
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Subject: Re: [AVT] WGLC comments on draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 2 Unicast RTP session Termination
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OK, this looks good. It maybe needs some 2119 adjustment, though: e.g., =
if something we say here is already specified in 3550 with a 2119 =
keyword, I think we better repeat them without a 2119 keyword but do =
point to 3550.

-acbegen

> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Wednesday, January 19, 2011 10:54 AM
> To: Ali C. Begen (abegen)
> Cc: IETF AVT WG; =
draft-ietf-avt-ports-for-ucast-mcast-rtp@tools.ietf.org
> Subject: Re: WGLC comments on =
draft-ietf-avt-ports-for-ucast-mcast-rtp-11: Part 2 Unicast RTP session =
Termination
>=20
> Ali C. Begen (abegen) skrev 2011-01-19 16:22:
> > OK, let's not describe a specific method, and the following text:
> >
> > A specification using the Token-based authentication described in =
this document is REQUIRED to discuss the methods and
> rules for terminating/ending the unicast session.
> >
> > I propose to add the text above to the 5th parag. of section 3 as =
follows:
> >
> >    Upon successful validation and once the unicast session is
> >    established, all the RTP and RTCP rules specified in [RFC3550] =
and
> >    other relevant specifications also apply in this session until it =
is
> >    terminated.  >>>
> >
>=20
> Well based on this, I wouldn't come to the conclusion that one can =
stop
> sending RTCP as server at all unless there has been some explicit
> signalling for termination. this is Probably fine for the general
> introduction. And it do leave the question of termination wide open.
>=20
> So, I don't see other than that you have to discuss the termination in
> the draft.
>=20
> "The Unicast RTP session created using the port mapping method can be
> explicitly terminated by procedures specified by the using application
> or extension. Any unicast RTP session created using this method may be
> terminated by a timeout procedure. If either client or Retransmission
> Server has not seen any RTCP from any SSRC used by a peer for 5 =
regular
> transmission intervals, the client or server SHOULD consider the RTP
> session as terminated, and in that case not send any RTP or RTCP =
traffic
> in the future. The client will have to establish a new unicast session
> if required.
>=20
> If no procedure for explicit termination exist, the client MAY stop
> sending RTCP to the Retransmission Server to accomplish session
> termination. The Server SHALL NOT stop sending RTCP until the unicast
> RTP session is terminated. If Token Verification is signaled to be
> allowed also on in the unicast RTP session, i.e. the RTCP messages =
sent
> from c2 to P4, the client SHOULD prefer to terminate by sending RTCP =
BYE
> on all SSRCs it has used in this unicast RTP session."
>=20
> I think this works fine, and is way better than having no =
specification
> at all.
>=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  Wed Jan 19 09:03:23 2011
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Subject: Re: [AVT] draft-ietf-avt-ecn-for-rtp: Is AVPF required for leap-of-faith initialisation?
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Belling, Thomas (NSN - DE/Munich) skrev 2010-12-21 18:25:
> Dear Magnus,
> 
> allow me to continue a discussion we hadabout a month ago.
> 
> Youwere suggesting that the leap of faith initialisation method requires
> AVPF becausethis allows XR ECN summary reports can then be sent quicker
> and shorter reportingintervalthan the average 5secs canbe
> negotiated.Correct?
> 
> The draft contains the following relatedprocedurein6.2.3:
> 
>   If the sender marks all packets as ECT while transmitting on a path
> 
>    that contains an ECN-blocking middlebox, then receivers downstream of
> 
>    that middlebox will not receive any RTP data packets from the sender,
> 
>    and hence will not consider it to be an active RTP SSRC.  The sender
> 
>    can detect this and revert to sending packets without ECT marks,
> 
>    since RTCP SR/RR packets from such receivers will either not include
> 
>    a report for sender's SSRC, or will report that no packets have been
> 
>    received, but this takes at least one RTCP reporting interval.  It
> 
>    should be noted that a receiver might generate its first RTCP packet
> 
>    immediately on joining a unicast session, or very shortly after
> 
>    joining a RTP/AVPF session, before it has had chance to receive any
> 
>    data packets.  A sender that receives RTCP SR/RR packet indicating
> 
>    lack of reception by a receiver SHOULD therefore wait for a second
> 
>    RTCP report from that receiver to be sure that the lack of reception
> 
>    is due to ECT-marking.  Since this recovery process can take several
> 
>    tens of seconds, during which time the RTP session is unusable for
> 
>    media, it is NOT RECOMMENDED that the leap-of-faith ECT initiation
> 
>    method be used in environments where ECN-blocking middleboxes are
> 
>   likely to be present.
> 
> So there is noparticulartrigger for the receiver to sentSR/RR
> reportsthat would allow it to benefitfrom the more flexible timing rules
> of AVPF.

That is true for the case where you use leap-of-faith and have packet
dropping box in the path.

> 
> One could think about usingthe AVPFearly RTCP feedback if the sender
> does not receiveany packets for a whileat call setup, but the duration
> is quite open and it might also bepart of normal operationthat no
> packets are received–considerearly media that are usually only
> transmitted in backward direction as an example.

No, that would be a strange special case.

> 
> You suggested to negotiate a higher RTCP transmission rate as aremedy.In
> my understanding this would work both with AVP and AVPF usingSDP RTCP
> bandwidth modifiers.

Well, it is the minium interval value that needs to be set to something
else than 5 seconds to get better results. For AVP that is allowed but
no signalling exist. Using the formula for calculating the min interval
according to RFC 3550 could work.

  o  The RECOMMENDED value for the reduced minimum in seconds is 360
      divided by the session bandwidth in kilobits/second.  This minimum
      is smaller than 5 seconds for bandwidths greater than 72 kb/s.

AVPF allows one to set trr-int to a reasonable interval through
signalling for this case. And that is without any extreme setting of the
RTCP bandwidth. The whole point with AVPF here is to use the signalling
and improved rules to provide better RTCP functionality without
increasing the bandwidth.

> 
> But I am not convinced this is a good idea given the extra required
> bandwidth.There is a clear wordingalreadythat theleap of faith method
> isnot recommendedin environments whereECN-blocking middleboxes arelikely
> to be present. It is really meant for operator controlled environments
> where such problemswill hardly occur.Isit really worth spending more
> bandwidthto improve the handling of a very unlikelyevent?If there is a
> real risk that ECN-marked packets are blocked ina network, other
> initialisation methods are preferable anyway.

No, I agree that increasing the RTCP bandwidth is not the best method
here. Better utilization of the assigned resource is what should happen.
Thus AVPF.

> 
> As said, I alsostilldoubt that you would have any benefit from AVPFfor
> thatinitialisationmethod.ButevenIonly failto understand the advantages
> you had in mind (do youassume some particular receiverbehaviourdescribed
> nowhere up to now?), I doubt this would beworth the extra complexity,
> only to improve the handling of a very unlikely error. Again, in
> annetworkwherethere is a real risk that ECN-marked packets are blocked,
> other initialisation methods are preferable anyway.
> 

Even with leap-of-faith you need to send the ECN indications for sender
driven, thus you need it. For receiver driven reactions, you still need
to send the control message to the sender. If that is using RTCP, it
still needs AVPF for allowing for earliest possible transmission.

The only cases where I will agree that you don't need AVPF are:

- Codec mode adaptation for AMR

But, in 3GPP I would argue that AVPF is still needed in this case as you
have an APP packet for adjusting redundancy or number of frames in the
packet.

I will look into the formulations here. If there is a large number of
things that are needed to be fulfilled before AVPF is not requried, it
is simpler to require it always. But, I will look into that we have
accurate statements in the draft.

Cheers

Magnus Westerlund

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

From csp@csperkins.org  Wed Jan 19 14:11:16 2011
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Subject: Re: [AVT] New version of RTCP for IDMS
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On 19 Jan 2011, at 11:30, Brandenburg, R. (Ray) van wrote:
> Hi Peter,
>=20
>> I think the draft is in good shape and provides a lot of additional =
information.=20
>=20
> Thanks!
>=20
>> I am therefore suggesting that an additional data element be added to =
the end of the report - a free running local device time.=20
>=20
> I understand your concern and agree that free-running clocks would be =
a possible solution. Would it be OK with you if we rename the 7th and =
8th words of the XR block to '64bit timestamp' instead of 'NTP =
timestamp'?

In RFC3550, absolute timestamps of this sort are represented using the =
format of NTP, even though implementations are not required to run NTP =
[RFC3550 section 4]. I'd suggest this draft follow the same approach: =
use an NTP-format timestamp in the RTCP packets, with signalling to =
specify what clock is used to populate that timestamp. Saves receivers =
having to learn how to parse different timestamp formats.

> Applications can then decide whether this timestamp should be filled =
using NTP, local device times or another timing solution such as GPS or =
PTP (precision time protocol). This way the block becomes =
NTP-independent while not having to add additional elements making it =
unnecessarily big.=20


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




From csp@csperkins.org  Wed Jan 19 14:24:26 2011
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Subject: Re: [AVT] New version of RTCP for IDMS
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Hi,

On 18 Jan 2011, at 13:43, Brandenburg, R. (Ray) van wrote:
...
>> RTCP XR is appropriate for receivers to report on their =
synchronisation state. If IDMS were a peer-to-peer protocol, with each =
receiver directly determining its playout based on the reported state =
from receivers, then this would be sufficient for controlling playout =
too. However, if there is a server-to-client control message that =
informs receivers of their synchronisation point, that should, I =
believe, follow the previous RTCP approach, and be coded as a separate =
packet type.
>=20
> What exactly do you mean by 'previous RTCP approach'?

A separate RTCP packet type, in the same way RTCP SR packets are their =
own packet type.

> The main reason we went with an RTCP-XR report block was that we =
believed it would be less controversial to add an XR block type than a =
completely new RTCP packet type. In RFC5968 (which I believe you =
authored) it says that the only valid reason to create a new RTCP packet =
type is if the required functionality would not be appropriate as part =
of one of the current packet types (such as XR-). In the description of =
the XR- packet type in the same document it says the XR blocks are =
appropriate for "for transporting new metrics regarding media =
transmission or reception quality". So I think we should have a =
discussion about whether the IDMS block fits this description.=20

As I said, I think using RTCP XR is appropriate for receivers to provide =
feedback to the sender on receipt and presentation times for RTP =
packets.=20

I don't think an RTCP XR block is appropriate to, as your draft phrases =
it, "indicate synchronization settings instructions to a receiver". That =
seems to be providing guidance on when to playout the media, rather than =
conveying a metric on how well media playout is doing. This message from =
MSAS to SC would be better expressed as a new RTCP packet type, I think.

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




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From: "Brandenburg, R. (Ray) van" <ray.vanbrandenburg@tno.nl>
To: 'Colin Perkins' <csp@csperkins.org>
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On 19 Jan 2011, at 23:14, Colin Perkins wrote:=20

> In RFC3550, absolute timestamps of this sort are represented using the fo=
rmat of NTP, even though implementations are not required to run NTP [RFC35=
50 section 4]. I'd suggest this draft follow the same approach: use an NTP-=
format timestamp in the RTCP packets, with signalling to specify what clock=
 is used to populate that timestamp. Saves receivers having to learn how to=
 parse different timestamp formats.

I agree, that would make sense. I will start work on some SDP parameters to=
 specify which clock, both type and source (such as which NTP server), is u=
sed.=20

Ray
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From: "Brandenburg, R. (Ray) van" <ray.vanbrandenburg@tno.nl>
To: 'Colin Perkins' <csp@csperkins.org>
Thread-Topic: [AVT] New version of RTCP for IDMS
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Subject: Re: [AVT] New version of RTCP for IDMS
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Hi,

On 19 Jan 2011, at 23:27, Colin Perkins wrote:

...

>> The main reason we went with an RTCP-XR report block was that we believe=
d it would be less controversial to add an XR block type than a completely =
new RTCP packet type. In RFC5968 (which I believe you authored) it says tha=
t the only valid reason to create a new RTCP packet type is if the required=
 functionality would not be appropriate as part of one of the current packe=
t types (such as XR-). In the description of the XR- packet type in the sam=
e document it says the XR blocks are appropriate for "for transporting new =
metrics regarding media transmission or reception quality". So I think we s=
hould have a discussion about whether the IDMS block fits this description.=20

>As I said, I think using RTCP XR is appropriate for receivers to provide f=
eedback to the sender on receipt and presentation times for RTP packets.=20

>I don't think an RTCP XR block is appropriate to, as your draft phrases it=
, "indicate synchronization settings instructions to a receiver". That seem=
s to be providing guidance on when to playout the media, rather than convey=
ing a metric on how well media playout is doing. This message from MSAS to =
SC would be better expressed as a new RTCP packet type, I think.

I agree that this sentence is somewhat misleading and leads one to believe =
that this IDMS message is some sort of control/command protocol. That is no=
t our goal. The message is used to inform receivers of how media playout is=
 doing on their slowest peer receiver. They can then use this information t=
o possibly synchronize their own playout to this peer, although this is not=
 required. In practice, a receiver would probably only synchronize when the=
 playout difference crosses some threshold (e.g. for Social TV >150ms), to =
not cause unnecessary stutters for the user.

Ray
This e-mail and its contents are subject to the DISCLAIMER at http://www.tn=
o.nl/disclaimer/email.html


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Thread-Topic: draft-ietf-avt-ecn-for-rtp: Is AVPF required for leap-of-faith initialisation?
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From: "Belling, Thomas (NSN - DE/Munich)" <thomas.belling@nsn.com>
To: "ext Magnus Westerlund" <magnus.westerlund@ericsson.com>
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Subject: Re: [AVT] draft-ietf-avt-ecn-for-rtp: Is AVPF required for leap-of-faith initialisation?
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Dear Magnus,

 below


> >=20
> > You suggested to negotiate a higher RTCP transmission rate as
aremedy.In
> > my understanding this would work both with AVP and AVPF usingSDP
RTCP
> > bandwidth modifiers.
>
> Well, it is the minium interval value that needs to be set to
something
> else than 5 seconds to get better results. For AVP that is allowed but
> no signalling exist. Using the formula for calculating the min
interval
> according to RFC 3550 could work.
>=20
>  o  The RECOMMENDED value for the reduced minimum in seconds is 360
>      divided by the session bandwidth in kilobits/second.  This
minimum
>      is smaller than 5 seconds for bandwidths greater than 72 kb/s.
>
> AVPF allows one to set trr-int to a reasonable interval through
> signalling for this case. And that is without any extreme setting of
the
> RTCP bandwidth. The whole point with AVPF here is to use the
signalling
> and improved rules to provide better RTCP functionality without
> increasing the bandwidth.

I believe the "calculated interval" is important rather than the
"minimum interval" which is only a lower bound for the calculated
interval. The calculated Interval depends on number of participants and
available bandwidth.
Remember that there are no specific triggers to send RTCP reports
earlier.

> >=20
> > As said, I alsostilldoubt that you would have any benefit from
AVPFfor
> > thatinitialisationmethod.ButevenIonly failto understand the
advantages
> > you had in mind (do youassume some particular
receiverbehaviourdescribed
> > nowhere up to now?), I doubt this would beworth the extra
complexity,
> > only to improve the handling of a very unlikely error. Again, in
> > annetworkwherethere is a real risk that ECN-marked packets are
blocked,
> > other initialisation methods are preferable anyway.
> >=20
>
> Even with leap-of-faith you need to send the ECN indications for
sender
> driven, thus you need it. For receiver driven reactions, you still
need
> to send the control message to the sender. If that is using RTCP, it
> still needs AVPF for allowing for earliest possible transmission.
>=20
> The only cases where I will agree that you don't need AVPF are:
>=20
> - Codec mode adaptation for AMR

This case may not be the exception but the rule in 3GPP, where it is
agreed to send CMR in response to ECT bits for speech. AMR is used for
speech. And speech is still more frequent than video. Thus this case
deserves special attention.

> But, in 3GPP I would argue that AVPF is still needed in this case as
you
> have an APP packet for adjusting redundancy or number of frames in the
> packet.

This will not be used in response to ECT markings being received
according to agreements in 3GPP SA4.
In my understanding the rational is that ECT markings in 3GPP will be
set in the radio network where IP header compression is in place. Here,
changing the codec mode reduces the bandwidth considerably without extra
delay.

> I will look into the formulations here. If there is a large number of
> things that are needed to be fulfilled before AVPF is not requried, it
> is simpler to require it always. But, I will look into that we have
> accurate statements in the draft.

Thanks a lot.
My current proposal is to link it to the ECN RTCP FB message. Obviously
initialisation methods using the ECN FB message then will also require
AVPF, and this can be stated in addition.
In a sense particular application specific feedback is out-of-scope of
the draft but a NOTE stating that many application specific feedback
methods also require AVPF may be helpful.


Regards, Thomas

From magnus.westerlund@ericsson.com  Thu Jan 20 08:07:09 2011
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Cc: IETF AVT WG <avt@ietf.org>
Subject: Re: [AVT] draft-ietf-avt-ecn-for-rtp: Is AVPF required for leap-of-faith initialisation?
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Hi Thomas,


Belling, Thomas (NSN - DE/Munich) skrev 2011-01-20 15:50:
>>
>> Well, it is the minium interval value that needs to be set to
> something
>> else than 5 seconds to get better results. For AVP that is allowed but
>> no signalling exist. Using the formula for calculating the min
> interval
>> according to RFC 3550 could work.
>>
>>  o  The RECOMMENDED value for the reduced minimum in seconds is 360
>>      divided by the session bandwidth in kilobits/second.  This
> minimum
>>      is smaller than 5 seconds for bandwidths greater than 72 kb/s.
>>
>> AVPF allows one to set trr-int to a reasonable interval through
>> signalling for this case. And that is without any extreme setting of
> the
>> RTCP bandwidth. The whole point with AVPF here is to use the
> signalling
>> and improved rules to provide better RTCP functionality without
>> increasing the bandwidth.
> 
> I believe the "calculated interval" is important rather than the
> "minimum interval" which is only a lower bound for the calculated
> interval. The calculated Interval depends on number of participants and
> available bandwidth.
> Remember that there are no specific triggers to send RTCP reports
> earlier.

I wouldn't agree for most sensible settings of the RTCP bandwidth if you
are using AVP, then the minimal interval will be the limiting factor
unless there is a larger multi party group.

For example with a session bandwidht of 24 kbps then 5% bandwidth are
1200 bps. In a two party session, i.e. 2 SSRCs, then the deterministic
calculated RTCP interval becomes with the following assumptions: 200
bytes RTCP packets and both SSRCs send data. Which means they share the
full RTCP bandwidth, independent of how RS and RR is set.

n=2
C=200*8/1200

Td = max(Tmin, n*C = 2,667)

If Tmin is the default value of 5 seconds, then we can easily see that
for a point to point session we either needs twice as many session
members, double the size of the RTCP packet, or half the RTCP bandwidth
before Tmin isn't the limiting factor.

If one has a higher RTCP bandwidth without increasing the number of
session participants, the calculated Td is reduced correspondingly with
the RTCP bandwidth increase.

The whole point of this is that in MTSI type of use cases it will not be
calculated transmission interval that will be the limiting factor, it is
the default minimal interval of 5 seconds.


> 
>>>
>>> As said, I alsostilldoubt that you would have any benefit from
> AVPFfor
>>> thatinitialisationmethod.ButevenIonly failto understand the
> advantages
>>> you had in mind (do youassume some particular
> receiverbehaviourdescribed
>>> nowhere up to now?), I doubt this would beworth the extra
> complexity,
>>> only to improve the handling of a very unlikely error. Again, in
>>> annetworkwherethere is a real risk that ECN-marked packets are
> blocked,
>>> other initialisation methods are preferable anyway.
>>>
>>
>> Even with leap-of-faith you need to send the ECN indications for
> sender
>> driven, thus you need it. For receiver driven reactions, you still
> need
>> to send the control message to the sender. If that is using RTCP, it
>> still needs AVPF for allowing for earliest possible transmission.
>>
>> The only cases where I will agree that you don't need AVPF are:
>>
>> - Codec mode adaptation for AMR
> 
> This case may not be the exception but the rule in 3GPP, where it is
> agreed to send CMR in response to ECT bits for speech. AMR is used for
> speech. And speech is still more frequent than video. Thus this case
> deserves special attention.
> 


As I said before, this appears to be a real corner case:
1. It requires one to use Leap-of-faith init, anything else and the init
requires AVPF to work well.

2. Any sender based scheme congestion control algorithm needs rapid
report of CE marks, so it needs to be receiver based.

3. It needs to have a back channel that isn't RTCP based. I don't know
if anything outside of AMR, AMR-WB, EVRC and SMV has such mechanism.

I guess there are some points in that under the above restrictions it
isn't motivated by this proposal to require AVPF. We might still do it
because it is simpler to do and it is such a corner case.

>> But, in 3GPP I would argue that AVPF is still needed in this case as
> you
>> have an APP packet for adjusting redundancy or number of frames in the
>> packet.
> 
> This will not be used in response to ECT markings being received
> according to agreements in 3GPP SA4.
> In my understanding the rational is that ECT markings in 3GPP will be
> set in the radio network where IP header compression is in place. Here,
> changing the codec mode reduces the bandwidth considerably without extra
> delay.
> 

So, what is written in Section 7.3.2 in 26.114 version 9.4.0 at the end
of this section is not what is going to be used? So are you backtracking
that Rel 9 decision in Rel 10?

"For speech, RTCP APP packets are used for adaptation (see clause 10.2)."

Don't that include redundancy signalling also? Which do matter
independently if one uses cellular network with header compression or not.

>> I will look into the formulations here. If there is a large number of
>> things that are needed to be fulfilled before AVPF is not requried, it
>> is simpler to require it always. But, I will look into that we have
>> accurate statements in the draft.
> 
> Thanks a lot.
> My current proposal is to link it to the ECN RTCP FB message. Obviously
> initialisation methods using the ECN FB message then will also require
> AVPF, and this can be stated in addition.

> In a sense particular application specific feedback is out-of-scope of
> the draft but a NOTE stating that many application specific feedback
> methods also require AVPF may be helpful.

I will consider this in the editing pass and come back on this change
proposal.


Maybe we should spare AVT WG from the 3GPP SA4 discussion on this other
topic, related to if MTSI can use the above restrictions and if that
makes sense. I will however state my view here once:

1. MTSI uses RTCP for its bit-rate adaptation feedback, thus needs AVPF
even for speech only sessions.

2. AVPF has low cost and provides benefits for any RTCP handling. The
fact that one has explicit configuration of TRR-Int and the possibility
to use early mode to send important events is a great benefit. It
produces more rapid feedback when needed, and can lower the RTCP
bit-rate actually consumed.

3. Considering that you SA4 already has agreed to use AVPF in Rel9 it
seems strange to remove it in some cases in Rel 10.


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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Date: Mon, 24 Jan 2011 16:14:41 -0500
Thread-Topic: Errata report on erratum 2673 on RFC 4695 ("RTP Payload Format for MIDI") and erratum 2119 on RFC 5506 ("Support for Reduced-Size RTCP: Opportunities and Consequences")
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Subject: [AVT] Errata report on erratum 2673 on RFC 4695 ("RTP Payload Format for MIDI") and erratum 2119 on RFC 5506 ("Support for Reduced-Size RTCP: Opportunities and Consequences")
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=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
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=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
RFC4695, "RTP Payload Format for MIDI"
Source of RFC: avt (rai)

Errata ID: 2673

Status: Reported
Type: Technical

Reported By: John Lazzaro
Date Reported: 2010-12-18

Section 3 says:

   If the LEN header field is nonzero, the MIDI list has the structure
   shown in Figure 3.

      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |  Delta Time 0     (1-4 octets long, or 0 octets if Z =3D 1)     |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |  MIDI Command 0   (1 or more octets long)                     |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |  Delta Time 1     (1-4 octets long)                           |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |  MIDI Command 1   (1 or more octets long)                     |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |                              ...                              |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |  Delta Time N     (1-4 octets long)                           |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |  MIDI Command N   (0 or more octets long)                     |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

                       Figure 3 -- MIDI list structure

It should say:

   If the LEN header field is nonzero, the MIDI list has the structure
   shown in Figure 3.

      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |  Delta Time 0     (1-4 octets long, or 0 octets if Z =3D 0)     |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |  MIDI Command 0   (1 or more octets long)                     |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |  Delta Time 1     (1-4 octets long)                           |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |  MIDI Command 1   (1 or more octets long)                     |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |                              ...                              |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |  Delta Time N     (1-4 octets long)                           |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |  MIDI Command N   (0 or more octets long)                     |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

                       Figure 3 -- MIDI list structure

Notes:

The sole change here is in the parenthesized text that follows
the "Delta Time 0" name in the first bitfield shown in Figure 3.
RFC 4695 has "1-4 octets long, or 0 octets if Z =3D 1". This
does not match the definition of Z in the main text. The corrected
figure text is: "1-4 octets long, or 0 octets if Z =3D 0". Thankfully, all
known implementations follow the main text in this
regard, so this bug should not have resulted in interoperability
issues. This bug has been fixed in the bis I-D intended to
obsolete RFC 4695 that is currently in Last Call in AVT.
----------------------------------------------------------------------
Recommended status:  (correct) Hold for document update

The erratum appears to be correct, and given the submitter is an
author of both RFC 4695 and draft-ietf-avt-rfc4695-bis, there is no
reason to doubt that.  I recommend "Hold for document update" only
because a -bis I-D is in WGLC, otherwise "Publish" would be needed.
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
RFC5506, "Support for Reduced-Size RTCP: Opportunities and Consequences"
Source of RFC: avt (rai)

Errata ID: 2119

Status: Reported
Type: Editorial

Reported By: Alfred Hoenes
Date Reported: 2010-04-05

Section 3.4.5 says:

|  o  Compression of RTCP: No IETF-defined header compression method
      compress RTCP; however, if such methods are developed in the
|     future, these methods must take Reduced-Size RTCP in account.


It should say:

|  o  Compression of RTCP: No IETF-defined header compression methods
      compress RTCP; however, if such methods are developed in the
|     future, these methods must take Reduced-Size RTCP into account.


Notes:

Rationale: singular/plural mismatch; language improvement.
----------------------------------------------------------------------
Recommended status:  (correct) Hold for document update

Or one might correct the text to "No IETF-defined header compression
>>method compresses<< RTCP...".
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
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=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

Dale

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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.


	Title           : RTP Payload Format for MIDI
	Author(s)       : J. Lazzaro, J.  Wawrzynek
	Filename        : draft-ietf-avt-rfc4695-bis-10.txt
	Pages           : 171
	Date            : 2011-01-24

This memo describes a Real-time Transport Protocol (RTP) payload
format for the MIDI (Musical Instrument Digital Interface) command
language.  The format encodes all commands that may legally appear on
a MIDI 1.0 DIN cable.  The format is suitable for interactive
applications (such as network musical performance) and content-
delivery applications (such as file streaming).  The format may be
used over unicast and multicast UDP and TCP, and it defines tools for
graceful recovery from packet loss.  Stream behavior, including the
MIDI rendering method, may be customized during session setup.  The
format also serves as a mode for the mpeg4-generic format, to support
the MPEG 4 Audio Object Types for General MIDI, Downloadable Sounds
Level 2, and Structured Audio.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avt-rfc4695-bis-10.txt

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On Mon, Jan 24, 2011 at 3:15 PM,  <Internet-Drafts@ietf.org> wrote:
> A New Internet-Draft is available from the on-line Internet-Drafts direct=
ories.
>
> =A0 =A0 =A0 =A0Title =A0 =A0 =A0 =A0 =A0 : RTP Payload Format for MIDI
> =A0 =A0 =A0 =A0Author(s) =A0 =A0 =A0 : J. Lazzaro, J. =A0Wawrzynek
> =A0 =A0 =A0 =A0Filename =A0 =A0 =A0 =A0: draft-ietf-avt-rfc4695-bis-10.tx=
t


This fixes some IDnits, per Roni's request.

--=20
John Lazzaro
http://www.cs.berkeley.edu/~lazzaro
john [dot] lazzaro [at] gmail [dot] com

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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.


	Title           : Guidelines for Choosing RTP Control Protocol (RTCP) Canonical Names (CNAMEs)
	Author(s)       : A. Begen, et al.
	Filename        : draft-ietf-avt-rtp-cnames-05.txt
	Pages           : 10
	Date            : 2011-01-25

The RTP Control Protocol (RTCP) Canonical Name (CNAME) is a
persistent transport-level identifier for an RTP endpoint.  While the
Synchronization Source (SSRC) identifier of an RTP endpoint may
change if a collision is detected, or when the RTP application is
restarted, its RTCP CNAME is meant to stay unchanged, so that RTP
endpoints can be uniquely identified and associated with their RTP
media streams.  For proper functionality, RTCP CNAMEs should be
unique within the participants of an RTP session.  However, the
existing guidelines for choosing the RTCP CNAME provided in the RTP
standard are insufficient to achieve this uniqueness.  This memo
updates those guidelines to allow endpoints to choose unique RTCP
CNAMEs.

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http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-cnames-05.txt

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Subject: [AVT] WG Action: Audio/Video Transport Core Maintenence (AVTCORE)
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A new IETF working group has been formed in the Real-time Applications and
Infrastructure Area.  For additional information, please contact the Area
Directors or the WG Chairs.

Audio/Video Transport Core Maintenence (AVTCORE)
---------------------------------------------------
Current Status: Active Working Group

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

Real-time Applications and Infrastructure Area Directors:
  * Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>
  * Robert Sparks <rjsparks@nostrum.com>

Real-time Applications and Infrastructure Area Advisor:
  * Robert Sparks <rjsparks@nostrum.com>

Secretary:
  * Tom Taylor <tom111.taylor@bell.net>

Mailing Lists:
  General Discussion: avt@ietf.org
  To Subscribe: https://www.ietf.org/mailman/listinfo/avt
  Archive: http://www.ietf.org/mail-archive/web/avt/index.html


Description of Working Group:

The Real-time Transport Protocol, RTP, along with its associated
profiles and payload formats provides for real-time transmission of
audio and video over unicast and multicast transports. RTP itself has
been shepherded to Full Standard. Its associated profiles, extensions,
and payload formats are currently at various levels of standards
maturity.

The AVTCORE working group is chartered to maintain the core RTP/RTCP
specifications and the AVP, SAVP, AVPF, and SAVPF profiles. The group
will provide architectural guidance for extending the protocols and
guidelines for their proper use. While other working groups may be
chartered to work on application-specific extensions to the protocols,
extensions that are generally applicable will be developed in AVTCORE.

The AVTCORE working group will coordinate closely with the Security Area
while working on maintenance and enhancements to the SRTP Profile.

In addition to the milestones called out below, the AVTCORE working
group's initial tasks will include completing any remaining work
identified in those drafts from the AVT working group already in IESG
Evaluation, with the exception of the Rapid Acquisition of Multicast RTP
sessions, which will complete in the AVTEXT working group.

Goals and Milestones:

Done      Submit Guideline for Choosing RTCP CNAME as Proposed Standard
Done      Submit use of AES-192 and AES-256 in Secure RTP for Proposed 
          Standard
Jan 2011  Submit Port Mapping Between Unicast and Multicast RTP Sessions 
          for Proposed Standard
Feb 2011  Submit Monitoring Architecture for RTP for Informational
Feb 2011  Submit Guidelines for the use of Variable Bit Rate Audio with 
          Secure RTP as Informational (or possibly BCP)
Apr 2011  Submit in band keying mechanism for SRTP draft for Proposed 
          Standard
Apr 2011  Submit Explicit Congestion Notification (ECN) for RTP over UDP 
          for Proposed Standard
May 2011  Submit RTCP indication for retransmission suppression as 
          Proposed Standard
Sep 2011  Submit Encryption of Header Extensions in the Secure Real-Time 
          Transport Protocol (SRTP) for Proposed Standard
Dec 2011  Submit Real-Time Transport Control Protocol (RTCP) in Overlay 
          Multicast for Proposed Standard


From Even.roni@huawei.com  Wed Jan 26 00:39:13 2011
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Date: Wed, 26 Jan 2011 10:37:59 +0200
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Subject: [AVT] FW: WG Action: Audio/Video Transport Payloads (PAYLOAD)
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Hi,
Note the new payload WG.
If you are interested in this work please subscribe to the mailing list
Roni Even

-----Original Message-----
From: IESG Secretary [mailto:iesg-secretary@ietf.org] 
Sent: Wednesday, January 26, 2011 1:24 AM
To: IETF Announcement list
Cc: Even.roni@huawei.com; abegen@cisco.com; payload@ietf.org
Subject: WG Action: Audio/Video Transport Payloads (PAYLOAD)

A new IETF working group has been formed in the Real-time Applications and
Infrastructure Area.  For additional information, please contact the Area
Directors or the WG Chairs.

Audio/Video Transport Payloads (PAYLOAD)
---------------------------------------------------
Current Status: Active Working Group

Chairs:
  * Roni Even <even.roni@huawei.com>
  * Ali Begen  <abegen@cisco.com>

Real-time Applications and Infrastructure Area Director(s):
  * Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>
  * Robert Sparks <rjsparks@nostrum.com>

Real-time Applications and Infrastructure Area Advisor:
  * Robert Sparks <rjsparks@nostrum.com>

Mailing Lists:
  General Discussion: payload@ietf.org
  To Subscribe: https://www.ietf.org/mailman/listinfo/payload
  Archive: http://www.ietf.org/mail-archive/web/payload/index.html

Description of Working Group:

  The PAYLOAD working group is tasked with the specification and 
  maintenance of payload formats for use with the Real-Time Transport 
  Protocol, RTP [RFC3550]. The group will follow the guidelines 
  established in "Guidelines for Writers of RTP Payload Format 
  Specifications" [BCP 36], the "How to Write an RTP Payload Format" 
  (under development) and is responsible for maintaining those 
  guidelines.

  This working group will develop RTP payload formats for new media 
  codecs, review and revise existing payload formats to advance those 
  which are useful to Draft Standard or Standard, and declare others 
  Historic.

Goals and Milestones:

Jan 2011  Submit RTP Payload Format for MIDI for Proposed Standard
Feb 2011  Submit How to Write an RTP Payload Format for Informational
Feb 2011  Submit RTP Payload Format for MPEG-4 Audio/Visual Streams for 
          Proposed Standard
Mar 2011  Submit RTP Payload Format for EVBR/G.718 for Proposed Standard
Mar 2011  Submit RTP Payload Format for Enhanced Variable Rate 
          Narrowband-Wideband Codec (EVRC-NW) for Proposed Standard
Mar 2011  Submit RTP Payload Format for Bluetooth's SBC audio codec for 
          Proposed Standard
Apr 2011  Submit RTP Payload Format for MPEG2-TS preamble for Proposed 
          Standard
Apr 2011  Submit RTP Payload Format for DV (IEC 61834) Video for 
          Proposed Standard
Apr 2011  Submit RTP Payload Format for the iSAC codec for Proposed 
          Standard
Apr 2011  Submit RTP profile for the carriage of SMPTE 336M data for 
          Proposed Standard
Jun 2011  Submit RTP Payload Format for MVC Video for Proposed Standard


From Even.roni@huawei.com  Wed Jan 26 00:40:13 2011
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Date: Wed, 26 Jan 2011 10:39:01 +0200
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Subject: [AVT] FW: WG Action: Metric Blocks for use with RTCP's Extended Report Framework (XRBLOCK)
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Hi,
Please note the new XRblock WG.
If you have interest in this work please join the mailing list
Roni Even

-----Original Message-----
From: IESG Secretary [mailto:iesg-secretary@ietf.org] 
Sent: Wednesday, January 26, 2011 1:24 AM
To: IETF Announcement list
Cc: Even.roni@huawei.com; Peter.Musgrave@magorcorp.com; xrblock@ietf.org
Subject: WG Action: Metric Blocks for use with RTCP's Extended Report Framework (XRBLOCK)

A new IETF working group has been formed in the Real-time Applications and
Infrastructure Area.  For additional information, please contact the Area
Directors or the WG Chairs.

Metric Blocks for use with RTCP's Extended Report Framework (XRBLOCK)
---------------------------------------------------
Current Status: Active Working Group

Chairs:
  * Roni Even <even.roni@huawei.com>
  * Peter Musgrave <Peter.Musgrave@magorcorp.com>

Real-time Applications and Infrastructure Area Director(s):
  * Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>
  * Robert Sparks <rjsparks@nostrum.com>

Real-time Applications and Infrastructure Area Advisor:
  * Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>

Mailing Lists:
  General Discussion: xrblock@ietf.org
  To Subscribe: https://www.ietf.org/mailman/listinfo/xrblock
  Archive: http://www.ietf.org/mail-archive/web/xrblock/index.html

Description of Working Group:

RFC3611, "RTP Control Protocol Extended Reports (RTCP XR)" established
a framework to allow new information to be conveyed in RTCP,
supplementing the original report blocks defined in RFC 3550, "RTP: A
Transport Protocol for Real-Time Applications".

The XRBLOCK working group is chartered to work within this framework,
evaluating proposals for report block definitions containing new
metrics. The group will follow the guidelines established in RFC5968,
"Guidelines for Extending the RTP Control Protocol (RTCP)" and RTP
Monitoring Architecture being developed in the AVTCORE working group.
The group will consider the guidelines for performance metric
development currently being discussed in the PMOL working group.

Goals and Milestones:

Mar 2011 Submit RTCP XR Report Block for Measurement Identity
Mar 2011 Submit RTCP XR Report Block for Burst/Gap Discard metric 
         Reporting
Mar 2011 Submit RTCP XR Report Block for Burst/Gap Loss metric Reporting
Jun 2011 Submit RTCP XR Report Block for Concealed Seconds metric 
         Reporting
Jun 2011 Submit RTCP XR Report Block for Delay metric Reporting
Jun 2011 Submit RTCP XR Report Block for Discard metric Reporting
Sep 2011 Submit RTCP XR Report Block for Jitter Buffer Metric Reporting
Sep 2011 Submit RTCP XR Report Block for Loss Concealment metric 
         Reporting
Sep 2011 Submit RTCP XR Report Block for Packet Delay Variation Metric 
         Reporting
Dec 2011 Submit RTCP XR Report Block for Run Length Encodings of 
         Discarded Packets
Dec 2011 Submit RTCP XR Report Block for QoE Metrics 
         Reporting                


From keith.drage@alcatel-lucent.com  Wed Jan 26 02:11:17 2011
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Please note the new WG. This is one of four created out of the work of the =
AVT WG.

Noone has been automatically transferred to the new list so please subscrib=
e if you are interested in this continuing work.

Regards

Keith

-----Original Message-----
From: IESG Secretary [mailto:iesg-secretary@ietf.org]=20
Sent: 25 January 2011 23:23
To: IETF Announcement list
Cc: DRAGE, Keith (Keith); magnus.westerlund@ericsson.com; avtext@ietf.org
Subject: WG Action: Audio/Video Transport Extensions (AVTEXT)=20

A new IETF working group has been formed in the Real-time Applications and
Infrastructure Area.  For additional information, please contact the Area
Directors or the WG Chairs.

Audio/Video Transport Extensions (AVTEXT)
---------------------------------------------------
Current Status: Active Working Group

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

Real-time Applications and Infrastructure Area Director(s):
  * Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>
  * Robert Sparks <rjsparks@nostrum.com>

Real-time Applications and Infrastructure Area Advisor:
  * Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>

Mailing Lists:
  General Discussion: avtext@ietf.org
  To Subscribe: https://www.ietf.org/mailman/listinfo/avtext
  Archive: http://www.ietf.org/mail-archive/web/avtext/index.html

Description of Working Group:

The Full-Standard Real-time Transport Protocol, RTP [RFC 3550], along
with its associated profiles and payload formats provides for real-
time transmission of audio and video over unicast and multicast
transports. RTP is widely implemented, and deployed for a wide range
of applications, ranging from telephony to television over IP. As new
applications have emerged, the need for guidelines for the use of the
RTP/RTCP protocols and extensions specific to those applications
has been identified.

The AVTEXT working group is charted to develop application-specific
guidelines for the use of RTP/RTCP protocols with the AVP, SAVP,
AVPF, and SAVPF profiles, and extensions to those protocols that are
driven by application-specific needs. Proposals for extensions with
general applicability to many different RTP/RTCP usages should be
taken to the AVTCORE working group.

The AVTEXT working group is constrained to use the protocol extension
mechanisms defined in the core protocols (such as RTP Header
extensions [RFC5285], AVPF Feedback Messages [RFC4585], and
SDES Items [RFC3550]). If new ways to extend the core protocols are
needed, they will be developed in the AVTCORE working group.

In addition to the milestones called out below, the AVTEXT working
group's initial tasks will include completing any new work identified
during IESG evaluation for the Rapid Acquisition of Multicast RTP
Sessions.

Goals and Milestones:

Jun 2011  Submit Considerations for RAMS Scenarios for Informational
Oct 2011  Submit RTP Header extension for mixer to client audio level=20
          indication as Proposed Standard
Oct 2011  Submit RTP Header extension for client to mixer audio level=20
          indication as proposed standard
Dec 2011  Submit Support for multiple clock rates in an RTP session for=20
          Proposed Standard
Dec 2011  Submit SRTP Store-and-Forward Use Cases and Requirements for=20
          Informational
Dec 2011  Submit Use of the Secure Real-time Transport Protocol (SRTP)=20
          in Store-and-Forward Applications for Proposed Standard

From magnus.westerlund@ericsson.com  Wed Jan 26 07:31:47 2011
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Hi,

Here is a first email about how a some of the things is intended to work
going forward with the split of AVT.

Usage of email lists: As this list is now AVTcore's, we intended to be
strict on referring topics that doesn't belong to AVTcore to the other
WG's lists. So please go subscribe to the lists for the WG that you are
interested in.

AVTEXT:  https://www.ietf.org/mailman/listinfo/avtext
PAYLOAD: https://www.ietf.org/mailman/listinfo/payload
XRBLOCK: https://www.ietf.org/mailman/listinfo/xrblock

Document names: Soon, we will get the existing AVT documents assigned in
the database to the right WG. However, that is not 100% robust and does
not allow for determining the WG from the file name. Thus we intended to
get most document to change the WG name from AVT to the assigned WG as
they are updated. However, this MUST NOT be done for documents that
already have been publication requested. We don't want to confuse the IESG.

If you are an editor and are in doubt, ask your chair. If you don't know
which WG your document belongs to ask a WG chair.

Prague Meeting: The WG chairs for the WGs are discussing how we will do
these first meeting. More information will come.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
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From iesg-secretary@ietf.org  Wed Jan 26 11:35:40 2011
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Subject: [AVT] Protocol Action: 'Guidelines for Choosing RTP Control Protocol (RTCP)	Canonical Names (CNAMEs)' to Proposed Standard	(draft-ietf-avt-rtp-cnames-05.txt)
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The IESG has approved the following document:
- 'Guidelines for Choosing RTP Control Protocol (RTCP) Canonical Names
   (CNAMEs)'
  (draft-ietf-avt-rtp-cnames-05.txt) as a Proposed Standard

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

The IESG contact persons are Robert Sparks and Gonzalo Camarillo.

A URL of this Internet Draft is:
http://datatracker.ietf.org/doc/draft-ietf-avt-rtp-cnames/




Technical Summary

The RTP Control Protocol (RTCP) Canonical Name (CNAME) is a persistent
transport-level identifier for an RTP endpoint.  While the
Synchronization Source (SSRC) identifier of an RTP endpoint may change
if a collision is detected, or when the RTP application is restarted,
its RTCP CNAME is meant to stay unchanged, so that RTP endpoints can be
uniquely identified and associated with their RTP media streams.  For
proper functionality, RTCP CNAMEs should be unique within the
participants of an RTP session.  However, the existing guidelines for
choosing the RTCP CNAME provided in the RTP standard are insufficient
to achieve this uniqueness.  This memo updates these guidelines to
allow endpoints to choose unique RTCP CNAMEs.



Working Group Summary

The document achieved consensus in the AVT working group. 

Personnel

Keith Drage is the document shepherd. Robert Sparks is the responsible AD


From magnus.westerlund@ericsson.com  Thu Jan 27 01:26:14 2011
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Subject: Re: [AVT] Guidelines for the use of Variable Bit Rate Audio with Secure	RTP
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DRAGE, Keith (Keith) skrev 2011-01-08 19:08:
> (As WG co-chair)
> 
> In support of the milestone:
> 
> Feb 2011  Guidelines for the use of Variable Bit Rate Audio with Secure RTP as informational (or possibly BCP)
> 
> This is a WG consensus call to adopt all (or some) of the text of:
> 
> http://www.ietf.org/id/draft-perkins-avt-srtp-vbr-audio-05.txt
> 
> to fulfil that milestone. At this stage the text does not have to be regarded perfect, the question is more "Does this text form a better basis for completing the work in the working group that some text yet to be submitted?"
> 
> Please respond by Monday 18th January with your positions.
> 
> While the prime intention of this call is to adopt the entire text of the author draft, if you consider that some of the text is appropriate, but that some of the text does not form a reasonable start for the work, then please indicate that text and we will make a separate call on it (and adopt the rest).

There has been no comments on this call for adoption. Thus we will take
draft-perkins-avt-srtp-vbr-audio as basis for the WG document. We also
appoint authors of this document as editors.

Editors, please submit a new version of the document as AVTcore WG
document, i.e. draft-ietf-avtcore-srtp-vbr-audio

Regards

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
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From Internet-Drafts@ietf.org  Thu Jan 27 06:30:02 2011
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Core Maintenance Working Group of the IETF.


	Title           : Guidelines for the use of Variable Bit Rate Audio with Secure RTP
	Author(s)       : C. Perkins, J. Valin
	Filename        : draft-ietf-avtcore-srtp-vbr-audio-00.txt
	Pages           : 7
	Date            : 2011-01-27

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-00.txt

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From keith.drage@alcatel-lucent.com  Thu Jan 27 06:53:28 2011
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: IETF AVT WG <avt@ietf.org>, "avtext@ietf.org" <avtext@ietf.org>
Date: Thu, 27 Jan 2011 15:56:19 +0100
Thread-Topic: Call for adoption of text for audio level milestones
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Subject: [AVT] Call for adoption of text for audio level milestones
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(As AVTEXT WG cochair)

This message is copied to the AVTCORE list for those who are not yet on the=
 new list.

AVTEXT has two milestones relating to audio levels.

Oct 2011  Submit RTP Header extension for mixer to client audio level indic=
ation as Proposed Standard=20
=20
Oct 2011  Submit RTP Header extension for client to mixer audio level indic=
ation as proposed standard =20

We have the following author drafts providing candidate text for these mile=
stones:

A Real-Time Transport Protocol (RTP) Extension Header for Mixer-to-client A=
udio Level Indication
http://www.ietf.org/id/draft-ivov-avt-slic-04.txt

A Real-Time Transport Protocol (RTP) Header Extension for Client-to-Mixer A=
udio Level Indication
http://tools.ietf.org/id/draft-lennox-avt-rtp-audio-level-exthdr-02.txt

respectively.

This is a WG consensus call to adopt the above text as baseline text for th=
ese milestones. At this stage the text does not have to be regarded perfect=
, the question is more "Does this text form a better basis for completing t=
he work in the working group that some text yet to be submitted?"

While the prime intention of this call is to adopt the entire text of the a=
uthor draft, if you consider that some of the text is appropriate, but that=
 some of the text does not form a reasonable start for the work, then pleas=
e indicate that text and we will make a separate call on it (and adopt the =
rest).

Please respond by Thursday 10th February 2011 with your positions.=20

######### Please respond to the AVTEXT list only #############

Regards

Keith


From magnus.westerlund@ericsson.com  Thu Jan 27 06:54:20 2011
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Hi,

Additional things to consider:

Submission of individual documents: The submission of an individual
document will now require a bit of thinking from the authors. They
should match the documents content against the different charters. And
if you are in doubt about where to send a new draft, please contact any
of the chairs and they will help you.

Cheers

Magnus


Magnus Westerlund skrev 2011-01-26 16:34:
> Hi,
> 
> Here is a first email about how a some of the things is intended to work
> going forward with the split of AVT.
> 
> Usage of email lists: As this list is now AVTcore's, we intended to be
> strict on referring topics that doesn't belong to AVTcore to the other
> WG's lists. So please go subscribe to the lists for the WG that you are
> interested in.
> 
> AVTEXT:  https://www.ietf.org/mailman/listinfo/avtext
> PAYLOAD: https://www.ietf.org/mailman/listinfo/payload
> XRBLOCK: https://www.ietf.org/mailman/listinfo/xrblock
> 
> Document names: Soon, we will get the existing AVT documents assigned in
> the database to the right WG. However, that is not 100% robust and does
> not allow for determining the WG from the file name. Thus we intended to
> get most document to change the WG name from AVT to the assigned WG as
> they are updated. However, this MUST NOT be done for documents that
> already have been publication requested. We don't want to confuse the IESG.
> 
> If you are an editor and are in doubt, ask your chair. If you don't know
> which WG your document belongs to ask a WG chair.
> 
> Prague Meeting: The WG chairs for the WGs are discussing how we will do
> these first meeting. More information will come.
> 
> Cheers
> 
> Magnus Westerlund
> 
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> Färögatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
> _______________________________________________
> Audio/Video Transport Working Group
> 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 peter.musgrave@magorcorp.com  Thu Jan 27 06:58:33 2011
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This text is a good starting point.=20

Peter

On 2011-01-27, at 9:56 AM, DRAGE, Keith (Keith) wrote:

> (As AVTEXT WG cochair)
>=20
> This message is copied to the AVTCORE list for those who are not yet =
on the new list.
>=20
> AVTEXT has two milestones relating to audio levels.
>=20
> Oct 2011  Submit RTP Header extension for mixer to client audio level =
indication as Proposed Standard=20
>=20
> Oct 2011  Submit RTP Header extension for client to mixer audio level =
indication as proposed standard =20
>=20
> We have the following author drafts providing candidate text for these =
milestones:
>=20
> A Real-Time Transport Protocol (RTP) Extension Header for =
Mixer-to-client Audio Level Indication
> http://www.ietf.org/id/draft-ivov-avt-slic-04.txt
>=20
> A Real-Time Transport Protocol (RTP) Header Extension for =
Client-to-Mixer Audio Level Indication
> =
http://tools.ietf.org/id/draft-lennox-avt-rtp-audio-level-exthdr-02.txt
>=20
> respectively.
>=20
> This is a WG consensus call to adopt the above text as baseline text =
for these milestones. At this stage the text does not have to be =
regarded perfect, the question is more "Does this text form a better =
basis for completing the work in the working group that some text yet to =
be submitted?"
>=20
> While the prime intention of this call is to adopt the entire text of =
the author draft, if you consider that some of the text is appropriate, =
but that some of the text does not form a reasonable start for the work, =
then please indicate that text and we will make a separate call on it =
(and adopt the rest).
>=20
> Please respond by Thursday 10th February 2011 with your positions.=20
>=20
> ######### Please respond to the AVTEXT list only #############
>=20
> Regards
>=20
> Keith
>=20
> _______________________________________________
> avtext mailing list
> avtext@ietf.org
> https://www.ietf.org/mailman/listinfo/avtext


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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.


	Title           : AES-GCM and AES-CCM Authenticated Encryption in Secure RTP (SRTP)
	Author(s)       : D. McGrew
	Filename        : draft-ietf-avt-srtp-aes-gcm-01.txt
	Pages           : 15
	Date            : 2011-01-27

This document defines how AES-GCM, AES-CCM, and other Authenticated
Encryption with Associated Data (AEAD) algorithms, can be used to
provide confidentiality and data authentication mechanisms in the
SRTP protocol.

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From ray.vanbrandenburg@tno.nl  Fri Jan 28 03:00:14 2011
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From: "Brandenburg, R. (Ray) van" <ray.vanbrandenburg@tno.nl>
To: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
Thread-Topic: [avtext] Call for adoption of text for audio level milestones
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I agree.=20

Ray van Brandenburg=20

-----Original Message-----
From: avtext-bounces@ietf.org [mailto:avtext-bounces@ietf.org] On Behalf Of=
 Peter Musgrave
Sent: donderdag 27 januari 2011 16:01
To: DRAGE, Keith (Keith)
Cc: avtext@ietf.org; IETF AVT WG
Subject: Re: [avtext] Call for adoption of text for audio level milestones

This text is a good starting point.=20

Peter

On 2011-01-27, at 9:56 AM, DRAGE, Keith (Keith) wrote:

> (As AVTEXT WG cochair)
>=20
> This message is copied to the AVTCORE list for those who are not yet on t=
he new list.
>=20
> AVTEXT has two milestones relating to audio levels.
>=20
> Oct 2011  Submit RTP Header extension for mixer to client audio level=20
> indication as Proposed Standard
>=20
> Oct 2011  Submit RTP Header extension for client to mixer audio level=20
> indication as proposed standard
>=20
> We have the following author drafts providing candidate text for these mi=
lestones:
>=20
> A Real-Time Transport Protocol (RTP) Extension Header for=20
> Mixer-to-client Audio Level Indication=20
> http://www.ietf.org/id/draft-ivov-avt-slic-04.txt
>=20
> A Real-Time Transport Protocol (RTP) Header Extension for=20
> Client-to-Mixer Audio Level Indication=20
> http://tools.ietf.org/id/draft-lennox-avt-rtp-audio-level-exthdr-02.tx
> t
>=20
> respectively.
>=20
> This is a WG consensus call to adopt the above text as baseline text for =
these milestones. At this stage the text does not have to be regarded perfe=
ct, the question is more "Does this text form a better basis for completing=
 the work in the working group that some text yet to be submitted?"
>=20
> While the prime intention of this call is to adopt the entire text of the=
 author draft, if you consider that some of the text is appropriate, but th=
at some of the text does not form a reasonable start for the work, then ple=
ase indicate that text and we will make a separate call on it (and adopt th=
e rest).
>=20
> Please respond by Thursday 10th February 2011 with your positions.=20
>=20
> ######### Please respond to the AVTEXT list only #############
>=20
> Regards
>=20
> Keith
>=20
> _______________________________________________
> avtext mailing list
> avtext@ietf.org
> https://www.ietf.org/mailman/listinfo/avtext

_______________________________________________
avtext mailing list
avtext@ietf.org
https://www.ietf.org/mailman/listinfo/avtext
This e-mail and its contents are subject to the DISCLAIMER at http://www.tn=
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Core Maintenance Working Group of the IETF.


	Title           : Explicit Congestion Notification (ECN) for RTP over UDP
	Author(s)       : M. Westerlund, et al.
	Filename        : draft-ietf-avtcore-ecn-for-rtp-00.txt
	Pages           : 52
	Date            : 2011-01-28

This document specifies how explicit congestion notification (ECN)
can be used with RTP/UDP flows that use RTCP as feedback mechanism.
It defines one RTCP XR extension for ECN summary, a RTCP transport
feedback format for timely reporting of congestion events, and an
STUN extension used in the optional initilization method using ICE.
Signalling and procedures for negotiation of capabilities and
initilization methods are also defined.

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From magnus.westerlund@ericsson.com  Fri Jan 28 06:09:06 2011
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Subject: [AVT] WG adoption to meet milestone for indication of retransmission supression
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WG,

This is a call for adopting "RTCP Extension for Feedback Suppression
Indication" as the starting point for meeting our milestone. The draft
is available here:
https://datatracker.ietf.org/doc/draft-wu-avt-retransmission-supression-rtp/

The milestone:
May 2011 Submit RTCP indication for retransmission suppression as
Proposed Standard

Please provide any inut by Friday the 11th of February.

Regards

Magnus Westerlund

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

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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Core Maintenance Working Group of the IETF.


	Title           : Port Mapping Between Unicast and Multicast RTP Sessions
	Author(s)       : A. Begen, et al.
	Filename        : draft-ietf-avtcore-ports-for-ucast-mcast-rtp-00.txt
	Pages           : 36
	Date            : 2011-01-28

This document presents a port mapping solution that allows RTP
receivers to choose their own ports for an auxiliary unicast session
in RTP applications using both unicast and multicast services.  The
solution provides protection against denial-of-service or packet
amplification attacks that could be used to cause one or more RTP
packets to be sent to a victim client.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avtcore-ports-for-ucast-mcast-rtp-00.txt

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From abegen@cisco.com  Sat Jan 29 18:25:44 2011
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>, "IETF AVT WG" <avt@ietf.org>
X-OriginalArrivalTime: 30 Jan 2011 02:28:51.0007 (UTC) FILETIME=[6E7B04F0:01CBC025]
Subject: Re: [AVT] WG adoption to meet milestone for indication ofretransmission supression
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The draft is on the right track but some of the comments I provided =
earlier are still pending. We can them work them out during the document =
development.

-acbegen

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of =
Magnus Westerlund
> Sent: Friday, January 28, 2011 9:12 AM
> To: IETF AVT WG
> Subject: [AVT] WG adoption to meet milestone for indication =
ofretransmission supression
>=20
> WG,
>=20
> This is a call for adopting "RTCP Extension for Feedback Suppression
> Indication" as the starting point for meeting our milestone. The draft
> is available here:
> =
https://datatracker.ietf.org/doc/draft-wu-avt-retransmission-supression-r=
tp/
>=20
> The milestone:
> May 2011 Submit RTCP indication for retransmission suppression as
> Proposed Standard
>=20
> Please provide any inut by Friday the 11th of February.
>=20
> Regards
>=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
> ----------------------------------------------------------------------
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt

From gwz@net-zen.net  Sat Jan 29 23:53:34 2011
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From: "Glen Zorn" <gwz@net-zen.net>
To: "'Ali C. Begen \(abegen\)'" <abegen@cisco.com>
References: <4D42CEB8.1090806@ericsson.com> <04CAD96D4C5A3D48B1919248A8FE0D540E2D1D32@xmb-sjc-215.amer.cisco.com>
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Date: Sun, 30 Jan 2011 14:56:18 +0700
Organization: Network Zen
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Cc: 'IETF AVT WG' <avt@ietf.org>, avt-chairs@tools.ietf.org
Subject: Re: [AVT] WG adoption to meet milestone for indication	of retransmission supression
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Just to be clear, do you mean that you support the adoption of the =
document
as a WG item?

Hope this helps.

 ~gwz

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
> Ali C. Begen (abegen)
> Sent: Sunday, January 30, 2011 9:29 AM
> To: Magnus Westerlund; IETF AVT WG
> Subject: Re: [AVT] WG adoption to meet milestone for indication
> ofretransmission supression
>=20
> The draft is on the right track but some of the comments I provided
> earlier are still pending. We can them work them out during the =
document
> development.
>=20
> -acbegen
>=20
> > -----Original Message-----
> > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf =
Of
> Magnus Westerlund
> > Sent: Friday, January 28, 2011 9:12 AM
> > To: IETF AVT WG
> > Subject: [AVT] WG adoption to meet milestone for indication
> ofretransmission supression
> >
> > WG,
> >
> > This is a call for adopting "RTCP Extension for Feedback Suppression
> > Indication" as the starting point for meeting our milestone. The =
draft
> > is available here:
> > https://datatracker.ietf.org/doc/draft-wu-avt-retransmission-
> supression-rtp/
> >
> > The milestone:
> > May 2011 Submit RTCP indication for retransmission suppression as
> > Proposed Standard
> >
> > Please provide any inut by Friday the 11th of February.
> >
> > Regards
> >
> > 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 Working Group
> > avt@ietf.org
> > https://www.ietf.org/mailman/listinfo/avt
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt


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Date: Sun, 30 Jan 2011 18:16:48 +0900
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Subject: Re: [AVT] WG adoption to meet milestone for indication	of retransmission supression
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From gwz@net-zen.net  Sun Jan 30 02:04:38 2011
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To: "'Ali C. Begen \(abegen\)'" <abegen@cisco.com>
References: <4D42CEB8.1090806@ericsson.com> <04CAD96D4C5A3D48B1919248A8FE0D540E2D1D32@xmb-sjc-215.amer.cisco.com> <001201cbc053$30ebf620$92c3e260$@net> <97E01434-BE8D-4714-B5AF-460E6075ED27@cisco.com>
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Date: Sun, 30 Jan 2011 17:07:25 +0700
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Cc: 'IETF AVT WG' <avt@ietf.org>, avt-chairs@tools.ietf.org
Subject: Re: [AVT] WG adoption to meet milestone for indication	of retransmission supression
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Ali C. Begen (abegen) [mailto://abegen@cisco.com] writes:

> Sorry if it was not clear. Yes I do support adoption.

Maybe it _was_ clear & I'm just dense ;-).  In any case, thanks for =
humoring me.

>=20
> -acbegen
>=20
> On Jan 30, 2011, at 4:56 PM, "Glen Zorn" <gwz@net-zen.net> wrote:
>=20
> > Just to be clear, do you mean that you support the adoption of the
> document
> > as a WG item?
> >
> > Hope this helps.
> >
> > ~gwz
> >
> >> -----Original Message-----
> >> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf =
Of
> >> Ali C. Begen (abegen)
> >> Sent: Sunday, January 30, 2011 9:29 AM
> >> To: Magnus Westerlund; IETF AVT WG
> >> Subject: Re: [AVT] WG adoption to meet milestone for indication
> >> ofretransmission supression
> >>
> >> The draft is on the right track but some of the comments I provided
> >> earlier are still pending. We can them work them out during the
> document
> >> development.
> >>
> >> -acbegen
> >>
> >>> -----Original Message-----
> >>> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf
> Of
> >> Magnus Westerlund
> >>> Sent: Friday, January 28, 2011 9:12 AM
> >>> To: IETF AVT WG
> >>> Subject: [AVT] WG adoption to meet milestone for indication
> >> ofretransmission supression
> >>>
> >>> WG,
> >>>
> >>> This is a call for adopting "RTCP Extension for Feedback =
Suppression
> >>> Indication" as the starting point for meeting our milestone. The
> draft
> >>> is available here:
> >>> https://datatracker.ietf.org/doc/draft-wu-avt-retransmission-
> >> supression-rtp/
> >>>
> >>> The milestone:
> >>> May 2011 Submit RTCP indication for retransmission suppression as
> >>> Proposed Standard
> >>>
> >>> Please provide any inut by Friday the 11th of February.
> >>>
> >>> Regards
> >>>
> >>> Magnus Westerlund
> >>>
> >>> =
--------------------------------------------------------------------
> --
> >>> Multimedia Technologies, Ericsson Research EAB/TVM
> >>> =
--------------------------------------------------------------------
> --
> >>> Ericsson AB                | Phone  +46 10 7148287
> >>> F=C3=A4r=C3=B6gatan 6                | Mobile +46 73 0949079
> >>> SE-164 80 Stockholm, Sweden| mailto: =
magnus.westerlund@ericsson.com
> >>> =
--------------------------------------------------------------------
> --
> >>> _______________________________________________
> >>> Audio/Video Transport Working Group
> >>> avt@ietf.org
> >>> https://www.ietf.org/mailman/listinfo/avt
> >> _______________________________________________
> >> Audio/Video Transport Working Group
> >> avt@ietf.org
> >> https://www.ietf.org/mailman/listinfo/avt
> >


From gwz@net-zen.net  Sun Jan 30 02:07:47 2011
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To: "'Glen Zorn'" <gwz@net-zen.net>, "'Ali C. Begen \(abegen\)'" <abegen@cisco.com>
References: <4D42CEB8.1090806@ericsson.com>	<04CAD96D4C5A3D48B1919248A8FE0D540E2D1D32@xmb-sjc-215.amer.cisco.com>	<001201cbc053$30ebf620$92c3e260$@net>	<97E01434-BE8D-4714-B5AF-460E6075ED27@cisco.com> <003a01cbc065$80c90a40$825b1ec0$@net>
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Subject: Re: [AVT] WG adoption to meet milestone for indication	of	retransmission supression
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Glen Zorn [mailto://gwz@net-zen.net] writes:

> Ali C. Begen (abegen) [mailto://abegen@cisco.com] writes:
> 
> > Sorry if it was not clear. Yes I do support adoption.

BTW, so do I.

...


From Even.roni@huawei.com  Sun Jan 30 02:22:44 2011
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Date: Sun, 30 Jan 2011 12:21:20 +0200
From: Roni Even <Even.roni@huawei.com>
In-reply-to: <1C6EB1CB-3A15-4A74-9FDE-47EF194454D7@magorcorp.com>
To: 'Peter Musgrave' <peter.musgrave@magorcorp.com>, "'DRAGE, Keith (Keith)'" <keith.drage@alcatel-lucent.com>
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Cc: avtext@ietf.org, 'IETF AVT WG' <avt@ietf.org>
Subject: Re: [AVT] [avtext] Call for adoption of text for audio level milestones
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I agree
Roni Even

> -----Original Message-----
> From: avtext-bounces@ietf.org [mailto:avtext-bounces@ietf.org] On
> Behalf Of Peter Musgrave
> Sent: Thursday, January 27, 2011 5:01 PM
> To: DRAGE, Keith (Keith)
> Cc: avtext@ietf.org; IETF AVT WG
> Subject: Re: [avtext] Call for adoption of text for audio level
> milestones
> 
> This text is a good starting point.
> 
> Peter
> 
> On 2011-01-27, at 9:56 AM, DRAGE, Keith (Keith) wrote:
> 
> > (As AVTEXT WG cochair)
> >
> > This message is copied to the AVTCORE list for those who are not yet
> on the new list.
> >
> > AVTEXT has two milestones relating to audio levels.
> >
> > Oct 2011  Submit RTP Header extension for mixer to client audio level
> indication as Proposed Standard
> >
> > Oct 2011  Submit RTP Header extension for client to mixer audio level
> indication as proposed standard
> >
> > We have the following author drafts providing candidate text for
> these milestones:
> >
> > A Real-Time Transport Protocol (RTP) Extension Header for Mixer-to-
> client Audio Level Indication
> > http://www.ietf.org/id/draft-ivov-avt-slic-04.txt
> >
> > A Real-Time Transport Protocol (RTP) Header Extension for Client-to-
> Mixer Audio Level Indication
> > http://tools.ietf.org/id/draft-lennox-avt-rtp-audio-level-exthdr-
> 02.txt
> >
> > respectively.
> >
> > This is a WG consensus call to adopt the above text as baseline text
> for these milestones. At this stage the text does not have to be
> regarded perfect, the question is more "Does this text form a better
> basis for completing the work in the working group that some text yet
> to be submitted?"
> >
> > While the prime intention of this call is to adopt the entire text of
> the author draft, if you consider that some of the text is appropriate,
> but that some of the text does not form a reasonable start for the
> work, then please indicate that text and we will make a separate call
> on it (and adopt the rest).
> >
> > Please respond by Thursday 10th February 2011 with your positions.
> >
> > ######### Please respond to the AVTEXT list only #############
> >
> > Regards
> >
> > Keith
> >
> > _______________________________________________
> > avtext mailing list
> > avtext@ietf.org
> > https://www.ietf.org/mailman/listinfo/avtext
> 
> _______________________________________________
> avtext mailing list
> avtext@ietf.org
> https://www.ietf.org/mailman/listinfo/avtext


From magnus.westerlund@ericsson.com  Mon Jan 31 04:44:49 2011
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Subject: Re: [AVT] FW: New Version Notification for	draft-ietf-avtcore-ports-for-ucast-mcast-rtp-00
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Hi,

I have reviewed these changes and think they address all the issues I
had raised.

Cheers

Magnus

Ali C. Begen (abegen) skrev 2011-01-28 17:05:
> Following the wglc, we revised the token draft (and renamed it as well). Here is the new spec:
> https://datatracker.ietf.org/doc/draft-ietf-avtcore-ports-for-ucast-mcast-rtp/
> 
> For diff, please use:
> http://tools.ietf.org/rfcdiff?url1=http://www.ietf.org/id/draft-ietf-avt-ports-for-ucast-mcast-rtp-11.txt&url2=http://www.ietf.org/id/draft-ietf-avtcore-ports-for-ucast-mcast-rtp-00.txt
> 
> We hope this addressed all the comments received so far and we are ready to ship this.
> 
> Cheers,
> -acbegen
> 
> -----Original Message-----
> From: IETF I-D Submission Tool [mailto:idsubmission@ietf.org] 
> Sent: Friday, January 28, 2011 10:55 AM
> To: Ali C. Begen (abegen)
> Cc: Dan Wing (dwing); Tom.Van_Caenegem@alcatel-lucent.com
> Subject: New Version Notification for draft-ietf-avtcore-ports-for-ucast-mcast-rtp-00 
> 
> 
> A new version of I-D, draft-ietf-avtcore-ports-for-ucast-mcast-rtp-00.txt has been successfully submitted by Ali Begen and posted to the IETF repository.
> 
> Filename:	 draft-ietf-avtcore-ports-for-ucast-mcast-rtp
> Revision:	 00
> Title:		 Port Mapping Between Unicast and Multicast RTP Sessions
> Creation_date:	 2011-01-28
> WG ID:		 avtcore
> Number_of_pages: 36
> 
> Abstract:
> This document presents a port mapping solution that allows RTP
> receivers to choose their own ports for an auxiliary unicast session
> in RTP applications using both unicast and multicast services.  The
> solution provides protection against denial-of-service or packet
> amplification attacks that could be used to cause one or more RTP
> packets to be sent to a victim client.
>                                                                                   
> 
> 
> The IETF Secretariat.
> 
> 
> _______________________________________________
> Audio/Video Transport Working Group
> 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 Even.roni@huawei.com  Mon Jan 31 05:32:35 2011
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Date: Mon, 31 Jan 2011 15:31:11 +0200
From: Roni Even <Even.roni@huawei.com>
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Subject: [AVTCORE] Short WGLC on Port Mapping Between Unicast and Multicast RTP Sessions
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Hi,

I would like to start a one week working group last call on
http://tools.ietf.org/html/draft-ietf-avtcore-ports-for-ucast-mcast-rtp-00 .

This draft finished a WGLC in the AVT WG before the restructure and the new
00 revision reflects that changes that were done based on email discussion.

 

Please review the draft and send any comment to the AVTCore mailing list
(avt@ietf.org) till the end of Monday February 7th 2011.

Thanks

Roni Even

AVTCore co chair 


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class=3DMsoNormal>Hi,<o:p></o:p></p><p class=3DMsoNormal>I would like to =
start a one week working group last call on <a =
href=3D"http://tools.ietf.org/html/draft-ietf-avtcore-ports-for-ucast-mca=
st-rtp-00">http://tools.ietf.org/html/draft-ietf-avtcore-ports-for-ucast-=
mcast-rtp-00</a> .<o:p></o:p></p><p class=3DMsoNormal>This draft =
finished a WGLC in the AVT WG before the restructure and the new 00 =
revision reflects that changes that were done based on email =
discussion.<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>Please review the draft and send any comment to the =
AVTCore mailing list (<a href=3D"mailto:avt@ietf.org">avt@ietf.org</a>) =
till the end of Monday February 7<sup>th</sup> 2011.<o:p></o:p></p><p =
class=3DMsoNormal>Thanks<o:p></o:p></p><p class=3DMsoNormal>Roni =
Even<o:p></o:p></p><p class=3DMsoNormal>AVTCore co chair =
<o:p></o:p></p></div></body></html>=

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