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From: Bernard Aboba <bernard_aboba@hotmail.com>
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Date: Wed, 2 Jan 2013 07:24:27 -0800
To: Eric Rescorla <ekr@rtfm.com>
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Subject: Re: [MMUSIC] [rtcweb] Proposed Agenda for Interim
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This is a good agenda - for an MMUSIC WG interim (only day 3 has any RTCWEB i=
tems on it).

Since there is a notice requirement, can we get MMUSIC on board so the inter=
im can actually take place?

On Jan 2, 2013, at 7:13, "Eric Rescorla" <ekr@rtfm.com> wrote:

> Chairs,
>=20
> I've spent some time thinking about the agenda for the interim, and
> thought it might be helpful if I sent you a strawman based on the
> items that are blocking progress in terms of implementation and
> interoperability.
>=20
> Note:
> I am assuming 4 hours each day. The times have been selected with
> the objective of giving us enough time to really resolve the
> open issues. Honestly, we might even add more time for
> the issues on day 1.
>=20
> Hope this is helpful.
>=20
> Day 1
> 2:00    Bundle   [Holmberg]
> 	Objective: select an approach and work out enough
> 	details that the editors can work to produce a
> 	suitable draft.
>=20
> 	Drafts:
> 	- http://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiati=
on
> 	- http://tools.ietf.org/html/draft-alvestrand-one-rtp
> 	- http://tools.ietf.org/html/draft-holmberg-mmusic-sdp-mmt-negotiat=
ion
>=20
>=20
> 2:00    Multiple flow SDP representation [Jennings, Alvestrand]
> 	Objective: decide on an architecture for large numbers
> 	of similar streams (e.g., multiple video streams).
>=20
> 	Drafts:
> 	- http://tools.ietf.org/html/draft-alvestrand-rtcweb-msid
> 	- http://tools.ietf.org/html/draft-alvestrand-mmusic-msid
> 	- Do we have a "master" document for the single m-line case?
> 	- Multiple m-lines doesn't have a document
>=20
>=20
> Day 2
> 1:30    Mapping stream/track label concepts SDP identifiers [Alvestrand]
> 	Objective: determine requirements
> =09
> =09
> 1:30    Trickle ICE open issues [Ivov(?), Uberti, Rescorla]
> =09
> 	- Starting checks
> 	- Termination conditions
> 	- Error handling
> =09
> 	Drafts:
> 	http://tools.ietf.org/html/draft-rescorla-mmusic-ice-trickle	=
=09
>=20
> 1:00    SDP for Datachannels open issues [Jesup, Loreto]
> 	Objective: resolve open issues (post WGLC?)
>=20
> 	Drafts:
> 	http://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-02
>=20
> Day 3
> 1:00    Other recommended audio codecs [Roach]
> =09
> 	Objective: identify what other audio codecs we intend to
> 	recommend.=09
>=20
> 1:00    Process WGLC comments on documents [authors]
>=20
> 0:30    Other business
>=20
> 0:30    Wrapup/next steps [Chairs]
>=20
>=20
> Best,
> -Ekr
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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<html><head><meta http-equiv="content-type" content="text/html; charset=utf-8"></head><body dir="auto"><div>This is a good agenda - for an MMUSIC WG interim (only day 3 has any RTCWEB items on it).</div><div><br></div><div>Since there is a notice requirement, can we get MMUSIC on board so the interim can actually take place?</div><div><br>On Jan 2, 2013, at 7:13, "Eric Rescorla" &lt;<a href="mailto:ekr@rtfm.com">ekr@rtfm.com</a>&gt; wrote:<br><br></div><blockquote type="cite"><div><div>Chairs,</div><div><br></div><div>I've spent some time thinking about the agenda for the interim, and</div><div>thought it might be helpful if I sent you a strawman based on the</div><div>items that are blocking progress in terms of implementation and</div>

<div>interoperability.</div><div><br></div><div>Note:</div><div>I am assuming 4 hours each day. The times have been selected with</div><div>the objective of giving us enough time to really resolve the</div><div>open issues. Honestly, we might even add more time for</div>

<div>the issues on day 1.</div><div><br></div><div>Hope this is helpful.</div><div><br></div><div>Day 1</div><div>2:00 &nbsp; &nbsp;Bundle &nbsp; [Holmberg]</div><div><span class="Apple-tab-span" style="white-space:pre">	</span>Objective: select an approach and work out enough</div>

<div><span class="Apple-tab-span" style="white-space:pre">	</span>details that the editors can work to produce a</div><div><span class="Apple-tab-span" style="white-space:pre">	</span>suitable draft.</div><div><br></div>
<div>
<span class="Apple-tab-span" style="white-space:pre">	</span>Drafts:</div><div><span class="Apple-tab-span" style="white-space:pre">	</span>- <a href="http://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation">http://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation</a></div>

<div><span class="Apple-tab-span" style="white-space:pre">	</span>- <a href="http://tools.ietf.org/html/draft-alvestrand-one-rtp">http://tools.ietf.org/html/draft-alvestrand-one-rtp</a></div><div><span class="Apple-tab-span" style="white-space:pre">	</span>- <a href="http://tools.ietf.org/html/draft-holmberg-mmusic-sdp-mmt-negotiation">http://tools.ietf.org/html/draft-holmberg-mmusic-sdp-mmt-negotiation</a></div>

<div><br></div><div><br></div><div>2:00 &nbsp; &nbsp;Multiple flow SDP representation [Jennings, Alvestrand]</div><div><span class="Apple-tab-span" style="white-space:pre">	</span>Objective: decide on an architecture for large numbers</div>

<div><span class="Apple-tab-span" style="white-space:pre">	</span>of similar streams (e.g., multiple video streams).</div><div><br></div><div><span class="Apple-tab-span" style="white-space:pre">	</span>Drafts:</div><div>

<span class="Apple-tab-span" style="white-space:pre">	</span>- <a href="http://tools.ietf.org/html/draft-alvestrand-rtcweb-msid">http://tools.ietf.org/html/draft-alvestrand-rtcweb-msid</a></div><div><span class="Apple-tab-span" style="white-space:pre">	</span>- <a href="http://tools.ietf.org/html/draft-alvestrand-mmusic-msid">http://tools.ietf.org/html/draft-alvestrand-mmusic-msid</a></div>

<div><span class="Apple-tab-span" style="white-space:pre">	</span>- Do we have a "master" document for the single m-line case?</div><div><span class="Apple-tab-span" style="white-space:pre">	</span>- Multiple m-lines doesn't have a document</div>

<div><br></div><div><br></div><div>Day 2</div><div>1:30 &nbsp; &nbsp;Mapping stream/track label concepts SDP identifiers [Alvestrand]</div><div><span class="Apple-tab-span" style="white-space:pre">	</span>Objective: determine requirements</div>

<div><span class="Apple-tab-span" style="white-space:pre">	</span></div><div><span class="Apple-tab-span" style="white-space:pre">	</span></div><div>1:30 &nbsp; &nbsp;Trickle ICE open issues [Ivov(?), Uberti, Rescorla]</div><div><span class="Apple-tab-span" style="white-space:pre">	</span></div>

<div><span class="Apple-tab-span" style="white-space:pre">	</span>- Starting checks</div><div><span class="Apple-tab-span" style="white-space:pre">	</span>- Termination conditions</div><div><span class="Apple-tab-span" style="white-space:pre">	</span>- Error handling</div>

<div><span class="Apple-tab-span" style="white-space:pre">	</span></div><div><span class="Apple-tab-span" style="white-space:pre">	</span>Drafts:</div><div><span class="Apple-tab-span" style="white-space:pre">	</span><a href="http://tools.ietf.org/html/draft-rescorla-mmusic-ice-trickle">http://tools.ietf.org/html/draft-rescorla-mmusic-ice-trickle</a><span class="Apple-tab-span" style="white-space:pre">		</span></div>

<div><br></div><div>1:00 &nbsp; &nbsp;SDP for Datachannels open issues [Jesup, Loreto]</div><div><span class="Apple-tab-span" style="white-space:pre">	</span>Objective: resolve open issues (post WGLC?)</div><div><br></div><div><span class="Apple-tab-span" style="white-space:pre">	</span>Drafts:</div>

<div><span class="Apple-tab-span" style="white-space:pre">	</span><a href="http://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-02">http://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-02</a></div><div><br></div><div>Day 3</div>

<div>1:00 &nbsp; &nbsp;Other recommended audio codecs [Roach]</div><div><span class="Apple-tab-span" style="white-space:pre">	</span></div><div><span class="Apple-tab-span" style="white-space:pre">	</span>Objective: identify what other audio codecs we intend to</div>

<div><span class="Apple-tab-span" style="white-space:pre">	</span>recommend.<span class="Apple-tab-span" style="white-space:pre">	</span></div><div><br></div><div>1:00 &nbsp; &nbsp;Process WGLC comments on documents [authors]</div>

<div><br></div><div>0:30 &nbsp; &nbsp;Other business</div><div><br></div><div>0:30 &nbsp; &nbsp;Wrapup/next steps [Chairs]</div><div><br></div><div><br></div><div>Best,</div><div>-Ekr</div><div><br></div>
</div></blockquote><blockquote type="cite"><div><span>_______________________________________________</span><br><span>rtcweb mailing list</span><br><span><a href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br><span><a href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a></span><br></div></blockquote></body></html>
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_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

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From fluffy@cisco.com  Wed Jan  2 10:18:25 2013
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Bernard Aboba <bernard_aboba@hotmail.com>
Thread-Topic: [MMUSIC] [rtcweb] Proposed Agenda for Interim
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Date: Wed, 2 Jan 2013 18:18:21 +0000
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] [rtcweb] Proposed Agenda for Interim
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I'm not really sure what to do here - given a key role of the chairs is lar=
gely to facilitate the discussions and meetings that Wg participants want t=
o have,  guidance from participants of both MMUSIC and rtcweb is really imp=
ortant here.=20

I agree that=20

1) the items in EKR's email do look like the a bunch of the high runner ite=
ms that need to be resolved soon=20

2) the solutions of these problems are likely in to be mostly MMUSIC though=
, as Ted has pointed out, some amount of the requirements, architecture and=
 other discussion can happen in rtcweb.=20

3) if we are going to have an MMUSIC face to face interim meeting, it would=
 need AD approval and 4 weeks notices which puts us very close do the deadl=
ine so this needs to get sort out real soon.=20

Cullen



On Jan 2, 2013, at 8:24 AM, Bernard Aboba <bernard_aboba@hotmail.com> wrote=
:

> This is a good agenda - for an MMUSIC WG interim (only day 3 has any RTCW=
EB items on it).
>=20
> Since there is a notice requirement, can we get MMUSIC on board so the in=
terim can actually take place?
>=20
> On Jan 2, 2013, at 7:13, "Eric Rescorla" <ekr@rtfm.com> wrote:
>=20
>> Chairs,
>>=20
>> I've spent some time thinking about the agenda for the interim, and
>> thought it might be helpful if I sent you a strawman based on the
>> items that are blocking progress in terms of implementation and
>> interoperability.
>>=20
>> Note:
>> I am assuming 4 hours each day. The times have been selected with
>> the objective of giving us enough time to really resolve the
>> open issues. Honestly, we might even add more time for
>> the issues on day 1.
>>=20
>> Hope this is helpful.
>>=20
>> Day 1
>> 2:00    Bundle   [Holmberg]
>> 	Objective: select an approach and work out enough
>> 	details that the editors can work to produce a
>> 	suitable draft.
>>=20
>> 	Drafts:
>> 	- http://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation
>> 	- http://tools.ietf.org/html/draft-alvestrand-one-rtp
>> 	- http://tools.ietf.org/html/draft-holmberg-mmusic-sdp-mmt-negotiation
>>=20
>>=20
>> 2:00    Multiple flow SDP representation [Jennings, Alvestrand]
>> 	Objective: decide on an architecture for large numbers
>> 	of similar streams (e.g., multiple video streams).
>>=20
>> 	Drafts:
>> 	- http://tools.ietf.org/html/draft-alvestrand-rtcweb-msid
>> 	- http://tools.ietf.org/html/draft-alvestrand-mmusic-msid
>> 	- Do we have a "master" document for the single m-line case?
>> 	- Multiple m-lines doesn't have a document
>>=20
>>=20
>> Day 2
>> 1:30    Mapping stream/track label concepts SDP identifiers [Alvestrand]
>> 	Objective: determine requirements
>> =09
>> =09
>> 1:30    Trickle ICE open issues [Ivov(?), Uberti, Rescorla]
>> =09
>> 	- Starting checks
>> 	- Termination conditions
>> 	- Error handling
>> =09
>> 	Drafts:
>> 	http://tools.ietf.org/html/draft-rescorla-mmusic-ice-trickle	=09
>>=20
>> 1:00    SDP for Datachannels open issues [Jesup, Loreto]
>> 	Objective: resolve open issues (post WGLC?)
>>=20
>> 	Drafts:
>> 	http://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-02
>>=20
>> Day 3
>> 1:00    Other recommended audio codecs [Roach]
>> =09
>> 	Objective: identify what other audio codecs we intend to
>> 	recommend.=09
>>=20
>> 1:00    Process WGLC comments on documents [authors]
>>=20
>> 0:30    Other business
>>=20
>> 0:30    Wrapup/next steps [Chairs]
>>=20
>>=20
>> Best,
>> -Ekr
>>=20
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
> <Mail Attachment.txt>_______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From ted.ietf@gmail.com  Wed Jan  2 10:39:54 2013
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Date: Wed, 2 Jan 2013 10:39:53 -0800
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From: Ted Hardie <ted.ietf@gmail.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] [rtcweb] Proposed Agenda for Interim
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On Wed, Jan 2, 2013 at 10:18 AM, Cullen Jennings (fluffy)
<fluffy@cisco.com>wrote:

>
> 2) the solutions of these problems are likely in to be mostly MMUSIC
> though, as Ted has pointed out, some amount of the requirements,
> architecture and other discussion can happen in rtcweb.
>
>
I actually believe it can go beyond requirements and architecture; I
believe that if RTCWEB can come to a firm understanding of what solutions
are best for its use cases, it can communicate that to MMUSIC.  That
doesn't mean the final decision doesn't rest with MMUSIC--it does--but it
does mean that RTCWEB doesn't have to leave the technology discussion
completely off the table.  A good analysis can go to MMUSIC for discussion
and potential resolution in/by Orlando.

3) if we are going to have an MMUSIC face to face interim meeting, it would
> need AD approval and 4 weeks notices which puts us very close do the
> deadline so this needs to get sort out real soon.
>
>
There is also the option of a virtual interim, which is (somewhat) less of
a logistical issue.

Ted



> Cullen
>
>
>
> On Jan 2, 2013, at 8:24 AM, Bernard Aboba <bernard_aboba@hotmail.com>
> wrote:
>
> > This is a good agenda - for an MMUSIC WG interim (only day 3 has any
> RTCWEB items on it).
> >
> > Since there is a notice requirement, can we get MMUSIC on board so the
> interim can actually take place?
> >
> > On Jan 2, 2013, at 7:13, "Eric Rescorla" <ekr@rtfm.com> wrote:
> >
> >> Chairs,
> >>
> >> I've spent some time thinking about the agenda for the interim, and
> >> thought it might be helpful if I sent you a strawman based on the
> >> items that are blocking progress in terms of implementation and
> >> interoperability.
> >>
> >> Note:
> >> I am assuming 4 hours each day. The times have been selected with
> >> the objective of giving us enough time to really resolve the
> >> open issues. Honestly, we might even add more time for
> >> the issues on day 1.
> >>
> >> Hope this is helpful.
> >>
> >> Day 1
> >> 2:00    Bundle   [Holmberg]
> >>      Objective: select an approach and work out enough
> >>      details that the editors can work to produce a
> >>      suitable draft.
> >>
> >>      Drafts:
> >>      -
> http://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation
> >>      - http://tools.ietf.org/html/draft-alvestrand-one-rtp
> >>      -
> http://tools.ietf.org/html/draft-holmberg-mmusic-sdp-mmt-negotiation
> >>
> >>
> >> 2:00    Multiple flow SDP representation [Jennings, Alvestrand]
> >>      Objective: decide on an architecture for large numbers
> >>      of similar streams (e.g., multiple video streams).
> >>
> >>      Drafts:
> >>      - http://tools.ietf.org/html/draft-alvestrand-rtcweb-msid
> >>      - http://tools.ietf.org/html/draft-alvestrand-mmusic-msid
> >>      - Do we have a "master" document for the single m-line case?
> >>      - Multiple m-lines doesn't have a document
> >>
> >>
> >> Day 2
> >> 1:30    Mapping stream/track label concepts SDP identifiers [Alvestrand]
> >>      Objective: determine requirements
> >>
> >>
> >> 1:30    Trickle ICE open issues [Ivov(?), Uberti, Rescorla]
> >>
> >>      - Starting checks
> >>      - Termination conditions
> >>      - Error handling
> >>
> >>      Drafts:
> >>      http://tools.ietf.org/html/draft-rescorla-mmusic-ice-trickle
> >>
> >> 1:00    SDP for Datachannels open issues [Jesup, Loreto]
> >>      Objective: resolve open issues (post WGLC?)
> >>
> >>      Drafts:
> >>      http://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-02
> >>
> >> Day 3
> >> 1:00    Other recommended audio codecs [Roach]
> >>
> >>      Objective: identify what other audio codecs we intend to
> >>      recommend.
> >>
> >> 1:00    Process WGLC comments on documents [authors]
> >>
> >> 0:30    Other business
> >>
> >> 0:30    Wrapup/next steps [Chairs]
> >>
> >>
> >> Best,
> >> -Ekr
> >>
> >> _______________________________________________
> >> rtcweb mailing list
> >> rtcweb@ietf.org
> >> https://www.ietf.org/mailman/listinfo/rtcweb
> > <Mail Attachment.txt>_______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www.ietf.org/mailman/listinfo/mmusic
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>

--90e6ba1efbc274a4e504d2529282
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

On Wed, Jan 2, 2013 at 10:18 AM, Cullen Jennings (fluffy) <span dir=3D"ltr"=
>&lt;<a href=3D"mailto:fluffy@cisco.com" target=3D"_blank">fluffy@cisco.com=
</a>&gt;</span> wrote:<br><div class=3D"gmail_quote"><blockquote class=3D"g=
mail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-l=
eft:1ex">
<br>
2) the solutions of these problems are likely in to be mostly MMUSIC though=
, as Ted has pointed out, some amount of the requirements, architecture and=
 other discussion can happen in rtcweb.<br>
<br></blockquote><div><br>I actually believe it can go beyond requirements =
and architecture; I believe that if RTCWEB can come to a firm understanding=
 of what solutions are best for its use cases, it can communicate that to M=
MUSIC.=A0 That doesn&#39;t mean the final decision doesn&#39;t rest with MM=
USIC--it does--but it does mean that RTCWEB doesn&#39;t have to leave the t=
echnology discussion completely off the table.=A0 A good analysis can go to=
 MMUSIC for discussion and potential resolution in/by Orlando.<br>
<br></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bord=
er-left:1px #ccc solid;padding-left:1ex">
3) if we are going to have an MMUSIC face to face interim meeting, it would=
 need AD approval and 4 weeks notices which puts us very close do the deadl=
ine so this needs to get sort out real soon.<br>
<br></blockquote><div><br>There is also the option of a virtual interim, wh=
ich is (somewhat) less of a logistical issue.<br><br>Ted<br><br>=A0</div><b=
lockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px =
#ccc solid;padding-left:1ex">

Cullen<br>
<div><div class=3D"h5"><br>
<br>
<br>
On Jan 2, 2013, at 8:24 AM, Bernard Aboba &lt;<a href=3D"mailto:bernard_abo=
ba@hotmail.com">bernard_aboba@hotmail.com</a>&gt; wrote:<br>
<br>
&gt; This is a good agenda - for an MMUSIC WG interim (only day 3 has any R=
TCWEB items on it).<br>
&gt;<br>
&gt; Since there is a notice requirement, can we get MMUSIC on board so the=
 interim can actually take place?<br>
&gt;<br>
&gt; On Jan 2, 2013, at 7:13, &quot;Eric Rescorla&quot; &lt;<a href=3D"mail=
to:ekr@rtfm.com">ekr@rtfm.com</a>&gt; wrote:<br>
&gt;<br>
&gt;&gt; Chairs,<br>
&gt;&gt;<br>
&gt;&gt; I&#39;ve spent some time thinking about the agenda for the interim=
, and<br>
&gt;&gt; thought it might be helpful if I sent you a strawman based on the<=
br>
&gt;&gt; items that are blocking progress in terms of implementation and<br=
>
&gt;&gt; interoperability.<br>
&gt;&gt;<br>
&gt;&gt; Note:<br>
&gt;&gt; I am assuming 4 hours each day. The times have been selected with<=
br>
&gt;&gt; the objective of giving us enough time to really resolve the<br>
&gt;&gt; open issues. Honestly, we might even add more time for<br>
&gt;&gt; the issues on day 1.<br>
&gt;&gt;<br>
&gt;&gt; Hope this is helpful.<br>
&gt;&gt;<br>
&gt;&gt; Day 1<br>
&gt;&gt; 2:00 =A0 =A0Bundle =A0 [Holmberg]<br>
&gt;&gt; =A0 =A0 =A0Objective: select an approach and work out enough<br>
&gt;&gt; =A0 =A0 =A0details that the editors can work to produce a<br>
&gt;&gt; =A0 =A0 =A0suitable draft.<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 =A0Drafts:<br>
&gt;&gt; =A0 =A0 =A0- <a href=3D"http://tools.ietf.org/html/draft-ietf-mmus=
ic-sdp-bundle-negotiation" target=3D"_blank">http://tools.ietf.org/html/dra=
ft-ietf-mmusic-sdp-bundle-negotiation</a><br>
&gt;&gt; =A0 =A0 =A0- <a href=3D"http://tools.ietf.org/html/draft-alvestran=
d-one-rtp" target=3D"_blank">http://tools.ietf.org/html/draft-alvestrand-on=
e-rtp</a><br>
&gt;&gt; =A0 =A0 =A0- <a href=3D"http://tools.ietf.org/html/draft-holmberg-=
mmusic-sdp-mmt-negotiation" target=3D"_blank">http://tools.ietf.org/html/dr=
aft-holmberg-mmusic-sdp-mmt-negotiation</a><br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; 2:00 =A0 =A0Multiple flow SDP representation [Jennings, Alvestrand=
]<br>
&gt;&gt; =A0 =A0 =A0Objective: decide on an architecture for large numbers<=
br>
&gt;&gt; =A0 =A0 =A0of similar streams (e.g., multiple video streams).<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 =A0Drafts:<br>
&gt;&gt; =A0 =A0 =A0- <a href=3D"http://tools.ietf.org/html/draft-alvestran=
d-rtcweb-msid" target=3D"_blank">http://tools.ietf.org/html/draft-alvestran=
d-rtcweb-msid</a><br>
&gt;&gt; =A0 =A0 =A0- <a href=3D"http://tools.ietf.org/html/draft-alvestran=
d-mmusic-msid" target=3D"_blank">http://tools.ietf.org/html/draft-alvestran=
d-mmusic-msid</a><br>
&gt;&gt; =A0 =A0 =A0- Do we have a &quot;master&quot; document for the sing=
le m-line case?<br>
&gt;&gt; =A0 =A0 =A0- Multiple m-lines doesn&#39;t have a document<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; Day 2<br>
&gt;&gt; 1:30 =A0 =A0Mapping stream/track label concepts SDP identifiers [A=
lvestrand]<br>
&gt;&gt; =A0 =A0 =A0Objective: determine requirements<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; 1:30 =A0 =A0Trickle ICE open issues [Ivov(?), Uberti, Rescorla]<br=
>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 =A0- Starting checks<br>
&gt;&gt; =A0 =A0 =A0- Termination conditions<br>
&gt;&gt; =A0 =A0 =A0- Error handling<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 =A0Drafts:<br>
&gt;&gt; =A0 =A0 =A0<a href=3D"http://tools.ietf.org/html/draft-rescorla-mm=
usic-ice-trickle" target=3D"_blank">http://tools.ietf.org/html/draft-rescor=
la-mmusic-ice-trickle</a><br>
&gt;&gt;<br>
&gt;&gt; 1:00 =A0 =A0SDP for Datachannels open issues [Jesup, Loreto]<br>
&gt;&gt; =A0 =A0 =A0Objective: resolve open issues (post WGLC?)<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 =A0Drafts:<br>
&gt;&gt; =A0 =A0 =A0<a href=3D"http://tools.ietf.org/html/draft-ietf-mmusic=
-sctp-sdp-02" target=3D"_blank">http://tools.ietf.org/html/draft-ietf-mmusi=
c-sctp-sdp-02</a><br>
&gt;&gt;<br>
&gt;&gt; Day 3<br>
&gt;&gt; 1:00 =A0 =A0Other recommended audio codecs [Roach]<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 =A0Objective: identify what other audio codecs we intend t=
o<br>
&gt;&gt; =A0 =A0 =A0recommend.<br>
&gt;&gt;<br>
&gt;&gt; 1:00 =A0 =A0Process WGLC comments on documents [authors]<br>
&gt;&gt;<br>
&gt;&gt; 0:30 =A0 =A0Other business<br>
&gt;&gt;<br>
&gt;&gt; 0:30 =A0 =A0Wrapup/next steps [Chairs]<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; Best,<br>
&gt;&gt; -Ekr<br>
&gt;&gt;<br>
&gt;&gt; _______________________________________________<br>
&gt;&gt; rtcweb mailing list<br>
&gt;&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D=
"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div>&gt; &lt;Mail Attachment.txt&gt;_______________________________=
________________<br>
&gt; mmusic mailing list<br>
&gt; <a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/mmusic</a><br>
<br>
_______________________________________________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/mmusic</a><br>
</blockquote></div><br>

--90e6ba1efbc274a4e504d2529282--

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From: Bernard Aboba <bernard_aboba@hotmail.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
Date: Wed, 2 Jan 2013 11:44:53 -0800
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] [rtcweb] Proposed Agenda for Interim
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--_b4c74f88-3437-42e5-bbaf-f211f88f3d32_
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Cullen said:

"3) if we are going to have an MMUSIC face to face interim meeting=2C it wo=
uld need AD approval and 4 weeks notices which puts us very close do the de=
adline so this needs to get sort out real soon. "

[BA] I see no particularly good reason why this can't  happen quickly.  In =
fact=2C why not schedule two MMUSIC interims=2C 4-6 weeks apart from each o=
ther?  If travel is an issue=2C they can be virtual.  EKR's=0A=
 issue list is a fine starting point=2C and there is an informal IESG =0A=
telechat tomorrow (3-Jan) as well as one on 10-Jan-2013. =20

Ted's suggestion (to cover this in the RTCWEB interim) might do in a pinch=
=2C but really=2C MMUSIC WG has to develop a sense of urgency on this.  If =
things keep going as they have=2C by the time we get published RFCs out of =
MMUSIC=2C  implementations will have been in the field for years. =20

If MMUSIC were a Hollywood Celebrity=2C we'd be *way* past the time for "to=
ugh love" and on the front page of the National Enquirer for the Nth time.=
=20



 		 	   		  =

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<html>
<head>
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<body class=3D'hmmessage'><div dir=3D'ltr'>Cullen said:<br><br>"3) if we ar=
e going to have an MMUSIC face to face interim meeting=2C it would need AD =
approval and 4 weeks notices which puts us very close do the deadline so th=
is needs to get sort out real soon. "<br><div><br>[BA] I see no particularl=
y good reason why this can't&nbsp=3B happen quickly.&nbsp=3B In fact=2C why=
 not schedule two MMUSIC interims=2C 4-6 weeks apart from each other?&nbsp=
=3B If travel is an issue=2C they can be virtual.&nbsp=3B EKR's=0A=
 issue list is a fine starting point=2C and there is an informal IESG =0A=
telechat tomorrow (3-Jan) as well as one on 10-Jan-2013.&nbsp=3B <br><br>Te=
d's suggestion (to cover this in the RTCWEB interim) might do in a pinch=2C=
 but really=2C MMUSIC WG has to develop a sense of urgency on this.&nbsp=3B=
 If things keep going as they have=2C by the time we get published RFCs out=
 of MMUSIC=2C&nbsp=3B implementations will have been in the field for years=
.&nbsp=3B <br><br>If MMUSIC were a Hollywood Celebrity=2C we'd be *way* pas=
t the time for "tough love" and on the front page of the National Enquirer =
for the Nth time. <br><br><br><br></div> 		 	   		  </div></body>
</html>=

--_b4c74f88-3437-42e5-bbaf-f211f88f3d32_--

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Subject: Re: [MMUSIC] [rtcweb] Proposed Agenda for Interim
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On Wed, Jan 2, 2013 at 11:44 AM, Bernard Aboba <bernard_aboba@hotmail.com>wrote:

> Cullen said:
>
> "3) if we are going to have an MMUSIC face to face interim meeting, it
> would need AD approval and 4 weeks notices which puts us very close do the
> deadline so this needs to get sort out real soon. "
>
> [BA] I see no particularly good reason why this can't  happen quickly.  In
> fact, why not schedule two MMUSIC interims, 4-6 weeks apart from each
> other?  If travel is an issue, they can be virtual.  EKR's issue list is a
> fine starting point, and there is an informal IESG telechat tomorrow
> (3-Jan) as well as one on 10-Jan-2013.
>

I'm not in principle opposed to "virtual" interims, but my general feeling
is
that we should be using face-to-face time to cover the most important and
intractable issues. ISTM that those are the ones that I listed in my email.

Accordingly, I don't think it makes sense to have virtual MMUSIC interims
to discuss the important stuff and a F2F RTCWEB interim to discuss the
rest...

-Ekr


Ted's suggestion (to cover this in the RTCWEB interim) might do in a pinch,
> but really, MMUSIC WG has to develop a sense of urgency on this.  If things
> keep going as they have, by the time we get published RFCs out of MMUSIC,
> implementations will have been in the field for years.
>
> If MMUSIC were a Hollywood Celebrity, we'd be *way* past the time for
> "tough love" and on the front page of the National Enquirer for the Nth
> time.
>
>
>
>

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<br><br><div class=3D"gmail_quote">On Wed, Jan 2, 2013 at 11:44 AM, Bernard=
 Aboba <span dir=3D"ltr">&lt;<a href=3D"mailto:bernard_aboba@hotmail.com" t=
arget=3D"_blank">bernard_aboba@hotmail.com</a>&gt;</span> wrote:<br><blockq=
uote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc =
solid;padding-left:1ex">




<div><div dir=3D"ltr"><div class=3D"im">Cullen said:<br><br>&quot;3) if we =
are going to have an MMUSIC face to face interim meeting, it would need AD =
approval and 4 weeks notices which puts us very close do the deadline so th=
is needs to get sort out real soon. &quot;<br>

</div><div><br>[BA] I see no particularly good reason why this can&#39;t=A0=
 happen quickly.=A0 In fact, why not schedule two MMUSIC interims, 4-6 week=
s apart from each other?=A0 If travel is an issue, they can be virtual.=A0 =
EKR&#39;s
 issue list is a fine starting point, and there is an informal IESG=20
telechat tomorrow (3-Jan) as well as one on 10-Jan-2013.=A0<br></div></div>=
</div></blockquote><div><br></div><div>I&#39;m not in principle opposed to =
&quot;virtual&quot; interims, but my general feeling is</div><div>that we s=
hould be using face-to-face time to cover the most important and</div>

<div>intractable issues. ISTM that those are the ones that I listed in my e=
mail.</div><div><br></div><div>Accordingly, I don&#39;t think it makes sens=
e to have virtual MMUSIC interims</div><div>to discuss the important stuff =
and a F2F RTCWEB interim to discuss the</div>

<div>rest...</div><div><br></div><div>-Ekr</div><div><br></div><div><br></d=
iv><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left=
:1px #ccc solid;padding-left:1ex"><div><div dir=3D"ltr"><div>Ted&#39;s sugg=
estion (to cover this in the RTCWEB interim) might do in a pinch, but reall=
y, MMUSIC WG has to develop a sense of urgency on this.=A0 If things keep g=
oing as they have, by the time we get published RFCs out of MMUSIC,=A0 impl=
ementations will have been in the field for years.=A0 <br>

<br>If MMUSIC were a Hollywood Celebrity, we&#39;d be *way* past the time f=
or &quot;tough love&quot; and on the front page of the National Enquirer fo=
r the Nth time. <br><br><br><br></div> 		 	   		  </div></div>
</blockquote></div><br>

--e89a8f643556d946dd04d255bb0a--

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From: "Dan Wing" <dwing@cisco.com>
To: "'Paul Kyzivat'" <pkyzivat@alum.mit.edu>
References: <5093A2C9.9040001@alvestrand.no><50B9E3ED.6010604@ericsson.com>		<50BA19F9.4040701@alvestrand.no>		<50BD04D2.7090207@alum.mit.edu><50C6F800.1080500@ericsson.com>	 <50C7548C.3090807@alum.mit.edu><010401cdd7d0$d006d310$70147930$@cisco.com><50C786C8.2010403@alum.mit.edu><023f01cdd7d5$48e33c70$daa9b550$@cisco.com> <1.3e69ec2c964b01899213@alum.mit.edu>
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Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
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> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf
> Of Paul Kyzivat
> Sent: Tuesday, December 11, 2012 12:20 PM
> To: Dan Wing
> Cc: mmusic@ietf.org
> Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
> 
> Inline
> 
> On 12/11/12 2:25 PM, Dan Wing wrote:
> >> -----Original Message-----
> >> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
> >> Sent: Tuesday, December 11, 2012 11:17 AM
> >> To: Dan Wing
> >> Cc: mmusic@ietf.org
> >> Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
> >>
> >> Hi Dan,
> >>
> >> Comments inline
> >>
> >> On 12/11/12 1:53 PM, Dan Wing wrote:
> >>>> -----Original Message-----
> >>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> >>>> Behalf Of Paul Kyzivat
> >>>> Sent: Tuesday, December 11, 2012 7:43 AM
> >>>> To: mmusic@ietf.org
> >>>> Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
> >>>>
> >>>> More inline.
> >>>>
> >>>> On 12/11/12 4:08 AM, Salvatore Loreto wrote:
> >>>>> Hi Paul,
> >>>>>
> >>>>> see in line!
> >>>>>
> >>>>> On 12/3/12 10:00 PM, Paul Kyzivat wrote:
> >>>>>> Commenting on a different point
> >>>>>>
> >>>>>> On 12/1/12 9:53 AM, Harald Alvestrand wrote:
> >>>>>>
> >>>>>>> The interesting difference is that the multiplexing between
> >>>>>>> DTLS/SCTP traffic and BUNDLE multiplexing is that DTLS/SCTP
> >>>>>>> traffic is not carried in SSRCs, which means:
> >>>>>>>
> >>>>>>> - There can be only one DTLS/SCTP stream in a bundle (which may
> >>>>>>> have multiple associations, as you state below); you can't have
> >>>>>>> multiple lines with proto DTLS/SCTP in a bundle.
> >>>>>>
> >>>>>> I am not an SCTP expert. But IIUC, SCTP was designed to run
> >>>>>> directly over IP. It has its own notion of port used to demux
> >>>>>> multiple SCTP associations over the same IP address.
> >>>>>>
> >>>>>> I presume that that same mechanism is still there when SCTP is
> >>>>>> run over DTLS over UDP.
> >>>>>>
> >>>>>> So, the traffic coming over DTLS must first be demuxed into RTP
> >>>>>> traffic and SCTP traffic.
> >>>>> based on the current stack the SCTP traffic is the only traffic
> >>>>> that runs directly over the DTLS stack.
> >>>>
> >>>> Yes, that is what I thought. But Harald has been asking about
> >>>> multiplexing this with RTP traffic. (Actually I think it would be
> >>>> DTLS/SRTP traffic that it would be multiplexed with.)
> >>>>
> >>>>> What I am trying to do is to include the Randell Jesup (I am
> >>>>> including him in CC as I am not sure he is subscribed to this
> >> mailing list)
> >>>>> suggestion to give the possibility to have multiple SCTP
> >>>>> *associations* running on top of the same DTLS session and of
> >>>>> course providing a way to signal it in SDP.
> >>>>
> >>>> IIUC, SCTP (having been designed as a transport layer protocol)
> >>>> defines its own notion of port, and has fields in its protocol to
> >>>> carry the local and remote port number. Presumably those fields are
> >>>> still there when run over UDP or DTLS. So it should be possible to
> >>>> support multiple SCTP associations over the same DTLS connection,
> >>>> each distinguished by its own port pair.
> >>>
> >>> Yes, SCTP has its own notion of ports. How this works when SCTP is
> >>> carried over UDP is not quite clear, especially because SCTP assumes
> >>> a NAPT will not rewrite the SCTP port number (SCTP endeavors to make
> >>> such port rewriting difficult). But of course a NAPT (and the MAP
> >>> techniques) rewrite UDP port numbers. I believe SCTP would only be
> >>> able to successfully convey its UDP port numbers for a device that
> >>> is not behind a NAT (that is, a server sitting directly on the
> >>> Internet, rather than a remote peer that is behind a NAT). Creative
> >>> use of PCP, NAT-PMP, or UPnP IGD would improve that situation in the
> future.
> >>
> >> I find what you say above confusing.
> >
> > It is confusing, because SCTP's specifications require different
> > things of a NAPT than UDP, TCP, or DCCP. Specifically, SCTP's
> > specifications require the NAPT to not rewrite (not change) the source
> > port number of outgoing SCTP-over-IP packets. Reference
> > draft-ietf-behave-sctpnat.
> >
> >> SCTP running naked over IP would still have its own ports.
> >> If that ran over a path with a NAPT, then I could see why the NAPT
> >> might want to do the same as it does with UDP and TCP, and so replace
> >> addr/port pairs to minimize addr usage. (That's if the NAPT supported
> >> SCTP at all - I gather most don't.)
> >
> > I am not aware of any shipping NAPT that supports SCTP.
> >
> >
> >> But with SCTP over DTLS over UDP, there is a UDP port, and then
> >> presumably a separate SCTP port carried in the SCTP protocol within
> >> the UDP message.
> >
> > Right.
> >
> > And the new SCTP_REMOTE_UDP_ENCAPS_PORT socket option has some
> > relationship, but I do not understand how it works. Reference
> > draft-ietf-tsvwg-sctp-udp-encaps.
> 
> 1) draft-ietf-tsvwg-sctp-udp-encaps doesn't seem to cover SCTP over DTLS
> 
> 2) I guess that would only apply for a kernel implementation of SCTP
> over UDP, accessed via the socket interface.
> 
> So I don't think it is relevant for the immediate future plans for user
> mode implementations over UDP.


The SCTP_REMOTE_UDP_ENCAPS_PORT socket option exists in the user-mode
implementation, per the middle of this thread,

http://www.ietf.org/mail-archive/web/rtcweb/current/msg03302.html


> >> An NAPT on the path would presumably be messing with the UDP port.
> >> Given use of DTLS, the NAPT won't even be able to tell that SCTP is
> >> being used, much less mess with the SCTP port.
> >
> > Right.
> >
> >>>> That of course depends on having a signaling mechanism to set it
> up.
> >>>
> >>> After the initial SCTP association is created, SCTP could be allowed
> >>> to do its own thing (that is, chose to find and use other ports).
> >>
> >> Here you are talking about finding and using other UDP addr/port
> pairs?
> >
> > Yes, for mobility.
> >
> >> That is explicitly excluded in the DTLS/SCTP mapping.
> >
> > I am thinking about where to layer mobility and multipath for RTP
> > (MPRTP) and for SCTP. Do we do that with MICE
> > (draft-wing-mmusic-ice-mobility) or does SCTP do mobility itself? If
> > we do it with MICE, we need to disable or prevent SCTP from adding /
> > removing interfaces itself, which is why SCTP includes its own ability
> > to signal IP addresses and ports in SCTP itself.
> 
> OK. That will require extensions here and there over what drafts
> currently say.

Yeah, but requires first deciding what layer is best for such
mobility.

-d


> Thanks,
> Paul
> 
> > -d
> >
> >
> >> Thanks,
> >> Paul
> >>
> >>> SCTP's behavior in this regard is similar to MPRTP (multipath RTP),
> >>> but of course they are not identical.
> >>>
> >>>>> to be clear: at moment WebRTC allows only one SCTP association per
> >>>>> PC, so this is something that would be nice to define just to be
> >>>>> ready for the future.
> >>>>
> >>>> AFAIK WebRTC is just one possible user of this mechanism. The SDP
> >>>> mechanism shouldn't be limited by the constraints of WebRTC. It
> >>>> would be very difficult to define the SDP so that it was impossible
> >>>> to set up multiple SCTP associations over different 5-tuples.
> >>>>
> >>>>>> Then the RTP traffic can be demuxed based on SSRC, and SCTP
> >>>>>> traffic can be demuxed based on SCTP port. And once the traffic
> >>>>>> for a single SCTP port is identified, it can be demuxed based on
> >>>>>> stream
> >> number.
> >>>>>>
> >>>>>> Representing this in SDP is a challenge. Some variant of the
> >>>>>> bundle proposal might allow bundling together several RTP m-lines
> >>>>>> and some DTLS/SCTP m-lines. This would require a mechanism for
> >>>>>> specifying the SCTP port number - already an open issue (#3) in
> >>>>>> draft-ietf-mmusic-sctp-sdp-02.
> >>>>> I agree that it is a challange how then to bundle everything
> >>>>> together
> >>>>
> >>>> But it is a challenge that needs to be tackled if we are to realize
> >>>> Harald's dream.
> >>>
> >>> -d
> >>>
> >>>
> >>>
> >
> >
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic



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From: "Dan Wing" <dwing@cisco.com>
To: "'Paul Kyzivat'" <pkyzivat@alum.mit.edu>
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Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
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> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf
> Of Paul Kyzivat
> Sent: Tuesday, December 11, 2012 11:17 AM
> To: Dan Wing
> Cc: mmusic@ietf.org
> Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
> 
> Hi Dan,
> 
> Comments inline
> 
> On 12/11/12 1:53 PM, Dan Wing wrote:
> >> -----Original Message-----
> >> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> >> Behalf Of Paul Kyzivat
> >> Sent: Tuesday, December 11, 2012 7:43 AM
> >> To: mmusic@ietf.org
> >> Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
> >>
> >> More inline.
> >>
> >> On 12/11/12 4:08 AM, Salvatore Loreto wrote:
> >>> Hi Paul,
> >>>
> >>> see in line!
> >>>
> >>> On 12/3/12 10:00 PM, Paul Kyzivat wrote:
> >>>> Commenting on a different point
> >>>>
> >>>> On 12/1/12 9:53 AM, Harald Alvestrand wrote:
> >>>>
> >>>>> The interesting difference is that the multiplexing between
> >>>>> DTLS/SCTP traffic and BUNDLE multiplexing is that DTLS/SCTP
> >>>>> traffic is not carried in SSRCs, which means:
> >>>>>
> >>>>> - There can be only one DTLS/SCTP stream in a bundle (which may
> >>>>> have multiple associations, as you state below); you can't have
> >>>>> multiple lines with proto DTLS/SCTP in a bundle.
> >>>>
> >>>> I am not an SCTP expert. But IIUC, SCTP was designed to run
> >>>> directly over IP. It has its own notion of port used to demux
> >>>> multiple SCTP associations over the same IP address.
> >>>>
> >>>> I presume that that same mechanism is still there when SCTP is run
> >>>> over DTLS over UDP.
> >>>>
> >>>> So, the traffic coming over DTLS must first be demuxed into RTP
> >>>> traffic and SCTP traffic.
> >>> based on the current stack the SCTP traffic is the only traffic that
> >>> runs directly over the DTLS stack.
> >>
> >> Yes, that is what I thought. But Harald has been asking about
> >> multiplexing this with RTP traffic. (Actually I think it would be
> >> DTLS/SRTP traffic that it would be multiplexed with.)
> >>
> >>> What I am trying to do is to include the Randell Jesup (I am
> >>> including him in CC as I am not sure he is subscribed to this
> >>> mailing list) suggestion to give the possibility to have multiple
> >>> SCTP
> >>> *associations* running on top of the same DTLS session and of course
> >>> providing a way to signal it in SDP.
> >>
> >> IIUC, SCTP (having been designed as a transport layer protocol)
> >> defines its own notion of port, and has fields in its protocol to
> >> carry the local and remote port number. Presumably those fields are
> >> still there when run over UDP or DTLS. So it should be possible to
> >> support multiple SCTP associations over the same DTLS connection,
> >> each distinguished by its own port pair.
> >
> > Yes, SCTP has its own notion of ports. How this works when SCTP is
> > carried over UDP is not quite clear, especially because SCTP assumes a
> > NAPT will not rewrite the SCTP port number (SCTP endeavors to make
> > such port rewriting difficult). But of course a NAPT (and the MAP
> > techniques) rewrite UDP port numbers. I believe SCTP would only be
> > able to successfully convey its UDP port numbers for a device that is
> > not behind a NAT (that is, a server sitting directly on the Internet,
> > rather than a remote peer that is behind a NAT). Creative use of PCP,
> > NAT-PMP, or UPnP IGD would improve that situation in the future.
> 
> I find what you say above confusing.
> 
> SCTP running naked over IP would still have its own ports.
> If that ran over a path with a NAPT, then I could see why the NAPT might
> want to do the same as it does with UDP and TCP, and so replace
> addr/port pairs to minimize addr usage. (That's if the NAPT supported
> SCTP at all - I gather most don't.)

As I mentioned earlier, SCTP does not allow its source port number to
be changed.  

> But with SCTP over DTLS over UDP, there is a UDP port, and then
> presumably a separate SCTP port carried in the SCTP protocol within the
> UDP message.

Yep, I assume that is how it works.  Effectively, UDP is just used
to tunnel the SCTP.

> An NAPT on the path would presumably be messing with the UDP port. Given
> use of DTLS, the NAPT won't even be able to tell that SCTP is being
> used, much less mess with the SCTP port.
> 
> >> That of course depends on having a signaling mechanism to set it up.
> >
> > After the initial SCTP association is created, SCTP could be allowed
> > to do its own thing (that is, chose to find and use other ports).
> 
> Here you are talking about finding and using other UDP addr/port pairs?

No, I was referring to the SCTP ports.

-d


> That is explicitly excluded in the DTLS/SCTP mapping.
> 
> Thanks,
> Paul
> 
> > SCTP's behavior in this regard is similar to MPRTP (multipath RTP),
> > but of course they are not identical.
> >
> >>> to be clear: at moment WebRTC allows only one SCTP association per
> >>> PC, so this is something that would be nice to define just to be
> >>> ready for the future.
> >>
> >> AFAIK WebRTC is just one possible user of this mechanism. The SDP
> >> mechanism shouldn't be limited by the constraints of WebRTC. It would
> >> be very difficult to define the SDP so that it was impossible to set
> >> up multiple SCTP associations over different 5-tuples.
> >>
> >>>> Then the RTP traffic can be demuxed based on SSRC, and SCTP traffic
> >>>> can be demuxed based on SCTP port. And once the traffic for a
> >>>> single SCTP port is identified, it can be demuxed based on stream
> number.
> >>>>
> >>>> Representing this in SDP is a challenge. Some variant of the bundle
> >>>> proposal might allow bundling together several RTP m-lines and some
> >>>> DTLS/SCTP m-lines. This would require a mechanism for specifying
> >>>> the SCTP port number - already an open issue (#3) in
> >>>> draft-ietf-mmusic-sctp-sdp-02.
> >>> I agree that it is a challange how then to bundle everything
> >>> together
> >>
> >> But it is a challenge that needs to be tackled if we are to realize
> >> Harald's dream.
> >
> > -d
> >
> >
> >
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic



From pkyzivat@alum.mit.edu  Thu Jan  3 11:23:18 2013
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Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] RESTART: SCTP question: Where does it multiplex?
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I'm totally lost!
I looked back, and count 43 messages in this thread.
At this point I no longer understand the question(s), or what if 
anything has been answered.

Can we start over?
Sort out what the questions are, and then discuss whether we have 
answers to them?
(Preferably in separate, well identified, threads.)

I *think* Harald originally asserted that DTLS/SCTP *can* already be 
multiplexed with SRTP audio and video streams. And so he was asking 
*how* this could be specified in SDP. (E.g. Does "bundle" support it?)

Along the way, a question came up of how many DTLS/SCTP connections 
could be multiplexed with each other and with SRTP media. I think Harald 
assumed only one (and assumed there is only a need for one.) But reasons 
for more have been discussed. I suggested that it appears that multiple 
should be supportable, each with a different SCTP port.

I think Dan raised some additional issues related to multipath support.

	Thanks,
	Paul

On 1/2/13 10:32 PM, Dan Wing wrote:
>> -----Original Message-----
>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf
>> Of Paul Kyzivat
>> Sent: Tuesday, December 11, 2012 11:17 AM
>> To: Dan Wing
>> Cc: mmusic@ietf.org
>> Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
>>
>> Hi Dan,
>>
>> Comments inline
>>
>> On 12/11/12 1:53 PM, Dan Wing wrote:
>>>> -----Original Message-----
>>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
>>>> Behalf Of Paul Kyzivat
>>>> Sent: Tuesday, December 11, 2012 7:43 AM
>>>> To: mmusic@ietf.org
>>>> Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
>>>>
>>>> More inline.
>>>>
>>>> On 12/11/12 4:08 AM, Salvatore Loreto wrote:
>>>>> Hi Paul,
>>>>>
>>>>> see in line!
>>>>>
>>>>> On 12/3/12 10:00 PM, Paul Kyzivat wrote:
>>>>>> Commenting on a different point
>>>>>>
>>>>>> On 12/1/12 9:53 AM, Harald Alvestrand wrote:
>>>>>>
>>>>>>> The interesting difference is that the multiplexing between
>>>>>>> DTLS/SCTP traffic and BUNDLE multiplexing is that DTLS/SCTP
>>>>>>> traffic is not carried in SSRCs, which means:
>>>>>>>
>>>>>>> - There can be only one DTLS/SCTP stream in a bundle (which may
>>>>>>> have multiple associations, as you state below); you can't have
>>>>>>> multiple lines with proto DTLS/SCTP in a bundle.
>>>>>>
>>>>>> I am not an SCTP expert. But IIUC, SCTP was designed to run
>>>>>> directly over IP. It has its own notion of port used to demux
>>>>>> multiple SCTP associations over the same IP address.
>>>>>>
>>>>>> I presume that that same mechanism is still there when SCTP is run
>>>>>> over DTLS over UDP.
>>>>>>
>>>>>> So, the traffic coming over DTLS must first be demuxed into RTP
>>>>>> traffic and SCTP traffic.
>>>>> based on the current stack the SCTP traffic is the only traffic that
>>>>> runs directly over the DTLS stack.
>>>>
>>>> Yes, that is what I thought. But Harald has been asking about
>>>> multiplexing this with RTP traffic. (Actually I think it would be
>>>> DTLS/SRTP traffic that it would be multiplexed with.)
>>>>
>>>>> What I am trying to do is to include the Randell Jesup (I am
>>>>> including him in CC as I am not sure he is subscribed to this
>>>>> mailing list) suggestion to give the possibility to have multiple
>>>>> SCTP
>>>>> *associations* running on top of the same DTLS session and of course
>>>>> providing a way to signal it in SDP.
>>>>
>>>> IIUC, SCTP (having been designed as a transport layer protocol)
>>>> defines its own notion of port, and has fields in its protocol to
>>>> carry the local and remote port number. Presumably those fields are
>>>> still there when run over UDP or DTLS. So it should be possible to
>>>> support multiple SCTP associations over the same DTLS connection,
>>>> each distinguished by its own port pair.
>>>
>>> Yes, SCTP has its own notion of ports. How this works when SCTP is
>>> carried over UDP is not quite clear, especially because SCTP assumes a
>>> NAPT will not rewrite the SCTP port number (SCTP endeavors to make
>>> such port rewriting difficult). But of course a NAPT (and the MAP
>>> techniques) rewrite UDP port numbers. I believe SCTP would only be
>>> able to successfully convey its UDP port numbers for a device that is
>>> not behind a NAT (that is, a server sitting directly on the Internet,
>>> rather than a remote peer that is behind a NAT). Creative use of PCP,
>>> NAT-PMP, or UPnP IGD would improve that situation in the future.
>>
>> I find what you say above confusing.
>>
>> SCTP running naked over IP would still have its own ports.
>> If that ran over a path with a NAPT, then I could see why the NAPT might
>> want to do the same as it does with UDP and TCP, and so replace
>> addr/port pairs to minimize addr usage. (That's if the NAPT supported
>> SCTP at all - I gather most don't.)
>
> As I mentioned earlier, SCTP does not allow its source port number to
> be changed.
>
>> But with SCTP over DTLS over UDP, there is a UDP port, and then
>> presumably a separate SCTP port carried in the SCTP protocol within the
>> UDP message.
>
> Yep, I assume that is how it works.  Effectively, UDP is just used
> to tunnel the SCTP.
>
>> An NAPT on the path would presumably be messing with the UDP port. Given
>> use of DTLS, the NAPT won't even be able to tell that SCTP is being
>> used, much less mess with the SCTP port.
>>
>>>> That of course depends on having a signaling mechanism to set it up.
>>>
>>> After the initial SCTP association is created, SCTP could be allowed
>>> to do its own thing (that is, chose to find and use other ports).
>>
>> Here you are talking about finding and using other UDP addr/port pairs?
>
> No, I was referring to the SCTP ports.
>
> -d
>
>
>> That is explicitly excluded in the DTLS/SCTP mapping.
>>
>> Thanks,
>> Paul
>>
>>> SCTP's behavior in this regard is similar to MPRTP (multipath RTP),
>>> but of course they are not identical.
>>>
>>>>> to be clear: at moment WebRTC allows only one SCTP association per
>>>>> PC, so this is something that would be nice to define just to be
>>>>> ready for the future.
>>>>
>>>> AFAIK WebRTC is just one possible user of this mechanism. The SDP
>>>> mechanism shouldn't be limited by the constraints of WebRTC. It would
>>>> be very difficult to define the SDP so that it was impossible to set
>>>> up multiple SCTP associations over different 5-tuples.
>>>>
>>>>>> Then the RTP traffic can be demuxed based on SSRC, and SCTP traffic
>>>>>> can be demuxed based on SCTP port. And once the traffic for a
>>>>>> single SCTP port is identified, it can be demuxed based on stream
>> number.
>>>>>>
>>>>>> Representing this in SDP is a challenge. Some variant of the bundle
>>>>>> proposal might allow bundling together several RTP m-lines and some
>>>>>> DTLS/SCTP m-lines. This would require a mechanism for specifying
>>>>>> the SCTP port number - already an open issue (#3) in
>>>>>> draft-ietf-mmusic-sctp-sdp-02.
>>>>> I agree that it is a challange how then to bundle everything
>>>>> together
>>>>
>>>> But it is a challenge that needs to be tackled if we are to realize
>>>> Harald's dream.
>>>
>>> -d
>>>
>>>
>>>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>
>
>


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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: The rtp-duplication related drafts
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Hi everyone,

The following two drafts have been adopted by MMUSIC a while ago and the ma=
in draft discussing the RTP duplication issues is being finalized in AVTEXT=
 (this was also adopted as a WG item a while ago). It is simply waiting for=
 the WG to make a decision on the srcname proposal.

http://datatracker.ietf.org/doc/draft-ietf-mmusic-delayed-duplication/
http://datatracker.ietf.org/doc/draft-ietf-mmusic-duplication-grouping/

These two mmusic drafts are essentially completed and they are needed to be=
 done for the avtext draft. I hope they will be WGLC'ed some time soon. In =
the mean time, I would like to get any comments, feedback that you might ha=
ve.

Thanks.
-acbegen

PS Here is the AVTEXT draft.
http://datatracker.ietf.org/doc/draft-ietf-avtext-rtp-duplication/ =

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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] Draft-ietf-mmusic-rfc4566bis-05: session-level attributes
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We clearly need an escape hatch as there are several examples of 
attributes with a given name that may occur multiple times to provide 
different values by use of additional qualifiers inside the attribute, 
and some of those attributes may appear at both the session and media 
level (capability negotiation for example uses this approach, as does 
"rtpmap", "fmtp", etc. inside a media description). In fact, I'd argue 
it's more than an escape hatch and probably more of a "default rule" in 
the absence of anything else specified for the attribute.

In terms of the "choice" wording below, I think it's underspecified what 
that means, and I'm also not sure we really need a general rule here. Do 
we have other examples than the direction attributes currently ? If not 
(or we can't think of those), I'd rather just call those out explicitly.

Thanks

-- Flemming (as Individual)



On 12/29/12 6:03 PM, Ali C. Begen (abegen) wrote:
> I don¹t remember us reaching a consensus regarding the text proposed by
> Brett. One thing certain is that we don't wanna break anything that
> complies with 4566 but we should also make the text as clear as possible.
>
> I am OK with Brett's suggestion as I don't recall an attribute like what
> Paul might have seen. But, even if such an attribute exists, the current
> text would be incorrect, too (This would indicate a problem with that
> particular attribute not 4566 or 4566bis).
>
> Brett's proposal does not make the text any worse. So, I am planning to
> incorporate this change.
>
> Any objections?
>
> -acbegen
>
> -----Original Message-----
> From: Paul Kyzivat <pkyzivat@alum.mit.edu>
> Date: Monday, August 27, 2012 12:03 PM
> To: "mmusic@ietf.org" <mmusic@ietf.org>
> Subject: Re: [MMUSIC] Draft-ietf-mmusic-rfc4566bis-05:
> session-level	attributes
>
>> I've always thought this should be the way things work, for all
>> attributes. But somewhere I came across an attribute (sorry, don't
>> remember which one) where the attribute at session level means something
>> different than the attribute at media level, and so can't be treated as
>> a default for media level. (E.g. the session level is an aggregate limit
>> for something, while the media level is a limit for a single m-line.)
>>
>> Am I just dreaming this? Or can someone identify specific attributes
>> that behave this way?
>>
>> If there are existing attributes that behave this way, then we can't
>> make a blanket statement like this. There will need to be an escape
>> hatch. For example:
>>
>> "The connection ("c=") information in the
>>   session-level section applies to all the media of that session unless
>>   overridden by connection information
>>   in the media description.  For instance, in the example
>>   below, TBD..."
>>
>> "The attribute ("a=") information in the
>>   session-level section applies to all the media of that session unless
>>   overridden by connection information or an attribute of the same name
>>   or choice in the media description, or the specification of that
>>   attribute calls for a different treatment.
>>   For instance, in the example
>>   below, each media behaves as if it were given a "recvonly" attribute.
>>   If the example had additionally contained a "sendrecv" media-level
>>   attribute for audio, only the video media would behave as if it were
>>   given a "recvonly" attribute."
>>
>> Even this could have some backward compatibility consequences. I think
>> there are some attributes that are only defined for media level. This
>> revision would mean that a session level value could override a default
>> for media level. Implementations that aren't prepared for that might
>> break.
>>
>> 	Thanks,
>> 	Paul
>>
>> On 8/27/12 12:04 PM, Ali C. Begen (abegen) wrote:
>>> Brett suggested the following change in the SDP draft. Any objections?
>>>
>>> -acbegen
>>>
>>> Current text:
>>>
>>> "The connection ("c=") and attribute ("a=") information in the
>>>    session-level section applies to all the media of that session unless
>>>    overridden by connection information or an attribute of the same name
>>>    in the media description.  For instance, in the example below, each
>>>    media behaves as if it were given a "recvonly" attribute."
>>>
>>> Proposed text:
>>>
>>> "The connection ("c=") and attribute ("a=") information in the
>>>    session-level section applies to all the media of that session unless
>>>    overridden by connection information or an attribute of the same name
>>>    or choice in the media description.  For instance, in the example
>>>    below, each media behaves as if it were given a "recvonly" attribute.
>>>    If the example had additionally contained a "sendrecv" media-level
>>>    attribute for audio, only the video media would behave as if it were
>>>    given a "recvonly" attribute."
>>>
>>>> -----Original Message-----
>>>> From: Brett Tate [mailto:brett@broadsoft.com]
>>>> Sent: Wednesday, August 08, 2012 7:54 AM
>>>> To: Ali C. Begen (abegen); mmusic@ietf.org
>>>> Subject: RE: Draft-ietf-mmusic-rfc4566bis-05: session-level attributes
>>>>
>>>>>> Draft-ietf-mmusic-rfc4566bis-05 section 5 indicates the
>>>>>> following.
>>>>>>
>>>>>> "In general, session-level values are the default for all
>>>>>> media unless overridden by an equivalent media-level value."
>>>>>>
>>>>>> "The connection ("c=") and attribute ("a=") information
>>>>>> in the session-level section applies to all the media of
>>>>>> that session unless overridden by connection information
>>>>>> or an attribute of the same name in the media description.
>>>>>> For instance, in the example below, each media behaves
>>>>>> as if it were given a "recvonly" attribute."
>>>>>>
>>>>>> If the attributes are related but with different names
>>>>>> (such as "inactive" and "sendrecv"), does the attribute
>>>>>> at the media-level have precedence over the value at the
>>>>>> session-level?  If so, I recommend that the draft be
>>>>>> updated to clarify the topic.
>>>>> I believe so. at least that is my understanding from the
>>>>> text. Does it really need clarification? Thoughts?
>>>> I think that it should be clarified because it is unclear if "an
>>>> attribute of the same name" indicates the typical situation or only
>>>> situation concerning application of "unless overridden by an
>>>> equivalent media-level value".  For instance if I sent "a=inactive"
>>>> at session-level and "a=sendrecv" at media-level, I'd be concerned
>>>> that the receiver may interpret the SDP as being
>>>> abnormal (such as including both values at media-level) instead of
>>>> allowing "a=sendrecv" to override "a=inactive".
>>>>
>>>> In case it matters from an extendibility perspective, requiring the
>>>> same name might make it easier to recognize the override.
>>>> However since section 6 already introduces complications with
>>>> defaulting to "sendrecv" if a new option is defined/used and
>>>> unknown, allowing the override even when the names are not the same
>>>> doesn't make things too much worse; it actually
>>>> allows communication of a backup attribute (such as "a=inactive") if
>>>> they don't understand the preferred new attribute (such
>>>> as "a=foo").
>>>>
>>>> "If none of the attributes "sendonly", "recvonly", "inactive",
>>>> and "sendrecv" is present, "sendrecv" SHOULD be assumed as the
>>>> default for sessions that are not of the conference type
>>>> "broadcast" or "H332" (see below)."
>>>>
>>>> Thanks for the response,
>>>> Brett
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
> .
>


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Cc: "Cullen Jennings \(fluffy\)" <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "mmusic@ietf.org" <mmusic@ietf.org>
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On 1/2/13 2:44 PM, Bernard Aboba wrote:
> Cullen said:
>
> "3) if we are going to have an MMUSIC face to face interim meeting, it 
> would need AD approval and 4 weeks notices which puts us very close do 
> the deadline so this needs to get sort out real soon. "
>
> [BA] I see no particularly good reason why this can't  happen 
> quickly.  In fact, why not schedule two MMUSIC interims, 4-6 weeks 
> apart from each other?  If travel is an issue, they can be virtual.  
> EKR's issue list is a fine starting point, and there is an informal 
> IESG telechat tomorrow (3-Jan) as well as one on 10-Jan-2013.
>
> Ted's suggestion (to cover this in the RTCWEB interim) might do in a 
> pinch, but really, MMUSIC WG has to develop a sense of urgency on 
> this.  If things keep going as they have, by the time we get published 
> RFCs out of MMUSIC,  implementations will have been in the field for 
> years.
>
The MMUSIC WG is driven by the contributions, discussions, and 
ultimately rough consensus of its participants, just like any other WG. 
The active participants on the topics of interest here seem to be 
largely the same in the two groups, so I'm not sure exactly what you are 
looking for or proposing here, but constructive suggestions are always 
welcome.

Thanks

-- Flemming (MMUSIC co-chair)


> If MMUSIC were a Hollywood Celebrity, we'd be *way* past the time for 
> "tough love" and on the front page of the National Enquirer for the 
> Nth time.
>
>
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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    <br>
    <div class="moz-cite-prefix">On 1/2/13 2:44 PM, Bernard Aboba wrote:<br>
    </div>
    <blockquote cite="mid:BLU002-W7BBB1FEC4D1BEEDF1498C93220@phx.gbl"
      type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=ISO-8859-1">
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      <div dir="ltr">Cullen said:<br>
        <br>
        "3) if we are going to have an MMUSIC face to face interim
        meeting, it would need AD approval and 4 weeks notices which
        puts us very close do the deadline so this needs to get sort out
        real soon. "<br>
        <div><br>
          [BA] I see no particularly good reason why this can't&nbsp; happen
          quickly.&nbsp; In fact, why not schedule two MMUSIC interims, 4-6
          weeks apart from each other?&nbsp; If travel is an issue, they can
          be virtual.&nbsp; EKR's issue list is a fine starting point, and
          there is an informal IESG telechat tomorrow (3-Jan) as well as
          one on 10-Jan-2013.&nbsp; <br>
          <br>
          Ted's suggestion (to cover this in the RTCWEB interim) might
          do in a pinch, but really, MMUSIC WG has to develop a sense of
          urgency on this.&nbsp; If things keep going as they have, by the
          time we get published RFCs out of MMUSIC,&nbsp; implementations
          will have been in the field for years.&nbsp; <br>
          <br>
        </div>
      </div>
    </blockquote>
    The MMUSIC WG is driven by the contributions, discussions, and
    ultimately rough consensus of its participants, just like any other
    WG. The active participants on the topics of interest here seem to
    be largely the same in the two groups, so I'm not sure exactly what
    you are looking for or proposing here, but constructive suggestions
    are always welcome. <br>
    <br>
    Thanks <br>
    <br>
    -- Flemming (MMUSIC co-chair)<br>
    <br>
    <br>
    <blockquote cite="mid:BLU002-W7BBB1FEC4D1BEEDF1498C93220@phx.gbl"
      type="cite">
      <div dir="ltr">
        <div>If MMUSIC were a Hollywood Celebrity, we'd be *way* past
          the time for "tough love" and on the front page of the
          National Enquirer for the Nth time. <br>
          <br>
          <br>
          <br>
        </div>
      </div>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
mmusic mailing list
<a class="moz-txt-link-abbreviated" href="mailto:mmusic@ietf.org">mmusic@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a>
</pre>
    </blockquote>
    <br>
  </body>
</html>

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From pkyzivat@alum.mit.edu  Thu Jan  3 18:09:32 2013
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Date: Thu, 03 Jan 2013 21:09:26 -0500
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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To: Flemming Andreasen <fandreas@cisco.com>
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Subject: Re: [MMUSIC] Draft-ietf-mmusic-rfc4566bis-05: session-level attributes
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Once upon a time I thought that rule actually existed and worked, and 
that all attributes ought to have the "session is default for media 
level" property.

But then it became apparent that there were cases that didn't fit that rule.

When writing my last reply to this thread, and examining things more 
closely, it became apparent to me that this defaulting behavior doesn't 
work in many/most cases. The interesting cases that work something like 
that need to be described individually to cover the special cases that 
seem to apply almost all the time.

So I agree we are probably better off just not asserting it.

And at the same time we should polish up the definitions of the 
attributes defined in this document so that they are very clear how 
session and media level interact.

	Thanks,
	Paul

On 1/3/13 6:01 PM, Flemming Andreasen wrote:
> We clearly need an escape hatch as there are several examples of
> attributes with a given name that may occur multiple times to provide
> different values by use of additional qualifiers inside the attribute,
> and some of those attributes may appear at both the session and media
> level (capability negotiation for example uses this approach, as does
> "rtpmap", "fmtp", etc. inside a media description). In fact, I'd argue
> it's more than an escape hatch and probably more of a "default rule" in
> the absence of anything else specified for the attribute.
>
> In terms of the "choice" wording below, I think it's underspecified what
> that means, and I'm also not sure we really need a general rule here. Do
> we have other examples than the direction attributes currently ? If not
> (or we can't think of those), I'd rather just call those out explicitly.
>
> Thanks
>
> -- Flemming (as Individual)
>
>
>
> On 12/29/12 6:03 PM, Ali C. Begen (abegen) wrote:
>> I don¹t remember us reaching a consensus regarding the text proposed by
>> Brett. One thing certain is that we don't wanna break anything that
>> complies with 4566 but we should also make the text as clear as possible.
>>
>> I am OK with Brett's suggestion as I don't recall an attribute like what
>> Paul might have seen. But, even if such an attribute exists, the current
>> text would be incorrect, too (This would indicate a problem with that
>> particular attribute not 4566 or 4566bis).
>>
>> Brett's proposal does not make the text any worse. So, I am planning to
>> incorporate this change.
>>
>> Any objections?
>>
>> -acbegen
>>
>> -----Original Message-----
>> From: Paul Kyzivat <pkyzivat@alum.mit.edu>
>> Date: Monday, August 27, 2012 12:03 PM
>> To: "mmusic@ietf.org" <mmusic@ietf.org>
>> Subject: Re: [MMUSIC] Draft-ietf-mmusic-rfc4566bis-05:
>> session-level    attributes
>>
>>> I've always thought this should be the way things work, for all
>>> attributes. But somewhere I came across an attribute (sorry, don't
>>> remember which one) where the attribute at session level means something
>>> different than the attribute at media level, and so can't be treated as
>>> a default for media level. (E.g. the session level is an aggregate limit
>>> for something, while the media level is a limit for a single m-line.)
>>>
>>> Am I just dreaming this? Or can someone identify specific attributes
>>> that behave this way?
>>>
>>> If there are existing attributes that behave this way, then we can't
>>> make a blanket statement like this. There will need to be an escape
>>> hatch. For example:
>>>
>>> "The connection ("c=") information in the
>>>   session-level section applies to all the media of that session unless
>>>   overridden by connection information
>>>   in the media description.  For instance, in the example
>>>   below, TBD..."
>>>
>>> "The attribute ("a=") information in the
>>>   session-level section applies to all the media of that session unless
>>>   overridden by connection information or an attribute of the same name
>>>   or choice in the media description, or the specification of that
>>>   attribute calls for a different treatment.
>>>   For instance, in the example
>>>   below, each media behaves as if it were given a "recvonly" attribute.
>>>   If the example had additionally contained a "sendrecv" media-level
>>>   attribute for audio, only the video media would behave as if it were
>>>   given a "recvonly" attribute."
>>>
>>> Even this could have some backward compatibility consequences. I think
>>> there are some attributes that are only defined for media level. This
>>> revision would mean that a session level value could override a default
>>> for media level. Implementations that aren't prepared for that might
>>> break.
>>>
>>>     Thanks,
>>>     Paul
>>>
>>> On 8/27/12 12:04 PM, Ali C. Begen (abegen) wrote:
>>>> Brett suggested the following change in the SDP draft. Any objections?
>>>>
>>>> -acbegen
>>>>
>>>> Current text:
>>>>
>>>> "The connection ("c=") and attribute ("a=") information in the
>>>>    session-level section applies to all the media of that session
>>>> unless
>>>>    overridden by connection information or an attribute of the same
>>>> name
>>>>    in the media description.  For instance, in the example below, each
>>>>    media behaves as if it were given a "recvonly" attribute."
>>>>
>>>> Proposed text:
>>>>
>>>> "The connection ("c=") and attribute ("a=") information in the
>>>>    session-level section applies to all the media of that session
>>>> unless
>>>>    overridden by connection information or an attribute of the same
>>>> name
>>>>    or choice in the media description.  For instance, in the example
>>>>    below, each media behaves as if it were given a "recvonly"
>>>> attribute.
>>>>    If the example had additionally contained a "sendrecv" media-level
>>>>    attribute for audio, only the video media would behave as if it were
>>>>    given a "recvonly" attribute."
>>>>
>>>>> -----Original Message-----
>>>>> From: Brett Tate [mailto:brett@broadsoft.com]
>>>>> Sent: Wednesday, August 08, 2012 7:54 AM
>>>>> To: Ali C. Begen (abegen); mmusic@ietf.org
>>>>> Subject: RE: Draft-ietf-mmusic-rfc4566bis-05: session-level attributes
>>>>>
>>>>>>> Draft-ietf-mmusic-rfc4566bis-05 section 5 indicates the
>>>>>>> following.
>>>>>>>
>>>>>>> "In general, session-level values are the default for all
>>>>>>> media unless overridden by an equivalent media-level value."
>>>>>>>
>>>>>>> "The connection ("c=") and attribute ("a=") information
>>>>>>> in the session-level section applies to all the media of
>>>>>>> that session unless overridden by connection information
>>>>>>> or an attribute of the same name in the media description.
>>>>>>> For instance, in the example below, each media behaves
>>>>>>> as if it were given a "recvonly" attribute."
>>>>>>>
>>>>>>> If the attributes are related but with different names
>>>>>>> (such as "inactive" and "sendrecv"), does the attribute
>>>>>>> at the media-level have precedence over the value at the
>>>>>>> session-level?  If so, I recommend that the draft be
>>>>>>> updated to clarify the topic.
>>>>>> I believe so. at least that is my understanding from the
>>>>>> text. Does it really need clarification? Thoughts?
>>>>> I think that it should be clarified because it is unclear if "an
>>>>> attribute of the same name" indicates the typical situation or only
>>>>> situation concerning application of "unless overridden by an
>>>>> equivalent media-level value".  For instance if I sent "a=inactive"
>>>>> at session-level and "a=sendrecv" at media-level, I'd be concerned
>>>>> that the receiver may interpret the SDP as being
>>>>> abnormal (such as including both values at media-level) instead of
>>>>> allowing "a=sendrecv" to override "a=inactive".
>>>>>
>>>>> In case it matters from an extendibility perspective, requiring the
>>>>> same name might make it easier to recognize the override.
>>>>> However since section 6 already introduces complications with
>>>>> defaulting to "sendrecv" if a new option is defined/used and
>>>>> unknown, allowing the override even when the names are not the same
>>>>> doesn't make things too much worse; it actually
>>>>> allows communication of a backup attribute (such as "a=inactive") if
>>>>> they don't understand the preferred new attribute (such
>>>>> as "a=foo").
>>>>>
>>>>> "If none of the attributes "sendonly", "recvonly", "inactive",
>>>>> and "sendrecv" is present, "sendrecv" SHOULD be assumed as the
>>>>> default for sessions that are not of the conference type
>>>>> "broadcast" or "H332" (see below)."
>>>>>
>>>>> Thanks for the response,
>>>>> Brett
>>>> _______________________________________________
>>>> mmusic mailing list
>>>> mmusic@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>>
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>> .
>>
>
>


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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
Thread-Topic: [MMUSIC] Draft-ietf-mmusic-rfc4566bis-05: session-level attributes
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Cc: "Flemming Andreasen \(fandreas\)" <fandreas@cisco.com>, "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] Draft-ietf-mmusic-rfc4566bis-05: session-level attributes
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Hi Paul,

On Jan 3, 2013, at 9:09 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> Once upon a time I thought that rule actually existed and worked, and tha=
t all attributes ought to have the "session is default for media level" pro=
perty.
>=20
> But then it became apparent that there were cases that didn't fit that ru=
le.
>=20
> When writing my last reply to this thread, and examining things more clos=
ely, it became apparent to me that this defaulting behavior doesn't work in=
 many/most cases. The interesting cases that work something like that need =
to be described individually to cover the special cases that seem to apply =
almost all the time.
>=20
> So I agree we are probably better off just not asserting it.
>=20
Are you saying we don't change the text or remove it altogether?

> And at the same time we should polish up the definitions of the attribute=
s defined in this document so that they are very clear how session and medi=
a level interact.

Could you provide an example of what you have in mind?

>=20
> 	Thanks,
> 	Paul
>=20
> On 1/3/13 6:01 PM, Flemming Andreasen wrote:
>> We clearly need an escape hatch as there are several examples of
>> attributes with a given name that may occur multiple times to provide
>> different values by use of additional qualifiers inside the attribute,
>> and some of those attributes may appear at both the session and media
>> level (capability negotiation for example uses this approach, as does
>> "rtpmap", "fmtp", etc. inside a media description). In fact, I'd argue
>> it's more than an escape hatch and probably more of a "default rule" in
>> the absence of anything else specified for the attribute.
>>=20
>> In terms of the "choice" wording below, I think it's underspecified what
>> that means, and I'm also not sure we really need a general rule here. Do
>> we have other examples than the direction attributes currently ? If not
>> (or we can't think of those), I'd rather just call those out explicitly.
>>=20
>> Thanks
>>=20
>> -- Flemming (as Individual)
>>=20
>>=20
>>=20
>> On 12/29/12 6:03 PM, Ali C. Begen (abegen) wrote:
>>> I don=B9t remember us reaching a consensus regarding the text proposed =
by
>>> Brett. One thing certain is that we don't wanna break anything that
>>> complies with 4566 but we should also make the text as clear as possibl=
e.
>>>=20
>>> I am OK with Brett's suggestion as I don't recall an attribute like wha=
t
>>> Paul might have seen. But, even if such an attribute exists, the curren=
t
>>> text would be incorrect, too (This would indicate a problem with that
>>> particular attribute not 4566 or 4566bis).
>>>=20
>>> Brett's proposal does not make the text any worse. So, I am planning to
>>> incorporate this change.
>>>=20
>>> Any objections?
>>>=20
>>> -acbegen
>>>=20
>>> -----Original Message-----
>>> From: Paul Kyzivat <pkyzivat@alum.mit.edu>
>>> Date: Monday, August 27, 2012 12:03 PM
>>> To: "mmusic@ietf.org" <mmusic@ietf.org>
>>> Subject: Re: [MMUSIC] Draft-ietf-mmusic-rfc4566bis-05:
>>> session-level    attributes
>>>=20
>>>> I've always thought this should be the way things work, for all
>>>> attributes. But somewhere I came across an attribute (sorry, don't
>>>> remember which one) where the attribute at session level means somethi=
ng
>>>> different than the attribute at media level, and so can't be treated a=
s
>>>> a default for media level. (E.g. the session level is an aggregate lim=
it
>>>> for something, while the media level is a limit for a single m-line.)
>>>>=20
>>>> Am I just dreaming this? Or can someone identify specific attributes
>>>> that behave this way?
>>>>=20
>>>> If there are existing attributes that behave this way, then we can't
>>>> make a blanket statement like this. There will need to be an escape
>>>> hatch. For example:
>>>>=20
>>>> "The connection ("c=3D") information in the
>>>>  session-level section applies to all the media of that session unless
>>>>  overridden by connection information
>>>>  in the media description.  For instance, in the example
>>>>  below, TBD..."
>>>>=20
>>>> "The attribute ("a=3D") information in the
>>>>  session-level section applies to all the media of that session unless
>>>>  overridden by connection information or an attribute of the same name
>>>>  or choice in the media description, or the specification of that
>>>>  attribute calls for a different treatment.
>>>>  For instance, in the example
>>>>  below, each media behaves as if it were given a "recvonly" attribute.
>>>>  If the example had additionally contained a "sendrecv" media-level
>>>>  attribute for audio, only the video media would behave as if it were
>>>>  given a "recvonly" attribute."
>>>>=20
>>>> Even this could have some backward compatibility consequences. I think
>>>> there are some attributes that are only defined for media level. This
>>>> revision would mean that a session level value could override a defaul=
t
>>>> for media level. Implementations that aren't prepared for that might
>>>> break.
>>>>=20
>>>>    Thanks,
>>>>    Paul
>>>>=20
>>>> On 8/27/12 12:04 PM, Ali C. Begen (abegen) wrote:
>>>>> Brett suggested the following change in the SDP draft. Any objections=
?
>>>>>=20
>>>>> -acbegen
>>>>>=20
>>>>> Current text:
>>>>>=20
>>>>> "The connection ("c=3D") and attribute ("a=3D") information in the
>>>>>   session-level section applies to all the media of that session
>>>>> unless
>>>>>   overridden by connection information or an attribute of the same
>>>>> name
>>>>>   in the media description.  For instance, in the example below, each
>>>>>   media behaves as if it were given a "recvonly" attribute."
>>>>>=20
>>>>> Proposed text:
>>>>>=20
>>>>> "The connection ("c=3D") and attribute ("a=3D") information in the
>>>>>   session-level section applies to all the media of that session
>>>>> unless
>>>>>   overridden by connection information or an attribute of the same
>>>>> name
>>>>>   or choice in the media description.  For instance, in the example
>>>>>   below, each media behaves as if it were given a "recvonly"
>>>>> attribute.
>>>>>   If the example had additionally contained a "sendrecv" media-level
>>>>>   attribute for audio, only the video media would behave as if it wer=
e
>>>>>   given a "recvonly" attribute."
>>>>>=20
>>>>>> -----Original Message-----
>>>>>> From: Brett Tate [mailto:brett@broadsoft.com]
>>>>>> Sent: Wednesday, August 08, 2012 7:54 AM
>>>>>> To: Ali C. Begen (abegen); mmusic@ietf.org
>>>>>> Subject: RE: Draft-ietf-mmusic-rfc4566bis-05: session-level attribut=
es
>>>>>>=20
>>>>>>>> Draft-ietf-mmusic-rfc4566bis-05 section 5 indicates the
>>>>>>>> following.
>>>>>>>>=20
>>>>>>>> "In general, session-level values are the default for all
>>>>>>>> media unless overridden by an equivalent media-level value."
>>>>>>>>=20
>>>>>>>> "The connection ("c=3D") and attribute ("a=3D") information
>>>>>>>> in the session-level section applies to all the media of
>>>>>>>> that session unless overridden by connection information
>>>>>>>> or an attribute of the same name in the media description.
>>>>>>>> For instance, in the example below, each media behaves
>>>>>>>> as if it were given a "recvonly" attribute."
>>>>>>>>=20
>>>>>>>> If the attributes are related but with different names
>>>>>>>> (such as "inactive" and "sendrecv"), does the attribute
>>>>>>>> at the media-level have precedence over the value at the
>>>>>>>> session-level?  If so, I recommend that the draft be
>>>>>>>> updated to clarify the topic.
>>>>>>> I believe so. at least that is my understanding from the
>>>>>>> text. Does it really need clarification? Thoughts?
>>>>>> I think that it should be clarified because it is unclear if "an
>>>>>> attribute of the same name" indicates the typical situation or only
>>>>>> situation concerning application of "unless overridden by an
>>>>>> equivalent media-level value".  For instance if I sent "a=3Dinactive=
"
>>>>>> at session-level and "a=3Dsendrecv" at media-level, I'd be concerned
>>>>>> that the receiver may interpret the SDP as being
>>>>>> abnormal (such as including both values at media-level) instead of
>>>>>> allowing "a=3Dsendrecv" to override "a=3Dinactive".
>>>>>>=20
>>>>>> In case it matters from an extendibility perspective, requiring the
>>>>>> same name might make it easier to recognize the override.
>>>>>> However since section 6 already introduces complications with
>>>>>> defaulting to "sendrecv" if a new option is defined/used and
>>>>>> unknown, allowing the override even when the names are not the same
>>>>>> doesn't make things too much worse; it actually
>>>>>> allows communication of a backup attribute (such as "a=3Dinactive") =
if
>>>>>> they don't understand the preferred new attribute (such
>>>>>> as "a=3Dfoo").
>>>>>>=20
>>>>>> "If none of the attributes "sendonly", "recvonly", "inactive",
>>>>>> and "sendrecv" is present, "sendrecv" SHOULD be assumed as the
>>>>>> default for sessions that are not of the conference type
>>>>>> "broadcast" or "H332" (see below)."
>>>>>>=20
>>>>>> Thanks for the response,
>>>>>> Brett
>>>>> _______________________________________________
>>>>> mmusic mailing list
>>>>> mmusic@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>>>=20
>>>> _______________________________________________
>>>> mmusic mailing list
>>>> mmusic@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>> .
>>>=20
>>=20
>>=20
>=20


From pkyzivat@alum.mit.edu  Thu Jan  3 18:29:22 2013
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Subject: Re: [MMUSIC] The rtp-duplication related drafts
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Ali,

On 1/3/13 5:31 PM, Ali C. Begen (abegen) wrote:
> Hi everyone,
>
> The following two drafts have been adopted by MMUSIC a while ago and the main draft discussing the RTP duplication issues is being finalized in AVTEXT (this was also adopted as a WG item a while ago). It is simply waiting for the WG to make a decision on the srcname proposal.
>
> http://datatracker.ietf.org/doc/draft-ietf-mmusic-delayed-duplication/
> http://datatracker.ietf.org/doc/draft-ietf-mmusic-duplication-grouping/

I have a question about these:

Each example shows only one group. But couldn't there be multiple 
duplication groups?

If so, the current proposal has no mechanism for specifying separate 
duplication-delay for each group.

	Thanks,
	Paul


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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
Thread-Topic: [MMUSIC] The rtp-duplication related drafts
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Subject: Re: [MMUSIC] The rtp-duplication related drafts
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Thanks for the review.

On Jan 3, 2013, at 9:29 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> Ali,
>=20
> On 1/3/13 5:31 PM, Ali C. Begen (abegen) wrote:
>> Hi everyone,
>>=20
>> The following two drafts have been adopted by MMUSIC a while ago and the=
 main draft discussing the RTP duplication issues is being finalized in AVT=
EXT (this was also adopted as a WG item a while ago). It is simply waiting =
for the WG to make a decision on the srcname proposal.
>>=20
>> http://datatracker.ietf.org/doc/draft-ietf-mmusic-delayed-duplication/
>> http://datatracker.ietf.org/doc/draft-ietf-mmusic-duplication-grouping/
>=20
> I have a question about these:
>=20
> Each example shows only one group. But couldn't there be multiple duplica=
tion groups?
>=20
There could be.

> If so, the current proposal has no mechanism for specifying separate dupl=
ication-delay for each group.
>=20

a=3Dduplication-delay:X could be specified in each media session. For examp=
le:

	a=3Drtpmap:100 MP2T/90000
        a=3Dssrc:1000 cname:ch1@example.com
        a=3Dssrc:1010 cname:ch1@example.com
        a=3Dssrc-group:DUP 1000 1010
        a=3Dduplication-delay:100
        a=3Dmid:Group1
	a=3Drtpmap:101 MP2T/90000
        a=3Dssrc:2000 cname:ch2@example.com
        a=3Dssrc:2010 cname:ch2@example.com
        a=3Dssrc-group:DUP 2000 2010
        a=3Dduplication-delay:150
        a=3Dmid:Group2

(The SSRC need not be different as SSRC only need to be unique within each =
media session)

For the non-ssrc grouping, your comment holds. It does not support specifyi=
ng different delay values for different groups. Do you think we should supp=
ort distinct delay values for different groups (which probably means duplic=
ation-delay attribute has to be media-level only)?

-acbegen

> 	Thanks,
> 	Paul
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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On 1/2/13 5:25 PM, Eric Rescorla wrote:
>
>
> On Wed, Jan 2, 2013 at 11:44 AM, Bernard Aboba 
> <bernard_aboba@hotmail.com <mailto:bernard_aboba@hotmail.com>> wrote:
>
>     Cullen said:
>
>     "3) if we are going to have an MMUSIC face to face interim
>     meeting, it would need AD approval and 4 weeks notices which puts
>     us very close do the deadline so this needs to get sort out real
>     soon. "
>
>     [BA] I see no particularly good reason why this can't happen
>     quickly.  In fact, why not schedule two MMUSIC interims, 4-6 weeks
>     apart from each other?  If travel is an issue, they can be
>     virtual.  EKR's issue list is a fine starting point, and there is
>     an informal IESG telechat tomorrow (3-Jan) as well as one on
>     10-Jan-2013.
>
>
> I'm not in principle opposed to "virtual" interims, but my general 
> feeling is
> that we should be using face-to-face time to cover the most important and
> intractable issues. ISTM that those are the ones that I listed in my 
> email.
>
> Accordingly, I don't think it makes sense to have virtual MMUSIC interims
> to discuss the important stuff and a F2F RTCWEB interim to discuss the
> rest...
>
While I agree it would be helpful with more face-to-face time, I don't 
believe in relying on this as a mechanism for making continuous 
progress. These types of meetings are generally most productive when 
coupled with regular interaction among a core set of people that 
represent the different points of view in a particular matter and hence 
I would suggest formation of one or more design teams on select topics 
with regular conference calls (we'll need dedicated volunteers for 
those) - of course the output of these design teams will still need to 
be discussed in the WG at large.

One particular area to start with on the bundle/mmt discussion for 
example is to look at how the different proposals handle all existing 
SDP parameters and attributes to see how important their different 
approach really is.

Thanks

-- Flemming (as Individual)

> -Ekr
>
>
>     Ted's suggestion (to cover this in the RTCWEB interim) might do in
>     a pinch, but really, MMUSIC WG has to develop a sense of urgency
>     on this.  If things keep going as they have, by the time we get
>     published RFCs out of MMUSIC,  implementations will have been in
>     the field for years.
>
>     If MMUSIC were a Hollywood Celebrity, we'd be *way* past the time
>     for "tough love" and on the front page of the National Enquirer
>     for the Nth time.
>
>
>
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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    <br>
    <div class="moz-cite-prefix">On 1/2/13 5:25 PM, Eric Rescorla wrote:<br>
    </div>
    <blockquote
cite="mid:CABcZeBOeokx98Oq=bDRhQK=dc0GKeHf_snC=VCJaSd_M2Mv5og@mail.gmail.com"
      type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=ISO-8859-1">
      <br>
      <br>
      <div class="gmail_quote">On Wed, Jan 2, 2013 at 11:44 AM, Bernard
        Aboba <span dir="ltr">&lt;<a moz-do-not-send="true"
            href="mailto:bernard_aboba@hotmail.com" target="_blank">bernard_aboba@hotmail.com</a>&gt;</span>
        wrote:<br>
        <blockquote class="gmail_quote" style="margin:0 0 0
          .8ex;border-left:1px #ccc solid;padding-left:1ex">
          <div>
            <div dir="ltr">
              <div class="im">Cullen said:<br>
                <br>
                "3) if we are going to have an MMUSIC face to face
                interim meeting, it would need AD approval and 4 weeks
                notices which puts us very close do the deadline so this
                needs to get sort out real soon. "<br>
              </div>
              <div><br>
                [BA] I see no particularly good reason why this can't&nbsp;
                happen quickly.&nbsp; In fact, why not schedule two MMUSIC
                interims, 4-6 weeks apart from each other?&nbsp; If travel is
                an issue, they can be virtual.&nbsp; EKR's issue list is a
                fine starting point, and there is an informal IESG
                telechat tomorrow (3-Jan) as well as one on
                10-Jan-2013.&nbsp;<br>
              </div>
            </div>
          </div>
        </blockquote>
        <div><br>
        </div>
        <div>I'm not in principle opposed to "virtual" interims, but my
          general feeling is</div>
        <div>that we should be using face-to-face time to cover the most
          important and</div>
        <div>intractable issues. ISTM that those are the ones that I
          listed in my email.</div>
        <div><br>
        </div>
        <div>Accordingly, I don't think it makes sense to have virtual
          MMUSIC interims</div>
        <div>to discuss the important stuff and a F2F RTCWEB interim to
          discuss the</div>
        <div>rest...</div>
        <div><br>
        </div>
      </div>
    </blockquote>
    While I agree it would be helpful with more face-to-face time, I
    don't believe in relying on this as a mechanism for making
    continuous progress. These types of meetings are generally most
    productive when coupled with regular interaction among a core set of
    people that represent the different points of view in a particular
    matter and hence I would suggest formation of one or more design
    teams on select topics with regular conference calls (we'll need
    dedicated volunteers for those) - of course the output of these
    design teams will still need to be discussed in the WG at large.<br>
    <br>
    One particular area to start with on the bundle/mmt discussion for
    example is to look at how the different proposals handle all
    existing SDP parameters and attributes to see how important their
    different approach really is. <br>
    <br>
    Thanks <br>
    <br>
    -- Flemming (as Individual)<br>
    <br>
    <blockquote
cite="mid:CABcZeBOeokx98Oq=bDRhQK=dc0GKeHf_snC=VCJaSd_M2Mv5og@mail.gmail.com"
      type="cite">
      <div class="gmail_quote">
        <div>-Ekr</div>
        <div><br>
        </div>
        <div><br>
        </div>
        <blockquote class="gmail_quote" style="margin:0 0 0
          .8ex;border-left:1px #ccc solid;padding-left:1ex">
          <div>
            <div dir="ltr">
              <div>Ted's suggestion (to cover this in the RTCWEB
                interim) might do in a pinch, but really, MMUSIC WG has
                to develop a sense of urgency on this.&nbsp; If things keep
                going as they have, by the time we get published RFCs
                out of MMUSIC,&nbsp; implementations will have been in the
                field for years.&nbsp; <br>
                <br>
                If MMUSIC were a Hollywood Celebrity, we'd be *way* past
                the time for "tough love" and on the front page of the
                National Enquirer for the Nth time. <br>
                <br>
                <br>
                <br>
              </div>
            </div>
          </div>
        </blockquote>
      </div>
      <br>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
mmusic mailing list
<a class="moz-txt-link-abbreviated" href="mailto:mmusic@ietf.org">mmusic@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a>
</pre>
    </blockquote>
    <br>
  </body>
</html>

--------------020306090106000302040102--

From pkyzivat@alum.mit.edu  Thu Jan  3 19:31:48 2013
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Date: Thu, 03 Jan 2013 22:31:45 -0500
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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To: "Ali C. Begen (abegen)" <abegen@cisco.com>
References: <C15918F2FCDA0243A7C919DA7C4BE9940CDE1808@xmb-aln-x01.cisco.com> <50E60DDF.6070204@cisco.com> <50E639D6.90105@alum.mit.edu> <C15918F2FCDA0243A7C919DA7C4BE9940CDFB627@xmb-aln-x01.cisco.com>
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Cc: "Flemming Andreasen \(fandreas\)" <fandreas@cisco.com>, "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] Draft-ietf-mmusic-rfc4566bis-05: session-level attributes
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On 1/3/13 9:14 PM, Ali C. Begen (abegen) wrote:

> Are you saying we don't change the text or remove it altogether?

Major change.

OLD:

    The connection ("c=") and attribute ("a=") information in the
    session-level section applies to all the media of that session unless
    overridden by connection information or an attribute of the same name
    in the media description.  For instance, in the example below, each
    media behaves as if it were given a "recvonly" attribute.

    An example SDP description is:

       v=0
       o=jdoe 2890844526 2890842807 IN IP4 198.51.100.1
       s=SDP Seminar
       i=A Seminar on the session description protocol
       u=http://www.example.com/seminars/sdp.pdf
       e=j.doe@example.com (Jane Doe)
       c=IN IP4 233.252.0.1/127
       t=2873397496 2873404696
       a=recvonly
       m=audio 49170 RTP/AVP 0
       m=video 51372 RTP/AVP 99
       a=rtpmap:99 h263-1998/90000

NEW:

    The connection ("c=") information in the
    session-level section applies to all the media of that session unless
    overridden by connection information
    in the media description.  For instance, in the example below, each
    audio media behaves as if it were given a "c=IN IP4 192.0.2.3".

    An example SDP description is:

       v=0
       o=jdoe 2890844526 2890842807 IN IP4 198.51.100.1
       s=SDP Seminar
       i=A Seminar on the session description protocol
       u=http://www.example.com/seminars/sdp.pdf
       e=j.doe@example.com (Jane Doe)
       c=IN IP4 192.0.2.3
       t=2873397496 2873404696
       a=recvonly
       m=audio 49170 RTP/AVP 0
       m=audio 49180 RTP/AVP 0
       m=video 51372 RTP/AVP 99
       c=IN IP4 233.252.0.1/127
       a=rtpmap:99 h263-1998/90000

This drops all discussion of attributes and just describes the behavior 
for c-lines.

>> And at the same time we should polish up the definitions of the attributes defined in this document so that they are very clear how session and media level interact.
>
> Could you provide an example of what you have in mind?

I'll try. (But please review this carefully!) In section 6:

OLD:

       a=recvonly

          This specifies that the tools should be started in receive-only
          mode where applicable.  It can be either a session- or media-
          level attribute, and it is not dependent on charset.  Note that
          recvonly applies to the media only, not to any associated
          control protocol (e.g., an RTP-based system in recvonly mode
          SHOULD still send RTCP packets).

       a=sendrecv

          This specifies that the tools should be started in send and
          receive mode.  This is necessary for interactive conferences
          with tools that default to receive-only mode.  It can be either
          a session or media-level attribute, and it is not dependent on
          charset.

          If none of the attributes "sendonly", "recvonly", "inactive",
          and "sendrecv" is present, "sendrecv" SHOULD be assumed as the
          default for sessions that are not of the conference type
          "broadcast" or "H332" (see below).

       a=sendonly

          This specifies that the tools should be started in send-only
          mode.  An example may be where a different unicast address is
          to be used for a traffic destination than for a traffic source.
          In such a case, two media descriptions may be used, one
          sendonly and one recvonly.  It can be either a session- or
          media-level attribute, but would normally only be used as a
          media attribute.  It is not dependent on charset.  Note that
          sendonly applies only to the media, and any associated control
          protocol (e.g., RTCP) SHOULD still be received and processed as
          normal.

       a=inactive

          This specifies that the tools should be started in inactive
          mode.  This is necessary for interactive conferences where
          users can put other users on hold.  No media is sent over an
          inactive media stream.  Note that an RTP-based system SHOULD
          still send RTCP, even if started inactive.  It can be either a
          session or media-level attribute, and it is not dependent on
          charset.

NEW:

       a=recvonly, a=sendrecv, a=sendonly, a=inactive

          At most one of recvonly/sendrecv/sendonly/inactive MAY appear
          at session level, and at most one MAY appear in each media
          section.

          If any one of these appears in a media section then it
          applies for that media section. If none appear in a media
          section then the one from session level, if any, applies
          to that media section.

          If none of the attributes "sendonly", "recvonly", "inactive",
          and "sendrecv" is present at either session level or media
          level, "sendrecv" SHOULD be assumed as the
          default for sessions that are not of the conference type
          "broadcast" or "H332" (see below).

       For instance, in the
       example below, each media behaves as if it were given a
       "recvonly" attribute.

       An example SDP description is:

          v=0
          o=jdoe 2890844526 2890842807 IN IP4 198.51.100.1
          s=SDP Seminar
          i=A Seminar on the session description protocol
          u=http://www.example.com/seminars/sdp.pdf
          e=j.doe@example.com (Jane Doe)
          c=IN IP4 233.252.0.1/127
          t=2873397496 2873404696
          a=recvonly
          m=audio 49170 RTP/AVP 0
          m=video 51372 RTP/AVP 99
          a=rtpmap:99 h263-1998/90000
          a=recvonly

       a=recvonly

          This specifies that the tools should be started in receive-only
          mode where applicable.  It can be either a session- or media-
          level attribute, and it is not dependent on charset.  Note that
          recvonly applies to the media only, not to any associated
          control protocol (e.g., an RTP-based system in recvonly mode
          SHOULD still send RTCP packets).

       a=sendrecv

          This specifies that the tools should be started in send and
          receive mode.  This is necessary for interactive conferences
          with tools that default to receive-only mode.  It can be either
          a session or media-level attribute, and it is not dependent on
          charset.

       a=sendonly

          This specifies that the tools should be started in send-only
          mode.  An example may be where a different unicast address is
          to be used for a traffic destination than for a traffic source.
          In such a case, two media descriptions may be used, one
          sendonly and one recvonly.  It can be either a session- or
          media-level attribute.  It is not dependent on charset.
          Note that
          sendonly applies only to the media, and any associated control
          protocol (e.g., RTCP) SHOULD still be received and processed as
          normal.

       a=inactive

          This specifies that the tools should be started in inactive
          mode.  This is necessary for interactive conferences where
          users can put other users on hold.  No media is sent over an
          inactive media stream.  Note that an RTP-based system SHOULD
          still send RTCP, even if started inactive.  It can be either a
          session or media-level attribute, and it is not dependent on
          charset.

Also in section 6:

OLD:

       a=sdplang:<language tag>

          This can be a session-level attribute or a media-level
          attribute.  As a session-level attribute, it specifies the
          language for the session description.  As a media-level
          attribute, it specifies the language for any media-level SDP
          information field associated with that media.  Multiple sdplang
          attributes can be provided either at session or media level if
          the session description or media use multiple languages.

NEW:

       a=sdplang:<language tag>

          This can be a session-level attribute or a media-level
          attribute. Multiple sdplang attributes can be provided
          either at session or media level if the session description
          or media use multiple languages.

          As a session-level attribute, it specifies the
          language for the session description. As a media-level
          attribute, it specifies the language for any media-level SDP
          information field associated with that media, overriding any
          sdplang attributes specified at session-level.

OLD:

       a=lang:<language tag>

          This can be a session-level attribute or a media-level
          attribute.  As a session-level attribute, it specifies the
          default language for the session being described.  As a media-
          level attribute, it specifies the language for that media,
          overriding any session-level language specified.  Multiple lang
          attributes can be provided either at session or media level if
          the session or media use multiple languages, in which case the
          order of the attributes indicates the order of importance of
          the various languages in the session or media, from most
          important to least important.

NEW:

       a=lang:<language tag>

          This can be a session-level attribute or a media-level
          attribute. Multiple lang
          attributes can be provided either at session or media level if
          the session or media use multiple languages, in which case the
          order of the attributes indicates the order of importance of
          the various languages in the session or media, from most
          important to least important.

          As a session-level attribute, it specifies the
          default language for the session being described.
          As a media-level attribute, it specifies the language for
          that media, overriding any session-level languages specified.

That's all.

	Thanks,
	Paul

From pkyzivat@alum.mit.edu  Thu Jan  3 19:41:34 2013
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Subject: Re: [MMUSIC] The rtp-duplication related drafts
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On 1/3/13 9:47 PM, Ali C. Begen (abegen) wrote:
> Thanks for the review.
>
> On Jan 3, 2013, at 9:29 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>
>> Ali,
>>
>> On 1/3/13 5:31 PM, Ali C. Begen (abegen) wrote:
>>> Hi everyone,
>>>
>>> The following two drafts have been adopted by MMUSIC a while ago and the main draft discussing the RTP duplication issues is being finalized in AVTEXT (this was also adopted as a WG item a while ago). It is simply waiting for the WG to make a decision on the srcname proposal.
>>>
>>> http://datatracker.ietf.org/doc/draft-ietf-mmusic-delayed-duplication/
>>> http://datatracker.ietf.org/doc/draft-ietf-mmusic-duplication-grouping/
>>
>> I have a question about these:
>>
>> Each example shows only one group. But couldn't there be multiple duplication groups?
>>
> There could be.
>
>> If so, the current proposal has no mechanism for specifying separate duplication-delay for each group.
>>
>
> a=duplication-delay:X could be specified in each media session. For example:
>
> 	a=rtpmap:100 MP2T/90000
>          a=ssrc:1000 cname:ch1@example.com
>          a=ssrc:1010 cname:ch1@example.com
>          a=ssrc-group:DUP 1000 1010
>          a=duplication-delay:100
>          a=mid:Group1
> 	a=rtpmap:101 MP2T/90000
>          a=ssrc:2000 cname:ch2@example.com
>          a=ssrc:2010 cname:ch2@example.com
>          a=ssrc-group:DUP 2000 2010
>          a=duplication-delay:150
>          a=mid:Group2

This makes the attributes very order dependent. If this is intended, 
then I think you need more words to say that. Currently it says:

    If used as a media-level
    attribute, it MUST be used with the 'ssrc-group' attribute and "DUP"
    grouping semantics...

What does "used with" mean? I assumed it simply meant that both 
attributes had to be present in the media section. But what you are 
saying now says that there is some very specific ordering required. E.g. 
that the a=duplicaation-delay must follow the a=ssrc-group:DUP to which 
it applies, and precede any other a=ssrc-group:DUP. This is possible, 
but ugly to specify.

> (The SSRC need not be different as SSRC only need to be unique within each media session)
>
> For the non-ssrc grouping, your comment holds. It does not support specifying different delay values for different groups. Do you think we should support distinct delay values for different groups (which probably means duplication-delay attribute has to be media-level only)?

I don't know. Nothing seems appealing.

	Thanks,
	Paul

> -acbegen
>
>> 	Thanks,
>> 	Paul
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>
>


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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
Thread-Topic: [MMUSIC] The rtp-duplication related drafts
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Subject: Re: [MMUSIC] The rtp-duplication related drafts
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On Jan 3, 2013, at 10:41 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> On 1/3/13 9:47 PM, Ali C. Begen (abegen) wrote:
>> Thanks for the review.
>>=20
>> On Jan 3, 2013, at 9:29 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>>=20
>>> Ali,
>>>=20
>>> On 1/3/13 5:31 PM, Ali C. Begen (abegen) wrote:
>>>> Hi everyone,
>>>>=20
>>>> The following two drafts have been adopted by MMUSIC a while ago and t=
he main draft discussing the RTP duplication issues is being finalized in A=
VTEXT (this was also adopted as a WG item a while ago). It is simply waitin=
g for the WG to make a decision on the srcname proposal.
>>>>=20
>>>> http://datatracker.ietf.org/doc/draft-ietf-mmusic-delayed-duplication/
>>>> http://datatracker.ietf.org/doc/draft-ietf-mmusic-duplication-grouping=
/
>>>=20
>>> I have a question about these:
>>>=20
>>> Each example shows only one group. But couldn't there be multiple dupli=
cation groups?
>>>=20
>> There could be.
>>=20
>>> If so, the current proposal has no mechanism for specifying separate du=
plication-delay for each group.
>>>=20
>>=20
>> a=3Dduplication-delay:X could be specified in each media session. For ex=
ample:
>>=20
>> 	a=3Drtpmap:100 MP2T/90000
>>         a=3Dssrc:1000 cname:ch1@example.com
>>         a=3Dssrc:1010 cname:ch1@example.com
>>         a=3Dssrc-group:DUP 1000 1010
>>         a=3Dduplication-delay:100
>>         a=3Dmid:Group1
>> 	a=3Drtpmap:101 MP2T/90000
>>         a=3Dssrc:2000 cname:ch2@example.com
>>         a=3Dssrc:2010 cname:ch2@example.com
>>         a=3Dssrc-group:DUP 2000 2010
>>         a=3Dduplication-delay:150
>>         a=3Dmid:Group2
>=20
> This makes the attributes very order dependent. If this is intended, then=
 I think you need more words to say that. Currently it says:
>=20
>   If used as a media-level
>   attribute, it MUST be used with the 'ssrc-group' attribute and "DUP"
>   grouping semantics...

My mistake, did not copy/paste the whole thing.

	m=3Dvideo 30000 RTP/AVP 100
        c=3DIN IP4 233.252.0.1/127
        a=3Dsource-filter:incl IN IP4 233.252.0.1 198.51.100.1
        a=3Drtpmap:100 MP2T/90000
        a=3Dssrc:1000 cname:ch1@example.com
        a=3Dssrc:1010 cname:ch1@example.com
        a=3Dssrc-group:DUP 1000 1010
        a=3Dduplication-delay:100
        a=3Dmid:Group1
	m=3Dvideo 40000 RTP/AVP 101
        c=3DIN IP4 233.252.0.2/127
        a=3Dsource-filter:incl IN IP4 233.252.0.2 198.51.100.1
        a=3Drtpmap:101 MP2T/90000
        a=3Dssrc:2000 cname:ch2@example.com
        a=3Dssrc:2010 cname:ch2@example.com
        a=3Dssrc-group:DUP 2000 2010
        a=3Dduplication-delay:150
        a=3Dmid:Group2

so there is one duplication-delay attribute in each media session (specifie=
d by a m=3D line). It does not matter whether it is before or after the ssr=
c-group line. It is OK as long as it is somewhere before the next m line.

>=20
> What does "used with" mean? I assumed it simply meant that both attribute=
s had to be present in the media section. But what you are saying now says =
that there is some very specific ordering required. E.g. that the a=3Ddupli=
caation-delay must follow the a=3Dssrc-group:DUP to which it applies, and p=
recede any other a=3Dssrc-group:DUP. This is possible, but ugly to specify.

I hope it is clear now?

>=20
>> (The SSRC need not be different as SSRC only need to be unique within ea=
ch media session)
>>=20
>> For the non-ssrc grouping, your comment holds. It does not support speci=
fying different delay values for different groups. Do you think we should s=
upport distinct delay values for different groups (which probably means dup=
lication-delay attribute has to be media-level only)?
>=20
> I don't know. Nothing seems appealing.
>=20
> 	Thanks,
> 	Paul
>=20
>> -acbegen
>>=20
>>> 	Thanks,
>>> 	Paul
>>>=20
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>=20
>>=20
>=20


From internet-drafts@ietf.org  Fri Jan  4 15:16:41 2013
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-media-capabilities-17.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Multiparty Multimedia Session Control Wor=
king Group of the IETF.

	Title           : Session Description Protocol (SDP) Media Capabilities Ne=
gotiation
	Author(s)       : Robert R Gilman
                          Roni Even
                          Flemming Andreasen
	Filename        : draft-ietf-mmusic-sdp-media-capabilities-17.txt
	Pages           : 64
	Date            : 2013-01-04

Abstract:
   Session Description Protocol (SDP) capability negotiation provides a
   general framework for indicating and negotiating capabilities in SDP.
   The base framework defines only capabilities for negotiating
   transport protocols and attributes.  In this document, we extend the
   framework by defining media capabilities that can be used to
   negotiate media types and their associated parameters.

   This document updates the IANA Considerations of RFC 5939.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-media-capabilities

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-sdp-media-capabilities-17

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sdp-media-capabilities=
-17


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


From iesg-secretary@ietf.org  Mon Jan  7 07:39:49 2013
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Subject: [MMUSIC] Protocol Action: 'Session Description Protocol (SDP) Media	Capabilities Negotiation' to Proposed Standard	(draft-ietf-mmusic-sdp-media-capabilities-17.txt)
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The IESG has approved the following document:
- 'Session Description Protocol (SDP) Media Capabilities Negotiation'
  (draft-ietf-mmusic-sdp-media-capabilities-17.txt) as Proposed Standard

This document is the product of the Multiparty Multimedia Session Control
Working Group.

The IESG contact persons are Gonzalo Camarillo and Robert Sparks.

A URL of this Internet Draft is:
http://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-media-capabilities/




Technical Summary

The Session Description Protocol (SDP) has a general framework for
endpoints to indicate and negotiate capabilities within SDP. However,
the base framework defines capabilities for negotiating transport
protocols and attributes. In this document, the SDP capability
negotiation framework is extended with the ability to additionally
indicate and negotiate media types and their associated parameters.

Working Group Summary

The first version of the document is dating February 2007. Since
then, the MMUSIC working group has been progressing the document
until getting satisfied with the current version.

Document Quality

The document has been reviewed and contributed by many participants
of the working group, among others: Culleng Jennings, Christer
Holmberg, Matt Lepinski, Joerg Ott, Colin Perkins, Thomas Stach,
Ingemar Johansson, Andrew Allen, and Magnus Westerlund.

The document was first WGLCed on version 10 (July 2010) and subsequently
on version 14 (July 2012). All the open issues have been addressed
and the WG has got consensus on version 15.

Personnel

Miguel Garcia is the Document Shepherd. Gonzalo Camarillo is the
Responsible Area Director. 

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Subject: [MMUSIC] RTCWEB & MMUSIC WG Joint Interim Meeting, February 5-7, 2013
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The RTCWEB and MMUSIC working groups would like to announce an upcoming
joint interim meeting.

The meeting will be held in the Boston, Massachusetts area on February
5-7th, 2013, in conjunction with meetings of the W3C WEBRTC working
group. Full details, including meeting site, nearby hotels, and other
logistics have been posted to both the rtcweb and mmusic mailing lists =

(http://www.ietf.org/mail-archive/web/rtcweb/current/maillist.html, =

http://www.ietf.org/mail-archive/web/mmusic/current/maillist.html).

From fandreas@cisco.com  Mon Jan  7 10:36:54 2013
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Subject: [MMUSIC] Fwd: RTCWEB & MMUSIC WG Joint Interim Meeting, February 5-7, 2013
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Re-sending to MMUSIC list only.

Thanks

-- Flemming


-------- Original Message --------
Subject: 	RTCWEB & MMUSIC WG Joint Interim Meeting, February 5-7, 2013
Date: 	Mon, 7 Jan 2013 09:30:18 -0800
From: 	IESG Secretary <iesg-secretary@ietf.org>
To: 	IETF Announcement List <ietf-announce@ietf.org>
CC: 	<rtcweb@ietf.org>, <mmusic@ietf.org>



The RTCWEB and MMUSIC working groups would like to announce an upcoming
joint interim meeting.

The meeting will be held in the Boston, Massachusetts area on February
5-7th, 2013, in conjunction with meetings of the W3C WEBRTC working
group. Full details, including meeting site, nearby hotels, and other
logistics have been posted to both the rtcweb and mmusic mailing lists
(http://www.ietf.org/mail-archive/web/rtcweb/current/maillist.html,
http://www.ietf.org/mail-archive/web/mmusic/current/maillist.html).
.




--------------060009060205040302040807
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    Re-sending to MMUSIC list only. <br>
    <br>
    Thanks <br>
    <br>
    -- Flemming <br>
    <div class="moz-forward-container"><br>
      <br>
      -------- Original Message --------
      <table class="moz-email-headers-table" border="0" cellpadding="0"
        cellspacing="0">
        <tbody>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject:
            </th>
            <td>RTCWEB &amp; MMUSIC WG Joint Interim Meeting, February
              5-7, 2013</td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date: </th>
            <td>Mon, 7 Jan 2013 09:30:18 -0800</td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From: </th>
            <td>IESG Secretary <a class="moz-txt-link-rfc2396E" href="mailto:iesg-secretary@ietf.org">&lt;iesg-secretary@ietf.org&gt;</a></td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To: </th>
            <td>IETF Announcement List <a class="moz-txt-link-rfc2396E" href="mailto:ietf-announce@ietf.org">&lt;ietf-announce@ietf.org&gt;</a></td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">CC: </th>
            <td><a class="moz-txt-link-rfc2396E" href="mailto:rtcweb@ietf.org">&lt;rtcweb@ietf.org&gt;</a>, <a class="moz-txt-link-rfc2396E" href="mailto:mmusic@ietf.org">&lt;mmusic@ietf.org&gt;</a></td>
          </tr>
        </tbody>
      </table>
      <br>
      <br>
      <pre>The RTCWEB and MMUSIC working groups would like to announce an upcoming
joint interim meeting.

The meeting will be held in the Boston, Massachusetts area on February
5-7th, 2013, in conjunction with meetings of the W3C WEBRTC working
group. Full details, including meeting site, nearby hotels, and other
logistics have been posted to both the rtcweb and mmusic mailing lists 
(<a class="moz-txt-link-freetext" href="http://www.ietf.org/mail-archive/web/rtcweb/current/maillist.html">http://www.ietf.org/mail-archive/web/rtcweb/current/maillist.html</a>, 
<a class="moz-txt-link-freetext" href="http://www.ietf.org/mail-archive/web/mmusic/current/maillist.html">http://www.ietf.org/mail-archive/web/mmusic/current/maillist.html</a>).
.

</pre>
      <br>
    </div>
    <br>
  </body>
</html>

--------------060009060205040302040807--

From pkyzivat@alum.mit.edu  Mon Jan  7 12:06:44 2013
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Cc: "<mmusic@ietf.org>" <mmusic@ietf.org>
Subject: Re: [MMUSIC] The rtp-duplication related drafts
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On 1/3/13 11:20 PM, Ali C. Begen (abegen) wrote:
>
> On Jan 3, 2013, at 10:41 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>
>> On 1/3/13 9:47 PM, Ali C. Begen (abegen) wrote:
>>> Thanks for the review.
>>>
>>> On Jan 3, 2013, at 9:29 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>>>
>>>> Ali,
>>>>
>>>> On 1/3/13 5:31 PM, Ali C. Begen (abegen) wrote:
>>>>> Hi everyone,
>>>>>
>>>>> The following two drafts have been adopted by MMUSIC a while ago and the main draft discussing the RTP duplication issues is being finalized in AVTEXT (this was also adopted as a WG item a while ago). It is simply waiting for the WG to make a decision on the srcname proposal.
>>>>>
>>>>> http://datatracker.ietf.org/doc/draft-ietf-mmusic-delayed-duplication/
>>>>> http://datatracker.ietf.org/doc/draft-ietf-mmusic-duplication-grouping/
>>>>
>>>> I have a question about these:
>>>>
>>>> Each example shows only one group. But couldn't there be multiple duplication groups?
>>>>
>>> There could be.
>>>
>>>> If so, the current proposal has no mechanism for specifying separate duplication-delay for each group.
>>>>
>>>
>>> a=duplication-delay:X could be specified in each media session. For example:
>>>
>>> 	a=rtpmap:100 MP2T/90000
>>>          a=ssrc:1000 cname:ch1@example.com
>>>          a=ssrc:1010 cname:ch1@example.com
>>>          a=ssrc-group:DUP 1000 1010
>>>          a=duplication-delay:100
>>>          a=mid:Group1
>>> 	a=rtpmap:101 MP2T/90000
>>>          a=ssrc:2000 cname:ch2@example.com
>>>          a=ssrc:2010 cname:ch2@example.com
>>>          a=ssrc-group:DUP 2000 2010
>>>          a=duplication-delay:150
>>>          a=mid:Group2
>>
>> This makes the attributes very order dependent. If this is intended, then I think you need more words to say that. Currently it says:
>>
>>    If used as a media-level
>>    attribute, it MUST be used with the 'ssrc-group' attribute and "DUP"
>>    grouping semantics...
>
> My mistake, did not copy/paste the whole thing.
>
> 	m=video 30000 RTP/AVP 100
>          c=IN IP4 233.252.0.1/127
>          a=source-filter:incl IN IP4 233.252.0.1 198.51.100.1
>          a=rtpmap:100 MP2T/90000
>          a=ssrc:1000 cname:ch1@example.com
>          a=ssrc:1010 cname:ch1@example.com
>          a=ssrc-group:DUP 1000 1010
>          a=duplication-delay:100
>          a=mid:Group1
> 	m=video 40000 RTP/AVP 101
>          c=IN IP4 233.252.0.2/127
>          a=source-filter:incl IN IP4 233.252.0.2 198.51.100.1
>          a=rtpmap:101 MP2T/90000
>          a=ssrc:2000 cname:ch2@example.com
>          a=ssrc:2010 cname:ch2@example.com
>          a=ssrc-group:DUP 2000 2010
>          a=duplication-delay:150
>          a=mid:Group2
>
> so there is one duplication-delay attribute in each media session (specified by a m= line). It does not matter whether it is before or after the ssrc-group line. It is OK as long as it is somewhere before the next m line.

But then, duplication-delay applies to a=ssrc-group:DUP groups within 
that media section. You could have two groups of ssrcs in there. So the 
problem is the same as the problem across different m-lines. E.g.

	m=video 30000 RTP/AVP 100
         c=IN IP4 233.252.0.1/127
         a=source-filter:incl IN IP4 233.252.0.1 198.51.100.1
         a=rtpmap:100 MP2T/90000
         a=ssrc:1000 cname:ch1@example.com
         a=ssrc:1010 cname:ch1@example.com
         a=ssrc:1020 cname:ch1@example.com
         a=ssrc:1030 cname:ch1@example.com
         a=ssrc-group:DUP 1000 1010
         a=duplication-delay:100
         a=ssrc-group:DUP 1020 1030
         a=duplication-delay:200
	m=video 40000 RTP/AVP 101
         c=IN IP4 233.252.0.2/127
         a=source-filter:incl IN IP4 233.252.0.2 198.51.100.1
         a=rtpmap:101 MP2T/90000
         a=ssrc:2000 cname:ch2@example.com
         a=ssrc:2010 cname:ch2@example.com
         a=ssrc-group:DUP 2000 2010
         a=ssrc:2020 cname:ch2@example.com
         a=ssrc:2030 cname:ch2@example.com
         a=ssrc-group:DUP 2020 2030
         a=duplication-delay:250

 From what you say, the two duplication-delay attributes in one media 
section would make no sense.

>> What does "used with" mean? I assumed it simply meant that both attributes had to be present in the media section. But what you are saying now says that there is some very specific ordering required. E.g. that the a=duplicaation-delay must follow the a=ssrc-group:DUP to which it applies, and precede any other a=ssrc-group:DUP. This is possible, but ugly to specify.
>
> I hope it is clear now?
>
>>
>>> (The SSRC need not be different as SSRC only need to be unique within each media session)
>>>
>>> For the non-ssrc grouping, your comment holds. It does not support specifying different delay values for different groups. Do you think we should support distinct delay values for different groups (which probably means duplication-delay attribute has to be media-level only)?

Well, for the non-ssrc grouping you could put the group name in the 
a=duplication-delay attribute value.

And for the ssrc-grouping case you could put one of the grouped ssrcs in 
the a=duplication-delay attr value.

	Thanks,
	Paul


From fandreas@cisco.com  Mon Jan  7 17:08:00 2013
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Date: Mon, 07 Jan 2013 20:07:58 -0500
From: Flemming Andreasen <fandreas@cisco.com>
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Subject: Re: [MMUSIC] Fwd: RTCWEB & MMUSIC WG Joint Interim Meeting, February 5-7, 2013
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For further details on meeting site, hotels, etc., please see:

	http://www.ietf.org/mail-archive/web/rtcweb/current/msg05826.html


Stay tuned for more details around agenda, etc.

Thanks

-- Flemming


On 1/7/13 1:36 PM, Flemming Andreasen wrote:
> Re-sending to MMUSIC list only.
>
> Thanks
>
> -- Flemming
>
>
> -------- Original Message --------
> Subject: 	RTCWEB & MMUSIC WG Joint Interim Meeting, February 5-7, 2013
> Date: 	Mon, 7 Jan 2013 09:30:18 -0800
> From: 	IESG Secretary <iesg-secretary@ietf.org>
> To: 	IETF Announcement List <ietf-announce@ietf.org>
> CC: 	<rtcweb@ietf.org>, <mmusic@ietf.org>
>
>
>
> The RTCWEB and MMUSIC working groups would like to announce an upcoming
> joint interim meeting.
>
> The meeting will be held in the Boston, Massachusetts area on February
> 5-7th, 2013, in conjunction with meetings of the W3C WEBRTC working
> group. Full details, including meeting site, nearby hotels, and other
> logistics have been posted to both the rtcweb and mmusic mailing lists
> (http://www.ietf.org/mail-archive/web/rtcweb/current/maillist.html,
> http://www.ietf.org/mail-archive/web/mmusic/current/maillist.html).
> .
>
>
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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      http-equiv="Content-Type">
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  <body bgcolor="#FFFFFF" text="#000000">
    For further details on meeting site, hotels, etc., please see:<br>
    <pre wrap="">	<a class="moz-txt-link-freetext" href="http://www.ietf.org/mail-archive/web/rtcweb/current/msg05826.html">http://www.ietf.org/mail-archive/web/rtcweb/current/msg05826.html</a></pre>
    <br>
    Stay tuned for more details around agenda, etc. <br>
    <br>
    Thanks <br>
    <br>
    -- Flemming <br>
    <br>
    <br>
    <div class="moz-cite-prefix">On 1/7/13 1:36 PM, Flemming Andreasen
      wrote:<br>
    </div>
    <blockquote cite="mid:50EB15C3.2010707@cisco.com" type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=ISO-8859-1">
      Re-sending to MMUSIC list only. <br>
      <br>
      Thanks <br>
      <br>
      -- Flemming <br>
      <div class="moz-forward-container"><br>
        <br>
        -------- Original Message --------
        <table class="moz-email-headers-table" border="0"
          cellpadding="0" cellspacing="0">
          <tbody>
            <tr>
              <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject:


              </th>
              <td>RTCWEB &amp; MMUSIC WG Joint Interim Meeting, February
                5-7, 2013</td>
            </tr>
            <tr>
              <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date:
              </th>
              <td>Mon, 7 Jan 2013 09:30:18 -0800</td>
            </tr>
            <tr>
              <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From:
              </th>
              <td>IESG Secretary <a moz-do-not-send="true"
                  class="moz-txt-link-rfc2396E"
                  href="mailto:iesg-secretary@ietf.org">&lt;iesg-secretary@ietf.org&gt;</a></td>
            </tr>
            <tr>
              <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To: </th>
              <td>IETF Announcement List <a moz-do-not-send="true"
                  class="moz-txt-link-rfc2396E"
                  href="mailto:ietf-announce@ietf.org">&lt;ietf-announce@ietf.org&gt;</a></td>
            </tr>
            <tr>
              <th align="RIGHT" nowrap="nowrap" valign="BASELINE">CC: </th>
              <td><a moz-do-not-send="true"
                  class="moz-txt-link-rfc2396E"
                  href="mailto:rtcweb@ietf.org">&lt;rtcweb@ietf.org&gt;</a>,
                <a moz-do-not-send="true" class="moz-txt-link-rfc2396E"
                  href="mailto:mmusic@ietf.org">&lt;mmusic@ietf.org&gt;</a></td>
            </tr>
          </tbody>
        </table>
        <br>
        <br>
        <pre>The RTCWEB and MMUSIC working groups would like to announce an upcoming
joint interim meeting.

The meeting will be held in the Boston, Massachusetts area on February
5-7th, 2013, in conjunction with meetings of the W3C WEBRTC working
group. Full details, including meeting site, nearby hotels, and other
logistics have been posted to both the rtcweb and mmusic mailing lists 
(<a moz-do-not-send="true" class="moz-txt-link-freetext" href="http://www.ietf.org/mail-archive/web/rtcweb/current/maillist.html">http://www.ietf.org/mail-archive/web/rtcweb/current/maillist.html</a>, 
<a moz-do-not-send="true" class="moz-txt-link-freetext" href="http://www.ietf.org/mail-archive/web/mmusic/current/maillist.html">http://www.ietf.org/mail-archive/web/mmusic/current/maillist.html</a>).
.

</pre>
        <br>
      </div>
      <br>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
mmusic mailing list
<a class="moz-txt-link-abbreviated" href="mailto:mmusic@ietf.org">mmusic@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a>
</pre>
    </blockquote>
    <br>
  </body>
</html>

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Date: Mon, 07 Jan 2013 22:23:49 -0500
From: Flemming Andreasen <fandreas@cisco.com>
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Cc: mmusic <mmusic@ietf.org>, Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
Subject: [MMUSIC] SDP Directorate: Review of draft-boucadair-mmusic-altc-07
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Hi

I am the assigned SDP directorate reviewer for 
draft-boucadair-mmusic-altc-07

For background on the SDP directorate, please see the FAQ at

     http://www.ietf.org/iesg/directorate/sdp.html

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

Summary:
---------
- The document does not adequately cover how it relates to the existing 
Standards Track mechanism for solving the similar problem in a different 
and more general way (namely ICE [RFC 5245])
- The document has technical issues
- The document has a few minor editorial issues


New SDP Information Elements:
---------------------------
The draft defines a new "a=altc" attribute


Technical Comments:
------------------
1) The Abstract positions the document as an alternative to ANAT, but 
without certain problems experienced by ANAT. The abstract fails to 
mention that a general Standards Track solution in the form of ICE (RFC 
5245) has been defined whereas the solution defined in this document has 
a narrower scope and various limitations.

2) Section 1.1, third paragraph. The document states that ICE is 
processing intensive and has seen limited deployment. While this is 
perhaps a matter of debate, it would be helpful if the document would 
point out the limitations of using instead of ICE. Areas that come to 
mind here are
- lack of NAT traversal (without network/SBC support),
- no connectivity verification (verifying functioning path as well as 
authenticating/correlating initial incoming media packets with signaling),
- support for non-contiguous RTP/RTCP port pairs in alternatives
There are probably others and it would be helpful to point those out to 
potential implementers (the simplicity comes with trade-offs).

3) Section 1.1, third paragraph: The document claims that ICE is not 
usable in closed networks. This is incorrect (it is however correct that 
there are scenarios where ICE does not work, however those scenarios 
should then be explained instead).

4) Section 1.1, fourth paragraph: The document claims to be compliant 
with RFC 6157 because RFC 6157 includes the sentence
<quote>
The use of ICE can be avoided for signaling messages that stay within 
such managed networks
</quote>
The quote is taken out of context however as it applies to a scenario in 
RFC 6157 where the entire network is IPv6 based, whereas the mechanism 
defined here is for scenarios where it is not.

5) Section 3.2, first paragraph
<quote>
A compliant SDP parser will ignore any session description that contains 
attribute lines it does not support.
<quote>
No - only the unsupported attribute lines will be ignored (which is also 
what the rest of the draft implies)

6) Section 4.1, attribute definition
The optional trailing integer part is not defined.

7) Section 4.1, Bullet list, "port"
This does not seem to work with the "a=rtcp" attribute [RFC 4566], and 
hence there may be issues with non-contiguous RTP/RTCP port pair support 
in alternatives.

8) Section 4.1, 4th last paragraph
The mechanism relies on implicit ordering of attribute lines. If an 
intermediary changes the order, it would have unintended side-effects. 
It would be much better to have an explicit ordering mechanism. In the 
absence of that, the potential reordering issue should be called out.


Editorial Comments:
-----------------
1) Section 1.1, second paragraph. Add reference after "DS-Lite"

2) sdp-cap-neg should be SDPCapNeg (multiple occurrences)

3) Section 1.3, first paragraph: Add reference to RFC 4091.

4) Section 1.3, second paragraph: Add text and reference to RFC 6157 for 
a solution to "SIP layer address family interworking"

5) Section 4.2.3
Add RFC reference after "ICE"

6) Section 4.2.4
Add RFC reference after "SDPCapNeg"


Thanks

-- Flemming


From mohamed.boucadair@orange.com  Tue Jan  8 01:59:22 2013
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From: <mohamed.boucadair@orange.com>
To: Flemming Andreasen <fandreas@cisco.com>, "draft-boucadair-mmusic-altc@tools.ietf.org" <draft-boucadair-mmusic-altc@tools.ietf.org>
Date: Tue, 8 Jan 2013 10:59:15 +0100
Thread-Topic: [MMUSIC] SDP Directorate: Review of draft-boucadair-mmusic-altc-07
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Cc: mmusic <mmusic@ietf.org>, Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
Subject: Re: [MMUSIC] SDP Directorate: Review of draft-boucadair-mmusic-altc-07
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Dear Flemming,

Thank you for the review.=20

Please see inline.=20

Cheers,
Med=20

>-----Message d'origine-----
>De : mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org]=20
>De la part de Flemming Andreasen
>Envoy=E9 : mardi 8 janvier 2013 04:24
>=C0 : draft-boucadair-mmusic-altc@tools.ietf.org
>Cc : mmusic; Gonzalo Camarillo
>Objet : [MMUSIC] SDP Directorate: Review of=20
>draft-boucadair-mmusic-altc-07
>
>Hi
>
>I am the assigned SDP directorate reviewer for=20
>draft-boucadair-mmusic-altc-07
>
>For background on the SDP directorate, please see the FAQ at
>
>     http://www.ietf.org/iesg/directorate/sdp.html
>
>Please wait for direction from your document shepherd or AD before=20
>posting a new version of the draft.
>
>Summary:
>---------
>- The document does not adequately cover how it relates to the=20
>existing=20
>Standards Track mechanism for solving the similar problem in a=20
>different=20
>and more general way (namely ICE [RFC 5245])
>- The document has technical issues
>- The document has a few minor editorial issues
>
>
>New SDP Information Elements:
>---------------------------
>The draft defines a new "a=3Daltc" attribute
>
>
>Technical Comments:
>------------------
>1) The Abstract positions the document as an alternative to ANAT, but=20
>without certain problems experienced by ANAT. The abstract fails to=20
>mention that a general Standards Track solution in the form of=20
>ICE (RFC=20
>5245) has been defined whereas the solution defined in this=20
>document has=20
>a narrower scope and various limitations.

Med: I updated the abstract as follows:=20

OLD:

   This document proposes a mechanism which allows to carry multiple IP
   addresses, of different address families (e.g., IPv4, IPv6), in the
   same SDP offer.  The proposed attribute solves the backward
   compatibility problem which plagued ANAT, due to its syntax.

NEW:

   This document proposes a mechanism which allows to carry multiple IP
   addresses, of different address families (e.g., IPv4, IPv6), in the
   same SDP offer.  The proposed attribute solves the backward
   compatibility problem which plagued ANAT, due to its syntax.

   The proposed solution is applicable to scenarios where connectivity
   checks are not required.

>
>2) Section 1.1, third paragraph. The document states that ICE is=20
>processing intensive and has seen limited deployment. While this is=20
>perhaps a matter of debate, it would be helpful if the document would=20
>point out the limitations of using instead of ICE. Areas that come to=20
>mind here are
>- lack of NAT traversal (without network/SBC support),
>- no connectivity verification (verifying functioning path as well as=20
>authenticating/correlating initial incoming media packets with=20
>signaling),
>- support for non-contiguous RTP/RTCP port pairs in alternatives
>There are probably others and it would be helpful to point=20
>those out to=20
>potential implementers (the simplicity comes with trade-offs).

Med: The scope of ALTC is very clear: scenarios where there is no need for =
connectivity checks. This is already mentioned in Section 2 and I tried to =
make it clear in the new proposed abstract text. =20

>
>3) Section 1.1, third paragraph: The document claims that ICE is not=20
>usable in closed networks. This is incorrect (it is however=20
>correct that=20
>there are scenarios where ICE does not work, however those scenarios=20
>should then be explained instead).

Med: closed networks are provided only as an example. Anyway, I updated the=
 text as follows:=20

OLD:

(e.g., closed networks)

NEW:

(e.g., closed networks making use of SBCs)

>
>4) Section 1.1, fourth paragraph: The document claims to be compliant=20
>with RFC 6157 because RFC 6157 includes the sentence
><quote>
>The use of ICE can be avoided for signaling messages that stay within=20
>such managed networks
></quote>
>The quote is taken out of context however as it applies to a=20
>scenario in=20
>RFC 6157 where the entire network is IPv6 based, whereas the mechanism=20
>defined here is for scenarios where it is not.

Med: The text from RFC6157 is:

      Implementations are encouraged to use ICE; however, the normative
      strength of the text above is left at a SHOULD since in some
      managed networks (such as a closed enterprise network) it is
      possible for the administrator to have control over the IP version
      utilized in all nodes and thus deploy an IPv6-only network, for
      example.  The use of ICE can be avoided for signaling messages
      that stay within such managed networks.

IPv6-only is mentioned as an example. The key word IMHO in the above text i=
s "managed networks".=20

>
>5) Section 3.2, first paragraph
><quote>
>A compliant SDP parser will ignore any session description=20
>that contains=20
>attribute lines it does not support.
><quote>
>No - only the unsupported attribute lines will be ignored=20
>(which is also=20
>what the rest of the draft implies)

Med: OK. I updated the text.

>
>6) Section 4.1, attribute definition
>The optional trailing integer part is not defined.

Med: Thanks for catching this. I updated section 4.1.

>
>7) Section 4.1, Bullet list, "port"
>This does not seem to work with the "a=3Drtcp" attribute [RFC 4566], and=20
>hence there may be issues with non-contiguous RTP/RTCP port=20
>pair support=20
>in alternatives.

Med: We updated the text as follows:

OLD:

   Note that for RTCP, if applicable for the given media types, each
   side would act as if the chosen "altc" attribute's port number was in
   the 'm=3D' media line.  Typically, this would mean RTCP is sent to the
   odd +1 of the port number, unless some other attribute determines
   otherwise.

NEW:

   Note that for RTCP, if applicable for the given media types, each
   side would act as if the chosen "altc" attribute's port number was in
   the 'm=3D' media line.  Typically, this would mean RTCP is sent to the
   odd +1 of the port number, unless some other attribute determines
   otherwise.

   For example the RTP/RTCP multiplexing mechanism defined in [RFC5761]
   can still be used with ALTC, such that if both sides support
   multiplexing they will indicate so using the 'a=3Drtcp-mux' attribute
   as defined in [RFC5761]; but the IP connection address and port they
   use may be chosen using the ALTC mechanism.

   If the SDP offerer wishes to use the RTCP attribute defined in
   [RFC3605], a complication can arise since it may not be clear which
   address choice the 'a=3Drtcp' attribute applies to, relative the
   choices offered by ALTC.  Technically RFC 3605 allows indicating the
   address for RTCP explicitly in the 'a=3Drtcp' attribute itself, but
   this is optional and rarely used.  Since RFC 3605 did not prevent
   encoding multiple 'a=3Drtcp' attributes per media description, this
   document recommends encoding the first 'a=3Drtcp' attribute to be for
   the address choice encoded in the "m=3D" line, and a second 'a=3Drtcp'
   attribute for the alternate address provided by the 'a=3Daltc'
   attribute.  In other words, if the 'a=3Drtcp' attribute explicitly
   encodes an address in its attribute, then that applies for ALTC as
   per [RFC3605]; if it does not, then ALTC assumes the first one is for
   the address in the "m=3D" line, and the second is for the alternate.

>
>8) Section 4.1, 4th last paragraph
>The mechanism relies on implicit ordering of attribute lines. If an=20
>intermediary changes the order, it would have unintended side-effects.=20
>It would be much better to have an explicit ordering mechanism. In the=20
>absence of that, the potential reordering issue should be called out.

Med: Is this harmful? If an intermediary entity change the order, this does=
 not impact the session establishment. Note the document already specify ho=
w a remote UA can detect the presence of an intermediary entity changing th=
e "c" line is present in the path.=20

>
>
>Editorial Comments:
>-----------------
>1) Section 1.1, second paragraph. Add reference after "DS-Lite"
>
>2) sdp-cap-neg should be SDPCapNeg (multiple occurrences)
>
>3) Section 1.3, first paragraph: Add reference to RFC 4091.
>
>4) Section 1.3, second paragraph: Add text and reference to=20
>RFC 6157 for=20
>a solution to "SIP layer address family interworking"
>
>5) Section 4.2.3
>Add RFC reference after "ICE"
>
>6) Section 4.2.4
>Add RFC reference after "SDPCapNeg"

Med: I implemented all the proposed changed. Thanks.=20

>
>
>Thanks
>
>-- Flemming
>
>_______________________________________________
>mmusic mailing list
>mmusic@ietf.org
>https://www.ietf.org/mailman/listinfo/mmusic
>=

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: SDP Directorate: Review of draft-ietf-rmt-flute-sdp-03
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Subject: [MMUSIC] SDP Directorate: Review of draft-ietf-rmt-flute-sdp-03
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Hi,



I am the assigned SDP directorate reviewer for

draft-ietf-rmt-flute-sdp-03



For background on the SDP directorate, please see the FAQ at



     http://www.ietf.org/iesg/directorate/sdp.html



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

SUMMARY:
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

I have technical concerns with a number of the technical content of this do=
cument, and at least the "Composite Session" concept, as currently
defined in the document, would need to be discussed in MMUSIC. I think they=
 need to be resolved/clarified before this document can be published.

Regards,

Christer



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




TECHNICAL
=3D=3D=3D=3D=3D=3D=3D=3D=3D

General:
------------

T-GEN-1:              The draft seems to define a new generic SDP mechanism=
, "Composite Session", for grouping media streams into protocol-specific se=
ssions. I am not really sure what that means, but it seems to go outside th=
e scope of FLUTE. I also think it overlaps with the SDP BUNDLE work, and su=
ch work shall be done in MMUSIC.

                "The Composite Session mechanism enables the grouping of me=
dia lines
                in to distinct sessions.  The complete Composite Session se=
mantics
                are protocol-specific - as determined by the protocol id of=
 the
                grouped media lines."


T-GEN-2:              Maybe I've missed it, but it is a little unclear to m=
e why the grouping framework is needed in the first place. If that is descr=
ibed somewhere, please tell me :)

In section 3.2.1, the text says:

                "The Composite Session provides an unambiguous way to defin=
e multiple
                FLUTE sessions as distinct from multiple the media-sessions=
 semantics
                of RTP."

But, what is the technical need for this?


T-GEN-3:              The draft uses the same source address for every m- l=
ine associated with a group, but it uses different addresses in each m- lin=
e. Aren't the media streams symmetric?

T-GEN-4:              There is chapter describing the SDP Offer/Answer aspe=
cts. How is the answer generated? What if the answerer does not support (or=
, supports, but don't want to use) the mechanism?

If the mechanism is not to be used with SDP Offer/Answer, but e.g. for some=
 kind of declaration, I think that needs to be described.


Section 1:
-------------

T-1-1:                    The text says:

                "Note, this document may also be used to describe sessions =
of the
                experimental FLUTE specification [RFC3926]."

I think you need to indicate whether or not there will be any changes in th=
e usage of SDP in that case.


Section 3.2:
---------------

T-3-2:                    I would like to have some input from the MMUSIC c=
ommunity whether "FLUTE/UDP/ESP" is appropriate to describe a IPSec encrypt=
ed FLUTE channel, or whether something like "SFLUTE/UDP" should be used ins=
tead.


Section 3.11:
-----------------

T-3_11-1:             As the syntax mandates at least one fmt value, I sugg=
est that the document specifies which value SHALL be used.



EDITORIAL
=3D=3D=3D=3D=3D=3D=3D=3D=3D

General:
-----------

E-GEN-1:              The Abstract and Introduction needs to explicitly sta=
te that the specification defines a new SDP grouping framework value. The A=
bstract also needs to mention the new SDP protocol values.


Section 1:
-------------

E-1-1:                    s/defines two new protocol identifiers/defines tw=
o new SDP protocol values

E-1-2:                    I suggest to remove "The formal ABNF syntax [RFC5=
234] is used for the attributes." It only needs to be stated in the section=
 defining the new SDP attributes.


Section 3:
-------------

E-3-1:                    I think the second last paragraph, comparing FLUT=
E with RTP media streams, sessions etc is confusing. I think it is enough t=
o describe that each SDP m- line represents a FLUTE channel.


Section 3.1:
---------------

E-3_1-1:               I would suggest to have a new parent chapter, "SDP E=
xtensions", with sub-chapters defining the new SDP protocol values, attribu=
tes etc.


Section 3.11:
-----------------

E-3_11-1:             I suggest to remove this section. It's not needed, an=
d I don't even agree with the content.


Section 4:
-------------

E-4-1:                    Instead of saying "a=3DFEC-declaration line", I s=
uggest to say "SDP FEC-declaration attribute". (Also applies to the descrip=
tion of the other attributes).







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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">Hi,<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">I am the assigned SDP directorate reviewer for<o:=
p></o:p></p>
<p class=3D"MsoPlainText">draft-ietf-rmt-flute-sdp-03<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">For background on the SDP directorate, please see=
 the FAQ at<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">&nbsp;&nbsp;&nbsp;&nbsp; <a href=3D"http://www.ie=
tf.org/iesg/directorate/sdp.html">
http://www.ietf.org/iesg/directorate/sdp.html</a><o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">Please wait for direction from your document shep=
herd or AD before posting a new version of the draft.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">SUMMARY:<o:p></o:p></p>
<p class=3D"MsoNormal">=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">I have technical concerns with a number of the techn=
ical content of this document, and at least the &#8220;Composite Session&#8=
221; concept, as currently
<o:p></o:p></p>
<p class=3D"MsoNormal">defined in the document, would need to be discussed =
in MMUSIC. I think they need to be resolved/clarified before this document =
can be published.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Christer<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">----------------------------------------------------=
----<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">TECHNICAL<o:p></o:p></p>
<p class=3D"MsoNormal">=3D=3D=3D=3D=3D=3D=3D=3D=3D<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">General:<o:p></o:p></p>
<p class=3D"MsoNormal">------------<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">T-GEN-1:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; The draft seems to define a new generic =
SDP mechanism, &#8220;Composite Session&#8221;, for grouping media streams =
into protocol-specific sessions. I am not really sure what that means, but =
it seems to go outside the scope of
 FLUTE. I also think it overlaps with the SDP BUNDLE work, and such work sh=
all be done in MMUSIC.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &#8220;The Composite Session mechani=
sm enables the grouping of media lines<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; in to distinct sessions.&nbsp; The comple=
te Composite Session semantics<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; are protocol-specific - as determined by =
the protocol id of the<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; grouped media lines.&#8221;<o:p></o:p></p=
>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">T-GEN-2:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Maybe I&#8217;ve missed it, but it is a =
little unclear to me why the grouping framework is needed in the first plac=
e. If that is described somewhere, please tell me :)<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">In section 3.2.1, the text says:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &#8220;The Composite Session provide=
s an unambiguous way to define multiple<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; FLUTE sessions as distinct from multiple =
the media-sessions semantics<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; of RTP.&#8221;<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">But, what is the technical need for this?<o:p></o:p>=
</p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">T-GEN-3:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; The draft uses the same source address f=
or every m- line associated with a group, but it uses different addresses i=
n each m- line. Aren&#8217;t the media streams symmetric?<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">T-GEN-4:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; There is chapter describing the SDP Offe=
r/Answer aspects. How is the answer generated? What if the answerer does no=
t support (or, supports, but don&#8217;t want to use) the mechanism?<o:p></=
o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">If the mechanism is not to be used with SDP Offer/An=
swer, but e.g. for some kind of declaration, I think that needs to be descr=
ibed.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Section 1:<o:p></o:p></p>
<p class=3D"MsoNormal">-------------<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">T-1-1:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; The te=
xt says:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &#8220;Note, this document may also =
be used to describe sessions of the<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; experimental FLUTE specification [RFC3926=
].&#8221;<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">I think you need to indicate whether or not there wi=
ll be any changes in the usage of SDP in that case.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Section 3.2:<o:p></o:p></p>
<p class=3D"MsoNormal">---------------<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">T-3-2:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; I woul=
d like to have some input from the MMUSIC community whether &#8220;FLUTE/UD=
P/ESP&#8221; is appropriate to describe a IPSec encrypted FLUTE channel, or=
 whether something like &#8220;SFLUTE/UDP&#8221; should be used instead.<o:=
p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Section 3.11:<o:p></o:p></p>
<p class=3D"MsoNormal">-----------------<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">T-3_11-1:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp; As the syntax mandates at least one fmt value=
, I suggest that the document specifies which value SHALL be used.<o:p></o:=
p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">EDITORIAL<o:p></o:p></p>
<p class=3D"MsoNormal">=3D=3D=3D=3D=3D=3D=3D=3D=3D<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">General:<o:p></o:p></p>
<p class=3D"MsoNormal">-----------<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">E-GEN-1:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; The Abstract and Introduction needs to e=
xplicitly state that the specification defines a new SDP grouping framework=
 value. The Abstract also needs to mention the new SDP protocol values.<o:p=
></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Section 1:<o:p></o:p></p>
<p class=3D"MsoNormal">-------------<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">E-1-1:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; s/defi=
nes two new protocol identifiers/defines two new SDP protocol values<o:p></=
o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">E-1-2:&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; I suggest t=
o remove &#8220;The formal ABNF syntax [RFC5234] is used for the attributes=
.&#8221; It only needs to be stated in the section defining the new SDP att=
ributes.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Section 3:<o:p></o:p></p>
<p class=3D"MsoNormal">-------------<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">E-3-1:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; I thin=
k the second last paragraph, comparing FLUTE with RTP media streams, sessio=
ns etc is confusing. I think it is enough to describe that each SDP m- line=
 represents a FLUTE channel.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Section 3.1:<o:p></o:p></p>
<p class=3D"MsoNormal">---------------<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">E-3_1-1:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; I would suggest to have a new pare=
nt chapter, &#8220;SDP Extensions&#8221;, with sub-chapters defining the ne=
w SDP protocol values, attributes etc.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Section 3.11:<o:p></o:p></p>
<p class=3D"MsoNormal">-----------------<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">E-3_11-1:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp; I suggest to remove this section. It&#8217;s =
not needed, and I don&#8217;t even agree with the content.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Section 4:<o:p></o:p></p>
<p class=3D"MsoNormal">-------------<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">E-4-1:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Instea=
d of saying &#8220;a=3DFEC-declaration line&#8221;, I suggest to say &#8220=
;SDP FEC-declaration attribute&#8221;. (Also applies to the description of =
the other attributes).<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B09423BESESSMB209ericsso_--

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Ali C. Begen (abegen)" <abegen@cisco.com>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: The rtp-duplication related drafts - Initial comments on draft-ietf-mmusic-delayed-duplication
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Subject: Re: [MMUSIC] The rtp-duplication related drafts - Initial comments on draft-ietf-mmusic-delayed-duplication
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Hi Ali,

Some initial comments on draft-ietf-mmusic-delayed-duplication:

Q1:	I assume the attribute indicates the SENDING delay, or? I think it need=
s to be more explicit.

Q2:	There are no SDP Offer/Answer considerations. Does the Offer have impac=
t on the Answer? What is the expected behavior if the Answerer does not sup=
port the mechanism? Etc.

Regards,

Christer


-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 Ali C. Begen (abegen)
Sent: 4. tammikuuta 2013 0:31
To: mmusic@ietf.org
Subject: [MMUSIC] The rtp-duplication related drafts

Hi everyone,

The following two drafts have been adopted by MMUSIC a while ago and the ma=
in draft discussing the RTP duplication issues is being finalized in AVTEXT=
 (this was also adopted as a WG item a while ago). It is simply waiting for=
 the WG to make a decision on the srcname proposal.

http://datatracker.ietf.org/doc/draft-ietf-mmusic-delayed-duplication/
http://datatracker.ietf.org/doc/draft-ietf-mmusic-duplication-grouping/

These two mmusic drafts are essentially completed and they are needed to be=
 done for the avtext draft. I hope they will be WGLC'ed some time soon. In =
the mean time, I would like to get any comments, feedback that you might ha=
ve.

Thanks.
-acbegen

PS Here is the AVTEXT draft.
http://datatracker.ietf.org/doc/draft-ietf-avtext-rtp-duplication/=20
_______________________________________________
mmusic mailing list
mmusic@ietf.org
https://www.ietf.org/mailman/listinfo/mmusic

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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
Thread-Topic: The rtp-duplication related drafts - Initial comments on draft-ietf-mmusic-delayed-duplication
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Subject: Re: [MMUSIC] The rtp-duplication related drafts - Initial comments on draft-ietf-mmusic-delayed-duplication
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Hi Christer,

On Jan 8, 2013, at 7:56 AM, Christer Holmberg <christer.holmberg@ericsson.c=
om> wrote:

> Hi Ali,
>=20
> Some initial comments on draft-ietf-mmusic-delayed-duplication:
>=20
> Q1:	I assume the attribute indicates the SENDING delay, or? I think it ne=
eds to be more explicit.
>=20
Yes, the dup delay refers to the delay applied during sending. The delay be=
tween the dup packets might be different on the receiving side. Is this wha=
t you asked?

> Q2:	There are no SDP Offer/Answer considerations. Does the Offer have imp=
act on the Answer? What is the expected behavior if the Answerer does not s=
upport the mechanism? Etc.
>=20

This attribute is used with the grouping defined in the other draft. And O/=
A stuff applies to that grouping semantics, which the draft explains. Maybe=
, I should put a note regarding this in this draft?

Cheers,
-acbegen

> Regards,
>=20
> Christer
>=20
>=20
> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf =
Of Ali C. Begen (abegen)
> Sent: 4. tammikuuta 2013 0:31
> To: mmusic@ietf.org
> Subject: [MMUSIC] The rtp-duplication related drafts
>=20
> Hi everyone,
>=20
> The following two drafts have been adopted by MMUSIC a while ago and the =
main draft discussing the RTP duplication issues is being finalized in AVTE=
XT (this was also adopted as a WG item a while ago). It is simply waiting f=
or the WG to make a decision on the srcname proposal.
>=20
> http://datatracker.ietf.org/doc/draft-ietf-mmusic-delayed-duplication/
> http://datatracker.ietf.org/doc/draft-ietf-mmusic-duplication-grouping/
>=20
> These two mmusic drafts are essentially completed and they are needed to =
be done for the avtext draft. I hope they will be WGLC'ed some time soon. I=
n the mean time, I would like to get any comments, feedback that you might =
have.
>=20
> Thanks.
> -acbegen
>=20
> PS Here is the AVTEXT draft.
> http://datatracker.ietf.org/doc/draft-ietf-avtext-rtp-duplication/=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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Cc: "draft-ietf-rmt-flute-sdp@tools.ietf.org" <draft-ietf-rmt-flute-sdp@tools.ietf.org>, "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] SDP Directorate: Review of draft-ietf-rmt-flute-sdp-03
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When the request for review of this came out, I deferred because I knew 
nothing of flute. After seeing Christer's comments I was motivated to 
take a look.

I see many issues here, over and above those that Christer identified:

First, IIUC flute is intended primarily for use with multicast, though 
its spec says it can also be used with unicast. This bias towards 
multicast shows in this document. The examples are all multicast, and 
unicast is mentioned only twice.

Also, offer/answer is only mentioned in passing. There is no discussion 
of how all the constructs defined in this document should work in an O/A 
usage. I'm pretty sure they won't work.

Of course this isn't necessarily a problem if this functionality is not 
intended for use in O/A. But if that is the case then it needs to be 
clearly spelled out. If it *is* intended to be used with O/A, and 
especially with O/A negotiation of unicast, then a lot more work is 
needed to specify how this will work.

As Christer noted, the Composite Session grouping defined here seems to 
be addressing much the same problem as BUNDLE and MMT. But of course it 
has a somewhat different spin on things. This grouping introduces an 
another level of attribute inheritance that is kind of "interesting". It 
shares the problem of when and how to aggregate attribute values in the 
group, but only addresses it for a few attributes.  IMO we *really* 
don't want a bunch of grouping mechanisms, each focused on a different 
application.

This spec has a need to identify the source address. It repurposes 
a=source-address to satisfy this need. This use is quite different from 
the intended use. Because of limitations on source addresses for flute, 
it imposes limitations on the use of the a=source-address attribute. 
(See section 3.3.) I think these limitations may break using 
source-address for its intended purpose.

	Thanks,
	Paul

On 1/8/13 7:30 AM, Christer Holmberg wrote:
> Hi,
>
> I am the assigned SDP directorate reviewer for
>
> draft-ietf-rmt-flute-sdp-03
>
> For background on the SDP directorate, please see the FAQ at
>
> http://www.ietf.org/iesg/directorate/sdp.html
>
> Please wait for direction from your document shepherd or AD before
> posting a new version of the draft.
>
> SUMMARY:
>
> ==========
>
> I have technical concerns with a number of the technical content of this
> document, and at least the “Composite Session” concept, as currently
>
> defined in the document, would need to be discussed in MMUSIC. I think
> they need to be resolved/clarified before this document can be published.
>
> Regards,
>
> Christer
>
> --------------------------------------------------------
>
> TECHNICAL
>
> =========
>
> General:
>
> ------------
>
> T-GEN-1:              The draft seems to define a new generic SDP
> mechanism, “Composite Session”, for grouping media streams into
> protocol-specific sessions. I am not really sure what that means, but it
> seems to go outside the scope of FLUTE. I also think it overlaps with
> the SDP BUNDLE work, and such work shall be done in MMUSIC.
>
>                  “The Composite Session mechanism enables the grouping
> of media lines
>
>                  in to distinct sessions.  The complete Composite
> Session semantics
>
>                  are protocol-specific - as determined by the protocol
> id of the
>
>                  grouped media lines.”
>
> T-GEN-2:              Maybe I’ve missed it, but it is a little unclear
> to me why the grouping framework is needed in the first place. If that
> is described somewhere, please tell me :)
>
> In section 3.2.1, the text says:
>
>                  “The Composite Session provides an unambiguous way to
> define multiple
>
>                  FLUTE sessions as distinct from multiple the
> media-sessions semantics
>
>                  of RTP.”
>
> But, what is the technical need for this?
>
> T-GEN-3:              The draft uses the same source address for every
> m- line associated with a group, but it uses different addresses in each
> m- line. Aren’t the media streams symmetric?
>
> T-GEN-4:              There is chapter describing the SDP Offer/Answer
> aspects. How is the answer generated? What if the answerer does not
> support (or, supports, but don’t want to use) the mechanism?
>
> If the mechanism is not to be used with SDP Offer/Answer, but e.g. for
> some kind of declaration, I think that needs to be described.
>
> Section 1:
>
> -------------
>
> T-1-1:                    The text says:
>
>                  “Note, this document may also be used to describe
> sessions of the
>
>                  experimental FLUTE specification [RFC3926].”
>
> I think you need to indicate whether or not there will be any changes in
> the usage of SDP in that case.
>
> Section 3.2:
>
> ---------------
>
> T-3-2:                    I would like to have some input from the
> MMUSIC community whether “FLUTE/UDP/ESP” is appropriate to describe a
> IPSec encrypted FLUTE channel, or whether something like “SFLUTE/UDP”
> should be used instead.
>
> Section 3.11:
>
> -----------------
>
> T-3_11-1:             As the syntax mandates at least one fmt value, I
> suggest that the document specifies which value SHALL be used.
>
> EDITORIAL
>
> =========
>
> General:
>
> -----------
>
> E-GEN-1:              The Abstract and Introduction needs to explicitly
> state that the specification defines a new SDP grouping framework value.
> The Abstract also needs to mention the new SDP protocol values.
>
> Section 1:
>
> -------------
>
> E-1-1:                    s/defines two new protocol identifiers/defines
> two new SDP protocol values
>
> E-1-2:                    I suggest to remove “The formal ABNF syntax
> [RFC5234] is used for the attributes.” It only needs to be stated in the
> section defining the new SDP attributes.
>
> Section 3:
>
> -------------
>
> E-3-1:                    I think the second last paragraph, comparing
> FLUTE with RTP media streams, sessions etc is confusing. I think it is
> enough to describe that each SDP m- line represents a FLUTE channel.
>
> Section 3.1:
>
> ---------------
>
> E-3_1-1:               I would suggest to have a new parent chapter,
> “SDP Extensions”, with sub-chapters defining the new SDP protocol
> values, attributes etc.
>
> Section 3.11:
>
> -----------------
>
> E-3_11-1:             I suggest to remove this section. It’s not needed,
> and I don’t even agree with the content.
>
> Section 4:
>
> -------------
>
> E-4-1:                    Instead of saying “a=FEC-declaration line”, I
> suggest to say “SDP FEC-declaration attribute”. (Also applies to the
> description of the other attributes).
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


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Cc: mmusic <mmusic@ietf.org>, Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>, "draft-boucadair-mmusic-altc@tools.ietf.org" <draft-boucadair-mmusic-altc@tools.ietf.org>
Subject: Re: [MMUSIC] SDP Directorate: Review of draft-boucadair-mmusic-altc-07
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On 1/8/13 4:59 AM, mohamed.boucadair@orange.com wrote:
> Dear Flemming,
>
> Thank you for the review.
>
> Please see inline.
>
> Cheers,
> Med
>
>> -----Message d'origine-----
>> De : mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org]
>> De la part de Flemming Andreasen
>> Envoyé : mardi 8 janvier 2013 04:24
>> À : draft-boucadair-mmusic-altc@tools.ietf.org
>> Cc : mmusic; Gonzalo Camarillo
>> Objet : [MMUSIC] SDP Directorate: Review of
>> draft-boucadair-mmusic-altc-07
>>
>> Hi
>>
>> I am the assigned SDP directorate reviewer for
>> draft-boucadair-mmusic-altc-07
>>
>> For background on the SDP directorate, please see the FAQ at
>>
>>      http://www.ietf.org/iesg/directorate/sdp.html
>>
>> Please wait for direction from your document shepherd or AD before
>> posting a new version of the draft.
>>
>> Summary:
>> ---------
>> - The document does not adequately cover how it relates to the
>> existing
>> Standards Track mechanism for solving the similar problem in a
>> different
>> and more general way (namely ICE [RFC 5245])
>> - The document has technical issues
>> - The document has a few minor editorial issues
>>
>>
>> New SDP Information Elements:
>> ---------------------------
>> The draft defines a new "a=altc" attribute
>>
>>
>> Technical Comments:
>> ------------------
>> 1) The Abstract positions the document as an alternative to ANAT, but
>> without certain problems experienced by ANAT. The abstract fails to
>> mention that a general Standards Track solution in the form of
>> ICE (RFC
>> 5245) has been defined whereas the solution defined in this
>> document has
>> a narrower scope and various limitations.
> Med: I updated the abstract as follows:
>
> OLD:
>
>     This document proposes a mechanism which allows to carry multiple IP
>     addresses, of different address families (e.g., IPv4, IPv6), in the
>     same SDP offer.  The proposed attribute solves the backward
>     compatibility problem which plagued ANAT, due to its syntax.
>
> NEW:
>
>     This document proposes a mechanism which allows to carry multiple IP
>     addresses, of different address families (e.g., IPv4, IPv6), in the
>     same SDP offer.  The proposed attribute solves the backward
>     compatibility problem which plagued ANAT, due to its syntax.
>
>     The proposed solution is applicable to scenarios where connectivity
>     checks are not required.
I'd suggest adding "If connectivity checks are required, ICE (RFC 5245) 
provides such a solution".

>> 2) Section 1.1, third paragraph. The document states that ICE is
>> processing intensive and has seen limited deployment. While this is
>> perhaps a matter of debate, it would be helpful if the document would
>> point out the limitations of using instead of ICE. Areas that come to
>> mind here are
>> - lack of NAT traversal (without network/SBC support),
>> - no connectivity verification (verifying functioning path as well as
>> authenticating/correlating initial incoming media packets with
>> signaling),
>> - support for non-contiguous RTP/RTCP port pairs in alternatives
>> There are probably others and it would be helpful to point
>> those out to
>> potential implementers (the simplicity comes with trade-offs).
> Med: The scope of ALTC is very clear: scenarios where there is no need for connectivity checks. This is already mentioned in Section 2 and I tried to make it clear in the new proposed abstract text.

I don't think it will be obvious to people looking at the document what 
the trade-offs with the Standards Track solution (ICE) are, and hence I 
think it's advisable to call them out explicitly so people can make an 
informed decision as to which solution they want to use.

>> 3) Section 1.1, third paragraph: The document claims that ICE is not
>> usable in closed networks. This is incorrect (it is however
>> correct that
>> there are scenarios where ICE does not work, however those scenarios
>> should then be explained instead).
> Med: closed networks are provided only as an example. Anyway, I updated the text as follows:
>
> OLD:
>
> (e.g., closed networks)
>
> NEW:
>
> (e.g., closed networks making use of SBCs)
The statement is still not entirely correct (the SBC can for example 
support ICE or it could be doing signaling processing only).

My understanding of where ICE does not work is in a scenario where an 
intermediay (proxy, SBC, or whatever) inserts a media relay of some sort 
in the m= line without updating/supporting ICE as part of that. This 
will lead to ice-mismatch (detected by ICE by design) and hence 
abandoning the ICE procedures.
a) Is this the scenario you were thinking of, and/or are there others ?
b) How does altc work and/or work better in that case (compared to ICE) ?
c) Whether using ICE or altc in the above scenario, you still have an 
issue with NAT traversal that may require some form of latching to work 
(or do you see altc somehow avoiding this ?)

The above should be clarified (or the overly broad statement should be 
removed instead).

>> 4) Section 1.1, fourth paragraph: The document claims to be compliant
>> with RFC 6157 because RFC 6157 includes the sentence
>> <quote>
>> The use of ICE can be avoided for signaling messages that stay within
>> such managed networks
>> </quote>
>> The quote is taken out of context however as it applies to a
>> scenario in
>> RFC 6157 where the entire network is IPv6 based, whereas the mechanism
>> defined here is for scenarios where it is not.
> Med: The text from RFC6157 is:
>
>        Implementations are encouraged to use ICE; however, the normative
>        strength of the text above is left at a SHOULD since in some
>        managed networks (such as a closed enterprise network) it is
>        possible for the administrator to have control over the IP version
>        utilized in all nodes and thus deploy an IPv6-only network, for
>        example.  The use of ICE can be avoided for signaling messages
>        that stay within such managed networks.
>
> IPv6-only is mentioned as an example. The key word IMHO in the above text is "managed networks".
I think the key in the above is

"it is possible for the administrator to have control over the IP 
version utilized in all nodes"

and thus you know it's safe to use only one particular IP version and 
hence can avoid using ICE to actually pick a version that works. In any 
case, I don't think this reference to RFC 6157 compliance is critical 
for this document and hence it may be easier to simply remove it.

>> 5) Section 3.2, first paragraph
>> <quote>
>> A compliant SDP parser will ignore any session description
>> that contains
>> attribute lines it does not support.
>> <quote>
>> No - only the unsupported attribute lines will be ignored
>> (which is also
>> what the rest of the draft implies)
> Med: OK. I updated the text.
Ok
>> 6) Section 4.1, attribute definition
>> The optional trailing integer part is not defined.
> Med: Thanks for catching this. I updated section 4.1.
Ok.
>> 7) Section 4.1, Bullet list, "port"
>> This does not seem to work with the "a=rtcp" attribute [RFC 4566], and
>> hence there may be issues with non-contiguous RTP/RTCP port
>> pair support
>> in alternatives.
> Med: We updated the text as follows:
>
> OLD:
>
>     Note that for RTCP, if applicable for the given media types, each
>     side would act as if the chosen "altc" attribute's port number was in
>     the 'm=' media line.  Typically, this would mean RTCP is sent to the
>     odd +1 of the port number, unless some other attribute determines
>     otherwise.
>
> NEW:
>
>     Note that for RTCP, if applicable for the given media types, each
>     side would act as if the chosen "altc" attribute's port number was in
>     the 'm=' media line.  Typically, this would mean RTCP is sent to the
>     odd +1 of the port number, unless some other attribute determines
>     otherwise.
>
>     For example the RTP/RTCP multiplexing mechanism defined in [RFC5761]
>     can still be used with ALTC, such that if both sides support
>     multiplexing they will indicate so using the 'a=rtcp-mux' attribute
>     as defined in [RFC5761]; but the IP connection address and port they
>     use may be chosen using the ALTC mechanism.
>
>     If the SDP offerer wishes to use the RTCP attribute defined in
>     [RFC3605], a complication can arise since it may not be clear which
>     address choice the 'a=rtcp' attribute applies to, relative the
>     choices offered by ALTC.  Technically RFC 3605 allows indicating the
>     address for RTCP explicitly in the 'a=rtcp' attribute itself, but
>     this is optional and rarely used.  Since RFC 3605 did not prevent
>     encoding multiple 'a=rtcp' attributes per media description, this
>     document recommends encoding the first 'a=rtcp' attribute to be for
>     the address choice encoded in the "m=" line, and a second 'a=rtcp'
>     attribute for the alternate address provided by the 'a=altc'
>     attribute.  In other words, if the 'a=rtcp' attribute explicitly
>     encodes an address in its attribute, then that applies for ALTC as
>     per [RFC3605]; if it does not, then ALTC assumes the first one is for
>     the address in the "m=" line, and the second is for the alternate.
I don't think this is a good idea (multiple instances with different 
values, and also ordering specific). While RFC 3605 may not explicitly 
disallow the above, it's also not part of it's overall defined 
semantics, and it's unlikely to produce interoperable implementations if 
used this way IMO.

Rather than having multiple instances with some altc-specific 
interpretation associated with those, you could extend the altc 
attribute itself to include the information. Alternatively, you could 
limit the solution to not support non-contiguous RTP/RTCP ports for 
anything but the default value provided in the m= line (then add a rule 
that a=rtcp is to be ignored if another address family is used).

>> 8) Section 4.1, 4th last paragraph
>> The mechanism relies on implicit ordering of attribute lines. If an
>> intermediary changes the order, it would have unintended side-effects.
>> It would be much better to have an explicit ordering mechanism. In the
>> absence of that, the potential reordering issue should be called out.
> Med: Is this harmful? If an intermediary entity change the order, this does not impact the session establishment. Note the document already specify how a remote UA can detect the presence of an intermediary entity changing the "c" line is present in the path.
It doesn't produce the intended operation. I thought part of the draft 
motivation was to enable a preferential selection of the address family 
in a backwards compatible manner, especially for IPv4-only peers. If the 
attributes get reordered, the preference no longer works.

Since the document is Informational, I guess it's up to you whether this 
is problematic, but if left as-is, I think you should at least point out 
this potential issue.

>>
>> Editorial Comments:
>> -----------------
>> 1) Section 1.1, second paragraph. Add reference after "DS-Lite"
>>
>> 2) sdp-cap-neg should be SDPCapNeg (multiple occurrences)
>>
>> 3) Section 1.3, first paragraph: Add reference to RFC 4091.
>>
>> 4) Section 1.3, second paragraph: Add text and reference to
>> RFC 6157 for
>> a solution to "SIP layer address family interworking"
>>
>> 5) Section 4.2.3
>> Add RFC reference after "ICE"
>>
>> 6) Section 4.2.4
>> Add RFC reference after "SDPCapNeg"
> Med: I implemented all the proposed changed. Thanks.
Ok - thanks

-- Flemming


>>
>> Thanks
>>
>> -- Flemming
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>> .


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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-16.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Multiparty Multimedia Session Control Wor=
king Group of the IETF.

	Title           : Session Description Protocol (SDP) Extension For Setting=
 Up Audio and Video Media Streams Over Circuit-Switched Bearers In The Publ=
ic Switched Telephone Network (PSTN)
	Author(s)       : Miguel A. Garcia-Martin
                          Simo Veikkolainen
	Filename        : draft-ietf-mmusic-sdp-cs-16.txt
	Pages           : 37
	Date            : 2013-01-08

Abstract:
   This memo describes use cases, requirements, and protocol extensions
   for using the Session Description Protocol (SDP) Offer/Answer model
   for establishing audio and video media streams over circuit-switched
   bearers in the Public Switched Telephone Network (PSTN).


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-cs

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-16

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sdp-cs-16


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


From Simo.Veikkolainen@nokia.com  Wed Jan  9 00:06:01 2013
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From: <Simo.Veikkolainen@nokia.com>
To: <mmusic@ietf.org>
Thread-Topic: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-16.txt
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Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-16.txt
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Submitting version -16 which corrects some typos that were supposed to be i=
n already in version -15.

Regards,
Simo

-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 ext internet-drafts@ietf.org
Sent: 9. tammikuuta 2013 9:58
To: i-d-announce@ietf.org
Cc: mmusic@ietf.org
Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-16.txt


A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Multiparty Multimedia Session Control Wor=
king Group of the IETF.

	Title           : Session Description Protocol (SDP) Extension For Setting=
 Up Audio and Video Media Streams Over Circuit-Switched Bearers In The Publ=
ic Switched Telephone Network (PSTN)
	Author(s)       : Miguel A. Garcia-Martin
                          Simo Veikkolainen
	Filename        : draft-ietf-mmusic-sdp-cs-16.txt
	Pages           : 37
	Date            : 2013-01-08

Abstract:
   This memo describes use cases, requirements, and protocol extensions
   for using the Session Description Protocol (SDP) Offer/Answer model
   for establishing audio and video media streams over circuit-switched
   bearers in the Public Switched Telephone Network (PSTN).


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-cs

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-16

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sdp-cs-16


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

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

From harald@alvestrand.no  Wed Jan  9 11:25:07 2013
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Subject: [MMUSIC] Query: payload type collision with offer/answer
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Hello,
I have encountered an issue, and I'm not sure it's an issue or not. SDP 
wisdom is sought.

Suppose two entities exchange the following offer/answer:

offer from X:

m=video 97
a=rtpmap:97 foo

answer from Y:

m=video 98
a=rtpmap:98 foo

It's clear from RFC 3264 that now X must send codec foo with payload 
type 98, and Y must send codec foo with payload type 97. That's not the 
question.

But suppose these are the offer and answer:

offer from X:

m=video 97 98
a=rtpmap:97 foo
a=rtpmap:98 bar

answer from Y:

m=video 97 98
a=rtpmap:97 bar
a=rtpmap:98 foo

That is, the payload types collide.

It would be possible to write code so that X sends foo with 98 and bar 
with 97, while Y sends foo with 97 and bar with 98. But is it conformant 
with the specs to do so?

And if this has a clear answer - which paragraph of which RFC makes this 
clear?
(Yes, this discussion is triggered from a real world problem.)

                    Harald


From eckelcu@cisco.com  Wed Jan  9 13:01:47 2013
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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: Harald Alvestrand <harald@alvestrand.no>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Query: payload type collision with offer/answer
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Please see inline.

> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Harald Alvestrand
> Sent: Wednesday, January 09, 2013 11:25 AM
> To: mmusic@ietf.org
> Subject: [MMUSIC] Query: payload type collision with offer/answer
>=20
> Hello,
> I have encountered an issue, and I'm not sure it's an issue or not. SDP
> wisdom is sought.
>=20
> Suppose two entities exchange the following offer/answer:
>=20
> offer from X:
>=20
> m=3Dvideo 97
> a=3Drtpmap:97 foo
>=20
> answer from Y:
>=20
> m=3Dvideo 98
> a=3Drtpmap:98 foo
>=20
> It's clear from RFC 3264 that now X must send codec foo with payload
> type 98, and Y must send codec foo with payload type 97. That's not the
> question.
>=20
> But suppose these are the offer and answer:
>=20
> offer from X:
>=20
> m=3Dvideo 97 98
> a=3Drtpmap:97 foo
> a=3Drtpmap:98 bar
>=20
> answer from Y:
>=20
> m=3Dvideo 97 98
> a=3Drtpmap:97 bar
> a=3Drtpmap:98 foo
>=20
> That is, the payload types collide.
>=20
> It would be possible to write code so that X sends foo with 98 and bar
> with 97, while Y sends foo with 97 and bar with 98. But is it conformant
> with the specs to do so?
>=20
> And if this has a clear answer - which paragraph of which RFC makes this
> clear?
> (Yes, this discussion is triggered from a real world problem.)

It is recommended to avoid such mismatches when possible, but doing so may =
not always be possible. The examples you mention are conformant. The follow=
ing text from section 5.1 addresses this:

   For sendrecv RTP
   streams, the payload type numbers indicate the value of the payload
   type field the offerer expects to receive, and would prefer to send.
   However, for sendonly and sendrecv streams, the answer might indicate
   different payload type numbers for the same codecs, in which case,
   the offerer MUST send with the payload type numbers from the answer.

      Different payload type numbers may be needed in each direction
      because of interoperability concerns with H.323.

That said there are additional requirements placed on payload type usage by=
 some codecs, such as SVC.

Cheers,
Charles

=20
>                     Harald
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

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On 01/09/2013 10:01 PM, Charles Eckel (eckelcu) wrote:
> Please see inline.
>
> > -----Original Message-----
> > From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> > Behalf Of Harald Alvestrand
> > Sent: Wednesday, January 09, 2013 11:25 AM
> > To: mmusic@ietf.org
> > Subject: [MMUSIC] Query: payload type collision with offer/answer
> >
> > Hello,
> > I have encountered an issue, and I'm not sure it's an issue or not. SDP
> > wisdom is sought.
> >
> > Suppose two entities exchange the following offer/answer:
> >
> > offer from X:
> >
> > m=video 97
> > a=rtpmap:97 foo
> >
> > answer from Y:
> >
> > m=video 98
> > a=rtpmap:98 foo
> >
> > It's clear from RFC 3264 that now X must send codec foo with payload
> > type 98, and Y must send codec foo with payload type 97. That's not the
> > question.
> >
> > But suppose these are the offer and answer:
> >
> > offer from X:
> >
> > m=video 97 98
> > a=rtpmap:97 foo
> > a=rtpmap:98 bar
> >
> > answer from Y:
> >
> > m=video 97 98
> > a=rtpmap:97 bar
> > a=rtpmap:98 foo
> >
> > That is, the payload types collide.
> >
> > It would be possible to write code so that X sends foo with 98 and bar
> > with 97, while Y sends foo with 97 and bar with 98. But is it 
> conformant
> > with the specs to do so?
> >
> > And if this has a clear answer - which paragraph of which RFC makes 
> this
> > clear?
> > (Yes, this discussion is triggered from a real world problem.)
>
> It is recommended to avoid such mismatches when possible, but doing so 
> may not always be possible. The examples you mention are conformant. 
> The following text from section 5.1 addresses this:
>
> For sendrecv RTP
> streams, the payload type numbers indicate the value of the payload
> type field the offerer expects to receive, and would prefer to send.
> However, for sendonly and sendrecv streams, the answer might indicate
> different payload type numbers for the same codecs, in which case,
> the offerer MUST send with the payload type numbers from the answer.
>
> Different payload type numbers may be needed in each direction
> because of interoperability concerns with H.323.
I was reading that text, and it clearly says which ones to use if the 
payload types are different, but I don't think they clearly address the 
situation with using the same payload type for two different codecs in 
different directions.

In a multicast or transport relay scenario, this would clearly be 
untenable, since third parties would not be able to send streams that 
would be interpreted correctly by the two conflicting endpoints, but in 
a two-party conversation, it's possible to make this work.
I just don't know if it's worth the effort to make it so.



From emil@sip-communicator.org  Wed Jan  9 15:46:08 2013
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Message-ID: <50EE0139.8000909@jitsi.org>
Date: Thu, 10 Jan 2013 00:46:01 +0100
From: Emil Ivov <emcho@jitsi.org>
Organization: Jitsi
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Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] Query: payload type collision with offer/answer
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On 09.01.13, 23:52, Harald Alvestrand wrote:
> On 01/09/2013 10:01 PM, Charles Eckel (eckelcu) wrote:
>> Please see inline.
>>
>>> -----Original Message-----
>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
>>> Behalf Of Harald Alvestrand
>>> Sent: Wednesday, January 09, 2013 11:25 AM
>>> To: mmusic@ietf.org
>>> Subject: [MMUSIC] Query: payload type collision with offer/answer
>>>
>>> Hello,
>>> I have encountered an issue, and I'm not sure it's an issue or not. SDP
>>> wisdom is sought.
>>>
>>> Suppose two entities exchange the following offer/answer:
>>>
>>> offer from X:
>>>
>>> m=video 97
>>> a=rtpmap:97 foo
>>>
>>> answer from Y:
>>>
>>> m=video 98
>>> a=rtpmap:98 foo
>>>
>>> It's clear from RFC 3264 that now X must send codec foo with payload
>>> type 98, and Y must send codec foo with payload type 97. That's not the
>>> question.
>>>
>>> But suppose these are the offer and answer:
>>>
>>> offer from X:
>>>
>>> m=video 97 98
>>> a=rtpmap:97 foo
>>> a=rtpmap:98 bar
>>>
>>> answer from Y:
>>>
>>> m=video 97 98
>>> a=rtpmap:97 bar
>>> a=rtpmap:98 foo
>>>
>>> That is, the payload types collide.
>>>
>>> It would be possible to write code so that X sends foo with 98 and bar
>>> with 97, while Y sends foo with 97 and bar with 98. But is it 
>> conformant
>>> with the specs to do so?
>>>
>>> And if this has a clear answer - which paragraph of which RFC makes 
>> this
>>> clear?
>>> (Yes, this discussion is triggered from a real world problem.)
>>
>> It is recommended to avoid such mismatches when possible, but doing so 
>> may not always be possible. The examples you mention are conformant. 
>> The following text from section 5.1 addresses this:
>>
>> For sendrecv RTP
>> streams, the payload type numbers indicate the value of the payload
>> type field the offerer expects to receive, and would prefer to send.
>> However, for sendonly and sendrecv streams, the answer might indicate
>> different payload type numbers for the same codecs, in which case,
>> the offerer MUST send with the payload type numbers from the answer.
>>
>> Different payload type numbers may be needed in each direction
>> because of interoperability concerns with H.323.
> I was reading that text, and it clearly says which ones to use if the 
> payload types are different, but I don't think they clearly address the 
> situation with using the same payload type for two different codecs in 
> different directions.
> 
> In a multicast or transport relay scenario, this would clearly be 
> untenable, since third parties would not be able to send streams that 
> would be interpreted correctly by the two conflicting endpoints, but in 
> a two-party conversation, it's possible to make this work.
> I just don't know if it's worth the effort to make it so.

Well there's also section 8.3.2 in 3264. The section is about updating
offers but it seems like the text explicitly prohibits remapping payload
types in any scenarios:

   However, in the
   case of RTP, the mapping from a particular dynamic payload type
   number to a particular codec within that media stream MUST NOT change
   for the duration of a session.  For example, if A generates an offer
   with G.711 assigned to dynamic payload type number 46, payload type
   number 46 MUST refer to G.711 from that point forward in any offers
   or answers for that media stream within the session.  However, it is
   acceptable for multiple payload type numbers to be mapped to the same
   codec, so that an updated offer could also use payload type number 72
   for G.711.

Cheers,
Emil


-- 
https://jitsi.org

From pkyzivat@alum.mit.edu  Thu Jan 10 10:50:35 2013
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Subject: Re: [MMUSIC] Query: payload type collision with offer/answer
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To the best of my memory the issue Harald has raised is a new one. I 
don't think it has never been discussed on any of the lists I have been 
following since 2002.

More inline.

On 1/9/13 6:46 PM, Emil Ivov wrote:
>
>
> On 09.01.13, 23:52, Harald Alvestrand wrote:
>> On 01/09/2013 10:01 PM, Charles Eckel (eckelcu) wrote:
>>> Please see inline.
>>>
>>>> -----Original Message-----
>>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
>>>> Behalf Of Harald Alvestrand
>>>> Sent: Wednesday, January 09, 2013 11:25 AM
>>>> To: mmusic@ietf.org
>>>> Subject: [MMUSIC] Query: payload type collision with offer/answer
>>>>
>>>> Hello,
>>>> I have encountered an issue, and I'm not sure it's an issue or not. SDP
>>>> wisdom is sought.
>>>>
>>>> Suppose two entities exchange the following offer/answer:
>>>>
>>>> offer from X:
>>>>
>>>> m=video 97
>>>> a=rtpmap:97 foo
>>>>
>>>> answer from Y:
>>>>
>>>> m=video 98
>>>> a=rtpmap:98 foo
>>>>
>>>> It's clear from RFC 3264 that now X must send codec foo with payload
>>>> type 98, and Y must send codec foo with payload type 97. That's not the
>>>> question.
>>>>
>>>> But suppose these are the offer and answer:
>>>>
>>>> offer from X:
>>>>
>>>> m=video 97 98
>>>> a=rtpmap:97 foo
>>>> a=rtpmap:98 bar
>>>>
>>>> answer from Y:
>>>>
>>>> m=video 97 98
>>>> a=rtpmap:97 bar
>>>> a=rtpmap:98 foo
>>>>
>>>> That is, the payload types collide.
>>>>
>>>> It would be possible to write code so that X sends foo with 98 and bar
>>>> with 97, while Y sends foo with 97 and bar with 98. But is it
>>> conformant
>>>> with the specs to do so?
>>>>
>>>> And if this has a clear answer - which paragraph of which RFC makes
>>> this
>>>> clear?
>>>> (Yes, this discussion is triggered from a real world problem.)
>>>
>>> It is recommended to avoid such mismatches when possible, but doing so
>>> may not always be possible. The examples you mention are conformant.
>>> The following text from section 5.1 addresses this:
>>>
>>> For sendrecv RTP
>>> streams, the payload type numbers indicate the value of the payload
>>> type field the offerer expects to receive, and would prefer to send.
>>> However, for sendonly and sendrecv streams, the answer might indicate
>>> different payload type numbers for the same codecs, in which case,
>>> the offerer MUST send with the payload type numbers from the answer.
>>>
>>> Different payload type numbers may be needed in each direction
>>> because of interoperability concerns with H.323.
>> I was reading that text, and it clearly says which ones to use if the
>> payload types are different, but I don't think they clearly address the
>> situation with using the same payload type for two different codecs in
>> different directions.
>>
>> In a multicast or transport relay scenario, this would clearly be
>> untenable, since third parties would not be able to send streams that
>> would be interpreted correctly by the two conflicting endpoints, but in
>> a two-party conversation, it's possible to make this work.
>> I just don't know if it's worth the effort to make it so.

You are right that these arguments work differently for unicast and 
multicast. I don't recall (I think it was before my time) how much 
thought was given to multicast in 3264. (It doesn't seem to be very 
much. And I think the O/A discussions in new additions to SDP rarely 
consider it at all.)

> Well there's also section 8.3.2 in 3264. The section is about updating
> offers but it seems like the text explicitly prohibits remapping payload
> types in any scenarios:
>
>     However, in the
>     case of RTP, the mapping from a particular dynamic payload type
>     number to a particular codec within that media stream MUST NOT change
>     for the duration of a session.  For example, if A generates an offer
>     with G.711 assigned to dynamic payload type number 46, payload type
>     number 46 MUST refer to G.711 from that point forward in any offers
>     or answers for that media stream within the session.  However, it is
>     acceptable for multiple payload type numbers to be mapped to the same
>     codec, so that an updated offer could also use payload type number 72
>     for G.711.

The language above is underspecified.

It is my impression that the intent of this language was to ensure there 
is no ambiguity in the meaning of a payload type. Since the exchange of 
the SDP is not synchronized with the flow of RTP, if you were to send a 
new SDP that changed the mapping for a payload type, then the result 
would be almost certainly to interpret some media packets according to 
the wrong mapping.

But this argument is only about packets flowing in one direction. It 
isn't a valid argument for requiring the payload types to be consistent 
in the two directions. And its not clear to me that it even intended 
that this language apply to both directions. It could be clarified as 
follows:

                               ...  For example, if A generates an offer
    with G.711 assigned to dynamic payload type number 46, payload type
    number 46 MUST refer to G.711 from that point forward in any offers
    or answers ***by A*** for that media stream within the session.  ...

Interpreting it to apply to both directions brings it in conflict with 
the language elsewhere that permits it to be different in the two 
directions.

The use of different mappings in each direction has been discussed over 
the years, though I couldn't point you to where. IIRC one of the reasons 
had to do with gatewaying to other protocols. But I can't offer any details.

Bottom line: IMO, in the weird case Harald described X ought to follow 
Postel's Law and accept the answer from Y. But for the same reason I 
would strongly discourage implementations from acting as Y did.

	Thanks,
	Paul


From dwing@cisco.com  Thu Jan 10 14:06:42 2013
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From: "Dan Wing" <dwing@cisco.com>
To: "'Paul Kyzivat'" <pkyzivat@alum.mit.edu>
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Subject: Re: [MMUSIC] RESTART: SCTP question: Where does it multiplex?
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> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
> Sent: Thursday, January 03, 2013 11:23 AM
> To: Dan Wing
> Cc: mmusic@ietf.org
> Subject: Re: [MMUSIC] RESTART: SCTP question: Where does it multiplex?
> 
> I'm totally lost!
> I looked back, and count 43 messages in this thread.
> At this point I no longer understand the question(s), or what if
> anything has been answered.
> 
> Can we start over?
> Sort out what the questions are, and then discuss whether we have
> answers to them?
> (Preferably in separate, well identified, threads.)
> 
> I *think* Harald originally asserted that DTLS/SCTP *can* already be
> multiplexed with SRTP audio and video streams.

Yes, it can.

An UDP packet is either a STUN packet (for ICE connectivity checks),
DTLS, or SRTP.  This is depicted in Section 5.1.2 of RFC5764:

                   +----------------+
                   | 127 < B < 192 -+--> forward to RTP
                   |                |
       packet -->  |  19 < B < 64  -+--> forward to DTLS
                   |                |
                   |       B < 2   -+--> forward to STUN
                   +----------------+

A DTLS packet containing application_data is handed to the SCTP stack.

As I have mentioned previously, I am concerned with sending both realtime
interactive audio/video and non-realtime data over the same UDP 5-tuple,
because it becomes difficult (or impossible) to apply QoS to differentiate
those very different flows.  If everyone always had adequate bandwidth, that
concern would vanish.  But the industry has never been able to stay ahead of
the bandwidth curve.


> And so he was asking
> *how* this could be specified in SDP. (E.g. Does "bundle" support it?)

We need a way for SDP to signal:
  1. the UDP port for SCTP.  This feels like an SDP media ("m") line to me.
  2. SCTP native (native SCTP over IPv6 should be tenable, whereas native
SCTP over IPv4 is pretty much untenable), 
  3. the data carried over each SCTP port (e.g., port NNN for XMPP, port
NNNN for file sharing, port NNNNN for screen sharing, and so on).  This
feels like an SDP media ("m") line to me.

I don't think we can use media lines for both (1) and (3), so perhaps
bundling is the answer.  I don't know.


> Along the way, a question came up of how many DTLS/SCTP connections
> could be multiplexed with each other and with SRTP media. I think Harald
> assumed only one (and assumed there is only a need for one.) But reasons
> for more have been discussed. I suggested that it appears that multiple
> should be supportable, each with a different SCTP port.
> 
> I think Dan raised some additional issues related to multipath support.

SCTP can support multipath, so we need to decide if we allow SCTP to do its
own multipath, or if we require ICE to first validate a (new) path before
SCTP can try to do its multipath.  The same question and issues exist for
multipath RTP (draft-singh-avtcore-mprtp) and how multipath RTP should 
split its functions between ICE, ICE extensions, and SDP.  I know that
multipath RTP is not a working group document, but it might be a good
basis for analyzing multipath SCTP and reaching a conclusion on multipath
SCTP that makes sense for both multipath SCTP and multipath RTP (and
presumably multipath TCP, RFC6182 and its related documents).

-d


> 	Thanks,
> 	Paul
> 
> On 1/2/13 10:32 PM, Dan Wing wrote:
> >> -----Original Message-----
> >> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> >> Behalf Of Paul Kyzivat
> >> Sent: Tuesday, December 11, 2012 11:17 AM
> >> To: Dan Wing
> >> Cc: mmusic@ietf.org
> >> Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
> >>
> >> Hi Dan,
> >>
> >> Comments inline
> >>
> >> On 12/11/12 1:53 PM, Dan Wing wrote:
> >>>> -----Original Message-----
> >>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> >>>> Behalf Of Paul Kyzivat
> >>>> Sent: Tuesday, December 11, 2012 7:43 AM
> >>>> To: mmusic@ietf.org
> >>>> Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
> >>>>
> >>>> More inline.
> >>>>
> >>>> On 12/11/12 4:08 AM, Salvatore Loreto wrote:
> >>>>> Hi Paul,
> >>>>>
> >>>>> see in line!
> >>>>>
> >>>>> On 12/3/12 10:00 PM, Paul Kyzivat wrote:
> >>>>>> Commenting on a different point
> >>>>>>
> >>>>>> On 12/1/12 9:53 AM, Harald Alvestrand wrote:
> >>>>>>
> >>>>>>> The interesting difference is that the multiplexing between
> >>>>>>> DTLS/SCTP traffic and BUNDLE multiplexing is that DTLS/SCTP
> >>>>>>> traffic is not carried in SSRCs, which means:
> >>>>>>>
> >>>>>>> - There can be only one DTLS/SCTP stream in a bundle (which may
> >>>>>>> have multiple associations, as you state below); you can't have
> >>>>>>> multiple lines with proto DTLS/SCTP in a bundle.
> >>>>>>
> >>>>>> I am not an SCTP expert. But IIUC, SCTP was designed to run
> >>>>>> directly over IP. It has its own notion of port used to demux
> >>>>>> multiple SCTP associations over the same IP address.
> >>>>>>
> >>>>>> I presume that that same mechanism is still there when SCTP is
> >>>>>> run over DTLS over UDP.
> >>>>>>
> >>>>>> So, the traffic coming over DTLS must first be demuxed into RTP
> >>>>>> traffic and SCTP traffic.
> >>>>> based on the current stack the SCTP traffic is the only traffic
> >>>>> that runs directly over the DTLS stack.
> >>>>
> >>>> Yes, that is what I thought. But Harald has been asking about
> >>>> multiplexing this with RTP traffic. (Actually I think it would be
> >>>> DTLS/SRTP traffic that it would be multiplexed with.)
> >>>>
> >>>>> What I am trying to do is to include the Randell Jesup (I am
> >>>>> including him in CC as I am not sure he is subscribed to this
> >>>>> mailing list) suggestion to give the possibility to have multiple
> >>>>> SCTP
> >>>>> *associations* running on top of the same DTLS session and of
> >>>>> course providing a way to signal it in SDP.
> >>>>
> >>>> IIUC, SCTP (having been designed as a transport layer protocol)
> >>>> defines its own notion of port, and has fields in its protocol to
> >>>> carry the local and remote port number. Presumably those fields are
> >>>> still there when run over UDP or DTLS. So it should be possible to
> >>>> support multiple SCTP associations over the same DTLS connection,
> >>>> each distinguished by its own port pair.
> >>>
> >>> Yes, SCTP has its own notion of ports. How this works when SCTP is
> >>> carried over UDP is not quite clear, especially because SCTP assumes
> >>> a NAPT will not rewrite the SCTP port number (SCTP endeavors to make
> >>> such port rewriting difficult). But of course a NAPT (and the MAP
> >>> techniques) rewrite UDP port numbers. I believe SCTP would only be
> >>> able to successfully convey its UDP port numbers for a device that
> >>> is not behind a NAT (that is, a server sitting directly on the
> >>> Internet, rather than a remote peer that is behind a NAT). Creative
> >>> use of PCP, NAT-PMP, or UPnP IGD would improve that situation in the
> future.
> >>
> >> I find what you say above confusing.
> >>
> >> SCTP running naked over IP would still have its own ports.
> >> If that ran over a path with a NAPT, then I could see why the NAPT
> >> might want to do the same as it does with UDP and TCP, and so replace
> >> addr/port pairs to minimize addr usage. (That's if the NAPT supported
> >> SCTP at all - I gather most don't.)
> >
> > As I mentioned earlier, SCTP does not allow its source port number to
> > be changed.
> >
> >> But with SCTP over DTLS over UDP, there is a UDP port, and then
> >> presumably a separate SCTP port carried in the SCTP protocol within
> >> the UDP message.
> >
> > Yep, I assume that is how it works.  Effectively, UDP is just used to
> > tunnel the SCTP.
> >
> >> An NAPT on the path would presumably be messing with the UDP port.
> >> Given use of DTLS, the NAPT won't even be able to tell that SCTP is
> >> being used, much less mess with the SCTP port.
> >>
> >>>> That of course depends on having a signaling mechanism to set it
> up.
> >>>
> >>> After the initial SCTP association is created, SCTP could be allowed
> >>> to do its own thing (that is, chose to find and use other ports).
> >>
> >> Here you are talking about finding and using other UDP addr/port
> pairs?
> >
> > No, I was referring to the SCTP ports.
> >
> > -d
> >
> >
> >> That is explicitly excluded in the DTLS/SCTP mapping.
> >>
> >> Thanks,
> >> Paul
> >>
> >>> SCTP's behavior in this regard is similar to MPRTP (multipath RTP),
> >>> but of course they are not identical.
> >>>
> >>>>> to be clear: at moment WebRTC allows only one SCTP association per
> >>>>> PC, so this is something that would be nice to define just to be
> >>>>> ready for the future.
> >>>>
> >>>> AFAIK WebRTC is just one possible user of this mechanism. The SDP
> >>>> mechanism shouldn't be limited by the constraints of WebRTC. It
> >>>> would be very difficult to define the SDP so that it was impossible
> >>>> to set up multiple SCTP associations over different 5-tuples.
> >>>>
> >>>>>> Then the RTP traffic can be demuxed based on SSRC, and SCTP
> >>>>>> traffic can be demuxed based on SCTP port. And once the traffic
> >>>>>> for a single SCTP port is identified, it can be demuxed based on
> >>>>>> stream
> >> number.
> >>>>>>
> >>>>>> Representing this in SDP is a challenge. Some variant of the
> >>>>>> bundle proposal might allow bundling together several RTP m-lines
> >>>>>> and some DTLS/SCTP m-lines. This would require a mechanism for
> >>>>>> specifying the SCTP port number - already an open issue (#3) in
> >>>>>> draft-ietf-mmusic-sctp-sdp-02.
> >>>>> I agree that it is a challange how then to bundle everything
> >>>>> together
> >>>>
> >>>> But it is a challenge that needs to be tackled if we are to realize
> >>>> Harald's dream.
> >>>
> >>> -d
> >>>
> >>>
> >>>
> >>
> >> _______________________________________________
> >> mmusic mailing list
> >> mmusic@ietf.org
> >> https://www.ietf.org/mailman/listinfo/mmusic
> >
> >
> >


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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-media-path-middleboxes-06.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Multiparty Multimedia Session Control Wor=
king Group of the IETF.

	Title           : Analysis of Middlebox Interactions for Signaling Protoco=
l Communication along the Media Path
	Author(s)       : Brian Stucker
                          Hannes Tschofenig
                          Gonzalo Salgueiro
	Filename        : draft-ietf-mmusic-media-path-middleboxes-06.txt
	Pages           : 22
	Date            : 2013-01-10

Abstract:
   Middleboxes are defined as any intermediary box performing functions
   apart from normal, standard functions of an IP router on the data
   path between a source host and destination host.  Two such functions
   are network address translation and firewalling.

   When Application Layer Gateways, such as SIP entities, interact with
   NATs and firewalls, as described in the MIDCOM architecture, then
   problems may occur in the transport of media traffic when signaling
   protocol interaction takes place along the media path, as it is the
   case for recent key exchange proposals (such as DTLS-SRTP).  This
   document highlights problems that may arise.  Unfortunately, it is
   difficult for the end points to detect or predict problematic
   behavior and to determine whether the media path is reliably
   available for packet exchange.

   This document aims to summarize the various sources and effects of
   NAT and firewall control, the reasons that they exist, and possible
   means of improving their behavior to allow protocols that rely upon
   signaling along the media path to operate effectively.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-media-path-middleboxes

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-media-path-middleboxes-06

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-media-path-middleboxes=
-06


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


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MMUSIC -=20

I have simply updated the version of this draft to prevent expiration =
because recent discussions with the chairs have confirmed that we are =
ready now to progress this milestone. As part of that effort the chairs =
requested we clearly define the purpose, scope, and intended audience =
for the document.=20

I came a bit late to the party but from my perspective this draft's =
purpose is to provide guidance and best practices on mitigating the =
issues middleboxes (particularly NATs and firewalls) can cause to =
end-to-end SIP/SDP-signaled media streams in different architectures.  =
The target audience is anyone (i.e. NAT/FW vendors, protocol designers, =
etc.) wanting such information ond that might benefit from the design =
tips to make their network/protocol/device more middlebox friendly.

I believe Flemming's recent comments (as an individual) masterfully =
summarized the scope, purpose and target audience for this document. =
Thus, I don't want to repeat (less eloquently) what I consider to be the =
right approach to this document.  For your convenience, here are those =
comments:

=3D=3D=3D=3D=3D

A lot of the SIP and SDP work started out with a somewhat "purist" =
Internet view of the world, where things such as middleboxes were not =
considered. As other standards and industry organizations became =
interested in using SIP and SDP, different architectures were defined =
and some of those arcitectures did include the notion of middleboxes =
(e.g. 3GPP IMS and CableLabs PacketCable). =46rom an IETF point of view, =
there was, at least initially, not a great deal of knowledge about these =
architectures and what the middleboxes defined by them were doing. =
Conversely, there was (is) also a concern that such middleboxes may =
break end-to-end transparency and hence there was a desire to try and =
alleviate that.=20

To that effect, it was seen as useful to have a document that could
a) Explain what/how middleboxes might be operating in these =
architectures
b) Provide guidelines as to how such middleboxes could minimize (ideally =
avoid) impacting end-to-end transparency and/or how protocols could be =
used to try and alleviate any impact such middleboxes might have.=20

The middleboxes draft is trying to address the above as it relates to =
the media path. The target audience is thus IETF participants that would =
like an overview of how these middleboxes may affect SIP/SDP-signaled =
media streams as well as what can be done to try and overcome that. =
Similarly, the document is targeted at people involved in these "other" =
architecture efforts, with a goal of making it clear how middleboxes may =
affect the operation of SIP/SDP-signaled media streams and hence provide =
some "design principles". This is not unlike some of the NAT work that =
was done in BEHAVE.=20

It could be argued that these architectures have now been around for so =
long that producing the above document will not make any difference at =
this point. While I have some sympathy for this, I also think we have to =
recognize the importance of documenting what we know and to make that =
readily available for new people that will be working in this space.=20

In other words, I still believe there is value in pursuing this =
document. As to whether it should be a BCP or Informational, I don't =
have any strong opinions.=20

=3D=3D=3D=3D=3D

As a group can we agree on this scope and that is worthwhile to finally =
complete this milestone?  Feedback welcome.

Cheers,

Gonzalo


On Jan 10, 2013, at 5:45 PM, Internet-Drafts@ietf.org wrote:

>=20
> A New Internet-Draft is available from the on-line Internet-Drafts =
directories.
> This draft is a work item of the Multiparty Multimedia Session Control =
Working Group of the IETF.
>=20
> 	Title           : Analysis of Middlebox Interactions for =
Signaling Protocol Communication along the Media Path
> 	Author(s)       : Brian Stucker
>                          Hannes Tschofenig
>                          Gonzalo Salgueiro
> 	Filename        : =
draft-ietf-mmusic-media-path-middleboxes-06.txt
> 	Pages           : 22
> 	Date            : 2013-01-10
>=20
> Abstract:
>   Middleboxes are defined as any intermediary box performing functions
>   apart from normal, standard functions of an IP router on the data
>   path between a source host and destination host.  Two such functions
>   are network address translation and firewalling.
>=20
>   When Application Layer Gateways, such as SIP entities, interact with
>   NATs and firewalls, as described in the MIDCOM architecture, then
>   problems may occur in the transport of media traffic when signaling
>   protocol interaction takes place along the media path, as it is the
>   case for recent key exchange proposals (such as DTLS-SRTP).  This
>   document highlights problems that may arise.  Unfortunately, it is
>   difficult for the end points to detect or predict problematic
>   behavior and to determine whether the media path is reliably
>   available for packet exchange.
>=20
>   This document aims to summarize the various sources and effects of
>   NAT and firewall control, the reasons that they exist, and possible
>   means of improving their behavior to allow protocols that rely upon
>   signaling along the media path to operate effectively.
>=20
>=20
> The IETF datatracker status page for this draft is:
> =
https://datatracker.ietf.org/doc/draft-ietf-mmusic-media-path-middleboxes
>=20
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-mmusic-media-path-middleboxes-06
>=20
> A diff from the previous version is available at:
> =
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-media-path-middleboxe=
s-06
>=20
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> _______________________________________________
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>=20


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From: "Dan Wing" <dwing@cisco.com>
To: "'Gonzalo Salgueiro'" <gsalguei@cisco.com>, "'mmusic'" <mmusic@ietf.org>
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Subject: Re: [MMUSIC] I-D Action:	draft-ietf-mmusic-media-path-middleboxes-06.txt
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> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf
> Of Gonzalo Salgueiro
> Sent: Thursday, January 10, 2013 2:47 PM
> To: mmusic
> Cc: obsidian97@gmail.com
> Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-media-path-
> middleboxes-06.txt
> 
> MMUSIC -
> 
> I have simply updated the version of this draft to prevent expiration
> because recent discussions with the chairs have confirmed that we are
> ready now to progress this milestone. As part of that effort the chairs
> requested we clearly define the purpose, scope, and intended audience
> for the document.
> 
> I came a bit late to the party but from my perspective this draft's
> purpose is to provide guidance and best practices on mitigating the
> issues middleboxes (particularly NATs and firewalls) can cause to end-
> to-end SIP/SDP-signaled media streams in different architectures.  The
> target audience is anyone (i.e. NAT/FW vendors, protocol designers,
> etc.) wanting such information ond that might benefit from the design
> tips to make their network/protocol/device more middlebox friendly.
> 
> I believe Flemming's recent comments (as an individual) masterfully
> summarized the scope, purpose and target audience for this document.
> Thus, I don't want to repeat (less eloquently) what I consider to be the
> right approach to this document.  For your convenience, here are those
> comments:
> 
> =====
> 
> A lot of the SIP and SDP work started out with a somewhat "purist"
> Internet view of the world, where things such as middleboxes were not
> considered. As other standards and industry organizations became
> interested in using SIP and SDP, different architectures were defined
> and some of those arcitectures did include the notion of middleboxes
> (e.g. 3GPP IMS and CableLabs PacketCable). From an IETF point of view,
> there was, at least initially, not a great deal of knowledge about these
> architectures and what the middleboxes defined by them were doing.
> Conversely, there was (is) also a concern that such middleboxes may
> break end-to-end transparency and hence there was a desire to try and
> alleviate that.
> 
> To that effect, it was seen as useful to have a document that could
> a) Explain what/how middleboxes might be operating in these
> architectures
> b) Provide guidelines as to how such middleboxes could minimize (ideally
> avoid) impacting end-to-end transparency and/or how protocols could be
> used to try and alleviate any impact such middleboxes might have.
> 
> The middleboxes draft is trying to address the above as it relates to
> the media path. The target audience is thus IETF participants that would
> like an overview of how these middleboxes may affect SIP/SDP-signaled
> media streams as well as what can be done to try and overcome that.
> Similarly, the document is targeted at people involved in these "other"
> architecture efforts, with a goal of making it clear how middleboxes may
> affect the operation of SIP/SDP-signaled media streams and hence provide
> some "design principles". This is not unlike some of the NAT work that
> was done in BEHAVE.
> 
> It could be argued that these architectures have now been around for so
> long that producing the above document will not make any difference at
> this point. While I have some sympathy for this, I also think we have to
> recognize the importance of documenting what we know and to make that
> readily available for new people that will be working in this space.
> 
> In other words, I still believe there is value in pursuing this
> document. As to whether it should be a BCP or Informational, I don't
> have any strong opinions.
> 
> =====
> 
> As a group can we agree on this scope and that is worthwhile to finally
> complete this milestone?  Feedback welcome.

I am glad this document was resurrected.  

Can the document say SBC instead of the awkward (and inaccurate) 
"SIP ALG combined with MIDCOM agent"?  Or is saying SBC impolitic.

-d


> Cheers,
> 
> Gonzalo
> 
> 
> On Jan 10, 2013, at 5:45 PM, Internet-Drafts@ietf.org wrote:
> 
> >
> > A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> > This draft is a work item of the Multiparty Multimedia Session Control
> Working Group of the IETF.
> >
> > 	Title           : Analysis of Middlebox Interactions for Signaling
> Protocol Communication along the Media Path
> > 	Author(s)       : Brian Stucker
> >                          Hannes Tschofenig
> >                          Gonzalo Salgueiro
> > 	Filename        : draft-ietf-mmusic-media-path-middleboxes-06.txt
> > 	Pages           : 22
> > 	Date            : 2013-01-10
> >
> > Abstract:
> >   Middleboxes are defined as any intermediary box performing functions
> >   apart from normal, standard functions of an IP router on the data
> >   path between a source host and destination host.  Two such functions
> >   are network address translation and firewalling.
> >
> >   When Application Layer Gateways, such as SIP entities, interact with
> >   NATs and firewalls, as described in the MIDCOM architecture, then
> >   problems may occur in the transport of media traffic when signaling
> >   protocol interaction takes place along the media path, as it is the
> >   case for recent key exchange proposals (such as DTLS-SRTP).  This
> >   document highlights problems that may arise.  Unfortunately, it is
> >   difficult for the end points to detect or predict problematic
> >   behavior and to determine whether the media path is reliably
> >   available for packet exchange.
> >
> >   This document aims to summarize the various sources and effects of
> >   NAT and firewall control, the reasons that they exist, and possible
> >   means of improving their behavior to allow protocols that rely upon
> >   signaling along the media path to operate effectively.
> >
> >
> > The IETF datatracker status page for this draft is:
> > https://datatracker.ietf.org/doc/draft-ietf-mmusic-media-path-middlebo
> > xes
> >
> > There's also a htmlized version available at:
> > http://tools.ietf.org/html/draft-ietf-mmusic-media-path-middleboxes-06
> >
> > A diff from the previous version is available at:
> > http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-media-path-middlebo
> > xes-06
> >
> >
> > Internet-Drafts are also available by anonymous FTP at:
> > ftp://ftp.ietf.org/internet-drafts/
> >
> > _______________________________________________
> > I-D-Announce mailing list
> > I-D-Announce@ietf.org
> > https://www.ietf.org/mailman/listinfo/i-d-announce
> > Internet-Draft directories: http://www.ietf.org/shadow.html or
> > ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> >
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: Emil Ivov <emcho@jitsi.org>, Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: [MMUSIC] Query: payload type collision with offer/answer
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Date: Thu, 10 Jan 2013 23:22:58 +0000
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References: <50EDC40F.3010501@alvestrand.no> <1.77712513fa3f074cccef@cisco.com>	<50EDF490.4000101@alvestrand.no> <50EE0139.8000909@jitsi.org>
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Subject: Re: [MMUSIC] Query: payload type collision with offer/answer
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Please see inline.

> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Emil Ivov
> Sent: Wednesday, January 09, 2013 3:46 PM
> To: Harald Alvestrand
> Cc: mmusic@ietf.org
> Subject: Re: [MMUSIC] Query: payload type collision with offer/answer
>=20
>=20
>=20
> On 09.01.13, 23:52, Harald Alvestrand wrote:
> > On 01/09/2013 10:01 PM, Charles Eckel (eckelcu) wrote:
> >> Please see inline.
> >>
> >>> -----Original Message-----
> >>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> >>> Behalf Of Harald Alvestrand
> >>> Sent: Wednesday, January 09, 2013 11:25 AM
> >>> To: mmusic@ietf.org
> >>> Subject: [MMUSIC] Query: payload type collision with offer/answer
> >>>
> >>> Hello,
> >>> I have encountered an issue, and I'm not sure it's an issue or not. S=
DP
> >>> wisdom is sought.
> >>>
> >>> Suppose two entities exchange the following offer/answer:
> >>>
> >>> offer from X:
> >>>
> >>> m=3Dvideo 97
> >>> a=3Drtpmap:97 foo
> >>>
> >>> answer from Y:
> >>>
> >>> m=3Dvideo 98
> >>> a=3Drtpmap:98 foo
> >>>
> >>> It's clear from RFC 3264 that now X must send codec foo with payload
> >>> type 98, and Y must send codec foo with payload type 97. That's not t=
he
> >>> question.
> >>>
> >>> But suppose these are the offer and answer:
> >>>
> >>> offer from X:
> >>>
> >>> m=3Dvideo 97 98
> >>> a=3Drtpmap:97 foo
> >>> a=3Drtpmap:98 bar
> >>>
> >>> answer from Y:
> >>>
> >>> m=3Dvideo 97 98
> >>> a=3Drtpmap:97 bar
> >>> a=3Drtpmap:98 foo
> >>>
> >>> That is, the payload types collide.
> >>>
> >>> It would be possible to write code so that X sends foo with 98 and ba=
r
> >>> with 97, while Y sends foo with 97 and bar with 98. But is it
> >> conformant
> >>> with the specs to do so?
> >>>
> >>> And if this has a clear answer - which paragraph of which RFC makes
> >> this
> >>> clear?
> >>> (Yes, this discussion is triggered from a real world problem.)
> >>
> >> It is recommended to avoid such mismatches when possible, but doing so
> >> may not always be possible. The examples you mention are conformant.
> >> The following text from section 5.1 addresses this:
> >>
> >> For sendrecv RTP
> >> streams, the payload type numbers indicate the value of the payload
> >> type field the offerer expects to receive, and would prefer to send.
> >> However, for sendonly and sendrecv streams, the answer might indicate
> >> different payload type numbers for the same codecs, in which case,
> >> the offerer MUST send with the payload type numbers from the answer.
> >>
> >> Different payload type numbers may be needed in each direction
> >> because of interoperability concerns with H.323.
> > I was reading that text, and it clearly says which ones to use if the
> > payload types are different, but I don't think they clearly address the
> > situation with using the same payload type for two different codecs in
> > different directions.
> >
> > In a multicast or transport relay scenario, this would clearly be
> > untenable, since third parties would not be able to send streams that
> > would be interpreted correctly by the two conflicting endpoints, but in
> > a two-party conversation, it's possible to make this work.
> > I just don't know if it's worth the effort to make it so.
>=20
> Well there's also section 8.3.2 in 3264. The section is about updating
> offers but it seems like the text explicitly prohibits remapping payload
> types in any scenarios:
>=20
>    However, in the
>    case of RTP, the mapping from a particular dynamic payload type
>    number to a particular codec within that media stream MUST NOT change
>    for the duration of a session.  For example, if A generates an offer
>    with G.711 assigned to dynamic payload type number 46, payload type
>    number 46 MUST refer to G.711 from that point forward in any offers
>    or answers for that media stream within the session.  However, it is
>    acceptable for multiple payload type numbers to be mapped to the same
>    codec, so that an updated offer could also use payload type number 72
>    for G.711.

I viewed this as being a requirement for all offers or answers sent by A to=
 B, but not necessarily the corresponding answer and offers from B to A. Ot=
herwise, it seems to conflict with the text in section 5.1. That said, I ca=
n certainly see where someone else may interpret it differently.

 Cheers,
Charles

> Cheers,
> Emil
>=20
>=20
> --
> https://jitsi.org
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

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Date: Fri, 11 Jan 2013 00:37:29 +0100
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From: Emil Ivov <emcho@jitsi.org>
To: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
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Subject: Re: [MMUSIC] Query: payload type collision with offer/answer
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On Jan 11, 2013 12:23 AM, "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
wrote:
>
> Please see inline.
>
> > -----Original Message-----
> > From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> > Behalf Of Emil Ivov
> > Sent: Wednesday, January 09, 2013 3:46 PM
> > To: Harald Alvestrand
> > Cc: mmusic@ietf.org
> > Subject: Re: [MMUSIC] Query: payload type collision with offer/answer
> >
> >
> >
> > On 09.01.13, 23:52, Harald Alvestrand wrote:
> > > On 01/09/2013 10:01 PM, Charles Eckel (eckelcu) wrote:
> > >> Please see inline.
> > >>
> > >>> -----Original Message-----
> > >>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietforg] On
> > >>> Behalf Of Harald Alvestrand
> > >>> Sent: Wednesday, January 09, 2013 11:25 AM
> > >>> To: mmusic@ietf.org
> > >>> Subject: [MMUSIC] Query: payload type collision with offer/answer
> > >>>
> > >>> Hello,
> > >>> I have encountered an issue, and I'm not sure it's an issue or not.
SDP
> > >>> wisdom is sought.
> > >>>
> > >>> Suppose two entities exchange the following offer/answer:
> > >>>
> > >>> offer from X:
> > >>>
> > >>> m=video 97
> > >>> a=rtpmap:97 foo
> > >>>
> > >>> answer from Y:
> > >>>
> > >>> m=video 98
> > >>> a=rtpmap:98 foo
> > >>>
> > >>> It's clear from RFC 3264 that now X must send codec foo with
payload
> > >>> type 98, and Y must send codec foo with payload type 97. That's not
the
> > >>> question.
> > >>>
> > >>> But suppose these are the offer and answer:
> > >>>
> > >>> offer from X:
> > >>>
> > >>> m=video 97 98
> > >>> a=rtpmap:97 foo
> > >>> a=rtpmap:98 bar
> > >>>
> > >>> answer from Y:
> > >>>
> > >>> m=video 97 98
> > >>> a=rtpmap:97 bar
> > >>> a=rtpmap:98 foo
> > >>>
> > >>> That is, the payload types collide.
> > >>>
> > >>> It would be possible to write code so that X sends foo with 98 and
bar
> > >>> with 97, while Y sends foo with 97 and bar with 98. But is it
> > >> conformant
> > >>> with the specs to do so?
> > >>>
> > >>> And if this has a clear answer - which paragraph of which RFC makes
> > >> this
> > >>> clear?
> > >>> (Yes, this discussion is triggered from a real world problem.)
> > >>
> > >> It is recommended to avoid such mismatches when possible, but doing
so
> > >> may not always be possible. The examples you mention are conformant.
> > >> The following text from section 5.1 addresses this:
> > >>
> > >> For sendrecv RTP
> > >> streams, the payload type numbers indicate the value of the payload
> > >> type field the offerer expects to receive, and would prefer to send.
> > >> However, for sendonly and sendrecv streams, the answer might
indicate
> > >> different payload type numbers for the same codecs, in which case,
> > >> the offerer MUST send with the payload type numbers from the answer.
> > >>
> > >> Different payload type numbers may be needed in each direction
> > >> because of interoperability concerns with H.323.
> > > I was reading that text, and it clearly says which ones to use if the
> > > payload types are different, but I don't think they clearly address
the
> > > situation with using the same payload type for two different codecs
in
> > > different directions.
> > >
> > > In a multicast or transport relay scenario, this would clearly be
> > > untenable, since third parties would not be able to send streams that
> > > would be interpreted correctly by the two conflicting endpoints, but
in
> > > a two-party conversation, it's possible to make this work.
> > > I just don't know if it's worth the effort to make it so.
> >
> > Well there's also section 8.3.2 in 3264. The section is about updating
> > offers but it seems like the text explicitly prohibits remapping
payload
> > types in any scenarios:
> >
> > However, in the
> > case of RTP, the mapping from a particular dynamic payload type
> > number to a particular codec within that media stream MUST NOT change
> > for the duration of a session. For example, if A generates an offer
> > with G.711 assigned to dynamic payload type number 46, payload type
> > number 46 MUST refer to G.711 from that point forward in any offers
> > or answers for that media stream within the session. However, it is
> > acceptable for multiple payload type numbers to be mapped to the same
> > codec, so that an updated offer could also use payload type number 72
> > for G.711.
>
> I viewed this as being a requirement for all offers or answers sent by
> A to B, but not necessarily the corresponding answer and offers from
> B to A.

Well, it does say: "from that point forward in _any_ offers or answers"

> Otherwise, it seems to conflict with the text in section 5.1.

Mmm ... which part of 5.1?

To me it just seems like a completion (admittedly, somewhat out of place).

> That said, I can certainly see where someone else may interpret it
> differently.

Agreed. Definitely 3264bis material.

Cheers,
Emil

--sent from my mobile
>
> Cheers,
> Charles
>
> > Cheers,
> > Emil
> >
> >
> > --
> > https://jitsi.org
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www.ietf.org/mailman/listinfo/mmusic
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

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<p dir="ltr"><br></p>
<p dir="ltr">On Jan 11, 2013 12:23 AM, &quot;Charles Eckel (eckelcu)&quot; &lt;<a href="mailto:eckelcu@cisco.com">eckelcu@cisco.com</a>&gt; wrote:<br>
&gt;<br>
&gt; Please see inline. <br>
&gt;<br>
&gt; &gt; -----Original Message----- <br>
&gt; &gt; From: <a href="mailto:mmusic-bounces@ietf.org">mmusic-bounces@ietf.org</a> [mailto:<a href="mailto:mmusic-bounces@ietf.org">mmusic-bounces@ietf.org</a>] On <br>
&gt; &gt; Behalf Of Emil Ivov <br>
&gt; &gt; Sent: Wednesday, January 09, 2013 3:46 PM <br>
&gt; &gt; To: Harald Alvestrand <br>
&gt; &gt; Cc: <a href="mailto:mmusic@ietf.org">mmusic@ietf.org</a> <br>
&gt; &gt; Subject: Re: [MMUSIC] Query: payload type collision with offer/answer <br>
&gt; &gt; <br>
&gt; &gt; <br>
&gt; &gt; <br>
&gt; &gt; On 09.01.13, 23:52, Harald Alvestrand wrote: <br>
&gt; &gt; &gt; On 01/09/2013 10:01 PM, Charles Eckel (eckelcu) wrote: <br>
&gt; &gt; &gt;&gt; Please see inline. <br>
&gt; &gt; &gt;&gt; <br>
&gt; &gt; &gt;&gt;&gt; -----Original Message----- <br>
&gt; &gt; &gt;&gt;&gt; From: <a href="mailto:mmusic-bounces@ietf.org">mmusic-bounces@ietf.org</a> [mailto:<a href="mailto:mmusic-bounces@ietforg">mmusic-bounces@ietforg</a>] On <br>
&gt; &gt; &gt;&gt;&gt; Behalf Of Harald Alvestrand <br>
&gt; &gt; &gt;&gt;&gt; Sent: Wednesday, January 09, 2013 11:25 AM <br>
&gt; &gt; &gt;&gt;&gt; To: <a href="mailto:mmusic@ietf.org">mmusic@ietf.org</a> <br>
&gt; &gt; &gt;&gt;&gt; Subject: [MMUSIC] Query: payload type collision with offer/answer <br>
&gt; &gt; &gt;&gt;&gt; <br>
&gt; &gt; &gt;&gt;&gt; Hello, <br>
&gt; &gt; &gt;&gt;&gt; I have encountered an issue, and I&#39;m not sure it&#39;s an issue or not. SDP <br>
&gt; &gt; &gt;&gt;&gt; wisdom is sought. <br>
&gt; &gt; &gt;&gt;&gt; <br>
&gt; &gt; &gt;&gt;&gt; Suppose two entities exchange the following offer/answer: <br>
&gt; &gt; &gt;&gt;&gt; <br>
&gt; &gt; &gt;&gt;&gt; offer from X: <br>
&gt; &gt; &gt;&gt;&gt; <br>
&gt; &gt; &gt;&gt;&gt; m=video 97 <br>
&gt; &gt; &gt;&gt;&gt; a=rtpmap:97 foo <br>
&gt; &gt; &gt;&gt;&gt; <br>
&gt; &gt; &gt;&gt;&gt; answer from Y: <br>
&gt; &gt; &gt;&gt;&gt; <br>
&gt; &gt; &gt;&gt;&gt; m=video 98 <br>
&gt; &gt; &gt;&gt;&gt; a=rtpmap:98 foo <br>
&gt; &gt; &gt;&gt;&gt; <br>
&gt; &gt; &gt;&gt;&gt; It&#39;s clear from RFC 3264 that now X must send codec foo with payload <br>
&gt; &gt; &gt;&gt;&gt; type 98, and Y must send codec foo with payload type 97. That&#39;s not the <br>
&gt; &gt; &gt;&gt;&gt; question. <br>
&gt; &gt; &gt;&gt;&gt; <br>
&gt; &gt; &gt;&gt;&gt; But suppose these are the offer and answer: <br>
&gt; &gt; &gt;&gt;&gt; <br>
&gt; &gt; &gt;&gt;&gt; offer from X: <br>
&gt; &gt; &gt;&gt;&gt; <br>
&gt; &gt; &gt;&gt;&gt; m=video 97 98 <br>
&gt; &gt; &gt;&gt;&gt; a=rtpmap:97 foo <br>
&gt; &gt; &gt;&gt;&gt; a=rtpmap:98 bar <br>
&gt; &gt; &gt;&gt;&gt; <br>
&gt; &gt; &gt;&gt;&gt; answer from Y: <br>
&gt; &gt; &gt;&gt;&gt; <br>
&gt; &gt; &gt;&gt;&gt; m=video 97 98 <br>
&gt; &gt; &gt;&gt;&gt; a=rtpmap:97 bar <br>
&gt; &gt; &gt;&gt;&gt; a=rtpmap:98 foo <br>
&gt; &gt; &gt;&gt;&gt; <br>
&gt; &gt; &gt;&gt;&gt; That is, the payload types collide. <br>
&gt; &gt; &gt;&gt;&gt; <br>
&gt; &gt; &gt;&gt;&gt; It would be possible to write code so that X sends foo with 98 and bar <br>
&gt; &gt; &gt;&gt;&gt; with 97, while Y sends foo with 97 and bar with 98. But is it <br>
&gt; &gt; &gt;&gt; conformant <br>
&gt; &gt; &gt;&gt;&gt; with the specs to do so? <br>
&gt; &gt; &gt;&gt;&gt; <br>
&gt; &gt; &gt;&gt;&gt; And if this has a clear answer - which paragraph of which RFC makes <br>
&gt; &gt; &gt;&gt; this <br>
&gt; &gt; &gt;&gt;&gt; clear? <br>
&gt; &gt; &gt;&gt;&gt; (Yes, this discussion is triggered from a real world problem.) <br>
&gt; &gt; &gt;&gt; <br>
&gt; &gt; &gt;&gt; It is recommended to avoid such mismatches when possible, but doing so <br>
&gt; &gt; &gt;&gt; may not always be possible. The examples you mention are conformant. <br>
&gt; &gt; &gt;&gt; The following text from section 5.1 addresses this: <br>
&gt; &gt; &gt;&gt; <br>
&gt; &gt; &gt;&gt; For sendrecv RTP <br>
&gt; &gt; &gt;&gt; streams, the payload type numbers indicate the value of the payload <br>
&gt; &gt; &gt;&gt; type field the offerer expects to receive, and would prefer to send. <br>
&gt; &gt; &gt;&gt; However, for sendonly and sendrecv streams, the answer might indicate <br>
&gt; &gt; &gt;&gt; different payload type numbers for the same codecs, in which case, <br>
&gt; &gt; &gt;&gt; the offerer MUST send with the payload type numbers from the answer. <br>
&gt; &gt; &gt;&gt; <br>
&gt; &gt; &gt;&gt; Different payload type numbers may be needed in each direction <br>
&gt; &gt; &gt;&gt; because of interoperability concerns with H.323. <br>
&gt; &gt; &gt; I was reading that text, and it clearly says which ones to use if the <br>
&gt; &gt; &gt; payload types are different, but I don&#39;t think they clearly address the <br>
&gt; &gt; &gt; situation with using the same payload type for two different codecs in <br>
&gt; &gt; &gt; different directions. <br>
&gt; &gt; &gt; <br>
&gt; &gt; &gt; In a multicast or transport relay scenario, this would clearly be <br>
&gt; &gt; &gt; untenable, since third parties would not be able to send streams that <br>
&gt; &gt; &gt; would be interpreted correctly by the two conflicting endpoints, but in <br>
&gt; &gt; &gt; a two-party conversation, it&#39;s possible to make this work. <br>
&gt; &gt; &gt; I just don&#39;t know if it&#39;s worth the effort to make it so. <br>
&gt; &gt; <br>
&gt; &gt; Well there&#39;s also section 8.3.2 in 3264. The section is about updating <br>
&gt; &gt; offers but it seems like the text explicitly prohibits remapping payload <br>
&gt; &gt; types in any scenarios: <br>
&gt; &gt; <br>
&gt; &gt; However, in the <br>
&gt; &gt; case of RTP, the mapping from a particular dynamic payload type <br>
&gt; &gt; number to a particular codec within that media stream MUST NOT change <br>
&gt; &gt; for the duration of a session. For example, if A generates an offer <br>
&gt; &gt; with G.711 assigned to dynamic payload type number 46, payload type <br>
&gt; &gt; number 46 MUST refer to G.711 from that point forward in any offers <br>
&gt; &gt; or answers for that media stream within the session. However, it is <br>
&gt; &gt; acceptable for multiple payload type numbers to be mapped to the same <br>
&gt; &gt; codec, so that an updated offer could also use payload type number 72 <br>
&gt; &gt; for G.711. <br>
&gt;<br>
&gt; I viewed this as being a requirement for all offers or answers sent by<br>
&gt; A to B, but not necessarily the corresponding answer and offers from<br>
&gt; B to A. </p>
<p dir="ltr">Well, it does say: &quot;from that point forward in _any_ offers or answers&quot;</p>
<p dir="ltr">&gt; Otherwise, it seems to conflict with the text in section 5.1.</p>
<p dir="ltr">Mmm ... which part of 5.1? </p>
<p dir="ltr">To me it just seems like a completion (admittedly, somewhat out of place).</p>
<p dir="ltr">&gt; That said, I can certainly see where someone else may interpret it<br>
&gt; differently. </p>
<p dir="ltr">Agreed. Definitely 3264bis material.</p>
<p dir="ltr">Cheers,<br>
Emil</p>
<p dir="ltr">--sent from my mobile<br>
&gt;<br>
&gt; Cheers, <br>
&gt; Charles <br>
&gt;<br>
&gt; &gt; Cheers, <br>
&gt; &gt; Emil <br>
&gt; &gt; <br>
&gt; &gt; <br>
&gt; &gt; -- <br>
&gt; &gt; <a href="https://jitsi.org">https://jitsi.org</a> <br>
&gt; &gt; _______________________________________________ <br>
&gt; &gt; mmusic mailing list <br>
&gt; &gt; <a href="mailto:mmusic@ietf.org">mmusic@ietf.org</a> <br>
&gt; &gt; <a href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a> <br>
&gt; _______________________________________________ <br>
&gt; mmusic mailing list <br>
&gt; <a href="mailto:mmusic@ietf.org">mmusic@ietf.org</a> <br>
&gt; <a href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a> <br>
</p>

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Subject: Re: [MMUSIC] Fwd: RTCWEB & MMUSIC WG Joint Interim Meeting, February 5-7, 2013
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Is there yet an agenda for the upcoming MMUSIC interim or at least a 
list of drafts we intend to discuss that I could review in advance? Just 
feels like we're running out of days for advance flight bookings.

Matthew Kaufman

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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: Emil Ivov <emcho@jitsi.org>
Thread-Topic: [MMUSIC] Query: payload type collision with offer/answer
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Date: Thu, 10 Jan 2013 23:47:40 +0000
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References: <50EDC40F.3010501@alvestrand.no> <1.77712513fa3f074cccef@cisco.com><50EDF490.4000101@alvestrand.no> <50EE0139.8000909@jitsi.org><1.3841c28e488bfa397d5b@cisco.com> <1.96f025c482c091c166cc@jitsi.org>
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Subject: Re: [MMUSIC] Query: payload type collision with offer/answer
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> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Emil Ivov
> Sent: Thursday, January 10, 2013 3:37 PM
> To: Charles Eckel (eckelcu)
> Cc: mmusic
> Subject: Re: [MMUSIC] Query: payload type collision with offer/answer
>=20
>=20
>=20
>=20
> On Jan 11, 2013 12:23 AM, "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
> wrote:
> >
> > Please see inline.
> >
> > > -----Original Message-----
> > > From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> > > Behalf Of Emil Ivov
> > > Sent: Wednesday, January 09, 2013 3:46 PM
> > > To: Harald Alvestrand
> > > Cc: mmusic@ietf.org
> > > Subject: Re: [MMUSIC] Query: payload type collision with offer/answer
> > >
> > >
> > >
> > > On 09.01.13, 23:52, Harald Alvestrand wrote:
> > > > On 01/09/2013 10:01 PM, Charles Eckel (eckelcu) wrote:
> > > >> Please see inline.
> > > >>
> > > >>> -----Original Message-----
> > > >>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietforg]
> On
> > > >>> Behalf Of Harald Alvestrand
> > > >>> Sent: Wednesday, January 09, 2013 11:25 AM
> > > >>> To: mmusic@ietf.org
> > > >>> Subject: [MMUSIC] Query: payload type collision with offer/answer
> > > >>>
> > > >>> Hello,
> > > >>> I have encountered an issue, and I'm not sure it's an issue or no=
t. SDP
> > > >>> wisdom is sought.
> > > >>>
> > > >>> Suppose two entities exchange the following offer/answer:
> > > >>>
> > > >>> offer from X:
> > > >>>
> > > >>> m=3Dvideo 97
> > > >>> a=3Drtpmap:97 foo
> > > >>>
> > > >>> answer from Y:
> > > >>>
> > > >>> m=3Dvideo 98
> > > >>> a=3Drtpmap:98 foo
> > > >>>
> > > >>> It's clear from RFC 3264 that now X must send codec foo with payl=
oad
> > > >>> type 98, and Y must send codec foo with payload type 97. That's n=
ot
> the
> > > >>> question.
> > > >>>
> > > >>> But suppose these are the offer and answer:
> > > >>>
> > > >>> offer from X:
> > > >>>
> > > >>> m=3Dvideo 97 98
> > > >>> a=3Drtpmap:97 foo
> > > >>> a=3Drtpmap:98 bar
> > > >>>
> > > >>> answer from Y:
> > > >>>
> > > >>> m=3Dvideo 97 98
> > > >>> a=3Drtpmap:97 bar
> > > >>> a=3Drtpmap:98 foo
> > > >>>
> > > >>> That is, the payload types collide.
> > > >>>
> > > >>> It would be possible to write code so that X sends foo with 98 an=
d bar
> > > >>> with 97, while Y sends foo with 97 and bar with 98. But is it
> > > >> conformant
> > > >>> with the specs to do so?
> > > >>>
> > > >>> And if this has a clear answer - which paragraph of which RFC mak=
es
> > > >> this
> > > >>> clear?
> > > >>> (Yes, this discussion is triggered from a real world problem.)
> > > >>
> > > >> It is recommended to avoid such mismatches when possible, but
> doing so
> > > >> may not always be possible. The examples you mention are
> conformant.
> > > >> The following text from section 5.1 addresses this:
> > > >>
> > > >> For sendrecv RTP
> > > >> streams, the payload type numbers indicate the value of the payloa=
d
> > > >> type field the offerer expects to receive, and would prefer to sen=
d.
> > > >> However, for sendonly and sendrecv streams, the answer might
> indicate
> > > >> different payload type numbers for the same codecs, in which case,
> > > >> the offerer MUST send with the payload type numbers from the
> answer.
> > > >>
> > > >> Different payload type numbers may be needed in each direction
> > > >> because of interoperability concerns with H.323.
> > > > I was reading that text, and it clearly says which ones to use if t=
he
> > > > payload types are different, but I don't think they clearly address=
 the
> > > > situation with using the same payload type for two different codecs=
 in
> > > > different directions.
> > > >
> > > > In a multicast or transport relay scenario, this would clearly be
> > > > untenable, since third parties would not be able to send streams th=
at
> > > > would be interpreted correctly by the two conflicting endpoints, bu=
t in
> > > > a two-party conversation, it's possible to make this work.
> > > > I just don't know if it's worth the effort to make it so.
> > >
> > > Well there's also section 8.3.2 in 3264. The section is about updatin=
g
> > > offers but it seems like the text explicitly prohibits remapping payl=
oad
> > > types in any scenarios:
> > >
> > > However, in the
> > > case of RTP, the mapping from a particular dynamic payload type
> > > number to a particular codec within that media stream MUST NOT change
> > > for the duration of a session. For example, if A generates an offer
> > > with G.711 assigned to dynamic payload type number 46, payload type
> > > number 46 MUST refer to G.711 from that point forward in any offers
> > > or answers for that media stream within the session. However, it is
> > > acceptable for multiple payload type numbers to be mapped to the same
> > > codec, so that an updated offer could also use payload type number 72
> > > for G.711.
> >
> > I viewed this as being a requirement for all offers or answers sent by
> > A to B, but not necessarily the corresponding answer and offers from
> > B to A.
>=20
> Well, it does say: "from that point forward in _any_ offers or answers"
>=20
> > Otherwise, it seems to conflict with the text in section 5.1.
>=20
> Mmm ... which part of 5.1?

This part:

   However, for sendonly and sendrecv streams, the answer might indicate
   different payload type numbers for the same codecs, in which case,
   the offerer MUST send with the payload type numbers from the answer.

      Different payload type numbers may be needed in each direction
      because of interoperability concerns with H.323.
=20
Cheers,
Charles

> To me it just seems like a completion (admittedly, somewhat out of place)=
.
>=20
> > That said, I can certainly see where someone else may interpret it
> > differently.
>=20
> Agreed. Definitely 3264bis material.
>=20
> Cheers,
> Emil
>=20
> --sent from my mobile
> >
> > Cheers,
> > Charles
> >
> > > Cheers,
> > > Emil
> > >
> > >
> > > --
> > > https://jitsi.org
> > > _______________________________________________
> > > mmusic mailing list
> > > mmusic@ietf.org
> > > https://www.ietf.org/mailman/listinfo/mmusic
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www.ietf.org/mailman/listinfo/mmusic
>=20


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References: <50EDC40F.3010501@alvestrand.no> <1.77712513fa3f074cccef@cisco.com> <50EDF490.4000101@alvestrand.no> <50EE0139.8000909@jitsi.org> <1.3841c28e488bfa397d5b@cisco.com> <1.96f025c482c091c166cc@jitsi.org> <1.a70a7a76ba63b579d824@cisco.com>
Date: Fri, 11 Jan 2013 00:54:43 +0100
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From: Emil Ivov <emcho@jitsi.org>
To: "Charles Eckel, (eckelcu)" <eckelcu@cisco.com>
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Subject: Re: [MMUSIC] Query: payload type collision with offer/answer
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On Jan 11, 2013 12:48 AM, "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
wrote:
>
> > -----Original Message-----
> > From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> > Behalf Of Emil Ivov
> > Sent: Thursday, January 10, 2013 3:37 PM
> > To: Charles Eckel (eckelcu)
> > Cc: mmusic
> > Subject: Re: [MMUSIC] Query: payload type collision with offer/answer
> >
> >
> >
> >
> > On Jan 11, 2013 12:23 AM, "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
> > wrote:
> > >
> > > Please see inline.
> > >
> > > > -----Original Message-----
> > > > From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> > > > Behalf Of Emil Ivov
> > > > Sent: Wednesday, January 09, 2013 3:46 PM
> > > > To: Harald Alvestrand
> > > > Cc: mmusic@ietf.org
> > > > Subject: Re: [MMUSIC] Query: payload type collision with
offer/answer
> > > >
> > > >
> > > >
> > > > On 09.01.13, 23:52, Harald Alvestrand wrote:
> > > > > On 01/09/2013 10:01 PM, Charles Eckel (eckelcu) wrote:
> > > > >> Please see inline.
> > > > >>
> > > > >>> -----Original Message-----
> > > > >>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietforg]
> > On
> > > > >>> Behalf Of Harald Alvestrand
> > > > >>> Sent: Wednesday, January 09, 2013 11:25 AM
> > > > >>> To: mmusic@ietf.org
> > > > >>> Subject: [MMUSIC] Query: payload type collision with
offer/answer
> > > > >>>
> > > > >>> Hello,
> > > > >>> I have encountered an issue, and I'm not sure it's an issue or
not. SDP
> > > > >>> wisdom is sought.
> > > > >>>
> > > > >>> Suppose two entities exchange the following offer/answer:
> > > > >>>
> > > > >>> offer from X:
> > > > >>>
> > > > >>> m=video 97
> > > > >>> a=rtpmap:97 foo
> > > > >>>
> > > > >>> answer from Y:
> > > > >>>
> > > > >>> m=video 98
> > > > >>> a=rtpmap:98 foo
> > > > >>>
> > > > >>> It's clear from RFC 3264 that now X must send codec foo with
payload
> > > > >>> type 98, and Y must send codec foo with payload type 97. That's
not
> > the
> > > > >>> question.
> > > > >>>
> > > > >>> But suppose these are the offer and answer:
> > > > >>>
> > > > >>> offer from X:
> > > > >>>
> > > > >>> m=video 97 98
> > > > >>> a=rtpmap:97 foo
> > > > >>> a=rtpmap:98 bar
> > > > >>>
> > > > >>> answer from Y:
> > > > >>>
> > > > >>> m=video 97 98
> > > > >>> a=rtpmap:97 bar
> > > > >>> a=rtpmap:98 foo
> > > > >>>
> > > > >>> That is, the payload types collide.
> > > > >>>
> > > > >>> It would be possible to write code so that X sends foo with 98
and bar
> > > > >>> with 97, while Y sends foo with 97 and bar with 98. But is it
> > > > >> conformant
> > > > >>> with the specs to do so?
> > > > >>>
> > > > >>> And if this has a clear answer - which paragraph of which RFC
makes
> > > > >> this
> > > > >>> clear?
> > > > >>> (Yes, this discussion is triggered from a real world problem.)
> > > > >>
> > > > >> It is recommended to avoid such mismatches when possible, but
> > doing so
> > > > >> may not always be possible. The examples you mention are
> > conformant.
> > > > >> The following text from section 5.1 addresses this:
> > > > >>
> > > > >> For sendrecv RTP
> > > > >> streams, the payload type numbers indicate the value of the
payload
> > > > >> type field the offerer expects to receive, and would prefer to
send.
> > > > >> However, for sendonly and sendrecv streams, the answer might
> > indicate
> > > > >> different payload type numbers for the same codecs, in which
case,
> > > > >> the offerer MUST send with the payload type numbers from the
> > answer.
> > > > >>
> > > > >> Different payload type numbers may be needed in each direction
> > > > >> because of interoperability concerns with H.323.
> > > > > I was reading that text, and it clearly says which ones to use if
the
> > > > > payload types are different, but I don't think they clearly
address the
> > > > > situation with using the same payload type for two different
codecs in
> > > > > different directions.
> > > > >
> > > > > In a multicast or transport relay scenario, this would clearly be
> > > > > untenable, since third parties would not be able to send streams
that
> > > > > would be interpreted correctly by the two conflicting endpoints,
but in
> > > > > a two-party conversation, it's possible to make this work.
> > > > > I just don't know if it's worth the effort to make it so.
> > > >
> > > > Well there's also section 8.3.2 in 3264. The section is about
updating
> > > > offers but it seems like the text explicitly prohibits remapping
payload
> > > > types in any scenarios:
> > > >
> > > > However, in the
> > > > case of RTP, the mapping from a particular dynamic payload type
> > > > number to a particular codec within that media stream MUST NOT
change
> > > > for the duration of a session. For example, if A generates an offer
> > > > with G.711 assigned to dynamic payload type number 46, payload type
> > > > number 46 MUST refer to G.711 from that point forward in any offers
> > > > or answers for that media stream within the session. However, it is
> > > > acceptable for multiple payload type numbers to be mapped to the
same
> > > > codec, so that an updated offer could also use payload type number
72
> > > > for G.711.
> > >
> > > I viewed this as being a requirement for all offers or answers sent
by
> > > A to B, but not necessarily the corresponding answer and offers from
> > > B to A.
> >
> > Well, it does say: "from that point forward in _any_ offers or answers"
> >
> > > Otherwise, it seems to conflict with the text in section 5.1.
> >
> > Mmm ... which part of 5.1?
>
> This part:
>
> However, for sendonly and sendrecv streams, the answer might indicate
> different payload type numbers for the same codecs, in which case,
> the offerer MUST send with the payload type numbers from the answer.
>
> Different payload type numbers may be needed in each direction
> because of interoperability concerns with H.323.

This part just says the PT numbers can be different in the offer and the
answer. It doesn't say they can be the same as other PT numbers that were
assigned to other formats, which is what section 8.3.2 prohibits.

This is the reason I was seeing 8.3.2 more as a completion rather than a
contradiction.

Emil
--sent from my mobile

>
> Cheers,
> Charles
>
> > To me it just seems like a completion (admittedly, somewhat out of
place).
> >
> > > That said, I can certainly see where someone else may interpret it
> > > differently.
> >
> > Agreed. Definitely 3264bis material.
> >
> > Cheers,
> > Emil
> >
> > --sent from my mobile
> > >
> > > Cheers,
> > > Charles
> > >
> > > > Cheers,
> > > > Emil
> > > >
> > > >
> > > > --
> > > > https://jitsi.org
> > > > _______________________________________________
> > > > mmusic mailing list
> > > > mmusic@ietf.org
> > > > https://www.ietf.org/mailman/listinfo/mmusic
> > > _______________________________________________
> > > mmusic mailing list
> > > mmusic@ietf.org
> > > https://www.ietf.org/mailman/listinfo/mmusic
> >
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

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<p dir="ltr"><br>
On Jan 11, 2013 12:48 AM, &quot;Charles Eckel (eckelcu)&quot; &lt;<a href="mailto:eckelcu@cisco.com">eckelcu@cisco.com</a>&gt; wrote:<br>
&gt;<br>
&gt; &gt; -----Original Message----- <br>
&gt; &gt; From: <a href="mailto:mmusic-bounces@ietf.org">mmusic-bounces@ietf.org</a> [mailto:<a href="mailto:mmusic-bounces@ietf.org">mmusic-bounces@ietf.org</a>] On <br>
&gt; &gt; Behalf Of Emil Ivov <br>
&gt; &gt; Sent: Thursday, January 10, 2013 3:37 PM <br>
&gt; &gt; To: Charles Eckel (eckelcu) <br>
&gt; &gt; Cc: mmusic <br>
&gt; &gt; Subject: Re: [MMUSIC] Query: payload type collision with offer/answer <br>
&gt; &gt; <br>
&gt; &gt; <br>
&gt; &gt; <br>
&gt; &gt; <br>
&gt; &gt; On Jan 11, 2013 12:23 AM, &quot;Charles Eckel (eckelcu)&quot; &lt;<a href="mailto:eckelcu@cisco.com">eckelcu@cisco.com</a>&gt; <br>
&gt; &gt; wrote: <br>
&gt; &gt; &gt; <br>
&gt; &gt; &gt; Please see inline. <br>
&gt; &gt; &gt; <br>
&gt; &gt; &gt; &gt; -----Original Message----- <br>
&gt; &gt; &gt; &gt; From: <a href="mailto:mmusic-bounces@ietf.org">mmusic-bounces@ietf.org</a> [mailto:<a href="mailto:mmusic-bounces@ietf.org">mmusic-bounces@ietf.org</a>] On <br>
&gt; &gt; &gt; &gt; Behalf Of Emil Ivov <br>
&gt; &gt; &gt; &gt; Sent: Wednesday, January 09, 2013 3:46 PM <br>
&gt; &gt; &gt; &gt; To: Harald Alvestrand <br>
&gt; &gt; &gt; &gt; Cc: <a href="mailto:mmusic@ietf.org">mmusic@ietf.org</a> <br>
&gt; &gt; &gt; &gt; Subject: Re: [MMUSIC] Query: payload type collision with offer/answer <br>
&gt; &gt; &gt; &gt; <br>
&gt; &gt; &gt; &gt; <br>
&gt; &gt; &gt; &gt; <br>
&gt; &gt; &gt; &gt; On 09.01.13, 23:52, Harald Alvestrand wrote: <br>
&gt; &gt; &gt; &gt; &gt; On 01/09/2013 10:01 PM, Charles Eckel (eckelcu) wrote: <br>
&gt; &gt; &gt; &gt; &gt;&gt; Please see inline. <br>
&gt; &gt; &gt; &gt; &gt;&gt; <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; -----Original Message----- <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; From: <a href="mailto:mmusic-bounces@ietf.org">mmusic-bounces@ietf.org</a> [mailto:<a href="mailto:mmusic-bounces@ietforg">mmusic-bounces@ietforg</a>] <br>
&gt; &gt; On <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; Behalf Of Harald Alvestrand <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; Sent: Wednesday, January 09, 2013 11:25 AM <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; To: <a href="mailto:mmusic@ietf.org">mmusic@ietf.org</a> <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; Subject: [MMUSIC] Query: payload type collision with offer/answer <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; Hello, <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; I have encountered an issue, and I&#39;m not sure it&#39;s an issue or not. SDP <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; wisdom is sought. <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; Suppose two entities exchange the following offer/answer: <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; offer from X: <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; m=video 97 <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; a=rtpmap:97 foo <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; answer from Y: <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; m=video 98 <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; a=rtpmap:98 foo <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; It&#39;s clear from RFC 3264 that now X must send codec foo with payload <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; type 98, and Y must send codec foo with payload type 97. That&#39;s not <br>
&gt; &gt; the <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; question. <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; But suppose these are the offer and answer: <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; offer from X: <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; m=video 97 98 <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; a=rtpmap:97 foo <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; a=rtpmap:98 bar <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; answer from Y: <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; m=video 97 98 <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; a=rtpmap:97 bar <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; a=rtpmap:98 foo <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; That is, the payload types collide. <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; It would be possible to write code so that X sends foo with 98 and bar <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; with 97, while Y sends foo with 97 and bar with 98. But is it <br>
&gt; &gt; &gt; &gt; &gt;&gt; conformant <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; with the specs to do so? <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; And if this has a clear answer - which paragraph of which RFC makes <br>
&gt; &gt; &gt; &gt; &gt;&gt; this <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; clear? <br>
&gt; &gt; &gt; &gt; &gt;&gt;&gt; (Yes, this discussion is triggered from a real world problem.) <br>
&gt; &gt; &gt; &gt; &gt;&gt; <br>
&gt; &gt; &gt; &gt; &gt;&gt; It is recommended to avoid such mismatches when possible, but <br>
&gt; &gt; doing so <br>
&gt; &gt; &gt; &gt; &gt;&gt; may not always be possible. The examples you mention are <br>
&gt; &gt; conformant. <br>
&gt; &gt; &gt; &gt; &gt;&gt; The following text from section 5.1 addresses this: <br>
&gt; &gt; &gt; &gt; &gt;&gt; <br>
&gt; &gt; &gt; &gt; &gt;&gt; For sendrecv RTP <br>
&gt; &gt; &gt; &gt; &gt;&gt; streams, the payload type numbers indicate the value of the payload <br>
&gt; &gt; &gt; &gt; &gt;&gt; type field the offerer expects to receive, and would prefer to send. <br>
&gt; &gt; &gt; &gt; &gt;&gt; However, for sendonly and sendrecv streams, the answer might <br>
&gt; &gt; indicate <br>
&gt; &gt; &gt; &gt; &gt;&gt; different payload type numbers for the same codecs, in which case, <br>
&gt; &gt; &gt; &gt; &gt;&gt; the offerer MUST send with the payload type numbers from the <br>
&gt; &gt; answer. <br>
&gt; &gt; &gt; &gt; &gt;&gt; <br>
&gt; &gt; &gt; &gt; &gt;&gt; Different payload type numbers may be needed in each direction <br>
&gt; &gt; &gt; &gt; &gt;&gt; because of interoperability concerns with H.323. <br>
&gt; &gt; &gt; &gt; &gt; I was reading that text, and it clearly says which ones to use if the <br>
&gt; &gt; &gt; &gt; &gt; payload types are different, but I don&#39;t think they clearly address the <br>
&gt; &gt; &gt; &gt; &gt; situation with using the same payload type for two different codecs in <br>
&gt; &gt; &gt; &gt; &gt; different directions. <br>
&gt; &gt; &gt; &gt; &gt; <br>
&gt; &gt; &gt; &gt; &gt; In a multicast or transport relay scenario, this would clearly be <br>
&gt; &gt; &gt; &gt; &gt; untenable, since third parties would not be able to send streams that <br>
&gt; &gt; &gt; &gt; &gt; would be interpreted correctly by the two conflicting endpoints, but in <br>
&gt; &gt; &gt; &gt; &gt; a two-party conversation, it&#39;s possible to make this work. <br>
&gt; &gt; &gt; &gt; &gt; I just don&#39;t know if it&#39;s worth the effort to make it so. <br>
&gt; &gt; &gt; &gt; <br>
&gt; &gt; &gt; &gt; Well there&#39;s also section 8.3.2 in 3264. The section is about updating <br>
&gt; &gt; &gt; &gt; offers but it seems like the text explicitly prohibits remapping payload <br>
&gt; &gt; &gt; &gt; types in any scenarios: <br>
&gt; &gt; &gt; &gt; <br>
&gt; &gt; &gt; &gt; However, in the <br>
&gt; &gt; &gt; &gt; case of RTP, the mapping from a particular dynamic payload type <br>
&gt; &gt; &gt; &gt; number to a particular codec within that media stream MUST NOT change <br>
&gt; &gt; &gt; &gt; for the duration of a session. For example, if A generates an offer <br>
&gt; &gt; &gt; &gt; with G.711 assigned to dynamic payload type number 46, payload type <br>
&gt; &gt; &gt; &gt; number 46 MUST refer to G.711 from that point forward in any offers <br>
&gt; &gt; &gt; &gt; or answers for that media stream within the session. However, it is <br>
&gt; &gt; &gt; &gt; acceptable for multiple payload type numbers to be mapped to the same <br>
&gt; &gt; &gt; &gt; codec, so that an updated offer could also use payload type number 72 <br>
&gt; &gt; &gt; &gt; for G.711. <br>
&gt; &gt; &gt; <br>
&gt; &gt; &gt; I viewed this as being a requirement for all offers or answers sent by <br>
&gt; &gt; &gt; A to B, but not necessarily the corresponding answer and offers from <br>
&gt; &gt; &gt; B to A. <br>
&gt; &gt; <br>
&gt; &gt; Well, it does say: &quot;from that point forward in _any_ offers or answers&quot; <br>
&gt; &gt; <br>
&gt; &gt; &gt; Otherwise, it seems to conflict with the text in section 5.1. <br>
&gt; &gt; <br>
&gt; &gt; Mmm ... which part of 5.1? <br>
&gt;<br>
&gt; This part: <br>
&gt;<br>
&gt; However, for sendonly and sendrecv streams, the answer might indicate <br>
&gt; different payload type numbers for the same codecs, in which case, <br>
&gt; the offerer MUST send with the payload type numbers from the answer. <br>
&gt;<br>
&gt; Different payload type numbers may be needed in each direction <br>
&gt; because of interoperability concerns with H.323. </p>
<p dir="ltr">This part just says the PT numbers can be different in the offer and the answer. It doesn&#39;t say they can be the same as other PT numbers that were assigned to other formats, which is what section 8.3.2 prohibits. </p>

<p dir="ltr">This is the reason I was seeing 8.3.2 more as a completion rather than a contradiction.</p>
<p dir="ltr">Emil<br>
--sent from my mobile</p>
<p dir="ltr">&gt;<br>
&gt; Cheers, <br>
&gt; Charles <br>
&gt;<br>
&gt; &gt; To me it just seems like a completion (admittedly, somewhat out of place). <br>
&gt; &gt; <br>
&gt; &gt; &gt; That said, I can certainly see where someone else may interpret it <br>
&gt; &gt; &gt; differently. <br>
&gt; &gt; <br>
&gt; &gt; Agreed. Definitely 3264bis material. <br>
&gt; &gt; <br>
&gt; &gt; Cheers, <br>
&gt; &gt; Emil <br>
&gt; &gt; <br>
&gt; &gt; --sent from my mobile <br>
&gt; &gt; &gt; <br>
&gt; &gt; &gt; Cheers, <br>
&gt; &gt; &gt; Charles <br>
&gt; &gt; &gt; <br>
&gt; &gt; &gt; &gt; Cheers, <br>
&gt; &gt; &gt; &gt; Emil <br>
&gt; &gt; &gt; &gt; <br>
&gt; &gt; &gt; &gt; <br>
&gt; &gt; &gt; &gt; -- <br>
&gt; &gt; &gt; &gt; <a href="https://jitsi.org">https://jitsi.org</a> <br>
&gt; &gt; &gt; &gt; _______________________________________________ <br>
&gt; &gt; &gt; &gt; mmusic mailing list <br>
&gt; &gt; &gt; &gt; <a href="mailto:mmusic@ietf.org">mmusic@ietf.org</a> <br>
&gt; &gt; &gt; &gt; <a href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a> <br>
&gt; &gt; &gt; _______________________________________________ <br>
&gt; &gt; &gt; mmusic mailing list <br>
&gt; &gt; &gt; <a href="mailto:mmusic@ietf.org">mmusic@ietf.org</a> <br>
&gt; &gt; &gt; <a href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a> <br>
&gt; &gt; <br>
&gt;<br>
&gt; _______________________________________________ <br>
&gt; mmusic mailing list <br>
&gt; <a href="mailto:mmusic@ietf.org">mmusic@ietf.org</a> <br>
&gt; <a href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a> <br>
</p>

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Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] RESTART: SCTP question: Where does it multiplex?
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On 1/10/13 5:06 PM, Dan Wing wrote:
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
>> Sent: Thursday, January 03, 2013 11:23 AM
>> To: Dan Wing
>> Cc: mmusic@ietf.org
>> Subject: Re: [MMUSIC] RESTART: SCTP question: Where does it multiplex?
>>
>> I'm totally lost!
>> I looked back, and count 43 messages in this thread.
>> At this point I no longer understand the question(s), or what if
>> anything has been answered.
>>
>> Can we start over?
>> Sort out what the questions are, and then discuss whether we have
>> answers to them?
>> (Preferably in separate, well identified, threads.)
>>
>> I *think* Harald originally asserted that DTLS/SCTP *can* already be
>> multiplexed with SRTP audio and video streams.
>
> Yes, it can.
>
> An UDP packet is either a STUN packet (for ICE connectivity checks),
> DTLS, or SRTP.  This is depicted in Section 5.1.2 of RFC5764:
>
>                     +----------------+
>                     | 127 < B < 192 -+--> forward to RTP
>                     |                |
>         packet -->  |  19 < B < 64  -+--> forward to DTLS
>                     |                |
>                     |       B < 2   -+--> forward to STUN
>                     +----------------+
>
> A DTLS packet containing application_data is handed to the SCTP stack.
>
> As I have mentioned previously, I am concerned with sending both realtime
> interactive audio/video and non-realtime data over the same UDP 5-tuple,
> because it becomes difficult (or impossible) to apply QoS to differentiate
> those very different flows.  If everyone always had adequate bandwidth, that
> concern would vanish.  But the industry has never been able to stay ahead of
> the bandwidth curve.

This is all about policy, not mechanism. It seems that some people want 
to use this mechanism and think it will be good for them. Perhaps there 
can be a section in some draft on this subject that points out your 
concern and recommends that people consider carefully what they 
multiplex on the same 5-tuple.

Similar arguments may apply to bundling audio and video onto the same 
5-tuple.

>> And so he was asking
>> *how* this could be specified in SDP. (E.g. Does "bundle" support it?)
>
> We need a way for SDP to signal:
>    1. the UDP port for SCTP.  This feels like an SDP media ("m") line to me.
>    2. SCTP native (native SCTP over IPv6 should be tenable, whereas native
> SCTP over IPv4 is pretty much untenable),
>    3. the data carried over each SCTP port (e.g., port NNN for XMPP, port
> NNNN for file sharing, port NNNNN for screen sharing, and so on).  This
> feels like an SDP media ("m") line to me.

I don't disagree with any of your points, but you have missed the point 
I was making.

Everything you list applies for SCTP on its own 5-tuple. But then there 
is the question of how to describe how this shares a 5-tuple with some 
audio and video streams. The current candidates for representing the 
bundling of audio and video are the bundle draft and the mmt draft. 
Neither of these currently supports including DTLS/SCTP with audio or 
video. So if bundling in the SCTP is a requirement, then both bundle and 
mmt should be extended cover it if they are to remain in the running.

A *native* SCTP session (directly over IPv*) runs on a 5-tuple, 
including the *SCTP* ports at each end. So the SDP for that needs to 
specify a port. ISTM that in this case the port field on the m-line 
would signify the SCTP port. And this is what draft-ietf-mmusic-sctp-sdp 
says.

A DTLS/SCTP session requires a 7-tuple, including both an SCTP port and 
a UDP port at each end. So the SDP for that needs to specify both an 
SCTP port and a UDP port. Only one of those can be in the port field of 
the m-line. (Which one that should be can be argued either way.) 
draft-ietf-mmusic-sctp-sdp puts the UDP port in the m-line, and has no 
way to specify the SCTP port. (It has an open issue about that. 
Presumably it is either defaulted or must go in an attribute.)

> I don't think we can use media lines for both (1) and (3), so perhaps
> bundling is the answer.  I don't know.

Even *with* bundling the problem remains. But I agree that if you want 
to run SCTP on two or more SCTP ports over the same UDP port, then 
bundle is one way to do it. Each of the bundled m-lines would have to 
specify the SCTP port, presumably as an attribute.

>> Along the way, a question came up of how many DTLS/SCTP connections
>> could be multiplexed with each other and with SRTP media. I think Harald
>> assumed only one (and assumed there is only a need for one.) But reasons
>> for more have been discussed. I suggested that it appears that multiple
>> should be supportable, each with a different SCTP port.
>>
>> I think Dan raised some additional issues related to multipath support.
>
> SCTP can support multipath, so we need to decide if we allow SCTP to do its
> own multipath, or if we require ICE to first validate a (new) path before
> SCTP can try to do its multipath.

This is outside my area of expertise.

> The same question and issues exist for
> multipath RTP (draft-singh-avtcore-mprtp) and how multipath RTP should
> split its functions between ICE, ICE extensions, and SDP.  I know that
> multipath RTP is not a working group document, but it might be a good
> basis for analyzing multipath SCTP and reaching a conclusion on multipath
> SCTP that makes sense for both multipath SCTP and multipath RTP (and
> presumably multipath TCP, RFC6182 and its related documents).

If you want to multiplex SCTP and RTP on the same connection, and you 
are going to make SCTP multipath work for that, then why would you not 
just do RTP over an SCTP channel??? Then you don't need to reinvent how 
to multipath for RTP.

	Thanks,
	Paul

> -d
>
>
>> 	Thanks,
>> 	Paul
>>
>> On 1/2/13 10:32 PM, Dan Wing wrote:
>>>> -----Original Message-----
>>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
>>>> Behalf Of Paul Kyzivat
>>>> Sent: Tuesday, December 11, 2012 11:17 AM
>>>> To: Dan Wing
>>>> Cc: mmusic@ietf.org
>>>> Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
>>>>
>>>> Hi Dan,
>>>>
>>>> Comments inline
>>>>
>>>> On 12/11/12 1:53 PM, Dan Wing wrote:
>>>>>> -----Original Message-----
>>>>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
>>>>>> Behalf Of Paul Kyzivat
>>>>>> Sent: Tuesday, December 11, 2012 7:43 AM
>>>>>> To: mmusic@ietf.org
>>>>>> Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
>>>>>>
>>>>>> More inline.
>>>>>>
>>>>>> On 12/11/12 4:08 AM, Salvatore Loreto wrote:
>>>>>>> Hi Paul,
>>>>>>>
>>>>>>> see in line!
>>>>>>>
>>>>>>> On 12/3/12 10:00 PM, Paul Kyzivat wrote:
>>>>>>>> Commenting on a different point
>>>>>>>>
>>>>>>>> On 12/1/12 9:53 AM, Harald Alvestrand wrote:
>>>>>>>>
>>>>>>>>> The interesting difference is that the multiplexing between
>>>>>>>>> DTLS/SCTP traffic and BUNDLE multiplexing is that DTLS/SCTP
>>>>>>>>> traffic is not carried in SSRCs, which means:
>>>>>>>>>
>>>>>>>>> - There can be only one DTLS/SCTP stream in a bundle (which may
>>>>>>>>> have multiple associations, as you state below); you can't have
>>>>>>>>> multiple lines with proto DTLS/SCTP in a bundle.
>>>>>>>>
>>>>>>>> I am not an SCTP expert. But IIUC, SCTP was designed to run
>>>>>>>> directly over IP. It has its own notion of port used to demux
>>>>>>>> multiple SCTP associations over the same IP address.
>>>>>>>>
>>>>>>>> I presume that that same mechanism is still there when SCTP is
>>>>>>>> run over DTLS over UDP.
>>>>>>>>
>>>>>>>> So, the traffic coming over DTLS must first be demuxed into RTP
>>>>>>>> traffic and SCTP traffic.
>>>>>>> based on the current stack the SCTP traffic is the only traffic
>>>>>>> that runs directly over the DTLS stack.
>>>>>>
>>>>>> Yes, that is what I thought. But Harald has been asking about
>>>>>> multiplexing this with RTP traffic. (Actually I think it would be
>>>>>> DTLS/SRTP traffic that it would be multiplexed with.)
>>>>>>
>>>>>>> What I am trying to do is to include the Randell Jesup (I am
>>>>>>> including him in CC as I am not sure he is subscribed to this
>>>>>>> mailing list) suggestion to give the possibility to have multiple
>>>>>>> SCTP
>>>>>>> *associations* running on top of the same DTLS session and of
>>>>>>> course providing a way to signal it in SDP.
>>>>>>
>>>>>> IIUC, SCTP (having been designed as a transport layer protocol)
>>>>>> defines its own notion of port, and has fields in its protocol to
>>>>>> carry the local and remote port number. Presumably those fields are
>>>>>> still there when run over UDP or DTLS. So it should be possible to
>>>>>> support multiple SCTP associations over the same DTLS connection,
>>>>>> each distinguished by its own port pair.
>>>>>
>>>>> Yes, SCTP has its own notion of ports. How this works when SCTP is
>>>>> carried over UDP is not quite clear, especially because SCTP assumes
>>>>> a NAPT will not rewrite the SCTP port number (SCTP endeavors to make
>>>>> such port rewriting difficult). But of course a NAPT (and the MAP
>>>>> techniques) rewrite UDP port numbers. I believe SCTP would only be
>>>>> able to successfully convey its UDP port numbers for a device that
>>>>> is not behind a NAT (that is, a server sitting directly on the
>>>>> Internet, rather than a remote peer that is behind a NAT). Creative
>>>>> use of PCP, NAT-PMP, or UPnP IGD would improve that situation in the
>> future.
>>>>
>>>> I find what you say above confusing.
>>>>
>>>> SCTP running naked over IP would still have its own ports.
>>>> If that ran over a path with a NAPT, then I could see why the NAPT
>>>> might want to do the same as it does with UDP and TCP, and so replace
>>>> addr/port pairs to minimize addr usage. (That's if the NAPT supported
>>>> SCTP at all - I gather most don't.)
>>>
>>> As I mentioned earlier, SCTP does not allow its source port number to
>>> be changed.
>>>
>>>> But with SCTP over DTLS over UDP, there is a UDP port, and then
>>>> presumably a separate SCTP port carried in the SCTP protocol within
>>>> the UDP message.
>>>
>>> Yep, I assume that is how it works.  Effectively, UDP is just used to
>>> tunnel the SCTP.
>>>
>>>> An NAPT on the path would presumably be messing with the UDP port.
>>>> Given use of DTLS, the NAPT won't even be able to tell that SCTP is
>>>> being used, much less mess with the SCTP port.
>>>>
>>>>>> That of course depends on having a signaling mechanism to set it
>> up.
>>>>>
>>>>> After the initial SCTP association is created, SCTP could be allowed
>>>>> to do its own thing (that is, chose to find and use other ports).
>>>>
>>>> Here you are talking about finding and using other UDP addr/port
>> pairs?
>>>
>>> No, I was referring to the SCTP ports.
>>>
>>> -d
>>>
>>>
>>>> That is explicitly excluded in the DTLS/SCTP mapping.
>>>>
>>>> Thanks,
>>>> Paul
>>>>
>>>>> SCTP's behavior in this regard is similar to MPRTP (multipath RTP),
>>>>> but of course they are not identical.
>>>>>
>>>>>>> to be clear: at moment WebRTC allows only one SCTP association per
>>>>>>> PC, so this is something that would be nice to define just to be
>>>>>>> ready for the future.
>>>>>>
>>>>>> AFAIK WebRTC is just one possible user of this mechanism. The SDP
>>>>>> mechanism shouldn't be limited by the constraints of WebRTC. It
>>>>>> would be very difficult to define the SDP so that it was impossible
>>>>>> to set up multiple SCTP associations over different 5-tuples.
>>>>>>
>>>>>>>> Then the RTP traffic can be demuxed based on SSRC, and SCTP
>>>>>>>> traffic can be demuxed based on SCTP port. And once the traffic
>>>>>>>> for a single SCTP port is identified, it can be demuxed based on
>>>>>>>> stream
>>>> number.
>>>>>>>>
>>>>>>>> Representing this in SDP is a challenge. Some variant of the
>>>>>>>> bundle proposal might allow bundling together several RTP m-lines
>>>>>>>> and some DTLS/SCTP m-lines. This would require a mechanism for
>>>>>>>> specifying the SCTP port number - already an open issue (#3) in
>>>>>>>> draft-ietf-mmusic-sctp-sdp-02.
>>>>>>> I agree that it is a challange how then to bundle everything
>>>>>>> together
>>>>>>
>>>>>> But it is a challenge that needs to be tackled if we are to realize
>>>>>> Harald's dream.
>>>>>
>>>>> -d
>>>>>
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> mmusic mailing list
>>>> mmusic@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>
>>>
>>>
>
>


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From: "Dan Wing" <dwing@cisco.com>
To: "'Paul Kyzivat'" <pkyzivat@alum.mit.edu>
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> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
> Sent: Thursday, January 10, 2013 4:02 PM
> To: Dan Wing
> Cc: mmusic@ietf.org
> Subject: Re: [MMUSIC] RESTART: SCTP question: Where does it multiplex?
> 
> On 1/10/13 5:06 PM, Dan Wing wrote:
> >> -----Original Message-----
> >> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
> >> Sent: Thursday, January 03, 2013 11:23 AM
> >> To: Dan Wing
> >> Cc: mmusic@ietf.org
> >> Subject: Re: [MMUSIC] RESTART: SCTP question: Where does it
> multiplex?
> >>
> >> I'm totally lost!
> >> I looked back, and count 43 messages in this thread.
> >> At this point I no longer understand the question(s), or what if
> >> anything has been answered.
> >>
> >> Can we start over?
> >> Sort out what the questions are, and then discuss whether we have
> >> answers to them?
> >> (Preferably in separate, well identified, threads.)
> >>
> >> I *think* Harald originally asserted that DTLS/SCTP *can* already be
> >> multiplexed with SRTP audio and video streams.
> >
> > Yes, it can.
> >
> > An UDP packet is either a STUN packet (for ICE connectivity checks),
> > DTLS, or SRTP.  This is depicted in Section 5.1.2 of RFC5764:
> >
> >                     +----------------+
> >                     | 127 < B < 192 -+--> forward to RTP
> >                     |                |
> >         packet -->  |  19 < B < 64  -+--> forward to DTLS
> >                     |                |
> >                     |       B < 2   -+--> forward to STUN
> >                     +----------------+
> >
> > A DTLS packet containing application_data is handed to the SCTP stack.
> >
> > As I have mentioned previously, I am concerned with sending both
> > realtime interactive audio/video and non-realtime data over the same
> > UDP 5-tuple, because it becomes difficult (or impossible) to apply QoS
> > to differentiate those very different flows.  If everyone always had
> > adequate bandwidth, that concern would vanish.  But the industry has
> > never been able to stay ahead of the bandwidth curve.
> 
> This is all about policy, not mechanism. It seems that some people want
> to use this mechanism and think it will be good for them. Perhaps there
> can be a section in some draft on this subject that points out your
> concern and recommends that people consider carefully what they
> multiplex on the same 5-tuple.
> 
> Similar arguments may apply to bundling audio and video onto the same 5-
> tuple.

I have been told there is a 3GPP document that might be worth citing
on that point.  I'll get a pointer soon.  I expect there is a long-standing
RFC on that point, too, but I don't know of one.  


> >> And so he was asking
> >> *how* this could be specified in SDP. (E.g. Does "bundle" support
> >> it?)
> >
> > We need a way for SDP to signal:
> >    1. the UDP port for SCTP.  This feels like an SDP media ("m") line
> to me.
> >    2. SCTP native (native SCTP over IPv6 should be tenable, whereas
> > native SCTP over IPv4 is pretty much untenable),
> >    3. the data carried over each SCTP port (e.g., port NNN for XMPP,
> > port NNNN for file sharing, port NNNNN for screen sharing, and so on).
> > This feels like an SDP media ("m") line to me.
> 
> I don't disagree with any of your points, but you have missed the point
> I was making.
> 
> Everything you list applies for SCTP on its own 5-tuple. But then there
> is the question of how to describe how this shares a 5-tuple with some
> audio and video streams.

(1), from my list, is the UDP port.  This could be the same UDP port
as the audio/video media.

> The current candidates for representing the
> bundling of audio and video are the bundle draft and the mmt draft.
> Neither of these currently supports including DTLS/SCTP with audio or
> video. So if bundling in the SCTP is a requirement, then both bundle and
> mmt should be extended cover it if they are to remain in the running.

Yep.

> A *native* SCTP session (directly over IPv*) runs on a 5-tuple,
> including the *SCTP* ports at each end. So the SDP for that needs to
> specify a port. 
>
> ISTM that in this case the port field on the m-line
> would signify the SCTP port. And this is what draft-ietf-mmusic-sctp-sdp
> says.
>
> A DTLS/SCTP session requires a 7-tuple, including both an SCTP port and
> a UDP port at each end. So the SDP for that needs to specify both an
> SCTP port and a UDP port. Only one of those can be in the port field of
> the m-line. (Which one that should be can be argued either way.) draft-
> ietf-mmusic-sctp-sdp puts the UDP port in the m-line, and has no way to
> specify the SCTP port. (It has an open issue about that.
> Presumably it is either defaulted or must go in an attribute.)

That's a good summary of the current state of the art.

> > I don't think we can use media lines for both (1) and (3), so perhaps
> > bundling is the answer.  I don't know.
> 
> Even *with* bundling the problem remains. But I agree that if you want
> to run SCTP on two or more SCTP ports over the same UDP port, then
> bundle is one way to do it. Each of the bundled m-lines would have to
> specify the SCTP port, presumably as an attribute.
> 
> >> Along the way, a question came up of how many DTLS/SCTP connections
> >> could be multiplexed with each other and with SRTP media. I think
> >> Harald assumed only one (and assumed there is only a need for one.)
> >> But reasons for more have been discussed. I suggested that it appears
> >> that multiple should be supportable, each with a different SCTP port.
> >>
> >> I think Dan raised some additional issues related to multipath
> support.
> >
> > SCTP can support multipath, so we need to decide if we allow SCTP to
> > do its own multipath, or if we require ICE to first validate a (new)
> > path before SCTP can try to do its multipath.
> 
> This is outside my area of expertise.
> 
> > The same question and issues exist for multipath RTP
> > (draft-singh-avtcore-mprtp) and how multipath RTP should split its
> > functions between ICE, ICE extensions, and SDP.  I know that multipath
> > RTP is not a working group document, but it might be a good basis for
> > analyzing multipath SCTP and reaching a conclusion on multipath SCTP
> > that makes sense for both multipath SCTP and multipath RTP (and
> > presumably multipath TCP, RFC6182 and its related documents).
> 
> If you want to multiplex SCTP and RTP on the same connection, and you
> are going to make SCTP multipath work for that, then why would you not
> just do RTP over an SCTP channel??? Then you don't need to reinvent how
> to multipath for RTP.

I guess to support remote endpoints that lack SCTP, yet want to do 
multipath RTP.

-d

> 	Thanks,
> 	Paul
> 
> > -d
> >
> >
> >> 	Thanks,
> >> 	Paul
> >>
> >> On 1/2/13 10:32 PM, Dan Wing wrote:
> >>>> -----Original Message-----
> >>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> >>>> Behalf Of Paul Kyzivat
> >>>> Sent: Tuesday, December 11, 2012 11:17 AM
> >>>> To: Dan Wing
> >>>> Cc: mmusic@ietf.org
> >>>> Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
> >>>>
> >>>> Hi Dan,
> >>>>
> >>>> Comments inline
> >>>>
> >>>> On 12/11/12 1:53 PM, Dan Wing wrote:
> >>>>>> -----Original Message-----
> >>>>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> >>>>>> Behalf Of Paul Kyzivat
> >>>>>> Sent: Tuesday, December 11, 2012 7:43 AM
> >>>>>> To: mmusic@ietf.org
> >>>>>> Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
> >>>>>>
> >>>>>> More inline.
> >>>>>>
> >>>>>> On 12/11/12 4:08 AM, Salvatore Loreto wrote:
> >>>>>>> Hi Paul,
> >>>>>>>
> >>>>>>> see in line!
> >>>>>>>
> >>>>>>> On 12/3/12 10:00 PM, Paul Kyzivat wrote:
> >>>>>>>> Commenting on a different point
> >>>>>>>>
> >>>>>>>> On 12/1/12 9:53 AM, Harald Alvestrand wrote:
> >>>>>>>>
> >>>>>>>>> The interesting difference is that the multiplexing between
> >>>>>>>>> DTLS/SCTP traffic and BUNDLE multiplexing is that DTLS/SCTP
> >>>>>>>>> traffic is not carried in SSRCs, which means:
> >>>>>>>>>
> >>>>>>>>> - There can be only one DTLS/SCTP stream in a bundle (which
> >>>>>>>>> may have multiple associations, as you state below); you can't
> >>>>>>>>> have multiple lines with proto DTLS/SCTP in a bundle.
> >>>>>>>>
> >>>>>>>> I am not an SCTP expert. But IIUC, SCTP was designed to run
> >>>>>>>> directly over IP. It has its own notion of port used to demux
> >>>>>>>> multiple SCTP associations over the same IP address.
> >>>>>>>>
> >>>>>>>> I presume that that same mechanism is still there when SCTP is
> >>>>>>>> run over DTLS over UDP.
> >>>>>>>>
> >>>>>>>> So, the traffic coming over DTLS must first be demuxed into RTP
> >>>>>>>> traffic and SCTP traffic.
> >>>>>>> based on the current stack the SCTP traffic is the only traffic
> >>>>>>> that runs directly over the DTLS stack.
> >>>>>>
> >>>>>> Yes, that is what I thought. But Harald has been asking about
> >>>>>> multiplexing this with RTP traffic. (Actually I think it would be
> >>>>>> DTLS/SRTP traffic that it would be multiplexed with.)
> >>>>>>
> >>>>>>> What I am trying to do is to include the Randell Jesup (I am
> >>>>>>> including him in CC as I am not sure he is subscribed to this
> >>>>>>> mailing list) suggestion to give the possibility to have
> >>>>>>> multiple SCTP
> >>>>>>> *associations* running on top of the same DTLS session and of
> >>>>>>> course providing a way to signal it in SDP.
> >>>>>>
> >>>>>> IIUC, SCTP (having been designed as a transport layer protocol)
> >>>>>> defines its own notion of port, and has fields in its protocol to
> >>>>>> carry the local and remote port number. Presumably those fields
> >>>>>> are still there when run over UDP or DTLS. So it should be
> >>>>>> possible to support multiple SCTP associations over the same DTLS
> >>>>>> connection, each distinguished by its own port pair.
> >>>>>
> >>>>> Yes, SCTP has its own notion of ports. How this works when SCTP is
> >>>>> carried over UDP is not quite clear, especially because SCTP
> >>>>> assumes a NAPT will not rewrite the SCTP port number (SCTP
> >>>>> endeavors to make such port rewriting difficult). But of course a
> >>>>> NAPT (and the MAP
> >>>>> techniques) rewrite UDP port numbers. I believe SCTP would only be
> >>>>> able to successfully convey its UDP port numbers for a device that
> >>>>> is not behind a NAT (that is, a server sitting directly on the
> >>>>> Internet, rather than a remote peer that is behind a NAT).
> >>>>> Creative use of PCP, NAT-PMP, or UPnP IGD would improve that
> >>>>> situation in the
> >> future.
> >>>>
> >>>> I find what you say above confusing.
> >>>>
> >>>> SCTP running naked over IP would still have its own ports.
> >>>> If that ran over a path with a NAPT, then I could see why the NAPT
> >>>> might want to do the same as it does with UDP and TCP, and so
> >>>> replace addr/port pairs to minimize addr usage. (That's if the NAPT
> >>>> supported SCTP at all - I gather most don't.)
> >>>
> >>> As I mentioned earlier, SCTP does not allow its source port number
> >>> to be changed.
> >>>
> >>>> But with SCTP over DTLS over UDP, there is a UDP port, and then
> >>>> presumably a separate SCTP port carried in the SCTP protocol within
> >>>> the UDP message.
> >>>
> >>> Yep, I assume that is how it works.  Effectively, UDP is just used
> >>> to tunnel the SCTP.
> >>>
> >>>> An NAPT on the path would presumably be messing with the UDP port.
> >>>> Given use of DTLS, the NAPT won't even be able to tell that SCTP is
> >>>> being used, much less mess with the SCTP port.
> >>>>
> >>>>>> That of course depends on having a signaling mechanism to set it
> >> up.
> >>>>>
> >>>>> After the initial SCTP association is created, SCTP could be
> >>>>> allowed to do its own thing (that is, chose to find and use other
> ports).
> >>>>
> >>>> Here you are talking about finding and using other UDP addr/port
> >> pairs?
> >>>
> >>> No, I was referring to the SCTP ports.
> >>>
> >>> -d
> >>>
> >>>
> >>>> That is explicitly excluded in the DTLS/SCTP mapping.
> >>>>
> >>>> Thanks,
> >>>> Paul
> >>>>
> >>>>> SCTP's behavior in this regard is similar to MPRTP (multipath
> >>>>> RTP), but of course they are not identical.
> >>>>>
> >>>>>>> to be clear: at moment WebRTC allows only one SCTP association
> >>>>>>> per PC, so this is something that would be nice to define just
> >>>>>>> to be ready for the future.
> >>>>>>
> >>>>>> AFAIK WebRTC is just one possible user of this mechanism. The SDP
> >>>>>> mechanism shouldn't be limited by the constraints of WebRTC. It
> >>>>>> would be very difficult to define the SDP so that it was
> >>>>>> impossible to set up multiple SCTP associations over different 5-
> tuples.
> >>>>>>
> >>>>>>>> Then the RTP traffic can be demuxed based on SSRC, and SCTP
> >>>>>>>> traffic can be demuxed based on SCTP port. And once the traffic
> >>>>>>>> for a single SCTP port is identified, it can be demuxed based
> >>>>>>>> on stream
> >>>> number.
> >>>>>>>>
> >>>>>>>> Representing this in SDP is a challenge. Some variant of the
> >>>>>>>> bundle proposal might allow bundling together several RTP
> >>>>>>>> m-lines and some DTLS/SCTP m-lines. This would require a
> >>>>>>>> mechanism for specifying the SCTP port number - already an open
> >>>>>>>> issue (#3) in draft-ietf-mmusic-sctp-sdp-02.
> >>>>>>> I agree that it is a challange how then to bundle everything
> >>>>>>> together
> >>>>>>
> >>>>>> But it is a challenge that needs to be tackled if we are to
> >>>>>> realize Harald's dream.
> >>>>>
> >>>>> -d
> >>>>>
> >>>>>
> >>>>>
> >>>>
> >>>> _______________________________________________
> >>>> mmusic mailing list
> >>>> mmusic@ietf.org
> >>>> https://www.ietf.org/mailman/listinfo/mmusic
> >>>
> >>>
> >>>
> >
> >



From gonzalo.camarillo@ericsson.com  Fri Jan 11 03:28:03 2013
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: [MMUSIC] AD review: draft-ietf-mmusic-sdp-cs-16
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Hi,

I just got a publication request for the following draft:

http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-16

Please, find below my AD review.

Cheers,

Gonzalo


draft-ietf-mmusic-sdp-cs-16:

Page 10: remove the indentation of the following paragraph. Indented
paragraphs are typically used to provide clarifications while this one
contains normative language:

      Note that <addrtype> and/or <connection-address> MUST NOT be
      omitted when unknown since this would violate basic syntax of SDP
      [RFC4566].  In such cases, they MUST be set to a "-".

Section 5.2.3.1 provides a "format" for the cs-correlation attribute.
However, its ABNF formal syntax is in Section 5.7. So, why does Section
5.2.3.1 also contain a syntax for the attribute... and what type of
syntax is that?

Page 14: the text says:

      To mitigate this problem implementations
      should consider only some of the rightmost digits from the E.164

Can we provide a more concrete advice (or at least some discussion) on
the number of digits to consider?

Page 15: we define how to encode a binary value in hexadecimal format
and represent it as a two-digit ASCII value. Can we point to an existing
specification that defines how to do that instead of specifying it inline?

Page 17: the text says:

      Implementations are advised to select a number of DTMF digits that
      provide enough assurance that the call is related, but on the
      other hand do not prolong the bearer setup time unnecessarily.

Can we provide a more concrete recommendation?

Page 17: Section 5.2.3.4 specifies what to do when the correlation
fails. However, that is described in general in Section 5.3.3. Why don't
we move all the statements about how to decide whether the sessions are
correlated to Section 5.3.3? Now, the following statement contradicts
what Section 5.3.3 says, for example:

   the passive side
   SHOULD treat the circuit-switched bearer as not correlated to the
   ongoing session.

The indented text right after the statement above talks about
suppressing alerting. Such a supression is applicable to any session,
not only those that are correlated using DTMF tones. So, we could also
move that discussion to Section 5.3.3.

Section 5.5: rephrase the following sentence (possibly splitting it into
more than one sentence) so that it is clearer:

      Note that this may result in the recipient of the initial Offer in
      rejecting the Offer if the recipient of the Offer is not aware of
      its own E.164 number, and thus concluding that it will not be
      possible to establish a circuit-switched bearer since neither
      party is aware of their E.164 number.





From gonzalo.camarillo@ericsson.com  Fri Jan 11 03:50:48 2013
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: [MMUSIC] AD review: draft-ietf-mmusic-sdp-cs-16
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Hi,

I just got a publication request for the following draft:

http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-16

Please, find below my AD review.

Cheers,

Gonzalo


draft-ietf-mmusic-sdp-cs-16:

Page 10: remove the indentation of the following paragraph. Indented
paragraphs are typically used to provide clarifications while this one
contains normative language:

      Note that <addrtype> and/or <connection-address> MUST NOT be
      omitted when unknown since this would violate basic syntax of SDP
      [RFC4566].  In such cases, they MUST be set to a "-".

Section 5.2.3.1 provides a "format" for the cs-correlation attribute.
However, its ABNF formal syntax is in Section 5.7. So, why does Section
5.2.3.1 also contain a syntax for the attribute... and what type of
syntax is that?

Page 14: the text says:

      To mitigate this problem implementations
      should consider only some of the rightmost digits from the E.164

Can we provide a more concrete advice (or at least some discussion) on
the number of digits to consider?

Page 15: we define how to encode a binary value in hexadecimal format
and represent it as a two-digit ASCII value. Can we point to an existing
specification that defines how to do that instead of specifying it inline?

Page 17: the text says:

      Implementations are advised to select a number of DTMF digits that
      provide enough assurance that the call is related, but on the
      other hand do not prolong the bearer setup time unnecessarily.

Can we provide a more concrete recommendation?

Page 17: Section 5.2.3.4 specifies what to do when the correlation
fails. However, that is described in general in Section 5.3.3. Why don't
we move all the statements about how to decide whether the sessions are
correlated to Section 5.3.3? Now, the following statement contradicts
what Section 5.3.3 says, for example:

   the passive side
   SHOULD treat the circuit-switched bearer as not correlated to the
   ongoing session.

The indented text right after the statement above talks about
suppressing alerting. Such a supression is applicable to any session,
not only those that are correlated using DTMF tones. So, we could also
move that discussion to Section 5.3.3.

Section 5.5: rephrase the following sentence (possibly splitting it into
more than one sentence) so that it is clearer:

      Note that this may result in the recipient of the initial Offer in
      rejecting the Offer if the recipient of the Offer is not aware of
      its own E.164 number, and thus concluding that it will not be
      possible to establish a circuit-switched bearer since neither
      party is aware of their E.164 number.





From gonzalo.camarillo@ericsson.com  Fri Jan 11 03:52:20 2013
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Date: Fri, 11 Jan 2013 13:52:14 +0200
From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: Re: [MMUSIC] AD review: draft-ietf-mmusic-sdp-cs-16
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Sorry for the duplication. I had an email problem when sending the email
below and it seems to have made it to the list twice.

Gonzalo

On 11/01/2013 1:50 PM, Gonzalo Camarillo wrote:
> Hi,
> 
> I just got a publication request for the following draft:
> 
> http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-16
> 
> Please, find below my AD review.
> 
> Cheers,
> 
> Gonzalo
> 
> 
> draft-ietf-mmusic-sdp-cs-16:
> 
> Page 10: remove the indentation of the following paragraph. Indented
> paragraphs are typically used to provide clarifications while this one
> contains normative language:
> 
>       Note that <addrtype> and/or <connection-address> MUST NOT be
>       omitted when unknown since this would violate basic syntax of SDP
>       [RFC4566].  In such cases, they MUST be set to a "-".
> 
> Section 5.2.3.1 provides a "format" for the cs-correlation attribute.
> However, its ABNF formal syntax is in Section 5.7. So, why does Section
> 5.2.3.1 also contain a syntax for the attribute... and what type of
> syntax is that?
> 
> Page 14: the text says:
> 
>       To mitigate this problem implementations
>       should consider only some of the rightmost digits from the E.164
> 
> Can we provide a more concrete advice (or at least some discussion) on
> the number of digits to consider?
> 
> Page 15: we define how to encode a binary value in hexadecimal format
> and represent it as a two-digit ASCII value. Can we point to an existing
> specification that defines how to do that instead of specifying it inline?
> 
> Page 17: the text says:
> 
>       Implementations are advised to select a number of DTMF digits that
>       provide enough assurance that the call is related, but on the
>       other hand do not prolong the bearer setup time unnecessarily.
> 
> Can we provide a more concrete recommendation?
> 
> Page 17: Section 5.2.3.4 specifies what to do when the correlation
> fails. However, that is described in general in Section 5.3.3. Why don't
> we move all the statements about how to decide whether the sessions are
> correlated to Section 5.3.3? Now, the following statement contradicts
> what Section 5.3.3 says, for example:
> 
>    the passive side
>    SHOULD treat the circuit-switched bearer as not correlated to the
>    ongoing session.
> 
> The indented text right after the statement above talks about
> suppressing alerting. Such a supression is applicable to any session,
> not only those that are correlated using DTMF tones. So, we could also
> move that discussion to Section 5.3.3.
> 
> Section 5.5: rephrase the following sentence (possibly splitting it into
> more than one sentence) so that it is clearer:
> 
>       Note that this may result in the recipient of the initial Offer in
>       rejecting the Offer if the recipient of the Offer is not aware of
>       its own E.164 number, and thus concluding that it will not be
>       possible to establish a circuit-switched bearer since neither
>       party is aware of their E.164 number.
> 
> 
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
> 
> 


From keith.drage@alcatel-lucent.com  Fri Jan 11 04:33:55 2013
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Subject: Re: [MMUSIC] Query: payload type collision with offer/answer
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At least to attempt to answer the history question here relating to multica=
st, my understanding is that the authors of RFC 2543 (the original SIP), di=
d make attempts to cover multicast, but this was abandoned (although existi=
ng RFC 2543 material was not removed) when RFC 3261 was produced.

RFC 3264 was a new document at the time RFC 3261 was produced and therefore=
 my understanding was that it was designed to cover unicast only, because S=
IP, as defined in RFC 3261, was the user of offer/answer.

Keith

> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf
> Of Paul Kyzivat
> Sent: 10 January 2013 18:51
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] Query: payload type collision with offer/answer
>=20
> To the best of my memory the issue Harald has raised is a new one. I
> don't think it has never been discussed on any of the lists I have been
> following since 2002.
>=20
> More inline.
>=20
> On 1/9/13 6:46 PM, Emil Ivov wrote:
> >
> >
> > On 09.01.13, 23:52, Harald Alvestrand wrote:
> >> On 01/09/2013 10:01 PM, Charles Eckel (eckelcu) wrote:
> >>> Please see inline.
> >>>
> >>>> -----Original Message-----
> >>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> >>>> Behalf Of Harald Alvestrand
> >>>> Sent: Wednesday, January 09, 2013 11:25 AM
> >>>> To: mmusic@ietf.org
> >>>> Subject: [MMUSIC] Query: payload type collision with offer/answer
> >>>>
> >>>> Hello,
> >>>> I have encountered an issue, and I'm not sure it's an issue or not.
> SDP
> >>>> wisdom is sought.
> >>>>
> >>>> Suppose two entities exchange the following offer/answer:
> >>>>
> >>>> offer from X:
> >>>>
> >>>> m=3Dvideo 97
> >>>> a=3Drtpmap:97 foo
> >>>>
> >>>> answer from Y:
> >>>>
> >>>> m=3Dvideo 98
> >>>> a=3Drtpmap:98 foo
> >>>>
> >>>> It's clear from RFC 3264 that now X must send codec foo with payload
> >>>> type 98, and Y must send codec foo with payload type 97. That's not
> the
> >>>> question.
> >>>>
> >>>> But suppose these are the offer and answer:
> >>>>
> >>>> offer from X:
> >>>>
> >>>> m=3Dvideo 97 98
> >>>> a=3Drtpmap:97 foo
> >>>> a=3Drtpmap:98 bar
> >>>>
> >>>> answer from Y:
> >>>>
> >>>> m=3Dvideo 97 98
> >>>> a=3Drtpmap:97 bar
> >>>> a=3Drtpmap:98 foo
> >>>>
> >>>> That is, the payload types collide.
> >>>>
> >>>> It would be possible to write code so that X sends foo with 98 and
> bar
> >>>> with 97, while Y sends foo with 97 and bar with 98. But is it
> >>> conformant
> >>>> with the specs to do so?
> >>>>
> >>>> And if this has a clear answer - which paragraph of which RFC makes
> >>> this
> >>>> clear?
> >>>> (Yes, this discussion is triggered from a real world problem.)
> >>>
> >>> It is recommended to avoid such mismatches when possible, but doing s=
o
> >>> may not always be possible. The examples you mention are conformant.
> >>> The following text from section 5.1 addresses this:
> >>>
> >>> For sendrecv RTP
> >>> streams, the payload type numbers indicate the value of the payload
> >>> type field the offerer expects to receive, and would prefer to send.
> >>> However, for sendonly and sendrecv streams, the answer might indicate
> >>> different payload type numbers for the same codecs, in which case,
> >>> the offerer MUST send with the payload type numbers from the answer.
> >>>
> >>> Different payload type numbers may be needed in each direction
> >>> because of interoperability concerns with H.323.
> >> I was reading that text, and it clearly says which ones to use if the
> >> payload types are different, but I don't think they clearly address th=
e
> >> situation with using the same payload type for two different codecs in
> >> different directions.
> >>
> >> In a multicast or transport relay scenario, this would clearly be
> >> untenable, since third parties would not be able to send streams that
> >> would be interpreted correctly by the two conflicting endpoints, but i=
n
> >> a two-party conversation, it's possible to make this work.
> >> I just don't know if it's worth the effort to make it so.
>=20
> You are right that these arguments work differently for unicast and
> multicast. I don't recall (I think it was before my time) how much
> thought was given to multicast in 3264. (It doesn't seem to be very
> much. And I think the O/A discussions in new additions to SDP rarely
> consider it at all.)
>=20
> > Well there's also section 8.3.2 in 3264. The section is about updating
> > offers but it seems like the text explicitly prohibits remapping payloa=
d
> > types in any scenarios:
> >
> >     However, in the
> >     case of RTP, the mapping from a particular dynamic payload type
> >     number to a particular codec within that media stream MUST NOT
> change
> >     for the duration of a session.  For example, if A generates an offe=
r
> >     with G.711 assigned to dynamic payload type number 46, payload type
> >     number 46 MUST refer to G.711 from that point forward in any offers
> >     or answers for that media stream within the session.  However, it i=
s
> >     acceptable for multiple payload type numbers to be mapped to the
> same
> >     codec, so that an updated offer could also use payload type number
> 72
> >     for G.711.
>=20
> The language above is underspecified.
>=20
> It is my impression that the intent of this language was to ensure there
> is no ambiguity in the meaning of a payload type. Since the exchange of
> the SDP is not synchronized with the flow of RTP, if you were to send a
> new SDP that changed the mapping for a payload type, then the result
> would be almost certainly to interpret some media packets according to
> the wrong mapping.
>=20
> But this argument is only about packets flowing in one direction. It
> isn't a valid argument for requiring the payload types to be consistent
> in the two directions. And its not clear to me that it even intended
> that this language apply to both directions. It could be clarified as
> follows:
>=20
>                                ...  For example, if A generates an offer
>     with G.711 assigned to dynamic payload type number 46, payload type
>     number 46 MUST refer to G.711 from that point forward in any offers
>     or answers ***by A*** for that media stream within the session.  ...
>=20
> Interpreting it to apply to both directions brings it in conflict with
> the language elsewhere that permits it to be different in the two
> directions.
>=20
> The use of different mappings in each direction has been discussed over
> the years, though I couldn't point you to where. IIRC one of the reasons
> had to do with gatewaying to other protocols. But I can't offer any
> details.
>=20
> Bottom line: IMO, in the weird case Harald described X ought to follow
> Postel's Law and accept the answer from Y. But for the same reason I
> would strongly discourage implementations from acting as Y did.
>=20
> 	Thanks,
> 	Paul
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

From harald@alvestrand.no  Fri Jan 11 04:47:17 2013
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Subject: Re: [MMUSIC] Query: payload type collision with offer/answer
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On 01/10/2013 07:50 PM, Paul Kyzivat wrote:
>
>> Well there's also section 8.3.2 in 3264. The section is about updating
>> offers but it seems like the text explicitly prohibits remapping payload
>> types in any scenarios:
>>
>>     However, in the
>>     case of RTP, the mapping from a particular dynamic payload type
>>     number to a particular codec within that media stream MUST NOT 
>> change
>>     for the duration of a session.  For example, if A generates an offer
>>     with G.711 assigned to dynamic payload type number 46, payload type
>>     number 46 MUST refer to G.711 from that point forward in any offers
>>     or answers for that media stream within the session. However, it is
>>     acceptable for multiple payload type numbers to be mapped to the 
>> same
>>     codec, so that an updated offer could also use payload type 
>> number 72
>>     for G.711.
>
> The language above is underspecified.
>
> It is my impression that the intent of this language was to ensure 
> there is no ambiguity in the meaning of a payload type. Since the 
> exchange of the SDP is not synchronized with the flow of RTP, if you 
> were to send a new SDP that changed the mapping for a payload type, 
> then the result would be almost certainly to interpret some media 
> packets according to the wrong mapping.
>
> But this argument is only about packets flowing in one direction. It 
> isn't a valid argument for requiring the payload types to be 
> consistent in the two directions. And its not clear to me that it even 
> intended that this language apply to both directions. It could be 
> clarified as follows:
>
>                               ...  For example, if A generates an offer
>    with G.711 assigned to dynamic payload type number 46, payload type
>    number 46 MUST refer to G.711 from that point forward in any offers
>    or answers ***by A*** for that media stream within the session.  ...
>
> Interpreting it to apply to both directions brings it in conflict with 
> the language elsewhere that permits it to be different in the two 
> directions.
More than one number for one codec is explicitly permitted.
More than one codec for one number in a single direction is clearly 
impossible.
More than one codec for one number in a single RTP session..... that's 
the question.

If one were to go with textual analysis, this would hinge on the meaning 
of "for that media stream" - which again hinges on whether a media 
stream is unidirectional or bidirectional.

RFC 3264's definition:

       Media Stream: From RTSP [8], a media stream is a single media
          instance, e.g., an audio stream or a video stream as well as a
          single whiteboard or shared application group.  In SDP, a media
          stream is described by an "m=" line and its associated
          attributes.

which points back to RFC 2326:

    (Media) stream:
           A single media instance, e.g., an audio stream or a video
           stream as well as a single whiteboard or shared application
           group. When using RTP, a stream consists of all RTP and RTCP
           packets created by a source within an RTP session. This is
           equivalent to the definition of a DSM-CC stream([5]).

So if we believe these definitions, there's no prohibition on reusing 
the payload type.
Definitely nice to clarify in a 3264bis.

>
> The use of different mappings in each direction has been discussed 
> over the years, though I couldn't point you to where. IIRC one of the 
> reasons had to do with gatewaying to other protocols. But I can't 
> offer any details.
>
> Bottom line: IMO, in the weird case Harald described X ought to follow 
> Postel's Law and accept the answer from Y. But for the same reason I 
> would strongly discourage implementations from acting as Y did.
>
Seems like a conclusion.


From christer.holmberg@ericsson.com  Fri Jan 11 05:01:05 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Query: payload type collision with offer/answer
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Subject: Re: [MMUSIC] Query: payload type collision with offer/answer
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Hi,

My understanding of the section 8.3.2 is that the mapping must not change P=
ER DIRECTION, and the text "in any offers or answers" refers to a particula=
r participant.

Regards,

Christer


-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 Harald Alvestrand
Sent: 11. tammikuuta 2013 14:47
To: mmusic@ietf.org
Subject: Re: [MMUSIC] Query: payload type collision with offer/answer

On 01/10/2013 07:50 PM, Paul Kyzivat wrote:
>
>> Well there's also section 8.3.2 in 3264. The section is about=20
>> updating offers but it seems like the text explicitly prohibits=20
>> remapping payload types in any scenarios:
>>
>>     However, in the
>>     case of RTP, the mapping from a particular dynamic payload type
>>     number to a particular codec within that media stream MUST NOT=20
>> change
>>     for the duration of a session.  For example, if A generates an offer
>>     with G.711 assigned to dynamic payload type number 46, payload type
>>     number 46 MUST refer to G.711 from that point forward in any offers
>>     or answers for that media stream within the session. However, it is
>>     acceptable for multiple payload type numbers to be mapped to the=20
>> same
>>     codec, so that an updated offer could also use payload type=20
>> number 72
>>     for G.711.
>
> The language above is underspecified.
>
> It is my impression that the intent of this language was to ensure=20
> there is no ambiguity in the meaning of a payload type. Since the=20
> exchange of the SDP is not synchronized with the flow of RTP, if you=20
> were to send a new SDP that changed the mapping for a payload type,=20
> then the result would be almost certainly to interpret some media=20
> packets according to the wrong mapping.
>
> But this argument is only about packets flowing in one direction. It=20
> isn't a valid argument for requiring the payload types to be=20
> consistent in the two directions. And its not clear to me that it even=20
> intended that this language apply to both directions. It could be=20
> clarified as follows:
>
>                               ...  For example, if A generates an offer
>    with G.711 assigned to dynamic payload type number 46, payload type
>    number 46 MUST refer to G.711 from that point forward in any offers
>    or answers ***by A*** for that media stream within the session.  ...
>
> Interpreting it to apply to both directions brings it in conflict with=20
> the language elsewhere that permits it to be different in the two=20
> directions.
More than one number for one codec is explicitly permitted.
More than one codec for one number in a single direction is clearly impossi=
ble.
More than one codec for one number in a single RTP session..... that's the =
question.

If one were to go with textual analysis, this would hinge on the meaning of=
 "for that media stream" - which again hinges on whether a media stream is =
unidirectional or bidirectional.

RFC 3264's definition:

       Media Stream: From RTSP [8], a media stream is a single media
          instance, e.g., an audio stream or a video stream as well as a
          single whiteboard or shared application group.  In SDP, a media
          stream is described by an "m=3D" line and its associated
          attributes.

which points back to RFC 2326:

    (Media) stream:
           A single media instance, e.g., an audio stream or a video
           stream as well as a single whiteboard or shared application
           group. When using RTP, a stream consists of all RTP and RTCP
           packets created by a source within an RTP session. This is
           equivalent to the definition of a DSM-CC stream([5]).

So if we believe these definitions, there's no prohibition on reusing the p=
ayload type.
Definitely nice to clarify in a 3264bis.

>
> The use of different mappings in each direction has been discussed=20
> over the years, though I couldn't point you to where. IIRC one of the=20
> reasons had to do with gatewaying to other protocols. But I can't=20
> offer any details.
>
> Bottom line: IMO, in the weird case Harald described X ought to follow=20
> Postel's Law and accept the answer from Y. But for the same reason I=20
> would strongly discourage implementations from acting as Y did.
>
Seems like a conclusion.

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

From emil@sip-communicator.org  Fri Jan 11 05:50:42 2013
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Date: Fri, 11 Jan 2013 14:50:35 +0100
From: Emil Ivov <emcho@jitsi.org>
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Subject: Re: [MMUSIC] Query: payload type collision with offer/answer
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On 11.01.13, 14:01, Christer Holmberg wrote:
> Hi,
> 
> My understanding of the section 8.3.2 is that the mapping must not change PER DIRECTION, and the text "in any offers or answers" refers to a particular participant.

I wasn't around then so it could very well be that this was the
intention. Currently however there's nothing in the text that directly
supports this and it could be argued that it actually says the exact
opposite thing. Or in other words: the payload type number cannot be
remapped in "any offers and answers for that media stream in that session".

If we want it to meant just "A's offers and answers" then it should be
updated to say that.

Emil


> Regards,
> 
> Christer
> 
> 
> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of Harald Alvestrand
> Sent: 11. tammikuuta 2013 14:47
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] Query: payload type collision with offer/answer
> 
> On 01/10/2013 07:50 PM, Paul Kyzivat wrote:
>>
>>> Well there's also section 8.3.2 in 3264. The section is about 
>>> updating offers but it seems like the text explicitly prohibits 
>>> remapping payload types in any scenarios:
>>>
>>>     However, in the
>>>     case of RTP, the mapping from a particular dynamic payload type
>>>     number to a particular codec within that media stream MUST NOT 
>>> change
>>>     for the duration of a session.  For example, if A generates an offer
>>>     with G.711 assigned to dynamic payload type number 46, payload type
>>>     number 46 MUST refer to G.711 from that point forward in any offers
>>>     or answers for that media stream within the session. However, it is
>>>     acceptable for multiple payload type numbers to be mapped to the 
>>> same
>>>     codec, so that an updated offer could also use payload type 
>>> number 72
>>>     for G.711.
>>
>> The language above is underspecified.
>>
>> It is my impression that the intent of this language was to ensure 
>> there is no ambiguity in the meaning of a payload type. Since the 
>> exchange of the SDP is not synchronized with the flow of RTP, if you 
>> were to send a new SDP that changed the mapping for a payload type, 
>> then the result would be almost certainly to interpret some media 
>> packets according to the wrong mapping.
>>
>> But this argument is only about packets flowing in one direction. It 
>> isn't a valid argument for requiring the payload types to be 
>> consistent in the two directions. And its not clear to me that it even 
>> intended that this language apply to both directions. It could be 
>> clarified as follows:
>>
>>                               ...  For example, if A generates an offer
>>    with G.711 assigned to dynamic payload type number 46, payload type
>>    number 46 MUST refer to G.711 from that point forward in any offers
>>    or answers ***by A*** for that media stream within the session.  ...
>>
>> Interpreting it to apply to both directions brings it in conflict with 
>> the language elsewhere that permits it to be different in the two 
>> directions.
> More than one number for one codec is explicitly permitted.
> More than one codec for one number in a single direction is clearly impossible.
> More than one codec for one number in a single RTP session..... that's the question.
> 
> If one were to go with textual analysis, this would hinge on the meaning of "for that media stream" - which again hinges on whether a media stream is unidirectional or bidirectional.
> 
> RFC 3264's definition:
> 
>        Media Stream: From RTSP [8], a media stream is a single media
>           instance, e.g., an audio stream or a video stream as well as a
>           single whiteboard or shared application group.  In SDP, a media
>           stream is described by an "m=" line and its associated
>           attributes.
> 
> which points back to RFC 2326:
> 
>     (Media) stream:
>            A single media instance, e.g., an audio stream or a video
>            stream as well as a single whiteboard or shared application
>            group. When using RTP, a stream consists of all RTP and RTCP
>            packets created by a source within an RTP session. This is
>            equivalent to the definition of a DSM-CC stream([5]).
> 
> So if we believe these definitions, there's no prohibition on reusing the payload type.
> Definitely nice to clarify in a 3264bis.
> 
>>
>> The use of different mappings in each direction has been discussed 
>> over the years, though I couldn't point you to where. IIRC one of the 
>> reasons had to do with gatewaying to other protocols. But I can't 
>> offer any details.
>>
>> Bottom line: IMO, in the weird case Harald described X ought to follow 
>> Postel's Law and accept the answer from Y. But for the same reason I 
>> would strongly discourage implementations from acting as Y did.
>>
> Seems like a conclusion.
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
> _______________________________________________
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> mmusic@ietf.org
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> 

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From christer.holmberg@ericsson.com  Fri Jan 11 06:27:23 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Emil Ivov <emcho@jitsi.org>
Thread-Topic: [MMUSIC] Query: payload type collision with offer/answer
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] Query: payload type collision with offer/answer
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Hi,

>> My understanding of the section 8.3.2 is that the mapping must not chang=
e PER DIRECTION, and the text "in any offers or answers" refers to a partic=
ular participant.
>
> I wasn't around then so it could very well be that this was the intention=
. Currently however there's nothing in the text that directly=20
> supports this and it could be argued that it actually says the exact oppo=
site thing. Or in other words: the payload type number cannot=20
> be remapped in "any offers and answers for that media stream in that sess=
ion".
>
> If we want it to meant just "A's offers and answers" then it should be up=
dated to say that.

I agree the text could be more clear.

Regards,

Christer



> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On=20
> Behalf Of Harald Alvestrand
> Sent: 11. tammikuuta 2013 14:47
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] Query: payload type collision with offer/answer
>=20
> On 01/10/2013 07:50 PM, Paul Kyzivat wrote:
>>
>>> Well there's also section 8.3.2 in 3264. The section is about=20
>>> updating offers but it seems like the text explicitly prohibits=20
>>> remapping payload types in any scenarios:
>>>
>>>     However, in the
>>>     case of RTP, the mapping from a particular dynamic payload type
>>>     number to a particular codec within that media stream MUST NOT=20
>>> change
>>>     for the duration of a session.  For example, if A generates an offe=
r
>>>     with G.711 assigned to dynamic payload type number 46, payload type
>>>     number 46 MUST refer to G.711 from that point forward in any offers
>>>     or answers for that media stream within the session. However, it is
>>>     acceptable for multiple payload type numbers to be mapped to the=20
>>> same
>>>     codec, so that an updated offer could also use payload type=20
>>> number 72
>>>     for G.711.
>>
>> The language above is underspecified.
>>
>> It is my impression that the intent of this language was to ensure=20
>> there is no ambiguity in the meaning of a payload type. Since the=20
>> exchange of the SDP is not synchronized with the flow of RTP, if you=20
>> were to send a new SDP that changed the mapping for a payload type,=20
>> then the result would be almost certainly to interpret some media=20
>> packets according to the wrong mapping.
>>
>> But this argument is only about packets flowing in one direction. It=20
>> isn't a valid argument for requiring the payload types to be=20
>> consistent in the two directions. And its not clear to me that it=20
>> even intended that this language apply to both directions. It could=20
>> be clarified as follows:
>>
>>                               ...  For example, if A generates an offer
>>    with G.711 assigned to dynamic payload type number 46, payload type
>>    number 46 MUST refer to G.711 from that point forward in any offers
>>    or answers ***by A*** for that media stream within the session.  ...
>>
>> Interpreting it to apply to both directions brings it in conflict=20
>> with the language elsewhere that permits it to be different in the=20
>> two directions.
> More than one number for one codec is explicitly permitted.
> More than one codec for one number in a single direction is clearly impos=
sible.
> More than one codec for one number in a single RTP session..... that's th=
e question.
>=20
> If one were to go with textual analysis, this would hinge on the meaning =
of "for that media stream" - which again hinges on whether a media stream i=
s unidirectional or bidirectional.
>=20
> RFC 3264's definition:
>=20
>        Media Stream: From RTSP [8], a media stream is a single media
>           instance, e.g., an audio stream or a video stream as well as a
>           single whiteboard or shared application group.  In SDP, a media
>           stream is described by an "m=3D" line and its associated
>           attributes.
>=20
> which points back to RFC 2326:
>=20
>     (Media) stream:
>            A single media instance, e.g., an audio stream or a video
>            stream as well as a single whiteboard or shared application
>            group. When using RTP, a stream consists of all RTP and RTCP
>            packets created by a source within an RTP session. This is
>            equivalent to the definition of a DSM-CC stream([5]).
>=20
> So if we believe these definitions, there's no prohibition on reusing the=
 payload type.
> Definitely nice to clarify in a 3264bis.
>=20
>>
>> The use of different mappings in each direction has been discussed=20
>> over the years, though I couldn't point you to where. IIRC one of the=20
>> reasons had to do with gatewaying to other protocols. But I can't=20
>> offer any details.
>>
>> Bottom line: IMO, in the weird case Harald described X ought to=20
>> follow Postel's Law and accept the answer from Y. But for the same=20
>> reason I would strongly discourage implementations from acting as Y did.
>>
> Seems like a conclusion.
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>=20

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Subject: Re: [MMUSIC] Fwd: RTCWEB & MMUSIC WG Joint Interim Meeting, February 5-7, 2013
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We are working on an agenda with the RTCWEB chairs - expect to see 
something early next week.

Thanks

-- Flemming


On 1/10/13 6:46 PM, Matthew Kaufman wrote:
> Is there yet an agenda for the upcoming MMUSIC interim or at least a 
> list of drafts we intend to discuss that I could review in advance? 
> Just feels like we're running out of days for advance flight bookings.
>
> Matthew Kaufman
>


From pkyzivat@alum.mit.edu  Fri Jan 11 08:57:06 2013
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Date: Fri, 11 Jan 2013 11:57:02 -0500
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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To: Dan Wing <dwing@cisco.com>
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Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] RESTART: SCTP question: Where does it multiplex?
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Commenting on one specific point...

On 1/11/13 12:51 AM, Dan Wing wrote:

>>> We need a way for SDP to signal:
>>>     1. the UDP port for SCTP.  This feels like an SDP media ("m") line
>> to me.
>>>     2. SCTP native (native SCTP over IPv6 should be tenable, whereas
>>> native SCTP over IPv4 is pretty much untenable),
>>>     3. the data carried over each SCTP port (e.g., port NNN for XMPP,
>>> port NNNN for file sharing, port NNNNN for screen sharing, and so on).
>>> This feels like an SDP media ("m") line to me.
>>
>> I don't disagree with any of your points, but you have missed the point
>> I was making.
>>
>> Everything you list applies for SCTP on its own 5-tuple. But then there
>> is the question of how to describe how this shares a 5-tuple with some
>> audio and video streams.
>
> (1), from my list, is the UDP port.  This could be the same UDP port
> as the audio/video media.

Yes. But SDP doesn't permit two m-lines with the same address and port. 
Bundle has been proposed to work around that. But currently bundle 
requires all the bundled m-lines to have the same <proto>. And that 
won't work for bundling SCTP and RTP.

The restriction to the same <proto> is dictated by the need for a way to 
demux. Bundle specifies rules for multiple RTP m-lines that ensures they 
can be demuxed. The demuxing of RTP from SCTP on the same UDP port uses 
a different mechanism. So for this to work bundle needs:

- more elaborate rules for what <proto> values may be bundled,
   and which may not

- explicit specification for the demuxing of each of the different
   m-lines that have been bundled.

This is not rocket science. Once you think about it a bit it is fairly 
evident which combinations work, and how to demux them. But it needs to 
be written down, directly or by reference.

	Thanks,
	Paul


From eckelcu@cisco.com  Fri Jan 11 09:25:35 2013
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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: Harald Alvestrand <harald@alvestrand.no>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Query: payload type collision with offer/answer
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Subject: Re: [MMUSIC] Query: payload type collision with offer/answer
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> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Harald Alvestrand
> Sent: Friday, January 11, 2013 4:47 AM
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] Query: payload type collision with offer/answer
>=20
> On 01/10/2013 07:50 PM, Paul Kyzivat wrote:
> >
> >> Well there's also section 8.3.2 in 3264. The section is about updating
> >> offers but it seems like the text explicitly prohibits remapping paylo=
ad
> >> types in any scenarios:
> >>
> >>     However, in the
> >>     case of RTP, the mapping from a particular dynamic payload type
> >>     number to a particular codec within that media stream MUST NOT
> >> change
> >>     for the duration of a session.  For example, if A generates an off=
er
> >>     with G.711 assigned to dynamic payload type number 46, payload typ=
e
> >>     number 46 MUST refer to G.711 from that point forward in any offer=
s
> >>     or answers for that media stream within the session. However, it i=
s
> >>     acceptable for multiple payload type numbers to be mapped to the
> >> same
> >>     codec, so that an updated offer could also use payload type
> >> number 72
> >>     for G.711.
> >
> > The language above is underspecified.
> >
> > It is my impression that the intent of this language was to ensure
> > there is no ambiguity in the meaning of a payload type. Since the
> > exchange of the SDP is not synchronized with the flow of RTP, if you
> > were to send a new SDP that changed the mapping for a payload type,
> > then the result would be almost certainly to interpret some media
> > packets according to the wrong mapping.
> >
> > But this argument is only about packets flowing in one direction. It
> > isn't a valid argument for requiring the payload types to be
> > consistent in the two directions. And its not clear to me that it even
> > intended that this language apply to both directions. It could be
> > clarified as follows:
> >
> >                               ...  For example, if A generates an offer
> >    with G.711 assigned to dynamic payload type number 46, payload type
> >    number 46 MUST refer to G.711 from that point forward in any offers
> >    or answers ***by A*** for that media stream within the session.  ...
> >
> > Interpreting it to apply to both directions brings it in conflict with
> > the language elsewhere that permits it to be different in the two
> > directions.
> More than one number for one codec is explicitly permitted.
> More than one codec for one number in a single direction is clearly
> impossible.
> More than one codec for one number in a single RTP session..... that's
> the question.
>=20
> If one were to go with textual analysis, this would hinge on the meaning
> of "for that media stream" - which again hinges on whether a media
> stream is unidirectional or bidirectional.
>=20
> RFC 3264's definition:
>=20
>        Media Stream: From RTSP [8], a media stream is a single media
>           instance, e.g., an audio stream or a video stream as well as a
>           single whiteboard or shared application group.  In SDP, a media
>           stream is described by an "m=3D" line and its associated
>           attributes.
>=20
> which points back to RFC 2326:
>=20
>     (Media) stream:
>            A single media instance, e.g., an audio stream or a video
>            stream as well as a single whiteboard or shared application
>            group. When using RTP, a stream consists of all RTP and RTCP
>            packets created by a source within an RTP session. This is
>            equivalent to the definition of a DSM-CC stream([5]).
>=20
> So if we believe these definitions, there's no prohibition on reusing
> the payload type.
> Definitely nice to clarify in a 3264bis.
>=20
> >
> > The use of different mappings in each direction has been discussed
> > over the years, though I couldn't point you to where. IIRC one of the
> > reasons had to do with gatewaying to other protocols. But I can't
> > offer any details.
> >
> > Bottom line: IMO, in the weird case Harald described X ought to follow
> > Postel's Law and accept the answer from Y. But for the same reason I
> > would strongly discourage implementations from acting as Y did.
> >
> Seems like a conclusion.

Agreed, this would be a useful clarification to add as an update to RFC 326=
4.

Cheers,
Charles

>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

From pkyzivat@alum.mit.edu  Fri Jan 11 10:05:32 2013
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Subject: Re: [MMUSIC] Some issues with draft-ietf-mmusic-sctp-sdp-02
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Replying to myself, since nobody else did.

There has been more discussion recently (subject: SCTP question: Where 
does it multiplex?) on the same issue I raise below.

More inline.

On 12/3/12 3:29 PM, Paul Kyzivat wrote:
> While commenting on some related discussions I had occasion to look at
> this draft again and I found a few issues:
>
> First, some very specific ones::
>
> Section 4.1 says:
>
>     If the <port> sub-field is 'SCTP' or ...
>
> This is a typo. s/port/proto/
>
> But the major issue is that this draft doesn't explain the semantics of
> the <port> field. The only things I can find that touch on this are:
>
> - section 10.1 refers to "SCTP port 54111" and then shows an m-line
>    with: m=application 54111 SCTP 0 1 2
>    But this is just an example, not normative text.
>
> - section 4.1 says:
>
>     [OPEN ISSUE 3] do we need also an attribute to specify the SCTP port
>     number for the SCTP over DTLS scenario? that can be useful in the
>     case where multiple association are running on top of the same DTLS
>     session.
>
>    This implies that the SCTP port number is *not* what is in the port
>    field in this case. Presumably because the UDP port number is in
>    that field.
>
> ISTM that there needs to be a clear discussion of all the required port
> numbers for each case, and how they are specified. When there is more
> than one (namely in the SCTP/DTLS and DTLS/SCTP cases), *one* of them
> can be in the m-line and the other one must be somewhere else. And if
> the one that is not in the m-line is optional, then I guess there must
> be a default.

For *native* SCTP (directly over IPv* - <proto>=SCTP) it seems obvious 
that the port in the m-line should be the SCTP port.

For <proto> of SCTP/DTLS (RFC6083) there is again only the SCTP port to 
consider, so it should presumably go in the m-line. (My earlier comment 
about this one was wrong.)

For <proto> of DTLS/SCTP (draft-tuexen-sctp-udp-encaps) we need both the 
SCTP port and the UDP port, so we must pick which one goes in the 
m-line. What arguments are there for which should go in the m-line?

- Consistency with the other SCTP protos argues for the SCTP port.
- The port in the m-line is mandatory. AFAIK there is no default
   SCTP port. But there *is* a default UDP port for DTLS/SCTP
   (9899, defined in draft-tuexen-sctp-udp-encaps). This is also
   an argument for the SCTP port in the m-line.
- are there any arguments in favor of the UDP port in the m-line?
   (I haven't thought of any.)

Absent some arguments on the other side, I propose that this document 
specify that the <port> field of the m-line denote the SCTP port for all 
of the SCTP <proto>s, and specify an attribute to carry the UDP port, 
with a default of 9899.

	Thanks,
	Paul

From miguel.a.garcia@ericsson.com  Sun Jan 13 01:40:39 2013
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Subject: [MMUSIC] Stepping down as MMUSIC co-chair
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Dear all:

It's been almost two years serving as MMUSIC co-chair. Unfortunately my 
other non-IETF assignments are preventing me to do the job with the 
commitment I would like to do. So, I have decided to step down as MMUSIC 
co-chair.

A new co-chair should be announced soon.

Cheers,

      Miguel
-- 
Miguel A. Garcia
+34-91-339-3608
Ericsson Spain

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Date: Sun, 13 Jan 2013 13:08:17 +0200
From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: Re: [MMUSIC] Stepping down as MMUSIC co-chair
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Hi,

I want to thank Miguel Angel for all his efforts during this period. We
were all sorry to hear he had to step down at this point. Ari Keranen
will be replacing Miguel Angel.

Cheers,

Gonzalo

On 13/01/2013 11:40 AM, Miguel A. Garcia wrote:
> Dear all:
> 
> It's been almost two years serving as MMUSIC co-chair. Unfortunately my
> other non-IETF assignments are preventing me to do the job with the
> commitment I would like to do. So, I have decided to step down as MMUSIC
> co-chair.
> 
> A new co-chair should be announced soon.
> 
> Cheers,
> 
>      Miguel


From internet-drafts@ietf.org  Sun Jan 13 21:47:19 2013
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Cc: mmusic@ietf.org
Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-media-loopback-26.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Multiparty Multimedia Session Control Wor=
king Group of the IETF.

	Title           : An Extension to the Session Description Protocol (SDP) a=
nd Real-time Transport Protocol (RTP) for Media Loopback
	Author(s)       : Hadriel Kaplan
                          Kaynam Hedayat
                          Nagarjuna Venna
                          Paul E. Jones
                          Nathan Stratton
	Filename        : draft-ietf-mmusic-media-loopback-26.txt
	Pages           : 34
	Date            : 2013-01-13

Abstract:
    The wide deployment of Voice over IP (VoIP), Text and Video over IP
    services has introduced new challenges in managing and maintaining
    real-time voice/text/video quality, reliability, and overall
    performance.  In particular, media delivery is an area that needs
    attention.  One method of meeting these challenges is monitoring
    the media delivery performance by looping media back to the
    transmitter.  This is typically referred to as "active monitoring"
    of services.   Media loopback is especially popular in ensuring the
    quality of transport to the edge of a given VoIP, Real-time Text or
    Video over IP service.  Today in networks that deliver real-time
    media, short of running 'ping' and 'traceroute' to the edge,
    administrators are left without the necessary tools to actively
    monitor, manage, and diagnose quality issues with their service.
    The extension defined herein adds new SDP media types and
    attributes, which enable establishment of media sessions where the
    media is looped back to the transmitter. Such media sessions will
    serve as monitoring and troubleshooting tools by providing the
    means for measurement of more advanced VoIP, Real-time Text and
    Video over IP performance metrics.



The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-media-loopback

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-media-loopback-26

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-media-loopback-26


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


From miguel.a.garcia@ericsson.com  Mon Jan 14 01:11:49 2013
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Subject: Re: [MMUSIC] AD review: draft-ietf-mmusic-sdp-cs-16
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Hi Gonzalo:

Here are some inline comments. I am about to post version -17 addressing 
your comments:


On 11/01/2013 12:50, Gonzalo Camarillo wrote:
> Hi,
>
> I just got a publication request for the following draft:
>
> http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-16
>
> Please, find below my AD review.
>
> Cheers,
>
> Gonzalo
>
>
> draft-ietf-mmusic-sdp-cs-16:
>
> Page 10: remove the indentation of the following paragraph. Indented
> paragraphs are typically used to provide clarifications while this one
> contains normative language:
>
>        Note that <addrtype> and/or <connection-address> MUST NOT be
>        omitted when unknown since this would violate basic syntax of SDP
>        [RFC4566].  In such cases, they MUST be set to a "-".

Ok.

>
> Section 5.2.3.1 provides a "format" for the cs-correlation attribute.
> However, its ABNF formal syntax is in Section 5.7. So, why does Section
> 5.2.3.1 also contain a syntax for the attribute... and what type of
> syntax is that?

The sort-of-syntax in Section 5.2.3.1 is simply an introduction to the 
three types of correlation mechanisms that can be used. So, the syntax is 
the way to introduce to Sections 5.2.3.2, 5.2.3.3, and 5.2.3.4.

But I understand your point. I suggest removing the syntax from there and 
including a pointer to Section 5.7, where the formal syntax is described.

>
> Page 14: the text says:
>
>        To mitigate this problem implementations
>        should consider only some of the rightmost digits from the E.164
>
> Can we provide a more concrete advice (or at least some discussion) on
> the number of digits to consider?

This is a hard problem even to provide guidance, because the PSTN E.164 
numbers have a variable length. So, if you want to do it right, you 
should become aware of the number plan in the network you are camping. 
But this is outside the scope of the specification.

We can add a note to refer to ITU-T E.164 specification in order for 
implementations to consider what the right number of digits is.

>
> Page 15: we define how to encode a binary value in hexadecimal format
> and represent it as a two-digit ASCII value. Can we point to an existing
> specification that defines how to do that instead of specifying it inline?
>

Yeap, and I found that there is an RFC specifying it (there is always an 
RFC specifying what you want :-). It is RFC 4648 and we are referring to 
the base16 (or "hex") encoding. We will change the text accordingly.


> Page 17: the text says:
>
>        Implementations are advised to select a number of DTMF digits that
>        provide enough assurance that the call is related, but on the
>        other hand do not prolong the bearer setup time unnecessarily.
>
> Can we provide a more concrete recommendation?

Ok, I guess we are talking of 5 to 10 digits.

>
> Page 17: Section 5.2.3.4 specifies what to do when the correlation
> fails. However, that is described in general in Section 5.3.3. Why don't
> we move all the statements about how to decide whether the sessions are
> correlated to Section 5.3.3? Now, the following statement contradicts
> what Section 5.3.3 says, for example:
>
>     the passive side
>     SHOULD treat the circuit-switched bearer as not correlated to the
>     ongoing session.

ok.

>
> The indented text right after the statement above talks about
> suppressing alerting. Such a supression is applicable to any session,
> not only those that are correlated using DTMF tones. So, we could also
> move that discussion to Section 5.3.3.

ok too.

>
> Section 5.5: rephrase the following sentence (possibly splitting it into
> more than one sentence) so that it is clearer:
>
>        Note that this may result in the recipient of the initial Offer in
>        rejecting the Offer if the recipient of the Offer is not aware of
>        its own E.164 number, and thus concluding that it will not be
>        possible to establish a circuit-switched bearer since neither
>        party is aware of their E.164 number.

Done



Thanks,


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

-- 
Miguel A. Garcia
+34-91-339-3608
Ericsson Spain

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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: Re: [MMUSIC] AD review: draft-ietf-mmusic-sdp-cs-16
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Hi Miguel Angel,

thanks for your quick response. I am OK with all your proposals below.
When you implement the changes, please post a new revision of the draft.

Thanks,

Gonzalo

On 14/01/2013 11:11 AM, Miguel A. Garcia wrote:
> Hi Gonzalo:
> 
> Here are some inline comments. I am about to post version -17 addressing
> your comments:
> 
> 
> On 11/01/2013 12:50, Gonzalo Camarillo wrote:
>> Hi,
>>
>> I just got a publication request for the following draft:
>>
>> http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-16
>>
>> Please, find below my AD review.
>>
>> Cheers,
>>
>> Gonzalo
>>
>>
>> draft-ietf-mmusic-sdp-cs-16:
>>
>> Page 10: remove the indentation of the following paragraph. Indented
>> paragraphs are typically used to provide clarifications while this one
>> contains normative language:
>>
>>        Note that <addrtype> and/or <connection-address> MUST NOT be
>>        omitted when unknown since this would violate basic syntax of SDP
>>        [RFC4566].  In such cases, they MUST be set to a "-".
> 
> Ok.
> 
>>
>> Section 5.2.3.1 provides a "format" for the cs-correlation attribute.
>> However, its ABNF formal syntax is in Section 5.7. So, why does Section
>> 5.2.3.1 also contain a syntax for the attribute... and what type of
>> syntax is that?
> 
> The sort-of-syntax in Section 5.2.3.1 is simply an introduction to the
> three types of correlation mechanisms that can be used. So, the syntax
> is the way to introduce to Sections 5.2.3.2, 5.2.3.3, and 5.2.3.4.
> 
> But I understand your point. I suggest removing the syntax from there
> and including a pointer to Section 5.7, where the formal syntax is
> described.
> 
>>
>> Page 14: the text says:
>>
>>        To mitigate this problem implementations
>>        should consider only some of the rightmost digits from the E.164
>>
>> Can we provide a more concrete advice (or at least some discussion) on
>> the number of digits to consider?
> 
> This is a hard problem even to provide guidance, because the PSTN E.164
> numbers have a variable length. So, if you want to do it right, you
> should become aware of the number plan in the network you are camping.
> But this is outside the scope of the specification.
> 
> We can add a note to refer to ITU-T E.164 specification in order for
> implementations to consider what the right number of digits is.
> 
>>
>> Page 15: we define how to encode a binary value in hexadecimal format
>> and represent it as a two-digit ASCII value. Can we point to an existing
>> specification that defines how to do that instead of specifying it
>> inline?
>>
> 
> Yeap, and I found that there is an RFC specifying it (there is always an
> RFC specifying what you want :-). It is RFC 4648 and we are referring to
> the base16 (or "hex") encoding. We will change the text accordingly.
> 
> 
>> Page 17: the text says:
>>
>>        Implementations are advised to select a number of DTMF digits that
>>        provide enough assurance that the call is related, but on the
>>        other hand do not prolong the bearer setup time unnecessarily.
>>
>> Can we provide a more concrete recommendation?
> 
> Ok, I guess we are talking of 5 to 10 digits.
> 
>>
>> Page 17: Section 5.2.3.4 specifies what to do when the correlation
>> fails. However, that is described in general in Section 5.3.3. Why don't
>> we move all the statements about how to decide whether the sessions are
>> correlated to Section 5.3.3? Now, the following statement contradicts
>> what Section 5.3.3 says, for example:
>>
>>     the passive side
>>     SHOULD treat the circuit-switched bearer as not correlated to the
>>     ongoing session.
> 
> ok.
> 
>>
>> The indented text right after the statement above talks about
>> suppressing alerting. Such a supression is applicable to any session,
>> not only those that are correlated using DTMF tones. So, we could also
>> move that discussion to Section 5.3.3.
> 
> ok too.
> 
>>
>> Section 5.5: rephrase the following sentence (possibly splitting it into
>> more than one sentence) so that it is clearer:
>>
>>        Note that this may result in the recipient of the initial Offer in
>>        rejecting the Offer if the recipient of the Offer is not aware of
>>        its own E.164 number, and thus concluding that it will not be
>>        possible to establish a circuit-switched bearer since neither
>>        party is aware of their E.164 number.
> 
> Done
> 
> 
> 
> Thanks,
> 
> 
>>
>>
>>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>>
> 


From internet-drafts@ietf.org  Mon Jan 14 01:20:03 2013
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Cc: mmusic@ietf.org
Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-17.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Multiparty Multimedia Session Control Wor=
king Group of the IETF.

	Title           : Session Description Protocol (SDP) Extension For Setting=
 Up Audio and Video Media Streams Over Circuit-Switched Bearers In The Publ=
ic Switched Telephone Network (PSTN)
	Author(s)       : Miguel A. Garcia-Martin
                          Simo Veikkolainen
	Filename        : draft-ietf-mmusic-sdp-cs-17.txt
	Pages           : 37
	Date            : 2013-01-14

Abstract:
   This memo describes use cases, requirements, and protocol extensions
   for using the Session Description Protocol (SDP) Offer/Answer model
   for establishing audio and video media streams over circuit-switched
   bearers in the Public Switched Telephone Network (PSTN).


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-cs

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-17

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sdp-cs-17


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


From miguel.a.garcia@ericsson.com  Mon Jan 14 01:20:36 2013
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To: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Cc: mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] AD review: draft-ietf-mmusic-sdp-cs-16
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I have just submitted version -17.

/Miguel

On 14/01/2013 10:19, Gonzalo Camarillo wrote:
> Hi Miguel Angel,
>
> thanks for your quick response. I am OK with all your proposals below.
> When you implement the changes, please post a new revision of the draft.
>
> Thanks,
>
> Gonzalo
>
> On 14/01/2013 11:11 AM, Miguel A. Garcia wrote:
>> Hi Gonzalo:
>>
>> Here are some inline comments. I am about to post version -17 addressing
>> your comments:
>>
>>
>> On 11/01/2013 12:50, Gonzalo Camarillo wrote:
>>> Hi,
>>>
>>> I just got a publication request for the following draft:
>>>
>>> http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-16
>>>
>>> Please, find below my AD review.
>>>
>>> Cheers,
>>>
>>> Gonzalo
>>>
>>>
>>> draft-ietf-mmusic-sdp-cs-16:
>>>
>>> Page 10: remove the indentation of the following paragraph. Indented
>>> paragraphs are typically used to provide clarifications while this one
>>> contains normative language:
>>>
>>>         Note that <addrtype> and/or <connection-address> MUST NOT be
>>>         omitted when unknown since this would violate basic syntax of SDP
>>>         [RFC4566].  In such cases, they MUST be set to a "-".
>>
>> Ok.
>>
>>>
>>> Section 5.2.3.1 provides a "format" for the cs-correlation attribute.
>>> However, its ABNF formal syntax is in Section 5.7. So, why does Section
>>> 5.2.3.1 also contain a syntax for the attribute... and what type of
>>> syntax is that?
>>
>> The sort-of-syntax in Section 5.2.3.1 is simply an introduction to the
>> three types of correlation mechanisms that can be used. So, the syntax
>> is the way to introduce to Sections 5.2.3.2, 5.2.3.3, and 5.2.3.4.
>>
>> But I understand your point. I suggest removing the syntax from there
>> and including a pointer to Section 5.7, where the formal syntax is
>> described.
>>
>>>
>>> Page 14: the text says:
>>>
>>>         To mitigate this problem implementations
>>>         should consider only some of the rightmost digits from the E.164
>>>
>>> Can we provide a more concrete advice (or at least some discussion) on
>>> the number of digits to consider?
>>
>> This is a hard problem even to provide guidance, because the PSTN E.164
>> numbers have a variable length. So, if you want to do it right, you
>> should become aware of the number plan in the network you are camping.
>> But this is outside the scope of the specification.
>>
>> We can add a note to refer to ITU-T E.164 specification in order for
>> implementations to consider what the right number of digits is.
>>
>>>
>>> Page 15: we define how to encode a binary value in hexadecimal format
>>> and represent it as a two-digit ASCII value. Can we point to an existing
>>> specification that defines how to do that instead of specifying it
>>> inline?
>>>
>>
>> Yeap, and I found that there is an RFC specifying it (there is always an
>> RFC specifying what you want :-). It is RFC 4648 and we are referring to
>> the base16 (or "hex") encoding. We will change the text accordingly.
>>
>>
>>> Page 17: the text says:
>>>
>>>         Implementations are advised to select a number of DTMF digits that
>>>         provide enough assurance that the call is related, but on the
>>>         other hand do not prolong the bearer setup time unnecessarily.
>>>
>>> Can we provide a more concrete recommendation?
>>
>> Ok, I guess we are talking of 5 to 10 digits.
>>
>>>
>>> Page 17: Section 5.2.3.4 specifies what to do when the correlation
>>> fails. However, that is described in general in Section 5.3.3. Why don't
>>> we move all the statements about how to decide whether the sessions are
>>> correlated to Section 5.3.3? Now, the following statement contradicts
>>> what Section 5.3.3 says, for example:
>>>
>>>      the passive side
>>>      SHOULD treat the circuit-switched bearer as not correlated to the
>>>      ongoing session.
>>
>> ok.
>>
>>>
>>> The indented text right after the statement above talks about
>>> suppressing alerting. Such a supression is applicable to any session,
>>> not only those that are correlated using DTMF tones. So, we could also
>>> move that discussion to Section 5.3.3.
>>
>> ok too.
>>
>>>
>>> Section 5.5: rephrase the following sentence (possibly splitting it into
>>> more than one sentence) so that it is clearer:
>>>
>>>         Note that this may result in the recipient of the initial Offer in
>>>         rejecting the Offer if the recipient of the Offer is not aware of
>>>         its own E.164 number, and thus concluding that it will not be
>>>         possible to establish a circuit-switched bearer since neither
>>>         party is aware of their E.164 number.
>>
>> Done
>>
>>
>>
>> Thanks,
>>
>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>
>>
>

-- 
Miguel A. Garcia
+34-91-339-3608
Ericsson Spain

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Date: Sun, 13 Jan 2013 20:44:24 -0500
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Cc: mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] Stepping down as MMUSIC co-chair
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I'd like to echo that. Many thanks to Miguel for a job well done.

Regards

-- Flemming

On 1/13/13 6:08 AM, Gonzalo Camarillo wrote:
> Hi,
>
> I want to thank Miguel Angel for all his efforts during this period. We
> were all sorry to hear he had to step down at this point. Ari Keranen
> will be replacing Miguel Angel.
>
> Cheers,
>
> Gonzalo
>
> On 13/01/2013 11:40 AM, Miguel A. Garcia wrote:
> > Dear all:
> >
> > It's been almost two years serving as MMUSIC co-chair. Unfortunately my
> > other non-IETF assignments are preventing me to do the job with the
> > commitment I would like to do. So, I have decided to step down as 
> MMUSIC
> > co-chair.
> >
> > A new co-chair should be announced soon.
> >
> > Cheers,
> >
> > Miguel
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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<html>
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    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
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  <body bgcolor="#FFFFFF" text="#000000">
    I'd like to echo that. Many thanks to Miguel for a job well done. <br>
    <br>
    Regards<br>
    <br>
    -- Flemming <br>
    <br>
    <div class="moz-cite-prefix">On 1/13/13 6:08 AM, Gonzalo Camarillo
      wrote:<br>
    </div>
    <blockquote cite="mid:1.c0aae1c83d24cfcd39cb@ericsson.com"
      type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=ISO-8859-1">
      Hi,
      <br>
      <br>
      I want to thank Miguel Angel for all his efforts during this
      period. We
      <br>
      were all sorry to hear he had to step down at this point. Ari
      Keranen
      <br>
      will be replacing Miguel Angel.
      <br>
      <br>
      Cheers,
      <br>
      <br>
      Gonzalo
      <br>
      <br>
      On 13/01/2013 11:40 AM, Miguel A. Garcia wrote:
      <br>
      &gt; Dear all:
      <br>
      &gt; <br>
      &gt; It's been almost two years serving as MMUSIC co-chair.
      Unfortunately my
      <br>
      &gt; other non-IETF assignments are preventing me to do the job
      with the
      <br>
      &gt; commitment I would like to do. So, I have decided to step
      down as MMUSIC
      <br>
      &gt; co-chair.
      <br>
      &gt; <br>
      &gt; A new co-chair should be announced soon.
      <br>
      &gt; <br>
      &gt; Cheers,
      <br>
      &gt; <br>
      &gt; Miguel
      <br>
      <br>
      _______________________________________________
      <br>
      mmusic mailing list
      <br>
      <a class="moz-txt-link-abbreviated" href="mailto:mmusic@ietf.org">mmusic@ietf.org</a>
      <br>
      <a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a>
      <br>
    </blockquote>
    <br>
  </body>
</html>

--------------020009030406020709040401--

From ted.ietf@gmail.com  Mon Jan 14 09:41:30 2013
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From: Ted Hardie <ted.ietf@gmail.com>
To: rtcweb@ietf.org, mmusic@ietf.org
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Cc: Cullen Jennings <fluffy@cisco.com>, Magnus Westerlund <magnus.westerlund@ericsson.com>, =?ISO-8859-1?Q?Ari_Ker=E4nen?= <ari.keranen@ericsson.com>, Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>, Flemming Andreasen <fandreas@cisco.com>
Subject: [MMUSIC] Proposed agenda for the joint MMUSIC/RTCWEB inteirm meeting
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Below is a proposed agenda which has been discussed among the RTCWEB
and MMUSIC chairs; comments welcome.

Ted Hardie


Day 1 (February 5th, 2013)

Multiple Flow SDP representation

       90 minutes: Framing discussion (Magnus is working on text)

      120 minutes: drafts discussion


      Objective: decide on architecture for large numbers of similar streams

SDP for Data channels

        60 minutes:  resolve open issues

Day 2 (February 6, 2013)

SDP Grouping (Bundle/MMT/one-rtp)

           60 Minutes: framing discussion

          120 Minutes:  drafts discussion

           Objective: select an approach and provide direction to editors

Mapping stream/track label concepts to SDP identifiers

           60 minutes

            Objective: determine requirements

Day 3: (February 7, 2013)

            Trickle ICE

           120 minutes

            Objective: review offer/answer semantic impact; establish
initiation, termination and error handling

            JSEP

             120 minutes

              Objective: review and resolve open issues with the document

From internet-drafts@ietf.org  Mon Jan 14 11:13:17 2013
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-media-loopback-27.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Multiparty Multimedia Session Control Wor=
king Group of the IETF.

	Title           : An Extension to the Session Description Protocol (SDP) a=
nd Real-time Transport Protocol (RTP) for Media Loopback
	Author(s)       : Hadriel Kaplan
                          Kaynam Hedayat
                          Nagarjuna Venna
                          Paul E. Jones
                          Nathan Stratton
	Filename        : draft-ietf-mmusic-media-loopback-27.txt
	Pages           : 33
	Date            : 2013-01-14

Abstract:
    The wide deployment of Voice over IP (VoIP), Text and Video over IP
    services has introduced new challenges in managing and maintaining
    real-time voice/text/video quality, reliability, and overall
    performance.  In particular, media delivery is an area that needs
    attention.  One method of meeting these challenges is monitoring
    the media delivery performance by looping media back to the
    transmitter.  This is typically referred to as "active monitoring"
    of services.   Media loopback is especially popular in ensuring the
    quality of transport to the edge of a given VoIP, Real-time Text or
    Video over IP service.  Today in networks that deliver real-time
    media, short of running 'ping' and 'traceroute' to the edge,
    administrators are left without the necessary tools to actively
    monitor, manage, and diagnose quality issues with their service.
    The extension defined herein adds new SDP media types and
    attributes, which enable establishment of media sessions where the
    media is looped back to the transmitter. Such media sessions will
    serve as monitoring and troubleshooting tools by providing the
    means for measurement of more advanced VoIP, Real-time Text and
    Video over IP performance metrics.



The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-media-loopback

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-media-loopback-27

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-media-loopback-27


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


From magnus.westerlund@ericsson.com  Tue Jan 15 02:48:29 2013
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Subject: Re: [MMUSIC] BUNDLE: Session bandwidth (Re: commentson draft-ietf-mmusic-sdp-bundle-negotiation-01)
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On 2012-12-14 18:29, Charles Eckel (eckelcu) wrote:
>> -----Original Message-----
>> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
>> Sent: Wednesday, December 12, 2012 1:55 AM
>> To: Charles Eckel (eckelcu)
>> Cc: Harald Alvestrand; mmusic@ietf.org
>> Subject: Re: [MMUSIC] BUNDLE: Session bandwidth (Re: comments on draft-
>> ietf-mmusic-sdp-bundle-negotiation-01)
>>
>> Hi,
>>
>> Going through old emails. One important comment:
>>
>> On 2012-11-05 21:04, Charles Eckel (eckelcu) wrote:
>> >>
>> >> Last time I looked b=CT was scoped to a whole conference ("supplied
> for
>> >> the session"), while b=AS was scoped to a single RTP session. But I
> may
>> >> be misinterpreting RFC 4566 section 5.8.
>> >
>> > I recommend using b=TIAS, as defined in RFC 3890. This parameter may be
>> used at either the session or the media level, and it scope accordingly.
>> > What needs to be avoided is summing up the b=TIAS/AS values from the
>> bundled media lines and using that as that bandwidth for the bundle
> even if
>> it exceeds the b=TIAS/AS/CT value from the session level.
>> >
>>
>> I would note that TIAS has no definition in Offer/Answer context. It is
>> soley defined for Declarative usage. Yes, I know it is being used in O/A
>> contexts, but I still don't understand how a receiver can state the
>> actual packet rate to be used by the sender to enable correct
> calculation.
> 
> Agreed, TIAS is a declaration of the maximum bandwidth the declarer is
> willing to receive. It does not say anything about what the declarer is
> going to send. The same is true for AS and CT.
> If we want something that indicates the maximum bandwidth the declarer
> is planning to send, we would need to extend an existing attribute or
> define a new attribute.
> However, the restriction I mentioned above regarding bundled streams not
> violating limits declared by the corresponding b=TIAS/AS/CT attributes
> still applies.
> 

But the thing is that TIAS is defined as the bandwidth one is going to
send including packet rate in the declarative usage.

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  Tue Jan 15 07:45:34 2013
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Cc: mmusic chair <mmusic-chairs@tools.ietf.org>, mmusic mailing list <mmusic@ietf.org>, RFC Editor <rfc-editor@rfc-editor.org>
Subject: [MMUSIC] Protocol Action: 'An Extension to the Session Description Protocol	(SDP) and Real-time Transport Protocol (RTP) for Media	Loopback' to Proposed Standard	(draft-ietf-mmusic-media-loopback-27.txt)
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The IESG has approved the following document:
- 'An Extension to the Session Description Protocol (SDP) and Real-time
   Transport Protocol (RTP) for Media Loopback'
  (draft-ietf-mmusic-media-loopback-27.txt) as Proposed Standard

This document is the product of the Multiparty Multimedia Session Control
Working Group.

The IESG contact persons are Gonzalo Camarillo and Robert Sparks.

A URL of this Internet Draft is:
http://datatracker.ietf.org/doc/draft-ietf-mmusic-media-loopback/




Technical Summary

This document provides extensions to the Session Description Protocol
(SDP) and the Real-time Transport Protocol (RTP) for enabling looping
of sent media. Media loopback is especially popular in ensuring the
quality of transport to the edge of a given VoIP, Real-time Text or
Video over IP service. The extension defined herein adds new SDP media
types and attributes, which enable establishment of media sessions
where the media is looped back to the transmitter. Such media sessions
will serve as monitoring and troubleshooting tools by providing the
means for measurement of more advanced VoIP, Real-time Text and Video
over IP performance metrics.

Working Group Summary

The first version of the document was available at the end of 2004.
Since that time, the working group has been discussing a number of
issues toward getting the consensus that is believed to be achieved in
version 22.

The description of a latching mechanism to traverse NATs and firewalls
was discussed and decided to be removed from the actual document in
favor of standard NAT traversal techniques such as ICE/STUN/TURN.

Document Quality

At least four implementations of this functionality are known. Four
other vendors have indicated they plan to implement it.

The document was WGLCed on version 16. Additionally, the PAYLOAD WG
also conducted a thorough review of the same 16 version. As a result of
these reviews, 34 changes were requested by, among others, Tolga
Asveren, Dan Wing, Emil Ivov, Flemming Andreasen, and Miguel A. Garcia.
These comments were subsequently introduced into versions 17 and 18. A
new WGLC was issued on version 18, bringing a new set of comments by
Flemming Andreasen, Magnus Westerlund, and Gunnar Hellstrom. These were
addressed in version 19.

At least Magnus Westerlund and Flemming Andreasen did a thorough review
of the document as part of the WGLC on version 18.

Version 18 was sent to the ietf-types mailing list for review on April
4, 2012. Bjorern Hoehrmann and Magnus Westerlund posted comments that
were addressed in version 19.

Personnel

Who is the Document Shepherd? Who is the Responsible Area
Director?

Miguel A. Garcia is the Document Shepherd. Gonzalo Camarillo is the
Responsible Area Director 

From pkyzivat@alum.mit.edu  Tue Jan 15 08:39:09 2013
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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [MMUSIC] Protocol Action: 'An Extension to the Session Description Protocol (SDP) and Real-time Transport Protocol (RTP) for Media Loopback' to Proposed Standard (draft-ietf-mmusic-media-loopback-27.txt)
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Wow - finally!

-00 was in December 2004!. Not a record, but up there.

	Paul

On 1/15/13 10:45 AM, The IESG wrote:
> The IESG has approved the following document:
> - 'An Extension to the Session Description Protocol (SDP) and Real-time
>     Transport Protocol (RTP) for Media Loopback'
>    (draft-ietf-mmusic-media-loopback-27.txt) as Proposed Standard
>
> This document is the product of the Multiparty Multimedia Session Control
> Working Group.
>
> The IESG contact persons are Gonzalo Camarillo and Robert Sparks.
>
> A URL of this Internet Draft is:
> http://datatracker.ietf.org/doc/draft-ietf-mmusic-media-loopback/
>
>
>
>
> Technical Summary
>
> This document provides extensions to the Session Description Protocol
> (SDP) and the Real-time Transport Protocol (RTP) for enabling looping
> of sent media. Media loopback is especially popular in ensuring the
> quality of transport to the edge of a given VoIP, Real-time Text or
> Video over IP service. The extension defined herein adds new SDP media
> types and attributes, which enable establishment of media sessions
> where the media is looped back to the transmitter. Such media sessions
> will serve as monitoring and troubleshooting tools by providing the
> means for measurement of more advanced VoIP, Real-time Text and Video
> over IP performance metrics.
>
> Working Group Summary
>
> The first version of the document was available at the end of 2004.
> Since that time, the working group has been discussing a number of
> issues toward getting the consensus that is believed to be achieved in
> version 22.
>
> The description of a latching mechanism to traverse NATs and firewalls
> was discussed and decided to be removed from the actual document in
> favor of standard NAT traversal techniques such as ICE/STUN/TURN.
>
> Document Quality
>
> At least four implementations of this functionality are known. Four
> other vendors have indicated they plan to implement it.
>
> The document was WGLCed on version 16. Additionally, the PAYLOAD WG
> also conducted a thorough review of the same 16 version. As a result of
> these reviews, 34 changes were requested by, among others, Tolga
> Asveren, Dan Wing, Emil Ivov, Flemming Andreasen, and Miguel A. Garcia.
> These comments were subsequently introduced into versions 17 and 18. A
> new WGLC was issued on version 18, bringing a new set of comments by
> Flemming Andreasen, Magnus Westerlund, and Gunnar Hellstrom. These were
> addressed in version 19.
>
> At least Magnus Westerlund and Flemming Andreasen did a thorough review
> of the document as part of the WGLC on version 18.
>
> Version 18 was sent to the ietf-types mailing list for review on April
> 4, 2012. Bjorern Hoehrmann and Magnus Westerlund posted comments that
> were addressed in version 19.
>
> Personnel
>
> Who is the Document Shepherd? Who is the Responsible Area
> Director?
>
> Miguel A. Garcia is the Document Shepherd. Gonzalo Camarillo is the
> Responsible Area Director
>


From brian.rosen@neustar.biz  Tue Jan 15 08:48:06 2013
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From: "Rosen, Brian" <Brian.Rosen@neustar.biz>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
Date: Tue, 15 Jan 2013 11:48:01 -0500
Thread-Topic: [MMUSIC] Protocol Action: 'An Extension to the Session Description Protocol (SDP) and Real-time Transport Protocol (RTP) for Media Loopback' to Proposed Standard (draft-ietf-mmusic-media-loopback-27.txt)
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Thank you Hadriel for finally getting this finished!

Brian
On Jan 15, 2013, at 11:39 AM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> Wow - finally!
>=20
> -00 was in December 2004!. Not a record, but up there.
>=20
> 	Paul
>=20
> On 1/15/13 10:45 AM, The IESG wrote:
>> The IESG has approved the following document:
>> - 'An Extension to the Session Description Protocol (SDP) and Real-time
>>    Transport Protocol (RTP) for Media Loopback'
>>   (draft-ietf-mmusic-media-loopback-27.txt) as Proposed Standard
>>=20
>> This document is the product of the Multiparty Multimedia Session Contro=
l
>> Working Group.
>>=20
>> The IESG contact persons are Gonzalo Camarillo and Robert Sparks.
>>=20
>> A URL of this Internet Draft is:
>> http://datatracker.ietf.org/doc/draft-ietf-mmusic-media-loopback/
>>=20
>>=20
>>=20
>>=20
>> Technical Summary
>>=20
>> This document provides extensions to the Session Description Protocol
>> (SDP) and the Real-time Transport Protocol (RTP) for enabling looping
>> of sent media. Media loopback is especially popular in ensuring the
>> quality of transport to the edge of a given VoIP, Real-time Text or
>> Video over IP service. The extension defined herein adds new SDP media
>> types and attributes, which enable establishment of media sessions
>> where the media is looped back to the transmitter. Such media sessions
>> will serve as monitoring and troubleshooting tools by providing the
>> means for measurement of more advanced VoIP, Real-time Text and Video
>> over IP performance metrics.
>>=20
>> Working Group Summary
>>=20
>> The first version of the document was available at the end of 2004.
>> Since that time, the working group has been discussing a number of
>> issues toward getting the consensus that is believed to be achieved in
>> version 22.
>>=20
>> The description of a latching mechanism to traverse NATs and firewalls
>> was discussed and decided to be removed from the actual document in
>> favor of standard NAT traversal techniques such as ICE/STUN/TURN.
>>=20
>> Document Quality
>>=20
>> At least four implementations of this functionality are known. Four
>> other vendors have indicated they plan to implement it.
>>=20
>> The document was WGLCed on version 16. Additionally, the PAYLOAD WG
>> also conducted a thorough review of the same 16 version. As a result of
>> these reviews, 34 changes were requested by, among others, Tolga
>> Asveren, Dan Wing, Emil Ivov, Flemming Andreasen, and Miguel A. Garcia.
>> These comments were subsequently introduced into versions 17 and 18. A
>> new WGLC was issued on version 18, bringing a new set of comments by
>> Flemming Andreasen, Magnus Westerlund, and Gunnar Hellstrom. These were
>> addressed in version 19.
>>=20
>> At least Magnus Westerlund and Flemming Andreasen did a thorough review
>> of the document as part of the WGLC on version 18.
>>=20
>> Version 18 was sent to the ietf-types mailing list for review on April
>> 4, 2012. Bjorern Hoehrmann and Magnus Westerlund posted comments that
>> were addressed in version 19.
>>=20
>> Personnel
>>=20
>> Who is the Document Shepherd? Who is the Responsible Area
>> Director?
>>=20
>> Miguel A. Garcia is the Document Shepherd. Gonzalo Camarillo is the
>> Responsible Area Director
>>=20
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From harald@alvestrand.no  Wed Jan 16 04:11:20 2013
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Subject: Re: [MMUSIC] Proposed agenda for the joint MMUSIC/RTCWEB inteirm meeting
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Thanks for getting this out!

This is basically 2 days for grouping in SDP (with a slice off for data 
channels), and 1 day for trickle ICE and all the other issues in JSEP.

I would be happy if the chairs could update this draft agenda with links 
to the current drafts that pertain to each topic - in particular, that 
will give me some hints on what drafts I need to refresh before they 
expire; this discussion has been going on for quite some time!

                  Harald

On 01/14/2013 06:41 PM, Ted Hardie wrote:
> Below is a proposed agenda which has been discussed among the RTCWEB
> and MMUSIC chairs; comments welcome.
>
> Ted Hardie
>
>
> Day 1 (February 5th, 2013)
>
> Multiple Flow SDP representation
>
>         90 minutes: Framing discussion (Magnus is working on text)
>
>        120 minutes: drafts discussion
>
>
>        Objective: decide on architecture for large numbers of similar streams
>
> SDP for Data channels
>
>          60 minutes:  resolve open issues
>
> Day 2 (February 6, 2013)
>
> SDP Grouping (Bundle/MMT/one-rtp)
>
>             60 Minutes: framing discussion
>
>            120 Minutes:  drafts discussion
>
>             Objective: select an approach and provide direction to editors
>
> Mapping stream/track label concepts to SDP identifiers
>
>             60 minutes
>
>              Objective: determine requirements
>
> Day 3: (February 7, 2013)
>
>              Trickle ICE
>
>             120 minutes
>
>              Objective: review offer/answer semantic impact; establish
> initiation, termination and error handling
>
>              JSEP
>
>               120 minutes
>
>                Objective: review and resolve open issues with the document
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From emcho@sip-communicator.org  Wed Jan 16 05:09:14 2013
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References: <CA+9kkMAB8SG1ojzYDxQJq4TX44qZEkjfYcLutnKbdRwu+3UcbA@mail.gmail.com> <50F698DD.6020505@alvestrand.no>
Date: Wed, 16 Jan 2013 14:09:09 +0100
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From: Emil Ivov <emcho@jitsi.org>
To: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [MMUSIC] Proposed agenda for the joint MMUSIC/RTCWEB inteirm meeting
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--047d7b3438ea80eebd04d36795c7
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On Jan 16, 2013 1:11 PM, "Harald Alvestrand" <harald@alvestrand.no> wrote:
>
> Thanks for getting this out!
>
> This is basically 2 days for grouping in SDP (with a slice off for data
> channels), and 1 day for trickle ICE and all the other issues in JSEP.

FWIW, we will be submitting a new trickle ICE revision in the following
days.

Incidentally, are there going to be arrangements for remote
participation/presentation?

Emil

--sent from my mobile

> I would be happy if the chairs could update this draft agenda with
> links to the current drafts that pertain to each topic - in particular,
> that will give me some hints on what drafts I need to refresh before
> they expire; this discussion has been going on for quite some time!
>
>                  Harald
>
>
> On 01/14/2013 06:41 PM, Ted Hardie wrote:
>>
>> Below is a proposed agenda which has been discussed among the RTCWEB
>> and MMUSIC chairs; comments welcome.
>>
>> Ted Hardie
>>
>>
>> Day 1 (February 5th, 2013)
>>
>> Multiple Flow SDP representation
>>
>>         90 minutes: Framing discussion (Magnus is working on text)
>>
>>        120 minutes: drafts discussion
>>
>>
>>        Objective: decide on architecture for large numbers of similar
streams
>>
>> SDP for Data channels
>>
>>          60 minutes:  resolve open issues
>>
>> Day 2 (February 6, 2013)
>>
>> SDP Grouping (Bundle/MMT/one-rtp)
>>
>>             60 Minutes: framing discussion
>>
>>            120 Minutes:  drafts discussion
>>
>>             Objective: select an approach and provide direction to
editors
>>
>> Mapping stream/track label concepts to SDP identifiers
>>
>>             60 minutes
>>
>>              Objective: determine requirements
>>
>> Day 3: (February 7, 2013)
>>
>>              Trickle ICE
>>
>>             120 minutes
>>
>>              Objective: review offer/answer semantic impact; establish
>> initiation, termination and error handling
>>
>>              JSEP
>>
>>               120 minutes
>>
>>                Objective: review and resolve open issues with the
document
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

--047d7b3438ea80eebd04d36795c7
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<p dir=3D"ltr">On Jan 16, 2013 1:11 PM, &quot;Harald Alvestrand&quot; &lt;<=
a href=3D"mailto:harald@alvestrand.no">harald@alvestrand.no</a>&gt; wrote:<=
br>
&gt;<br>
&gt; Thanks for getting this out!<br>
&gt;<br>
&gt; This is basically 2 days for grouping in SDP (with a slice off for dat=
a<br>
&gt; channels), and 1 day for trickle ICE and all the other issues in JSEP.=
</p>
<p dir=3D"ltr">FWIW, we will be submitting a new trickle ICE revision in th=
e following days.</p>
<p dir=3D"ltr">Incidentally, are there going to be arrangements for remote =
participation/presentation?</p>
<p dir=3D"ltr">Emil</p>
<p dir=3D"ltr">--sent from my mobile</p>
<p dir=3D"ltr">&gt; I would be happy if the chairs could update this draft =
agenda with<br>
&gt; links to the current drafts that pertain to each topic - in particular=
,<br>
&gt; that will give me some hints on what drafts I need to refresh before<b=
r>
&gt; they expire; this discussion has been going on for quite some time!<br=
>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Harald<br>
&gt;<br>
&gt;<br>
&gt; On 01/14/2013 06:41 PM, Ted Hardie wrote:<br>
&gt;&gt;<br>
&gt;&gt; Below is a proposed agenda which has been discussed among the RTCW=
EB<br>
&gt;&gt; and MMUSIC chairs; comments welcome.<br>
&gt;&gt;<br>
&gt;&gt; Ted Hardie<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; Day 1 (February 5th, 2013)<br>
&gt;&gt;<br>
&gt;&gt; Multiple Flow SDP representation<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 =A0 =A0 90 minutes: Framing discussion (Magnus is working =
on text)<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 =A0 =A0120 minutes: drafts discussion<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 =A0 =A0Objective: decide on architecture for large numbers=
 of similar streams<br>
&gt;&gt;<br>
&gt;&gt; SDP for Data channels<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A060 minutes: =A0resolve open issues<br>
&gt;&gt;<br>
&gt;&gt; Day 2 (February 6, 2013)<br>
&gt;&gt;<br>
&gt;&gt; SDP Grouping (Bundle/MMT/one-rtp)<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 =A0 60 Minutes: framing discussion<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 =A0120 Minutes: =A0drafts discussion<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 =A0 Objective: select an approach and provide =
direction to editors<br>
&gt;&gt;<br>
&gt;&gt; Mapping stream/track label concepts to SDP identifiers<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 =A0 60 minutes<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0Objective: determine requirements<br>
&gt;&gt;<br>
&gt;&gt; Day 3: (February 7, 2013)<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0Trickle ICE<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 =A0 120 minutes<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0Objective: review offer/answer semantic=
 impact; establish<br>
&gt;&gt; initiation, termination and error handling<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0JSEP<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 120 minutes<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Objective: review and resolve open =
issues with the document<br>
&gt;&gt; _______________________________________________<br>
&gt;&gt; mmusic mailing list<br>
&gt;&gt; <a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/mmusic">https://w=
ww.ietf.org/mailman/listinfo/mmusic</a><br>
&gt;<br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; mmusic mailing list<br>
&gt; <a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/mmusic">https://www.i=
etf.org/mailman/listinfo/mmusic</a><br>
</p>

--047d7b3438ea80eebd04d36795c7--

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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "mmusic@ietf.org WG" <mmusic@ietf.org>
Thread-Topic: draft-reddy-mmusic-stun-auth-fw-traversal-00
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Why is this draft in music - it seems like a STUN extension. (I understand =
an AD might have sent it here but I'm still not getting why?)



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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>, "mmusic@ietf.org WG" <mmusic@ietf.org>
Thread-Topic: draft-reddy-mmusic-stun-auth-fw-traversal-00
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Subject: Re: [MMUSIC] draft-reddy-mmusic-stun-auth-fw-traversal-00
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The draft was initially submitted to RTCWEB since it describes a WebRTC use=
-case. The authors were informed by the AD that all the ICE extensions, inc=
luding all RTCWEB-related ICE extensions, will be done in MMUSIC and the dr=
aft needs to be discussed in MMUSIC.

Is BEHAVE a better option?

Muthu

|-----Original Message-----
|From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf O=
f Cullen Jennings (fluffy)
|Sent: Thursday, January 17, 2013 12:10 PM
|To: mmusic@ietf.org WG
|Subject: [MMUSIC] draft-reddy-mmusic-stun-auth-fw-traversal-00
|
|
|Why is this draft in music - it seems like a STUN extension. (I understand=
 an AD might have sent it
|here but I'm still not getting why?)
|
|
|_______________________________________________
|mmusic mailing list
|mmusic@ietf.org
|https://www.ietf.org/mailman/listinfo/mmusic

From tireddy@cisco.com  Thu Jan 17 03:19:57 2013
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From: "Tirumaleswar Reddy (tireddy)" <tireddy@cisco.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: New Version Notification for draft-wing-mmusic-ice-mobility-03.txt
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Date: Thu, 17 Jan 2013 11:19:55 +0000
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Subject: [MMUSIC] FW: New Version Notification for draft-wing-mmusic-ice-mobility-03.txt
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From iesg-secretary@ietf.org  Fri Jan 18 07:31:47 2013
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Subject: [MMUSIC] Last Call: <draft-ietf-mmusic-sdp-cs-17.txt> (Session Description	Protocol (SDP) Extension For Setting Up Audio and Video Media	Streams Over Circuit-Switched Bearers In The Public Switched	Telephone Network (PSTN)) to Proposed Standard
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The IESG has received a request from the Multiparty Multimedia Session
Control WG (mmusic) to consider the following document:
- 'Session Description Protocol (SDP) Extension For Setting Up Audio and
   Video Media Streams Over Circuit-Switched Bearers In The Public
   Switched Telephone Network (PSTN)'
  <draft-ietf-mmusic-sdp-cs-17.txt> as Proposed Standard

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action. Please send substantive comments to the
ietf@ietf.org mailing lists by 2013-02-01. Exceptionally, comments may be
sent to iesg@ietf.org instead. In either case, please retain the
beginning of the Subject line to allow automated sorting.

Abstract


   This memo describes use cases, requirements, and protocol extensions
   for using the Session Description Protocol (SDP) Offer/Answer model
   for establishing audio and video media streams over circuit-switched
   bearers in the Public Switched Telephone Network (PSTN).




The file can be obtained via
http://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-cs/

IESG discussion can be tracked via
http://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-cs/ballot/


No IPR declarations have been submitted directly on this I-D.



From btv1==7306b30e559==HKaplan@acmepacket.com  Fri Jan 18 11:33:27 2013
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>, "mmusic@ietf.org (E-mail)" <mmusic@ietf.org>, "public-media-capture@w3.org" <public-media-capture@w3.org>
Thread-Topic: February Interim registration and logistics
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Cc: Dominique Hazael-Massieux <dom@w3.org>, "rai-ads@tools.ietf.org" <rai-ads@tools.ietf.org>, "rtcweb-chairs@tools.ietf.org" <rtcweb-chairs@tools.ietf.org>, =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>, Harald Alvestrand <hta@google.com>, "<mmusic-chairs@tools.ietf.org>" <mmusic-chairs@tools.ietf.org>
Subject: [MMUSIC] February Interim registration and logistics
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Howdy,
We're pleased to host the upcoming interim meetings of IETF RTCWeb, IETF MM=
USIC, W3C WebRTC and W3C Media Capture groups, on February 5-7, 2013, at ou=
r Corporate Galactic Headquarters in Bedford, Massachusetts.=20

Please register at the following page if you plan to attend:
http://info.acmepacket.com/acme-packet-hosting-ietf-and-w3c

The logistics/venue info page is here:
http://info.acmepacket.com/acme-packet-hosting-ietf-and-w3c/logistics

You should NOT register if you're only going to attend remotely.  Details f=
or remote participation will be sent out later on the relevant mailing list=
s.

We hope to see you here!
And bring warm clothing - it's slightly chilly. ;)

-hadriel

p.s. We've tested the registration page on several browsers and it works ge=
nerally, except for on Lynx, Links, or Elinks. (Lynx and Links because they=
 don't support Javascript, but why it doesn't work on Elinks w/SpiderMonkey=
 I haven't looked into)  The logistics page seems to work on all browsers.=
=20
:)


From internet-drafts@ietf.org  Mon Jan 21 05:53:25 2013
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-03.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Multiparty Multimedia Session Control Wor=
king Group of the IETF.

	Title           : Stream Control Transmission Protocol (SCTP)-Based Media =
Transport in the Session Description Protocol (SDP)
	Author(s)       : Salvatore Loreto
                          Gonzalo Camarillo
	Filename        : draft-ietf-mmusic-sctp-sdp-03.txt
	Pages           : 14
	Date            : 2013-01-21

Abstract:
   SCTP (Stream Control Transmission Protocol) is a transport protocol
   used to establish associations between two endpoints.  This document
   describes how to express media transport over SCTP in SDP (Session
   Description Protocol).  This document defines the 'SCTP', 'SCTP/DTLS'
   and 'DTLS/SCTP' protocol identifiers for SDP.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-sctp-sdp

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-03

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sctp-sdp-03


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


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Subject: Re: [MMUSIC] Proposed agenda for the joint MMUSIC/RTCWEB inteirm meeting
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Hi,
I am not sure what will be discussed in multiple flow SDP representation =
and
Mapping stream/track label concepts to SDP identifiers. Are there drafts =
or
is it an open discussion.
What about the relation of these two topics to CLUE?

Thanks
Roni Even


-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf =
Of
Ted Hardie
Sent: 14 January, 2013 7:41 PM
To: rtcweb@ietf.org; mmusic@ietf.org
Cc: Cullen Jennings; Magnus Westerlund; Ari Ker=E4nen; Gonzalo =
Camarillo;
Flemming Andreasen
Subject: [MMUSIC] Proposed agenda for the joint MMUSIC/RTCWEB inteirm
meeting

Below is a proposed agenda which has been discussed among the RTCWEB and
MMUSIC chairs; comments welcome.

Ted Hardie


Day 1 (February 5th, 2013)

Multiple Flow SDP representation

       90 minutes: Framing discussion (Magnus is working on text)

      120 minutes: drafts discussion


      Objective: decide on architecture for large numbers of similar =
streams

SDP for Data channels

        60 minutes:  resolve open issues

Day 2 (February 6, 2013)

SDP Grouping (Bundle/MMT/one-rtp)

           60 Minutes: framing discussion

          120 Minutes:  drafts discussion

           Objective: select an approach and provide direction to =
editors

Mapping stream/track label concepts to SDP identifiers

           60 minutes

            Objective: determine requirements

Day 3: (February 7, 2013)

            Trickle ICE

           120 minutes

            Objective: review offer/answer semantic impact; establish
initiation, termination and error handling

            JSEP

             120 minutes

              Objective: review and resolve open issues with the =
document
_______________________________________________
mmusic mailing list
mmusic@ietf.org
https://www.ietf.org/mailman/listinfo/mmusic


From magnus.westerlund@ericsson.com  Tue Jan 22 00:16:10 2013
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References: <CA+9kkMAB8SG1ojzYDxQJq4TX44qZEkjfYcLutnKbdRwu+3UcbA@mail.gmail.com> <1.e0d2fbab2393f09b6fda@gmail.com>
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Subject: Re: [MMUSIC] Proposed agenda for the joint MMUSIC/RTCWEB inteirmmeeting
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On 2013-01-22 06:49, Roni Even wrote:
> Hi,
> I am not sure what will be discussed in multiple flow SDP representation
> and

This is a very fundamental choice of how SDP is going to deal with RTP
sessions that include multiple media streams, i.e. SSRCs carrying
different media sources and negotiation of media stream related
information and parameters.

I understand some drafts are under preparation here.

This is definitely something that will impact CLUE also.

> Mapping stream/track label concepts to SDP identifiers. Are there drafts or
> is it an open discussion.

This is more an WebRTC focused issues, but it has clearly overlap with
the relations discussion on how you identify media sources, different
encodings of the same media stream (in simulcast and scalable coding)
their repair streams all the way up to WebRTC specific concepts of
MediaStream.

For this my colleagues and I are preparing a draft. I don't expect it to
be available before second half of next week.

I think this is relevant for CLUE in the perspective of trying to
understand what is common abstractions and what is not.

Cheers

Magnus

> What about the relation of these two topics to CLUE?
> 
> Thanks
> Roni Even
> 
> 
> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of
> Ted Hardie
> Sent: 14 January, 2013 7:41 PM
> To: rtcweb@ietf.org; mmusic@ietf.org
> Cc: Cullen Jennings; Magnus Westerlund; Ari Keränen; Gonzalo Camarillo;
> Flemming Andreasen
> Subject: [MMUSIC] Proposed agenda for the joint MMUSIC/RTCWEB inteirm
> meeting
> 
> Below is a proposed agenda which has been discussed among the RTCWEB and
> MMUSIC chairs; comments welcome.
> 
> Ted Hardie
> 
> 
> Day 1 (February 5th, 2013)
> 
> Multiple Flow SDP representation
> 
> 90 minutes: Framing discussion (Magnus is working on text)
> 
> 120 minutes: drafts discussion
> 
> 
> Objective: decide on architecture for large numbers of similar streams
> 
> SDP for Data channels
> 
> 60 minutes: resolve open issues
> 
> Day 2 (February 6, 2013)
> 
> SDP Grouping (Bundle/MMT/one-rtp)
> 
> 60 Minutes: framing discussion
> 
> 120 Minutes: drafts discussion
> 
> Objective: select an approach and provide direction to editors
> 
> Mapping stream/track label concepts to SDP identifiers
> 
> 60 minutes
> 
> Objective: determine requirements
> 
> Day 3: (February 7, 2013)
> 
> Trickle ICE
> 
> 120 minutes
> 
> Objective: review offer/answer semantic impact; establish
> initiation, termination and error handling
> 
> JSEP
> 
> 120 minutes
> 
> Objective: review and resolve open issues with the document
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


-- 

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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Thread-Topic: [MMUSIC] Proposed agenda for the joint MMUSIC/RTCWEB inteirm meeting
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Subject: Re: [MMUSIC] Proposed agenda for the joint MMUSIC/RTCWEB inteirm	meeting
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I think it would be better if the JSEP discussion could take place on the ei=
ther day 1 or day 2 so there is some time for offline discussion before the=
 end of the of the meeting. I think there are some important concerns here w=
hich we need to reach resolution on.

What are the times for the start and end schedule each day?

Adding up the hours it seems that it should be possible to fit more into Day=
 1 and Day 2 and still only do a full "IETF work day".

Andrew

-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 Roni Even
Sent: Tuesday, January 22, 2013 12:49 AM
To: 'Ted Hardie'; rtcweb@ietf.org; mmusic@ietf.org
Cc: 'Cullen Jennings'; 'Magnus Westerlund'; 'Ari Ker=E4nen'; 'Gonzalo Camari=
llo'; 'Flemming Andreasen'
Subject: Re: [MMUSIC] Proposed agenda for the joint MMUSIC/RTCWEB inteirm me=
eting

Hi,
I am not sure what will be discussed in multiple flow SDP representation and=
 Mapping stream/track label concepts to SDP identifiers. Are there drafts or=
 is it an open discussion.
What about the relation of these two topics to CLUE?

Thanks
Roni Even


-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 Ted Hardie
Sent: 14 January, 2013 7:41 PM
To: rtcweb@ietf.org; mmusic@ietf.org
Cc: Cullen Jennings; Magnus Westerlund; Ari Ker=E4nen; Gonzalo Camarillo; Fl=
emming Andreasen
Subject: [MMUSIC] Proposed agenda for the joint MMUSIC/RTCWEB inteirm meetin=
g

Below is a proposed agenda which has been discussed among the RTCWEB and MMU=
SIC chairs; comments welcome.

Ted Hardie


Day 1 (February 5th, 2013)

Multiple Flow SDP representation

       90 minutes: Framing discussion (Magnus is working on text)

      120 minutes: drafts discussion


      Objective: decide on architecture for large numbers of similar streams

SDP for Data channels

        60 minutes:  resolve open issues

Day 2 (February 6, 2013)

SDP Grouping (Bundle/MMT/one-rtp)

           60 Minutes: framing discussion

          120 Minutes:  drafts discussion

           Objective: select an approach and provide direction to editors

Mapping stream/track label concepts to SDP identifiers

           60 minutes

            Objective: determine requirements

Day 3: (February 7, 2013)

            Trickle ICE

           120 minutes

            Objective: review offer/answer semantic impact; establish initia=
tion, termination and error handling

            JSEP

             120 minutes

              Objective: review and resolve open issues with the document __=
_____________________________________________
mmusic mailing list
mmusic@ietf.org
https://www.ietf.org/mailman/listinfo/mmusic

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

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Subject: Re: [MMUSIC] Proposed agenda for the joint MMUSIC/RTCWEB inteirmmeeting
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Hi Magnus,
My concern is that it is not easy to provide a proposal to a subject =
that is
not clear. I also have views about these topics but seeing no =
requirements
or use cases it is difficult to write a draft.
I believe that the actual syntax depends on the bundle decision so any =
work
now need to assume a bundle architecture.
Are there any requirements for a solution even if we are looking for a =
new
way or to leverage existing mechanisms like RFC 5576.

I will try to write something based on the CLUE RTP mapping draft we are
discussing that tries to address a similar problem as far as I =
understand.

BTW: will these drafts be MMUSIC drafts or RTCweb?

Thanks
Roni Even

-----Original Message-----
From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]=20
Sent: 22 January, 2013 10:16 AM
To: Roni Even
Cc: 'Ted Hardie'; rtcweb@ietf.org; mmusic@ietf.org; 'Cullen Jennings'; =
'Ari
Ker=E4nen'; 'Gonzalo Camarillo'; 'Flemming Andreasen'
Subject: Re: [MMUSIC] Proposed agenda for the joint MMUSIC/RTCWEB
inteirmmeeting

On 2013-01-22 06:49, Roni Even wrote:
> Hi,
> I am not sure what will be discussed in multiple flow SDP=20
> representation and

This is a very fundamental choice of how SDP is going to deal with RTP
sessions that include multiple media streams, i.e. SSRCs carrying =
different
media sources and negotiation of media stream related information and
parameters.

I understand some drafts are under preparation here.

This is definitely something that will impact CLUE also.

> Mapping stream/track label concepts to SDP identifiers. Are there=20
> drafts or is it an open discussion.

This is more an WebRTC focused issues, but it has clearly overlap with =
the
relations discussion on how you identify media sources, different =
encodings
of the same media stream (in simulcast and scalable coding) their repair
streams all the way up to WebRTC specific concepts of MediaStream.

For this my colleagues and I are preparing a draft. I don't expect it to =
be
available before second half of next week.

I think this is relevant for CLUE in the perspective of trying to =
understand
what is common abstractions and what is not.

Cheers

Magnus

> What about the relation of these two topics to CLUE?
>=20
> Thanks
> Roni Even
>=20
>=20
> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On=20
> Behalf Of Ted Hardie
> Sent: 14 January, 2013 7:41 PM
> To: rtcweb@ietf.org; mmusic@ietf.org
> Cc: Cullen Jennings; Magnus Westerlund; Ari Ker=E4nen; Gonzalo=20
> Camarillo; Flemming Andreasen
> Subject: [MMUSIC] Proposed agenda for the joint MMUSIC/RTCWEB inteirm=20
> meeting
>=20
> Below is a proposed agenda which has been discussed among the RTCWEB=20
> and MMUSIC chairs; comments welcome.
>=20
> Ted Hardie
>=20
>=20
> Day 1 (February 5th, 2013)
>=20
> Multiple Flow SDP representation
>=20
> 90 minutes: Framing discussion (Magnus is working on text)
>=20
> 120 minutes: drafts discussion
>=20
>=20
> Objective: decide on architecture for large numbers of similar streams
>=20
> SDP for Data channels
>=20
> 60 minutes: resolve open issues
>=20
> Day 2 (February 6, 2013)
>=20
> SDP Grouping (Bundle/MMT/one-rtp)
>=20
> 60 Minutes: framing discussion
>=20
> 120 Minutes: drafts discussion
>=20
> Objective: select an approach and provide direction to editors
>=20
> Mapping stream/track label concepts to SDP identifiers
>=20
> 60 minutes
>=20
> Objective: determine requirements
>=20
> Day 3: (February 7, 2013)
>=20
> Trickle ICE
>=20
> 120 minutes
>=20
> Objective: review offer/answer semantic impact; establish initiation,=20
> termination and error handling
>=20
> JSEP
>=20
> 120 minutes
>=20
> Objective: review and resolve open issues with the document=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


--=20

Magnus Westerlund

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


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Subject: Re: [MMUSIC] Proposed agenda for the joint MMUSIC/RTCWEB inteirmmeeting
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On 2013-01-22 11:41, Roni Even wrote:
> Hi Magnus,
> My concern is that it is not easy to provide a proposal to a subject that is
> not clear. I also have views about these topics but seeing no requirements
> or use cases it is difficult to write a draft.
> I believe that the actual syntax depends on the bundle decision so any work
> now need to assume a bundle architecture.

And I think the question of how one deals with multiple media streams in
an RTP session is an important input into the BUNDLE discussion.

> Are there any requirements for a solution even if we are looking for a new
> way or to leverage existing mechanisms like RFC 5576.

To my understanding what we are trying to achieve is making progress
into three different subjects.

1) How multiple media streams are handled in SDP. I think there exist
two proposal from the MMUSIC discussion:
  A) One media description per RTP session and media type. Each media
     stream are identified and provided with additional information
     using stream specific parameters.

  B) Each media description is a media stream and RTP sessions are then
     expressed as Bundle of media descriptions.

2) How "Bundle" is going to be done. Which of the three proposals do we
   really believe in. Note that direction of 1) will affect how commonly
   required the usage of "Bundle" will be. Thus affect the trade-off
   concerns.

3) What is common relations exist between concepts in WebRTC, CLUE and
other multi-media communication contexts that needs to be identified and
signalled and at what scopes. This influences the solution for
MediaStream identities, our SRCNAME proposal in AVTEXT and CLUE work.
There are a number of commonalities here that should be considered.


> 
> I will try to write something based on the CLUE RTP mapping draft we are
> discussing that tries to address a similar problem as far as I understand.

Please do, the more informed everyone is, the better.

> 
> BTW: will these drafts be MMUSIC drafts or RTCweb?
> 

I don't know . I think ours is likely MMUSIC. I can't answer for the
other contributions I expect. I guess the important is that all
contributions are announced to the relevant WGs.

Cheers

Magnus Westerlund

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


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Subject: Re: [MMUSIC] Proposed agenda for the joint MMUSIC/RTCWEB inteirmmeeting
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Hi Magnus,
Thanks, I think that the first subject is what we were also discussing =
in
CLUE when looking at RTP mapping and we considered the also the =
individual
drafts on maxssrc, and srcname.
As for the two options I am not sure that we must chose. The first one =
"A"
is the current practice and I agree it is under specified today.
The second on "B" depends on how we define bundle. Looking at option "B" =
was
suggested also for RTP mapping in CLUE (for example with regards to
simulcast)

Roni=20

-----Original Message-----
From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]=20
Sent: 22 January, 2013 1:01 PM
To: Roni Even
Cc: 'Ted Hardie'; rtcweb@ietf.org; mmusic@ietf.org; 'Cullen Jennings'; =
'Ari
Ker=E4nen'; 'Gonzalo Camarillo'; 'Flemming Andreasen'
Subject: Re: [MMUSIC] Proposed agenda for the joint MMUSIC/RTCWEB
inteirmmeeting

On 2013-01-22 11:41, Roni Even wrote:
> Hi Magnus,
> My concern is that it is not easy to provide a proposal to a subject=20
> that is not clear. I also have views about these topics but seeing no=20
> requirements or use cases it is difficult to write a draft.
> I believe that the actual syntax depends on the bundle decision so any =

> work now need to assume a bundle architecture.

And I think the question of how one deals with multiple media streams in =
an
RTP session is an important input into the BUNDLE discussion.

> Are there any requirements for a solution even if we are looking for a =

> new way or to leverage existing mechanisms like RFC 5576.

To my understanding what we are trying to achieve is making progress =
into
three different subjects.

1) How multiple media streams are handled in SDP. I think there exist =
two
proposal from the MMUSIC discussion:
  A) One media description per RTP session and media type. Each media
     stream are identified and provided with additional information
     using stream specific parameters.

  B) Each media description is a media stream and RTP sessions are then
     expressed as Bundle of media descriptions.

2) How "Bundle" is going to be done. Which of the three proposals do we
   really believe in. Note that direction of 1) will affect how commonly
   required the usage of "Bundle" will be. Thus affect the trade-off
   concerns.

3) What is common relations exist between concepts in WebRTC, CLUE and =
other
multi-media communication contexts that needs to be identified and =
signalled
and at what scopes. This influences the solution for MediaStream =
identities,
our SRCNAME proposal in AVTEXT and CLUE work.
There are a number of commonalities here that should be considered.


>=20
> I will try to write something based on the CLUE RTP mapping draft we=20
> are discussing that tries to address a similar problem as far as I
understand.

Please do, the more informed everyone is, the better.

>=20
> BTW: will these drafts be MMUSIC drafts or RTCweb?
>=20

I don't know . I think ours is likely MMUSIC. I can't answer for the =
other
contributions I expect. I guess the important is that all contributions =
are
announced to the relevant WGs.

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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=20
I am sorry to say that the treatment of attributes in this I-D is a bit of =
a mess.=20

- In 4.3 the attributes are introduced with a colon-notation, e.g. "a=3Dstr=
eam:", but the ABNF definitions in 5.2 through 5.7 use a notation "a=3Dstre=
am=3D" etc. The colon-notation is the more common one for SDP attributes.

- Looking at the examples, I suspect that 'sctpmap' is the only real attrib=
ute and the rest is meant to be parameters of an attribute 'webrtc-DataChan=
nel', which appears in examples but is not defined in section 5. If this is=
 correct then 'webrtc-DataChannel' needs to be added as a new attribute in =
section 5, and what is now in 5.2 - 5.7 should become sub-sections of this =
attribute definition and be renamed as "parameters".

- Section 11 says "This document defines two SDP session and media-level at=
tributes", but lists only one (sctpmap). Was the missing one meant to be 'w=
ebrtc-DataChannel'?

Note: An SDP attribute 'label' is already defined (RFC 4574) and listed in =
the IANA registry. This would cause a name clash if 5.3 is really intended =
as attribute and not as parameter definition.

Thanks,
Ernst Horvath =

From Simo.Veikkolainen@nokia.com  Fri Jan 25 00:17:40 2013
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From: <Simo.Veikkolainen@nokia.com>
To: <mmusic@ietf.org>
Thread-Topic: SDP Miscellaneous capabilities: port numbers
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Date: Fri, 25 Jan 2013 08:17:31 +0000
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Subject: [MMUSIC] SDP Miscellaneous capabilities: port numbers
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Dear WG,

An issue came up during review of http://tools.ietf.org/html/draft-ietf-mmu=
sic-sdp-miscellaneous-caps-03 . Figures 6 to 8 present an example SDP which=
 uses the "a=3Dccap" connection address capability to present two alternati=
ve SDP configurations, one for IP based media, and the other for PSTN based=
 media.

The issue is that these two alternative configurations use a different port=
 number. However, there is currently no way to indicate alternative port nu=
mbers using the SDP capability negotiation mechanism.

For PSTN based media, the port number does not carry any meaningful informa=
tion, and http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-15 proposes t=
o use a fixed port number (9, the discard port) which is to be ignored at r=
eception. So, in this particular example we can add a note that since the a=
lternative offered media uses a PSTN bearer, the port should be hardcoded t=
o "9" by the implementer. However, for other types of media this does not w=
ork, nor is the example usable in the case where PSTN media is included in =
the "m=3D" line, and the address for alternative RTP based media using the =
"a=3Dccap" attribute.

In order to fix the issue, probably something like "port capability attribu=
te" should be defined. This has been discussed in the past, but it has its =
own challenges (because of overlap with ICE, among other things).

The authors' proposal consist of adding a paragraph in the example descript=
ion where we indicate that this particular example works because the CS des=
cription uses a well-known port number, but in general, the example is not =
applicable if the negotiated capabilities require a change in the port numb=
er.

Regards,
Simo & Miguel


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<p class=3D"MsoNormal">Dear WG,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">An issue came up during review of <a href=3D"http://=
tools.ietf.org/html/draft-ietf-mmusic-sdp-miscellaneous-caps-03">
http://tools.ietf.org/html/draft-ietf-mmusic-sdp-miscellaneous-caps-03</a> =
. Figures 6 to 8 present an example SDP which uses the &quot;a=3Dccap&quot;=
 connection address capability to present two alternative SDP configuration=
s, one for IP based media, and the other for
 PSTN based media.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The issue is that these two alternative configuratio=
ns use a different port number. However, there is currently no way to indic=
ate alternative port numbers using the SDP capability negotiation mechanism=
.
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">For PSTN based media, the port number does not carry=
 any meaningful information, and
<a href=3D"http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-15">http://t=
ools.ietf.org/html/draft-ietf-mmusic-sdp-cs-15</a> proposes to use a fixed =
port number (9, the discard port) which is to be ignored at reception. So, =
in this particular example we can
 add a note that since the alternative offered media uses a PSTN bearer, th=
e port should be hardcoded to &quot;9&quot; by the implementer. However, fo=
r other types of media this does not work, nor is the example usable in the=
 case where PSTN media is included in the
 &quot;m=3D&quot; line, and the address for alternative RTP based media usi=
ng the &quot;a=3Dccap&quot; attribute.
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">In order to fix the issue, probably something like &=
quot;port capability attribute&quot; should be defined. This has been discu=
ssed in the past, but it has its own challenges (because of overlap with IC=
E, among other things).
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The authors&#8217; proposal consist of adding a para=
graph in the example description where we indicate that this particular exa=
mple works because the CS description uses a well-known port number, but in=
 general, the example is not applicable
 if the negotiated capabilities require a change in the port number.<o:p></=
o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal">Simo &amp; Miguel<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</body>
</html>

--_000_D09DAE6B636851459F7575D146EFB54B1C826B64008AM1MPN1026mg_--

From prvs=7737eb7039=aallen@rim.com  Fri Jan 25 03:25:34 2013
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From: Andrew Allen <aallen@rim.com>
To: "Simo.Veikkolainen@nokia.com" <Simo.Veikkolainen@nokia.com>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: SDP Miscellaneous capabilities: port numbers
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Date: Fri, 25 Jan 2013 11:25:20 +0000
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Subject: Re: [MMUSIC] SDP Miscellaneous capabilities: port numbers
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I am fine with the proposal.

Andrew

From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 Simo.Veikkolainen@nokia.com
Sent: Friday, January 25, 2013 3:18 AM
To: mmusic@ietf.org
Subject: [MMUSIC] SDP Miscellaneous capabilities: port numbers

Dear WG,

An issue came up during review of http://tools.ietf.org/html/draft-ietf-mmus=
ic-sdp-miscellaneous-caps-03 . Figures 6 to 8 present an example SDP which u=
ses the "a=3Dccap" connection address capability to present two alternative=
 SDP configurations, one for IP based media, and the other for PSTN based me=
dia.

The issue is that these two alternative configurations use a different port=
 number. However, there is currently no way to indicate alternative port num=
bers using the SDP capability negotiation mechanism.

For PSTN based media, the port number does not carry any meaningful informat=
ion, and http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-15 proposes to=
 use a fixed port number (9, the discard port) which is to be ignored at rec=
eption. So, in this particular example we can add a note that since the alte=
rnative offered media uses a PSTN bearer, the port should be hardcoded to "9=
" by the implementer. However, for other types of media this does not work,=
 nor is the example usable in the case where PSTN media is included in the "=
m=3D" line, and the address for alternative RTP based media using the "a=3Dc=
cap" attribute.

In order to fix the issue, probably something like "port capability attribut=
e" should be defined. This has been discussed in the past, but it has its ow=
n challenges (because of overlap with ICE, among other things).

The authors' proposal consist of adding a paragraph in the example descripti=
on where we indicate that this particular example works because the CS descr=
iption uses a well-known port number, but in general, the example is not app=
licable if the negotiated capabilities require a change in the port number.

Regards,
Simo & Miguel


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<p class=3D"MsoNormal"><span style=3D"color:#1F497D">I am fine with the prop=
osal.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></span=
></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Andrew<o:p></o:p></span=
></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></span=
></p>
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0=
in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot;=
Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-siz=
e:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> mmusic-boun=
ces@ietf.org [mailto:mmusic-bounces@ietf.org]
<b>On Behalf Of </b>Simo.Veikkolainen@nokia.com<br>
<b>Sent:</b> Friday, January 25, 2013 3:18 AM<br>
<b>To:</b> mmusic@ietf.org<br>
<b>Subject:</b> [MMUSIC] SDP Miscellaneous capabilities: port numbers<o:p></=
o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Dear WG,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">An issue came up during review of <a href=3D"http://t=
ools.ietf.org/html/draft-ietf-mmusic-sdp-miscellaneous-caps-03">
http://tools.ietf.org/html/draft-ietf-mmusic-sdp-miscellaneous-caps-03</a> .=
 Figures 6 to 8 present an example SDP which uses the &quot;a=3Dccap&quot; c=
onnection address capability to present two alternative SDP configurations,=
 one for IP based media, and the other for
 PSTN based media.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The issue is that these two alternative configuration=
s use a different port number. However, there is currently no way to indicat=
e alternative port numbers using the SDP capability negotiation mechanism.
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">For PSTN based media, the port number does not carry=
 any meaningful information, and
<a href=3D"http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-15">http://to=
ols.ietf.org/html/draft-ietf-mmusic-sdp-cs-15</a> proposes to use a fixed po=
rt number (9, the discard port) which is to be ignored at reception. So, in=
 this particular example we can
 add a note that since the alternative offered media uses a PSTN bearer, the=
 port should be hardcoded to &quot;9&quot; by the implementer. However, for=
 other types of media this does not work, nor is the example usable in the c=
ase where PSTN media is included in the
 &quot;m=3D&quot; line, and the address for alternative RTP based media usin=
g the &quot;a=3Dccap&quot; attribute.
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">In order to fix the issue, probably something like &q=
uot;port capability attribute&quot; should be defined. This has been discuss=
ed in the past, but it has its own challenges (because of overlap with ICE,=
 among other things).
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The authors&#8217; proposal consist of adding a parag=
raph in the example description where we indicate that this particular examp=
le works because the CS description uses a well-known port number, but in ge=
neral, the example is not applicable
 if the negotiated capabilities require a change in the port number.<o:p></o=
:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal">Simo &amp; Miguel<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
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From ted.ietf@gmail.com  Fri Jan 25 08:47:55 2013
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From: Ted Hardie <ted.ietf@gmail.com>
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Subject: [MMUSIC] Start of a reading list for the IETF MMUSIC/RTCWEB interim
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http://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation
http://tools.ietf.org/html/draft-alvestrand-mmusic-msid
http://tools.ietf.org/html/draft-holmberg-mmusic-sdp-mmt-negotiation
http://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp
http://tools.ietf.org/html/draft-rescorla-mmusic-ice-trickle
http://tools.ietf.org/html/draft-ietf-rtcweb-jsep

You  may also want to look at the proceedings from the Atlanta meeting;
in particular, this was suggested as good background reading:
http://www.ietf.org/proceedings/85/slides/slides-85-mmusic-7.ppt

regards,

Ted Hardie

From btv1==7376abf850a==HKaplan@acmepacket.com  Fri Jan 25 12:53:30 2013
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: mmusic <mmusic@ietf.org>
Thread-Topic: [MMUSIC] I-D	Action: draft-ietf-mmusic-media-path-middleboxes-06.txt
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On Jan 10, 2013, at 6:02 PM, Dan Wing <dwing@cisco.com> wrote:

> I am glad this document was resurrected. =20
>=20
> Can the document say SBC instead of the awkward (and inaccurate)=20
> "SIP ALG combined with MIDCOM agent"?  Or is saying SBC impolitic.
>=20

I don't think it can or should say "SBC" - because this doc does not descri=
be what an SBC does.  At least not the first half (section 4), which is rea=
lly about an inline ALG in the router/firewall. (ie, it's not the target/so=
urce of the IP addresses of the packets, doesn't change SIP/SDP, etc.)

Anyway, I read this latest draft, and it looks exactly like the older ones,=
 so I don't see it fixing the issues with the older ones:
1) It claims to describe the impact on ICE, but doesn't
2) Doesn't describe the issues with forked or moved calls on early in-band =
media signaling (especially if you actually comply with recommendations #2+=
#4)
3) Uses terminology from ancient history (ie, RFC 3303) that no one else us=
es
4) Recommendation #5 is backwards I think - it should recommend protocols n=
ot to require/expect to exchange media-plane signaling after the session ha=
s been terminated. (in other words, since you can't rely on middleboxes bei=
ng lenient, don't design a protocol to expect them to be so)=20

-hadriel



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Subject: Re: [MMUSIC] I-D	Action:	draft-ietf-mmusic-media-path-middleboxes-06.txt
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> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf
> Of Hadriel Kaplan
> Sent: Friday, January 25, 2013 12:53 PM
> To: mmusic
> Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-media-path-
> middleboxes-06.txt
> 
> 
> On Jan 10, 2013, at 6:02 PM, Dan Wing <dwing@cisco.com> wrote:
> 
> > I am glad this document was resurrected.
> >
> > Can the document say SBC instead of the awkward (and inaccurate) "SIP
> > ALG combined with MIDCOM agent"?  Or is saying SBC impolitic.
> >
> 
> I don't think it can or should say "SBC" - because this doc does not
> describe what an SBC does.  At least not the first half (section 4),
> which is really about an inline ALG in the router/firewall. (ie, it's
> not the target/source of the IP addresses of the packets, doesn't change
> SIP/SDP, etc.)

Yes, you're right.  Thanks.

> Anyway, I read this latest draft, and it looks exactly like the older
> ones, so I don't see it fixing the issues with the older ones:
> 1) It claims to describe the impact on ICE, but doesn't
> 2) Doesn't describe the issues with forked or moved calls on early in-
> band media signaling (especially if you actually comply with
> recommendations #2+#4)
> 3) Uses terminology from ancient history (ie, RFC 3303) that no one else
> uses

Which is where I got confused.  I thought it was trying to describe
what is sometimes called a 'decomposed SBC', where the B2BUA is 
controlling the media relaying function using a 
protocol-similar-to-MIDCOM.

I agree nobody uses "MIDCOM" or "MIDCOM architecture" to describe
how an ALG within a firewall or an ALG within a NAT (or NAPT) 
functions.

-d


> 4) Recommendation #5 is backwards I think - it should recommend
> protocols not to require/expect to exchange media-plane signaling after
> the session has been terminated. (in other words, since you can't rely
> on middleboxes being lenient, don't design a protocol to expect them to
> be so)
> 
> -hadriel
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From fandreas@cisco.com  Mon Jan 28 09:12:44 2013
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Subject: Re: [MMUSIC] I-D	Action:	draft-ietf-mmusic-media-path-middleboxes-06.txt
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On 1/25/13 3:53 PM, Hadriel Kaplan wrote:
> On Jan 10, 2013, at 6:02 PM, Dan Wing <dwing@cisco.com> wrote:
>
>> I am glad this document was resurrected.
>>
>> Can the document say SBC instead of the awkward (and inaccurate)
>> "SIP ALG combined with MIDCOM agent"?  Or is saying SBC impolitic.
>>
> I don't think it can or should say "SBC" - because this doc does not describe what an SBC does.  At least not the first half (section 4), which is really about an inline ALG in the router/firewall. (ie, it's not the target/source of the IP addresses of the packets, doesn't change SIP/SDP, etc.)
>
> Anyway, I read this latest draft, and it looks exactly like the older ones, so I don't see it fixing the issues with the older ones:
> 1) It claims to describe the impact on ICE, but doesn't
> 2) Doesn't describe the issues with forked or moved calls on early in-band media signaling (especially if you actually comply with recommendations #2+#4)
> 3) Uses terminology from ancient history (ie, RFC 3303) that no one else uses
> 4) Recommendation #5 is backwards I think - it should recommend protocols not to require/expect to exchange media-plane signaling after the session has been terminated. (in other words, since you can't rely on middleboxes being lenient, don't design a protocol to expect them to be so)
Thanks for taking a look and providing your comments Hadriel. As per 
previous discussions, we wanted to first agree on the scope of the 
document and whether there was still value in pursuing such a document. 
As per Gonzalo's e-mail starting this thread, do you have any 
comments/concerns on that part (if so, please let the authors know what 
those are) ?

Once (/if) we get agreement on the scope, we can then get into detail 
around the technical aspects of the document as per your comments above.

Thanks

-- Flemming (MMUSIC co-chair)


> -hadriel
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


From fluffy@cisco.com  Tue Jan 29 08:49:48 2013
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "mmusic@ietf.org WG" <mmusic@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [MMUSIC] Start of a reading list for the IETF MMUSIC/RTCWEB interim
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Subject: Re: [MMUSIC] Start of a reading list for the IETF MMUSIC/RTCWEB interim
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On Jan 25, 2013, at 9:47 AM, Ted Hardie <ted.ietf@gmail.com> wrote:

> http://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation
> http://tools.ietf.org/html/draft-alvestrand-mmusic-msid
> http://tools.ietf.org/html/draft-holmberg-mmusic-sdp-mmt-negotiation
> http://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp
> http://tools.ietf.org/html/draft-rescorla-mmusic-ice-trickle
> http://tools.ietf.org/html/draft-ietf-rtcweb-jsep
>=20
> You  may also want to look at the proceedings from the Atlanta meeting;
> in particular, this was suggested as good background reading:
> http://www.ietf.org/proceedings/85/slides/slides-85-mmusic-7.ppt


Some more background reading that is useful context=20

draft-ietf-rtcweb-rtp-usage-05
draft-ietf-rtcweb-overview-05
draft-ietf-rtcweb-use-cases-and-requirements-10

Thanks,=20
<Many chairs>=20




From emil@sip-communicator.org  Tue Jan 29 09:28:32 2013
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Subject: [MMUSIC] Feedback for draft-ietf-mmusic-latching
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Hey all,

This is just reminder about the latching draft:

http://tools.ietf.org/html/draft-ietf-mmusic-latching

The authors believe that it is solid enough but comments would be
appreciated even if only to point that the text makes sense and does not
present major issues.

Thanks,
Emil


-- 
https://jitsi.org

From stefan.lk.hakansson@ericsson.com  Wed Jan 30 01:51:00 2013
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Subject: [MMUSIC] Boston meeting agenda
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All,

below is a merged agenda covering both W3C and IETF parts of the 
meeting. Details are likely to change - the main message is that W3C 
meetings will be before lunch, and IETF after. Before lunch could mean 
8:30 - 12:30, after lunch 13:30 - 17:30; but the exact times are TBC. A 
hard stop is planned for 17:00 on Feb 7th as many have planes to catch.

(The sources used to assemble the agenda below are the W3C meeting 
wiki's, [1][2], and the mail sent to mmusic and rtcweb Jan 14th [3]).

--Stefan (for all involved chairs)

Feb 5th
=======
Morning session: W3C Media Capture TF
-------------------------------------
* device enumeration
* error handling
* “immediate stream” gUM
* Resource reservation

Afternoon session: IETF mmusic and rtcweb WGs
---------------------------------------------
* Multiple Flow SDP representation
* SDP for Data channels

Feb 6th
=======
Morning session: W3C Media Capture TF + WebRTC WG
-------------------------------------------------
* Identity aspects of MediaStreams (MediaCap)
* Settings/constraints (MediaCap)
* Recording (MediaCap)
* Call flow (WebRTC)

Afternoon session: IETF mmusic and rtcweb WGs
---------------------------------------------
* SDP Grouping (Bundle/MMT/one-rtp)
* Mapping stream/track label concepts to SDP identifiers

Feb 7th
=======
Morning session: W3C WebRTC WG
------------------------------
* MediaStream and MediaStreamTrack - SDP interface
* Error handling
* Data channel
* Stats

Afternoon session: IETF mmusic and rtcweb WGs
---------------------------------------------
* Trickle ICE
* JSEP

[1] http://www.w3.org/wiki/Media_Capture/Feb_5-6_2013
[2] http://www.w3.org/2011/04/webrtc/wiki/February_6_-_February_7_2013
[3] http://www.ietf.org/mail-archive/web/rtcweb/current/msg06069.html





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Date: Wed, 30 Jan 2013 17:38:33 +0100
From: Emil Ivov <emcho@jitsi.org>
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Subject: [MMUSIC] Trickle ICE update and a new SIP usage draft.
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Hey all,

Justin, Eric and I have just submitted an update to the trickle ICE
draft, accommodating comments from Atlanta:

http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00

More details about the changes here:
http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00#appendix-B

Diff2:
http://www.ietf.org/rfcdiff?url1=draft-rescorla-mmusic-ice-trickle-01&difftype=--html&submit=Go!&url2=draft-ivov-mmusic-trickle-ice-00

During the MMUSIC presentation and the mailing list discussions it was
also pointed out that it is important for this work to happen in
parallel with a SIP usage for trickle ICE. Enrico, Christer and I have
therefore submitted the following document:

http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-sip-00

As always, comments are welcome!

Cheers,
Emil

-------- Original Message --------
Subject: New Version Notification for draft-ivov-mmusic-trickle-ice-00.txt
Date: Mon, 28 Jan 2013 11:39:21 -0800
From: internet-drafts@ietf.org
To: emcho@jitsi.org
CC: justin@uberti.name, ekr@rtfm.com


A new version of I-D, draft-ivov-mmusic-trickle-ice-00.txt
has been successfully submitted by Emil Ivov and posted to the
IETF repository.

Filename:	 draft-ivov-mmusic-trickle-ice
Revision:	 00
Title:		 Trickle ICE: Incremental Provisioning of Candidates for the
Interactive Connectivity Establishment (ICE) Protocol
Creation date:	 2013-01-28
WG ID:		 Individual Submission
Number of pages: 20
URL:
http://www.ietf.org/internet-drafts/draft-ivov-mmusic-trickle-ice-00.txt
Status:
http://datatracker.ietf.org/doc/draft-ivov-mmusic-trickle-ice
Htmlized:        http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00


Abstract:
   This document describes an extension to the Interactive Connectivity
   Establishment (ICE) protocol that allows ICE agents to send and
   receive candidates incrementally rather than exchanging complete
   lists.  With such incremental provisioning, ICE agents can begin
   connectivity checks while they are still gathering candidates and
   considerably shorten the time necessary for ICE processing to
   complete.

   The above mechanism is also referred to as "trickle ICE".





The IETF Secretariat





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Hi,

Attached is a draft on multimedia concepts and relations that we believe
relevant for the upcoming interim. This includes commonalties between
WebRTC and CLUE and other RTP usages.

Sorry for the late submission, i do hope you have time to review it. At
will also be properly submitted to IETF as soon as we figure out why the
submission tools refuses to process it.


Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
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Network Working Group                                          B. Burman
Internet-Draft                                             M. Westerlund
Intended status: Informational                                  Ericsson
Expires: August 4, 2013                                 January 31, 2013


                   Multi-Media Concepts and Relations
             draft-burman-rtcweb-mmusic-media-structure-00

Abstract

   There are currently significant efforts ongoing in IETF regarding
   more advanced multi-media functionalities, such as the work related
   to RTCWEB and CLUE.  This work includes use cases for both multi-
   party communication and multiple media streams from an individual
   end-point.  The usage of scalable encoding or simulcast encoding as
   well as different types of transport mechanisms have created
   additional needs to correctly identify different types of resources
   and describe their relations to achieve intended functionalities.

   The different usages have both commonalities and differences in needs
   and behavior.  This document attempts to review some usages and
   identify commonalities and needs.  It then continues to highlight
   important aspects that need to be considered in the definition of
   these usages.

Status of this Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on August 4, 2013.

Copyright Notice

   Copyright (c) 2013 IETF Trust and the persons identified as the
   document authors.  All rights reserved.




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   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Motivation . . . . . . . . . . . . . . . . . . . . . . . . . .  3
   3.  Use Cases  . . . . . . . . . . . . . . . . . . . . . . . . . .  4
     3.1.  Existing RTP Usages  . . . . . . . . . . . . . . . . . . .  4
       3.1.1.  Basic VoIP call  . . . . . . . . . . . . . . . . . . .  4
       3.1.2.  Audio and Video Conference . . . . . . . . . . . . . .  5
       3.1.3.  Audio and Video Switched Conference  . . . . . . . . .  7
     3.2.  WebRTC . . . . . . . . . . . . . . . . . . . . . . . . . .  8
       3.2.1.  Mesh-based Multi-party . . . . . . . . . . . . . . . .  9
       3.2.2.  Multi-source Endpoints . . . . . . . . . . . . . . . . 10
       3.2.3.  Media Relaying . . . . . . . . . . . . . . . . . . . . 11
       3.2.4.  Usage of Simulcast . . . . . . . . . . . . . . . . . . 11
     3.3.  CLUE Telepresence  . . . . . . . . . . . . . . . . . . . . 13
       3.3.1.  Telepresence Functionality . . . . . . . . . . . . . . 13
       3.3.2.  Distributed Endpoint . . . . . . . . . . . . . . . . . 14
   4.  Discussion . . . . . . . . . . . . . . . . . . . . . . . . . . 14
     4.1.  Commonalities in Use Cases . . . . . . . . . . . . . . . . 14
       4.1.1.  Media Source . . . . . . . . . . . . . . . . . . . . . 14
       4.1.2.  Encodings  . . . . . . . . . . . . . . . . . . . . . . 16
       4.1.3.  Synchronization contexts . . . . . . . . . . . . . . . 17
       4.1.4.  Distributed Endpoints  . . . . . . . . . . . . . . . . 18
     4.2.  Identified WebRTC issues . . . . . . . . . . . . . . . . . 18
     4.3.  Relevant to SDP evolution  . . . . . . . . . . . . . . . . 19
   5.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 20
   6.  Security Considerations  . . . . . . . . . . . . . . . . . . . 21
   7.  Informative References . . . . . . . . . . . . . . . . . . . . 21
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 22











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1.  Introduction

   This document concerns itself with the conceptual structures that can
   be found in different logical levels of a multi-media communication,
   from transport aspects to high-level needs of the communication
   application.  The intention is to provide considerations and guidance
   that can be used when discussing how to resolve issues in the RTCWEB
   and CLUE related standardization.  Typical use cases for those WG
   have commonalities that likely should be addressed similarly and in a
   way that allows to align them.

   The document starts with going deeper in the motivation why this has
   become an important problem at this time.  This is followed by
   studies of some use cases and what concepts they contain, and
   concludes with a discussion of observed commonalities and important
   aspects to consider.


2.  Motivation

   There has arisen a number of new needs and requirements lately from
   work such as WebRTC/RTCWEB [I-D.ietf-rtcweb-overview] and CLUE
   [I-D.ietf-clue-framework].  The applications considered in those WG
   has surfaced new requirements on the usage of both RTP [RFC3550] and
   existing signalling solutions.

   The main application aspects that have created new needs are:

   o  Multiple Media Streams from an end-point.  The fact that an end-
      point may have multiple media capture devices, such as cameras or
      microphone mixes.

   o  Group communications involving multiple end-points.  This is
      realized using both mesh based connections as well as centralized
      conference nodes.  These creating a need for dealing with multiple
      endpoints and/or multiple streams with different origins from a
      transport peer.

   o  Media Stream Adaptation, both to adjust network resource
      consumption as well as to handle varying end-point capabilities in
      group communication.

   o  Transport mechanisms including both higher levels of aggregation
      [I-D.ietf-mmusic-sdp-bundle-negotiation]
      [I-D.ietf-avtcore-multi-media-rtp-session] and the use of
      application-level transport repair mechanisms such as forward
      error correction (FEC) and/or retransmission.




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   The presence of multiple media resources or components creates a need
   to identify, handle and group those resources across multiple
   different instantiations or alternatives.


3.  Use Cases

3.1.  Existing RTP Usages

   There are many different existing RTP usages.  This section brings up
   some that we deem interesting in comparison to the other use cases.

3.1.1.  Basic VoIP call

   This use case is intended to function as a base-line to contrast
   against the rest of the use cases.

   The communication context is an audio-only bi-directional
   communication between two users, Alice and Bob. This communication
   uses a single multi-media session that can be established in a number
   of ways, but let's assume SIP/SDP [RFC3261][RFC3264].  This multi-
   media session contains two end-points, one for Alice and one for Bob.
   Each end-point has an audio capture device that is used to create a
   single audio media source at each end-point.

                        +-------+         +-------+
                        | Alice |<------->|  Bob  |
                        +-------+         +-------+

                      Figure 1: Point-to-point Audio

   The session establishment (SIP/SDP) negotiates the intent to
   communicate over RTP using only the audio media type.  Inherent in
   the application is an assumption of only a single media source in
   each direction.  The boundaries for the encodings are represented
   using RTP Payload types in conjunction with the SDP bandwidth
   parameter (b=).  The session establishment is also used to negotiate
   that RTP will be used, thus resulting in that an RTP session will be
   created for the audio.  The underlying transport flows, in this case
   a single bi-directional UDP flow for RTP, another for RTCP, is
   configured by each end-point providing its' IP address and port,
   which becomes source or destination depending on in which direction
   the packet is sent.

   The RTP session will have two RTP media streams, one in each
   direction, which carries the encoding of the media source the sending
   implementation has chosen based on the boundaries established by the
   RTP payload types and other SDP parameters, e.g. codec, and bit-



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   rates.  The streams are in the RTP context identified by their SSRCs.

3.1.2.  Audio and Video Conference

   This use case is a multi-party use case with a central conference
   node performing media mixing.  It also includes two media types, both
   audio and video.  The high level topology of the communication
   session is the following:

            +-------+         +------------+           +-------+
            |       |<-Audio->|            |<--Audio-->|       |
            | Alice |         |            |           |  Bob  |
            |       |<-Video->|            |<--Video-->|       |
            +-------+         |            |           +-------+
                              |   Mixer    |
            +-------+         |            |           +-------+
            |       |<-Audio->|            |<--Audio-->|       |
            |Charlie|         |            |           | David |
            |       |<-Video->|            |<--Video-->|       |
            +-------+         +------------+           +-------+

       Figure 2: Audio and Video Conference with Centralized Mixing

   The communication session is a multi-party conference including the
   four users Alice, Bob, Charlie, and David.  This communication
   session contains four end-points and one middlebox (the Mixer).  The
   communication session is established using four different multi-media
   sessions; one each between the user's endpoints and the middlebox.
   Each of these multi-media sessions uses a session establishment
   method, like SIP/SDP.

   Looking at a single multi-media session between a user, e.g.  Alice,
   and the Mixer, there exist two media types, audio and video.  Alice
   has two capture devices, one video camera giving her a video media
   source, and an audio capture device giving an audio media source.
   These two media sources are captured in the same room by the same
   end-point and thus have a strong timing relationship, requiring
   inter-media synchronization at playback to provide the correct
   fidelity.  Thus Alice's endpoint has a synchronization context that
   both her media sources use.

   These two media sources are encoded using encoding parameters within
   the boundaries that has been agreed between the end-point and the
   Mixer using the session establishment.  As has been common practice,
   each media type will use its own RTP session between the end-point
   and the mixer.  Thus a single audio stream using a single SSRC will
   flow from Alice to the Mixer in the Audio RTP session and a single
   video stream will flow in the Video RTP session.  Using this division



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   in separate RTP sessions, the bandwidth of both audio and video can
   be unambiguously and separately negotiated by the SDP bandwidth
   attributes exchanged between the end-points and the mixer.  Each RTP
   session is using its own Transport Flows.  The common synchronization
   context across Alice's two media streams is identified by binding
   both streams to the same CNAME, generated by Alice's endpoint.

   The mixer does not have any physical capture devices, instead it
   creates conceptual media sources.  It provides two media sources
   towards Alice; one audio being a mix of the audio from Bob, Charlie
   and David, the second one being a conceptual video source that
   contains a selection of one of the other video sources received from
   Bob, Charlie, or David depending on who is speaking.  The Mixer's
   audio and video sources are provided in an encoding using a codec
   that is supported by both Alice's endpoint and the mixer.  These
   streams are identified by a single SSRC in the respective RTP
   session.

   The mixer will have its own synchronization context and it will
   inject the media from Bob, Charlie and David in a synchronized way
   into the mixer's synchronization context to maintain the inter-media
   synchronization of the original media sources.

   The mixer establishes independent multimedia sessions with each of
   the participant's endpoints.  The mixer will in most cases also have
   unique conceptual media sources for each of the endpoints.  This as
   audio mixes and video selections typically exclude media sources
   originating from the receiving end-point.  For example, Bob's audio
   mix will be a mix of Alice, Charlie and David, and will not contain
   Bob's own audio.

   This use case may need unique user identities across the whole
   communication session.  An example functionality of this is a
   participant list which includes audio energy levels showing who is
   speaking within the audio mix.  If that information is carried in RTP
   using the RTP header extension for Mixer to audio clients [RFC6465]
   then contributing source identities in the form of CSRC need to be
   bound to the other end-point's media sources or user identities.
   This despite the fact that each RTP session towards a particular
   user's endpoint is terminated in the RTP mixer.  This points out the
   need for identifiers that exist in multiple multi-media session
   contexts.  In most cases this can easily be solved by the application
   having identities tailored specifically for its own needs, but some
   applications will benefit from having access to some commonly defined
   structure for media source identities.






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3.1.3.  Audio and Video Switched Conference

   This use case is similar to the one above (Section 3.1.2), with the
   difference that the mixer does not mix media streams by decoding,
   mixing and re-encoding them, but rather switches a selection of
   received media more or less unmodified towards receiving end-points.
   This difference may not be very apparent to the end-user, but the
   main motivations to eliminate the mixing operation and switch rather
   than mix are:

   o  Lower processing requirements in the mixer.

   o  Lower complexity in the mixer.

   o  Higher media quality at the receiver given a certain media
      bitrate.

   o  Lower end-to-end media delay.

   Without the mixing operation, the mixer has limited ability to create
   conceptual media sources that are customized for each receiver.  The
   reasons for such customizations comes from sender and receiver
   differences in available resources and preferences:

   o  Presenting multiple conference users simultaneously, like in a
      video mosaic.

   o  Alignment of sent media quality to receivers presentation needs.

   o  Alignment of codec type and configuration between sender and
      receiver.

   o  Alignment of encoded bitrate to the available end-to-end link
      bandwidth.

   To enable elimination of the mixing operation, media sent to the
   mixer must sufficiently well meet the above constraints for all
   intended receivers.  There are several ways to achieve this.  One way
   is to, by some system-wide design, ensure that all senders and
   receivers are basically identical in all the above aspects.  This may
   however prove unrealistic when variations in conditions and end-
   points are too large.  Another way is to let a sender provide a
   (small) set of alternative representations for each sent media
   source, enough to sufficiently well cover the expected range of
   variation.  If those media source representations, encodings, are
   independent from one another, they constitute a Simulcast of the
   media source.  If an encoding is instead dependent on and thus
   requires reception of one or more other encodings, the representation



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   of the media source jointly achieved by all dependent encodings is
   said to be Scalable.  Simulcast and Scalable encoding can also be
   combined.

   Both Simulcast and Scalable encodings result in that a single media
   source generates multiple RTP media streams of the same media type.
   The division of bandwidth between the Simulcast or Scalable streams
   for a single media source is application specific and will vary.  The
   total bandwidth for a Simulcast or a Scalable source is the sum of
   all included RTP media streams.  Since all streams in a Simulcast or
   Scalable source originate from the same capture device, they are
   closely related and should thus share synchronization context.

   The first and second customizations listed above, presenting multiple
   conference users simultaneously, aligned with the presentation needs
   in the receiver, can also be achieved without mixing operation by
   simply sending appropriate quality media from those users
   individually to each receiver.  The total bandwidth of this user
   presentation aggregate is the sum of all included RTP media streams.
   Audio and video from a single user share synchronization context and
   can be synchronized.  Streams that originate from different users do
   not have the same synchronization context, which is acceptable since
   they do not need to be synchronized, but just presented jointly.

   An actual mixer device need not be either mixing-only or switching-
   only, but may implement both mixing and switching and may also choose
   dynamically what to do for a specific media and a specific receiving
   user on a case-by-case basis or based on some policy.

3.2.  WebRTC

   This section brings up two different instantiations of WebRTC
   [ref-webrtc10] that stresses different aspects.  But let's start with
   reviewing some important aspects of WebRTC and the MediaStream
   [ref-media-capture] API.

   In WebRTC, an application gets access to a media source by calling
   getUserMedia(), which creates a MediaStream [ref-media-capture] (note
   the capitalization).  A MediaStream consists of zero or more
   MediaStreamTracks, where each MediaStreamTrack is associated with a
   media source.  These locally generated MediaStreams and their tracks
   are connected to local media sources, which can be media devices such
   as video cameras or microphones, but can also be files.

   An WebRTC PeerConnection (PC) is an association between two endpoints
   that is capable of communicating media from one end to the other.
   The PC concept includes establishment procedures, including media
   negotiation.  Thus a PC is an instantiation of a Multimedia Session.



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   When one end-point adds a MediaStream to a PC, the other endpoint
   will by default receive an encoded representation of the MediaStream
   and the active MediaStreamTracks.

3.2.1.  Mesh-based Multi-party

   This is a use case of WebRTC which establishes a multi-party
   communication session by establishing an individual PC with each
   participant in the communication session.

                              +---+      +---+
                              | A |<---->| B |
                              +---+      +---+
                                ^         ^
                                 \       /
                                  \     /
                                   v   v
                                   +---+
                                   | C |
                                   +---+

                  Figure 3: WebRTC Mesh-based Multi-party

   Users A, B and C want to have a joint communication session.  This
   communication session is created using a Web-application without any
   central conference functionality.  Instead, it uses a mesh of
   PeerConnections to connect each participant's endpoint with the other
   endpoints.  In this example, three double-ended connections are
   required to connect the three participants, and each endpoint has two
   PCs.

   This is an audio and video communication and each end-point has one
   video camera and one microphone as media sources.  Each endpoint
   creates its own MediaStream with one video MediaStreamTrack and one
   audio MediaStreamTrack.  The endpoints add their MediaStream to both
   of their PCs.

   Let's now focus on a single PC; in this case the one established
   between A and B. During the establishment of this PC, the two
   endpoints agree to use only a single transport flow for all media
   types, thus a single RTP session is created between A and B. A's
   MediaStream has one audio media source that is encoded according to
   the boundaries established by the PeerConnection establishment
   signalling, which includes the RTP payload types and thus Codecs
   supported as well as bit-rate boundaries.  The encoding of A's media
   source is then sent in an RTP stream identified by a unique SSRC.  In
   this case, as there are two media sources at A, two encodings will be
   created which will be transmitted using two different RTP streams



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   with their respective SSRC.  Both these streams will reference the
   same synchronization context through a common CNAME identifier used
   by A. B will have the same configuration, thus resulting in at least
   four SSRC being used in the RTP session part of the A-B PC.

   Depending on the configuration of the two PCs that A has, i.e. the
   A-B and the A-C ones, A could potentially reuse the encoding of a
   media source in both contexts, under certain conditions.  First, a
   common codec and configuration needs to exist and the boundaries for
   these configurations must allow a common work point.  In addition,
   the required bandwidth capacity needs to be available over the paths
   used by the different PCs.  Both of those conditions are not always
   true.  Thus it is quite likely that the endpoint will sometimes
   instead be required to produce two different encodings of the same
   media source.

   If an application needs to reference the media from a particular
   endpoint, it can use the MediaStream and MediaStreamTrack as they
   point back to the media sources at a particular endpoint.  This as
   the MediaStream has a scope that is not PeerConnection specific.

   The programmer can however implement this differently while
   supporting the same use case.  In this case the programmer creates
   two MediaStreams that each have MediaStreamTracks that share common
   media sources.  This can be done either by calling getUserMedia()
   twice, or by cloning the MediaStream obtained by the only
   getUserMedia() call.  In this example the result is two MediaStreams
   that are connected to different PCs.  From an identity perspective,
   the two MediaStreams are different but share common media sources.
   This fact is currently not made explicit in the API.

3.2.2.  Multi-source Endpoints

   This section concerns itself with endpoints that have more than one
   media source for a particular media type.  A straightforward example
   would be a laptop with a built in video camera used to capture the
   user and a second video camera, for example attached by USB, that is
   used to capture something else the user wants to show.  Both these
   cameras are typically present in the same sound field, so it will be
   common to have only a single audio media source.

   A possible way of representing this is to have two MediaStreams, one
   with the built in camera and the audio, and a second one with the USB
   camera and the audio.  Each MediaStream is intended to be played with
   audio and video synchronized, but the user (local or remote) or
   application is likely to switch between the two captures.

   It becomes important for a receiving endpoint that it can determine



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   that the audio in the two MediaStreams have the same synchronization
   context.  Otherwise a receiver may playback the same media source
   twice, with some time overlap, at a switch between playing the two
   MediaStreams.  Being able to determine that they are the same media
   source further allow for removing redundancy by having a single
   encoding if appropriate for both MediaStreamTracks.

3.2.3.  Media Relaying

   WebRTC endpoints can relay a received MediaStream from one PC to
   another by the simple API level maneuver of adding the received
   MediaStream to the other PC.  To realize this in the implementation
   is more complex.  This can also cause some issues from a media
   perspective.  If an application spanning across multiple endpoints
   that relay media between each other makes a mistake, a media loop can
   be created.  Media Loops could become a significant issue.  For
   example could an audio echo be created, i.e. an endpoint receives its
   own media without detecting that it is its own media and plays it
   back with some delay.  In case a WebRTC endpoint produces a
   conceptual media source by mixing incoming MediaStreams, if there is
   no loop detection, a feedback loop can be created.

   RTP has loop detection to detect and handle such cases within a
   single RTP session.  However, in the context of WebRTC, the RTP
   session is local to the PC and thus cannot rely on the RTP level loop
   detection.  Instead, if this protection is needed on the WebRTC
   MediaStream level, it could for example be achieved by having media
   source identifiers that can be preserved between the different
   MediaStreams in the PCs.

   When relaying media and in case one receives multiple encodings of
   the same source it is beneficial to know that.  For example, if one
   encoding arrives with a delay of 80 ms and another with 450 ms, being
   able to choose the one with 80 ms and not be forced to delay all
   media sources from the same synchronization context to the most
   delayed source improves performance.

3.2.4.  Usage of Simulcast

   In this section we look at a use case applying simulcast from each
   user's endpoint to a central conference node to avoid the need for an
   individual encoding to each receiving endpoint.  Instead, the central
   node chooses which of the available encodings that is forwarded to a
   particular receiver, like in Section 3.1.3.







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                +-----------+      +------------+ Enc2 +---+
                | A   +-Enc1|----->|            |----->| B |
                |     |     |      |            |      +---+
                | Src-+-Enc2|----->|            | Enc1 +---+
                +-----------+      |   Mixer    |----->| C |
                                   |            |      +---+
                                   |            | Enc2 +---+
                                   |            |----->| D |
                                   +------------+      +---+

                                 Figure 4

   In this Communication Session there are four users with endpoints and
   one middlebox (The Mixer).  This is an audio and video communication
   session.  The audio source is not simulcasted and the endpoint only
   needs to produce a single encoding.  For the video source, each
   endpoint will produce multiple encodings (Enc1 and Enc2 in Figure 4)
   and transfer them simultaneously to the mixer.  The mixer picks the
   most appropriate encoding for the path from the mixer to each
   receiving client.

   Currently there exists no specified way in WebRTC to realise the
   above, although use-cases and requirements discuss simulcast
   functionality.  The authors believe there exist two possible solution
   alternatives in the WebRTC context:

   Multiple Encodings within a PeerConnection:  The endpoint that wants
      to provide a simulcast creates one or more MediaStreams with the
      media sources it wants to transmit over a particular PC.  The
      WebRTC API provides functionality to enable multiple encodings to
      be produced for a particular MediaStreamTrack and have possibility
      to configure the desired quality levels and/or differences for
      each of the encodings.

   Using Multiple PeerConnections:  There exist capabilities to both
      negotiate and control the codec, bit-rate, video resolution,
      frame-rate, etc of a particular MediaStreamTrack in the context of
      one PeerConnection.  Thus one method to provide multiple encodings
      is to establish multiple PeerConnections between A and the Mixer,
      where each PC is configured to provide the desired quality.  Note
      that this solution comes in two flavors from an application
      perspective.  One is that the same MediaStream object is added to
      the two PeerConnections.  The second is that two different
      MediaStream objects, with the same number of MediaStreamTracks and
      representing the same sources, are created (e.g by cloning), one
      of them added to the first PeerConnection and the second one to
      the second PeerConnection.




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   Both of these solutions share a common requirement, the need to
   separate the received RTP streams not only based on media source, but
   also on the encoding.  However, on an API level the solutions appear
   different.  For Multiple Encodings within the context of a PC, the
   receiver will need new access methods for accessing and manipulating
   the different encodings.  Using multiple PC instead requires that one
   can easily determine the shared (simulcasted) media source despite
   receiving it in multiple MediaStreams on different PCs.  If the same
   MediaStream is added to both PC's the id's of the MediaStream and
   MediaStreamTracks will be the same, while they will be different if
   different MediaStream's (but representing the same sources) are added
   to the two PC's.

3.3.  CLUE Telepresence

   The CLUE framework [I-D.ietf-clue-framework] and use case
   [I-D.ietf-clue-telepresence-use-cases] documents make use of most, if
   not all, media concepts that were already discussed in previous
   sections, and adds a few more.

3.3.1.  Telepresence Functionality

   A communicating CLUE Endpoint can, compared to other types of
   Endpoints, be characterized by using multiple media resources:

   o  Multiple capture devices, such as cameras or microphones,
      generating the media for a media source.

   o  Multiple render devices, such as displays or speakers.

   o  Multiple Media Types, such as audio, video and presentation
      streams.

   o  Multiple remote Endpoints, since conference is a typical use case.

   o  Multiple Encodings (encoded representations) of a media source.

   o  Multiple Media Streams representing multiple media sources.

   To make the multitude of resources more manageable, CLUE introduces
   some additional structures.  For example, related media sources in a
   multimedia session are grouped into Scenes, which can generally be
   represented in different ways, described by alternative Scene
   Entries.  CLUE explicitly separates the concept of a media source
   from the encoded representations of it and a single media source can
   be used to create multiple Encodings.  It is also possible in CLUE to
   account for constraints in resource handling, like limitations in
   possible Encoding combinations due to physical device implementation.



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   The number of media resources typically differ between Endpoints.
   Specifically, the number of available media resources of a certain
   type used for sending at the sender side typically does not match the
   number of corresponding media resources used for receiving at the
   receiver side.  Some selection process must thus be applied either at
   the sender or the receiver to select a subset of resources to be
   used.  Hence, each resource that need to be part of that selection
   process must have some identification and characterization that can
   be understood by the selecting party.  In the CLUE model, the sender
   (Provider) announces available resources and the receiver (Consumer)
   chooses what to receive.  This choice is made independently in the
   two directions of a bi-directional communication.

3.3.2.  Distributed Endpoint

   The definition of a single CLUE Endpoint in the framework
   [I-D.ietf-clue-framework] says it can consist of several physical
   devices with source and sink media streams.  This means that each
   logical node of such distributed Endpoint can have a separate
   transport interface, and thus that media sources originating from the
   same Endpoint can have different transport addresses.


4.  Discussion

   This section discusses some conclusions the authors make based on the
   use cases.  First we will discuss commonalities between use cases.
   Secondly we will provide a summary of issues we see affect WebRTC.
   Lastly we consider aspects that need to be considered in the SDP
   evolution that is ongoing.

4.1.  Commonalities in Use Cases

   The above use cases illustrate a couple of concepts that are not well
   defined, nor have they fully specified standard mechanisms or
   behaviors.  This section contains a discussion of such concepts,
   which the authors believe are useful in more than one context and
   thus should be defined to provide a common function when needed by
   multi-media communication applications.

4.1.1.  Media Source

   In several of the above use cases there exist a need for a separation
   between the media source, the particular encoding and its transport
   stream.  In vanilla RTP there exist a one-to-one mapping between
   these; one media source is encoded in one particular way and
   transported as one RTP stream using a single SSRC in a particular RTP
   session.



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   The reason for not keeping a strict one-to-one mapping, allowing the
   media source to be identified separately from the RTP media stream
   (SSRC), varies depending on the application's needs and the desired
   functionalities:

   Simulcast:  Simulcast is a functionality to provide multiple
      simultaneous encodings of the same media source.  As each encoding
      is independent of the other, in contrast to scalable encoding,
      independent transport streams for each encoding is needed.  The
      receiver of a simulcast stream will need to be able to explicitly
      identify each encoding upon reception, as well as which media
      source it is an encoding of.  This is especially important in a
      context of multiple media sources being provided from the same
      endpoint.

   Mesh-based communication:  When a communication application
      implements multi-party communication through a mesh of transport
      flows, there exist a need for tracking the original media source,
      especially when relaying between nodes is possible.  It is likely
      that the encodings provided over the different transports are
      different.  If an application uses relaying between different
      transports, an endpoint may, intentionally or not, receive
      multiple encodings of the same media source over the same or
      different transports.  Some applications can handle the needed
      identification, but some can benefit from a standardized method to
      identify sources.

   The second argument above can be generalized into a common need in
   applications that utilize multiple multimedia sessions, such as
   multiple PeerConnections or multiple SIP/SDP-established RTP
   sessions, to form a larger communication session between multiple
   endpoints.  These applications commonly need to track media sources
   that occur in more than one multimedia session.

   Looking at both CLUE and WebRTC, they appear to contain their own
   variants of the concept that was above denoted a media source.  In
   CLUE it is called Media Capture.  In WebRTC each MediaStreamTrack is
   identifiable, however, several MediaStreamTracks can share the actual
   source, and there is no way for the application to realize this
   currently.  The identification of sources is being discussed, and
   there is a proposal [ref-leithead] that introduces the concept 'Track
   Source'.  Thus, in this document we see the media source as the
   generalized commonality between these two concepts.  Giving each
   media source a unique identifier in the communication session/context
   that is reused in all the PeerConnections or SIP/SDP-established RTP
   sessions would enable loop detection, correctly associate alternative
   encodings and provide a common name across the endpoints for
   application logic to reference the actual media source rather than a



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   particular encoding or transport stream.

   It is arguable if the application should really know a long term
   persistent source identification, such as based on hardware
   identities, for example due to fingerprinting issues, and it would
   likely be better to use an anonymous identification that is still
   unique in a sufficiently wide context, for example within the
   communication application instance.

4.1.2.  Encodings

   An Encoding is a particular encoded representation of a particular
   media source.  In the context of RTP and Signalling, a particular
   encoding must fit the established parameters, such as RTP payload
   types, media bandwidths, and other more or less codec-specific media
   constraints such as resolution, frame-rate, fidelity, audio
   bandwidth, etc.

   In the context of an application, it appears that there are primarily
   two considerations around the use of multiple encodings.

   The first is how many and what their defining parameters are.  This
   may require to be negotiated, something the existing signalling
   solutions, like SDP, currently lack support for.  For example in SDP,
   there exist no way to express that you would like to receive three
   different encodings of a particular video source.  In addition, if
   you for example prefer these three encodings to be 720p/25 Hz,
   360p/25 Hz and 180p/12.5 Hz, and even if you could define RTP payload
   types with these constraints, they must be linked to RTP streams
   carrying the encodings of the particular source.  Also, for some RTP
   payload types there exist difficulties to express encoding
   characteristics with the desired granularity.  The number of RTP
   payload types that can be used for a particular potential encoding
   can also be a constraint, especially as a single RTP payload type
   could well be used for all three target resolutions and frame rates
   in the example.  Using multiple encodings might even be desirable for
   multi-party conferences that switches video, rather than composites
   and re-encodes it.  It might be that SDP is not the most suitable
   place to negotiate this.  From an application perspective, utilizing
   clients that have standardized APIs or protocols to control them,
   there exist a need for the application to express what it prefers in
   number of encodings as well as what their primary target parameters
   are.

   Secondly, some applications may need explicit indication of what
   encoding a particular stream represents.  In some cases this can be
   deduced based on information such as RTP payload types and parameters
   received in the media stream, but such implicit information will not



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   always be detailed enough and it may also be time-consuming to
   extract.  For example, in SDP there is currently limitations for
   binding the relevant information about a particular encoding to the
   corresponding RTP stream, unless only a single RTP stream is defined
   per media description (m= line).

   The CLUE framework explicitly discusses encodings as constraints that
   are applied when transforming a media source (capture) into what CLUE
   calls a capture encoding.  This includes both explicit identification
   as well as a set of boundary parameters such as maximum width,
   height, frame rate as well as bandwidth.  In WebRTC nothing related
   has yet been defined, and we note this as an issue that needs to be
   resolved.  This as the authors expect that support for multiple
   encodings will be required to enable simulcast and scalability.

4.1.3.  Synchronization contexts

   The shortcomings around synchronization contexts appears rather
   limited.  In RTP, each RTP media stream is associated with a
   particular synchronization context through the CNAME session
   description item.  The main concerns here are likely twofold.

   The first concern is to avoid unnecessary creation of new contexts,
   and rather correctly associate with the contexts that actually exist.
   For example, WebRTC MediaStreams are defined so that all
   MediaStreamTracks within a particular MediaStream shall be
   synchronized.  An easy method for meeting this would be to assign a
   new CNAME for each MediaStream.  However, that would ignore the fact
   that several media sources from the same synchronization context may
   appear in different combinations across several MediaStreams.  Thus
   all these MediaStreams should share synchronization context to avoid
   playback glitches, like playing back different instantiations of a
   single media source out of sync because the media source was shared
   between two different MediaStreams.

   The second problem is that synchronization context identification in
   RTP, i.e.  CNAME, is overloaded as an endpoint identifier.  As an
   example, consider an endpoint that has two synchronization contexts;
   one for audio and video in the room and another for an audio and
   video presentation stream, like the output of an DVD player.  Relying
   on that an endpoint has only a single synchronization context and
   CNAME may be incorrect and could create issues that an application
   designer as well as RTP and signalling extension specifications need
   to watch out for.

   CLUE discusses so far quite little about synchronization, but clearly
   intends to enable lip synchronization between captures that have that
   relation.  The second issue is however quite likely to be encountered



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   in CLUE due to explicit inclusion of the Scene concept, where
   different Scenes do not require to share the same synchronization
   context, but is rather intended for situations where Scenes cannot
   share synchronization context.

4.1.4.  Distributed Endpoints

   When an endpoint consists of multiple nodes, the added complexity is
   often local to that endpoint, which is appropriate.  However, some
   few properties of distributed endpoints needs to be tolerated by all
   entities in a multimedia communication session.  The main item is to
   not assume that a single endpoint will only use a single network
   address.  This is a dangerous assumption even for non-distributed
   endpoints due to multi-homing and the common deployment of NATs,
   especially large scale NATs which in worst case uses multiple
   addresses for a single endpoint's transport flows.

   Distributed endpoints are brought up in the CLUE context.  They are
   not specifically discussed in the WebRTC context, instead the desire
   for transport level aggregation makes such endpoints problematic.
   However, WebRTC does allow for fallback to media type specific
   transport flows and can thus without issues support distributed
   endpoints.

4.2.  Identified WebRTC issues

   In the process of identifying commonalities and differences between
   the different use cases we have identified what to us appears to be
   issues in the current specification of WebRTC that needs to be
   reviewed.

   1.  If simulcast or scalability are to be supported at all, the
       WebRTC API will need to find a method to deal more explicitly
       with the existence of different encodings and how these are
       configured, accessed and referenced.  For simulcast, the authors
       see a quite straightforward solution where each PeerConnection is
       only allowed to contain a single encoding for a specific media
       source and the desired quality level can be negotiated for the
       full PeerConnection.  When multiple encodings are desired,
       multiple PeerConnections with differences in configuration are
       established.  That would only require that the underlying media
       source can explicitly be indicated and tracked by the receiver.

   2.  The current API structure allows to have multiple MediaStreams
       with fully or partially overlapping media sources.  This,
       combined with multiple PeerConnections and the likely possibility
       to do relaying, there appears to exist a significant need to
       determine the underlying media source, despite receiving



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       different MediaStreams with particular media sources encoded in
       different ways.  It is proposed that MediaSources are made
       possible to identify uniquely across multiple PeerConnections in
       the context of the communication application.  It is however
       likely that while being unique in a sufficiently large context,
       the identification should also be anonymous to avoid
       fingerprinting issues, similar to the situation discussed in
       Section 4.1.1.

   3.  Implementations of the MediaStream API must be careful in how
       they name and deal with synchronization contexts, so that the
       actual underlying synchronization context is preserved when
       possible.  It should be noted that cannot be done when a
       MediaStream is created that contains media sources from multiple
       synchronization contexts.  This will instead require
       resynchronization of contributing sources, creation of a new
       synchronization context, and inserting the sources into that
       synchronization context.

   These issues need to be discussed and an appropriate way to resolve
   them must be chosen.

4.3.  Relevant to SDP evolution

   The joint MMUSIC / RTCWeb WGs interim meeting in February 2013 will
   discuss a number of SDP related issues around the handling of
   multiple sources; the aggregation of multiple media types over the
   same RTP session as well as RTP sharing its transport flow not only
   with ICE/STUN but also with the WebRTC data channel using SCTP/DTLS/
   UDP.  These issues will potentially result in a significant impact on
   SDP.  It may also impact other ongoing work as well as existing
   usages and applications, making these discussions difficult.

   The above use cases and discussion points to the existence of a
   number of commonalities between WebRTC and CLUE, and that a solution
   should preferably be usable by both.  It is a very open question how
   much functionality CLUE requires from SDP, as CLUE WG plans to
   develop a protocol with a different usage model.  The appropriate
   division in functionality between SDP and this protocol is currently
   unknown.

   Based on this document, it is possible to express some protocol
   requirements when negotiating multimedia sessions and their media
   configurations.  Note that this is written as requirements to
   consider, given that one believes this functionality is needed in
   SDP.

   The Requirements:



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Internet-Draft        Media Concepts and Relations          January 2013


   Encoding negotiation:  For Simulcast and Scalability in applications,
      it must be possible to negotiate the number and the boundary
      conditions for the desired encodings created from a particular
      media source.

   Media Resource Identification:  SDP-based applications that need
      explicit information about media sources, multiple encodings and
      their related RTP media streams could benefit from a common way of
      providing this information.  This need can result in multiple
      different actual requirements.  Some require a common, explicit
      identification of media sources across multiple signalling
      contexts.  Some may require explicit indication of which set of
      encodings that has the same media source and thus which sets of
      RTP media streams (SSRCs) that are related to a particular media
      source.

   RTP media stream parameters:  With a greater heterogeneity of the
      possible encodings and their boundary conditions, situations may
      arise where some or sets of RTP media streams will need to have
      specific sets of parameters associated with them, compared to
      other (sets of) RTP media streams.

   The above are general requirements and in some cases the appropriate
   point to address the requirement may not even be SDP.  For example,
   media source identification could primarily be put in an RTCP Session
   Description (SDES) item, and only when so required by the application
   also be included in the signalling.

   The discussion in this document has impact on the high level decision
   regarding how to relate RTP media streams to SDP media descriptions.
   However, as it is currently presenting concepts rather than giving
   concrete proposals on how to enable these concepts as extensions to
   SDP or other protocols, it is difficult to determine the actual
   impact that a high level solution will have.  However, the authors
   are convinced that neither of the directions will prevent the
   definition of suitable concepts in SDP.


5.  IANA Considerations

   This document makes no request of IANA.

   Note to RFC Editor: this section may be removed on publication as an
   RFC.







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6.  Security Considerations

   The realization of the proposed concepts and the resolution will have
   security considerations.  However, at this stage it is unclear if any
   has not already common considerations regarding preserving privacy,
   confidentiality and ensure integrity to prevent denial of service or
   quality degradations.


7.  Informative References

   [I-D.ietf-avtcore-multi-media-rtp-session]
              Westerlund, M., Perkins, C., and J. Lennox, "Multiple
              Media Types in an RTP Session",
              draft-ietf-avtcore-multi-media-rtp-session-01 (work in
              progress), October 2012.

   [I-D.ietf-clue-framework]
              Duckworth, M., Pepperell, A., and S. Wenger, "Framework
              for Telepresence Multi-Streams",
              draft-ietf-clue-framework-08 (work in progress),
              December 2012.

   [I-D.ietf-clue-telepresence-use-cases]
              Romanow, A., Botzko, S., Duckworth, M., Even, R., and I.
              Communications, "Use Cases for Telepresence Multi-
              streams", draft-ietf-clue-telepresence-use-cases-04 (work
              in progress), August 2012.

   [I-D.ietf-mmusic-sdp-bundle-negotiation]
              Holmberg, C. and H. Alvestrand, "Multiplexing Negotiation
              Using Session Description Protocol (SDP) Port Numbers",
              draft-ietf-mmusic-sdp-bundle-negotiation-01 (work in
              progress), August 2012.

   [I-D.ietf-rtcweb-overview]
              Alvestrand, H., "Overview: Real Time Protocols for Brower-
              based Applications", draft-ietf-rtcweb-overview-05 (work
              in progress), December 2012.

   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.
              Schooler, "SIP: Session Initiation Protocol", RFC 3261,
              June 2002.

   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", RFC 3264,
              June 2002.



Burman & Westerlund      Expires August 4, 2013                [Page 21]

Internet-Draft        Media Concepts and Relations          January 2013


   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, July 2003.

   [RFC6465]  Ivov, E., Marocco, E., and J. Lennox, "A Real-time
              Transport Protocol (RTP) Header Extension for Mixer-to-
              Client Audio Level Indication", RFC 6465, December 2011.

   [ref-leithead]
              Microsoft, "Proposal: Media Capture and Streams Settings
              API v6, https://dvcs.w3.org/hg/dap/raw-file/tip/
              media-stream-capture/proposals/
              SettingsAPI_proposal_v6.html", December 2012.

   [ref-media-capture]
              "Media Capture and Streams,
              http://dev.w3.org/2011/webrtc/editor/getusermedia.html",
              December 2012.

   [ref-webrtc10]
              "WebRTC 1.0: Real-time Communication Between Browsers,
              http://dev.w3.org/2011/webrtc/editor/webrtc.html",
              January 2013.


Authors' Addresses

   Bo Burman
   Ericsson
   Farogatan 6
   SE-164 80 Kista
   Sweden

   Phone: +46 10 714 13 11
   Email: bo.burman@ericsson.com


   Magnus Westerlund
   Ericsson
   Farogatan 6
   SE-164 80 Kista
   Sweden

   Phone: +46 10 714 82 87
   Email: magnus.westerlund@ericsson.com






Burman & Westerlund      Expires August 4, 2013                [Page 22]


--------------080706010305080101000300--

From adam@nostrum.com  Thu Jan 31 16:17:37 2013
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From: Adam Roach <adam@nostrum.com>
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Subject: [MMUSIC] Thoughts on m= lines and media streams
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Based on the conversations we've had so far, I've put together the 
following summary of my thoughts on the topic of whether we want to put 
multiple media streams in a single m= line.

http://www.ietf.org/id/draft-roach-mmusic-mlines-00.txt

Sorry for the relatively short lead time before next week's meeting -- I 
had meant to put this kind of summary together a week or two ago, but 
have been far busier than I anticipated.

/a

From pkyzivat@alum.mit.edu  Thu Jan 31 16:57:35 2013
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Subject: Re: [MMUSIC] Thoughts on m= lines and media streams
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Adam,

Thanks for writing this. A few thoughts about it:

1) Putting session-level attributes in the first m-line has problems if 
that m-line is rejected and others are accepted. Maybe this can be 
solved with some more text in a new draft. But it will likely have 
backward compatibility problems.

2) For option 2, when you talk about a single stream per m= section, 
exactly what do you mean? Do you mean a single SSRC? If so, is a=ssrc 
used to specify which one? Or do you mean that the m= section applies to 
one RTP stream at a time, but might (serially) apply to different SSRCs 
without new O/A exchanges?

I can possibly buy in to having as many m-lines as concurrently active 
RTP-streams. But having to exchange SDP when SSRCs change is hard to accept.

	Thanks,
	Paul


On 1/31/13 7:17 PM, Adam Roach wrote:
> Based on the conversations we've had so far, I've put together the
> following summary of my thoughts on the topic of whether we want to put
> multiple media streams in a single m= line.
>
> http://www.ietf.org/id/draft-roach-mmusic-mlines-00.txt
>
> Sorry for the relatively short lead time before next week's meeting -- I
> had meant to put this kind of summary together a week or two ago, but
> have been far busier than I anticipated.
>
> /a
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


From thomas.stach@siemens-enterprise.com  Thu Jan 31 23:48:20 2013
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From: "Stach, Thomas" <thomas.stach@siemens-enterprise.com>
To: Adam Roach <adam@nostrum.com>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Thoughts on m= lines and media streams
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Adam,

thanks for analysing the various proposals and putting the draft together.
>From a quick read I have some issues with backward compatibility of your pr=
oposal to include=20
"common" attributes only in the first m-section.
Assuming that the examples in the draft are offers, an answerer that
didn't implement any of the mechanisms would probably show the
following behavior. =20

Alternativ 1a and 2 have ICE candidates only for the first m-section.
An answerer that doesn't understand the grouping semantics could -
dependent on the network topology - end up having audio, but no video
connectivity. =20
With having other attributes only in the first m-section we could get other=
 issues.


Alternative 1b doesn't have this problem, but the offer would most likely
be rejected resulting in no media at all.=20
I preferred the original mmt proposal as it had additional m-sections,
that could be accepted by a legacy answerer without the need for a GW that =
performs complex SDP operations.


Regards
Thomas


Adam Roach wrote:
> Based on the conversations we've had so far, I've put together the
> following summary of my thoughts on the topic of whether we
> want to put
> multiple media streams in a single m=3D line.
>=20
> http://www.ietf.org/id/draft-roach-mmusic-mlines-00.txt
>=20
> Sorry for the relatively short lead time before next week's
> meeting -- I
> had meant to put this kind of summary together a week or two ago, but
> have been far busier than I anticipated.
>=20
> /a
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic=
