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From: Salvatore Loreto <salvatore.loreto@ericsson.com>
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Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
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Hi Harald,

sorry to be late on this
and thanks a lot for raising the question

IMO how to multiplex DTLS/SCTP traffic with te rest of the "WebRTC" 
traffic is something that belongs
to the BUNDLE discussion, and there won't be anything about how to 
bundle in this draft.
To answer your question at present I don't think there should be 
anything special for DTLS/SCTP

However, what I have in my working copy on the draft (I will publish it 
at some point next week)
and it can be interesting to you is the possibility to have multiple 
SCTP associations on top of the same
DTLS connection (for future WebRTC scenarios, or non WebRTC scenarios):

        Running SCTP over DTLS make possible to have multiple SCTP
        associations on top of the same DTLS connection; each SCTP
        association make use of a distinct port number that is mainly
    used to
        demultiplex the associations.  If the <proto> sub-field is 'DTLS/
        SCTP' the <fmt> sub-fields contain SCTP association port numbers.
        When a list of port number identifiers is given, this implies that
        all of these associations MUST run on top of the DTLS connection.
        For the payload type assignments the "a=sctpmap:" attribute
        SHOULD be used to map from a port number to a media
        encoding name that identifies the payload format transported by the
        association or the actual application protocol running on top of it.


                   m=application 54111 DTLS/SCTP 5000 5001 5002
                   c=IN IP4 79.97.215.79
                   a=sctpmap:5000 protocol=webrtc-DataChannel;streams=16
                   a=sctpmap:5001 protocol=XXX;streams=2
                   a=sctpmap:5002 protocol=YYY;streams=1


best regards
Salvatore



On 11/2/12 12:39 PM, Harald Alvestrand wrote:
> As currently specified, the DTLS/SCTP protocol is carefully specified 
> in such a way that it can ride on the same 5-tuple as an SRTP session. 
> I'd like to keep that property when we SDP-negotiate it.
>
> The proposed syntax for DTLS/SCTP seems to be
>
> m=<something, likely application> <port number> DTLS/SCTP <fmt data, 
> mostly meaningless>
>
> I see two ways to define this as a multiplex:
>
> - Give the same port number as another m= line in the same SDP session
> - Give a meaningless port number, and use the a=group:BUNDLE (of 
> whatever flavor) to give which port number it's actually multiplexed 
> together with.
>
> The best way to do this may be obvious after the BUNDLE discussion, 
> but in case there are other considerations .... are there thoughts?
>
>                Harald
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


-- 
Salvatore Loreto, PhD
www.sloreto.com


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    <div class="moz-cite-prefix">Hi Harald,<br>
      <br>
      sorry to be late on this<br>
      and thanks a lot for raising the question<br>
      <br>
      IMO how to multiplex DTLS/SCTP traffic with te rest of the
      "WebRTC" traffic is something that belongs<br>
      to the BUNDLE discussion, and there won't be anything about how to
      bundle in this draft.<br>
      To answer your question at present I don't think there should be
      anything special for DTLS/SCTP<br>
      <br>
      However, what I have in my working copy on the draft (I will
      publish it at some point next week)<br>
      and it can be interesting to you is the possibility to have
      multiple SCTP associations on top of the same<br>
      DTLS connection (for future WebRTC scenarios, or non WebRTC
      scenarios):<br>
      <br>
      <blockquote>&nbsp;&nbsp; Running SCTP over DTLS make possible to have
        multiple SCTP<br>
        &nbsp;&nbsp; associations on top of the same DTLS connection; each SCTP<br>
        &nbsp;&nbsp; association make use of a distinct port number that is mainly
        used to<br>
        &nbsp;&nbsp; demultiplex the associations.&nbsp; If the &lt;proto&gt; sub-field
        is 'DTLS/<br>
        &nbsp;&nbsp; SCTP' the &lt;fmt&gt; sub-fields contain SCTP association
        port numbers.<br>
        &nbsp;&nbsp; When a list of port number identifiers is given, this implies
        that<br>
        &nbsp;&nbsp; all of these associations MUST run on top of the DTLS
        connection.<br>
        &nbsp;&nbsp; For the payload type assignments the "a=sctpmap:" attribute <br>
        &nbsp;&nbsp; SHOULD be used to map from a port number to a media<br>
        &nbsp;&nbsp; encoding name that identifies the payload format transported
        by the<br>
        &nbsp;&nbsp; association or the actual application protocol running on top
        of it.<br>
        <br>
        <br>
        &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; m=application 54111 DTLS/SCTP 5000 5001 5002<br>
        &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; c=IN IP4 79.97.215.79<br>
        &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=sctpmap:5000
        protocol=webrtc-DataChannel;streams=16<br>
        &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=sctpmap:5001 protocol=XXX;streams=2<br>
        &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=sctpmap:5002 protocol=YYY;streams=1<br>
      </blockquote>
      &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <br>
      best regards<br>
      Salvatore<br>
      <br>
      <br>
      <br>
      On 11/2/12 12:39 PM, Harald Alvestrand wrote:<br>
    </div>
    <blockquote cite="mid:5093A2C9.9040001@alvestrand.no" type="cite">As
      currently specified, the DTLS/SCTP protocol is carefully specified
      in such a way that it can ride on the same 5-tuple as an SRTP
      session. I'd like to keep that property when we SDP-negotiate it.
      <br>
      <br>
      The proposed syntax for DTLS/SCTP seems to be
      <br>
      <br>
      m=&lt;something, likely application&gt; &lt;port number&gt;
      DTLS/SCTP &lt;fmt data, mostly meaningless&gt;
      <br>
      <br>
      I see two ways to define this as a multiplex:
      <br>
      <br>
      - Give the same port number as another m= line in the same SDP
      session
      <br>
      - Give a meaningless port number, and use the a=group:BUNDLE (of
      whatever flavor) to give which port number it's actually
      multiplexed together with.
      <br>
      <br>
      The best way to do this may be obvious after the BUNDLE
      discussion, but in case there are other considerations .... are
      there thoughts?
      <br>
      <br>
      &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald
      <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>
      <br>
    </blockquote>
    <br>
    <br>
    <pre class="moz-signature" cols="72">-- 
Salvatore Loreto, PhD
<a class="moz-txt-link-abbreviated" href="http://www.sloreto.com">www.sloreto.com</a></pre>
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Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
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On 12/01/2012 12:03 PM, Salvatore Loreto wrote:
> Hi Harald,
>
> sorry to be late on this
> and thanks a lot for raising the question
>
> IMO how to multiplex DTLS/SCTP traffic with te rest of the "WebRTC" 
> traffic is something that belongs
> to the BUNDLE discussion, and there won't be anything about how to 
> bundle in this draft.
> To answer your question at present I don't think there should be 
> anything special for DTLS/SCTP
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.
- None of the ordinary BUNDLE considerations about colliding parameters 
apply

Since the ability to multiplex was thought an important factor for 
selecting this stacking, we'd better make sure we specify how it's being 
done.

>
> However, what I have in my working copy on the draft (I will publish 
> it at some point next week)
> and it can be interesting to you is the possibility to have multiple 
> SCTP associations on top of the same
> DTLS connection (for future WebRTC scenarios, or non WebRTC scenarios):
>
>        Running SCTP over DTLS make possible to have multiple SCTP
>        associations on top of the same DTLS connection; each SCTP
>        association make use of a distinct port number that is mainly
>     used to
>        demultiplex the associations.  If the <proto> sub-field is 'DTLS/
>        SCTP' the <fmt> sub-fields contain SCTP association port numbers.
>
that's a sentence fragment I can't parse, so I hope you revise that 
before submission. If there's a comma before "the <fmt> sub-fields" it 
parses.
>
>        When a list of port number identifiers is given, this implies that
>        all of these associations MUST run on top of the DTLS connection.
>        For the payload type assignments the "a=sctpmap:" attribute
>        SHOULD be used to map from a port number to a media
>        encoding name that identifies the payload format transported by the
>        association or the actual application protocol running on top
>     of it.
>
>
>                   m=application 54111 DTLS/SCTP 5000 5001 5002
>                   c=IN IP4 79.97.215.79
>                   a=sctpmap:5000 protocol=webrtc-DataChannel;streams=16
>                   a=sctpmap:5001 protocol=XXX;streams=2
>                   a=sctpmap:5002 protocol=YYY;streams=1
>
>
> best regards
> Salvatore
>
>
>
> On 11/2/12 12:39 PM, Harald Alvestrand wrote:
>> As currently specified, the DTLS/SCTP protocol is carefully specified 
>> in such a way that it can ride on the same 5-tuple as an SRTP 
>> session. I'd like to keep that property when we SDP-negotiate it.
>>
>> The proposed syntax for DTLS/SCTP seems to be
>>
>> m=<something, likely application> <port number> DTLS/SCTP <fmt data, 
>> mostly meaningless>
>>
>> I see two ways to define this as a multiplex:
>>
>> - Give the same port number as another m= line in the same SDP session
>> - Give a meaningless port number, and use the a=group:BUNDLE (of 
>> whatever flavor) to give which port number it's actually multiplexed 
>> together with.
>>
>> The best way to do this may be obvious after the BUNDLE discussion, 
>> but in case there are other considerations .... are there thoughts?
>>
>>                Harald
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>>
>
>
> -- 
> Salvatore Loreto, PhD
> www.sloreto.com
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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    <div class="moz-cite-prefix">On 12/01/2012 12:03 PM, Salvatore
      Loreto wrote:<br>
    </div>
    <blockquote cite="mid:50B9E3ED.6010604@ericsson.com" type="cite">
      <meta content="text/html; charset=ISO-8859-1"
        http-equiv="Content-Type">
      <div class="moz-cite-prefix">Hi Harald,<br>
        <br>
        sorry to be late on this<br>
        and thanks a lot for raising the question<br>
        <br>
        IMO how to multiplex DTLS/SCTP traffic with te rest of the
        "WebRTC" traffic is something that belongs<br>
        to the BUNDLE discussion, and there won't be anything about how
        to bundle in this draft.<br>
        To answer your question at present I don't think there should be
        anything special for DTLS/SCTP<br>
      </div>
    </blockquote>
    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:<br>
    <br>
    - 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.<br>
    - None of the ordinary BUNDLE considerations about colliding
    parameters apply<br>
    <br>
    Since the ability to multiplex was thought an important factor for
    selecting this stacking, we'd better make sure we specify how it's
    being done.<br>
    <br>
    <blockquote cite="mid:50B9E3ED.6010604@ericsson.com" type="cite">
      <div class="moz-cite-prefix"> <br>
        However, what I have in my working copy on the draft (I will
        publish it at some point next week)<br>
        and it can be interesting to you is the possibility to have
        multiple SCTP associations on top of the same<br>
        DTLS connection (for future WebRTC scenarios, or non WebRTC
        scenarios):<br>
        <br>
        <blockquote>&nbsp;&nbsp; Running SCTP over DTLS make possible to have
          multiple SCTP<br>
          &nbsp;&nbsp; associations on top of the same DTLS connection; each SCTP<br>
          &nbsp;&nbsp; association make use of a distinct port number that is
          mainly used to<br>
          &nbsp;&nbsp; demultiplex the associations.&nbsp; If the &lt;proto&gt;
          sub-field is 'DTLS/<br>
          &nbsp;&nbsp; SCTP' the &lt;fmt&gt; sub-fields contain SCTP association
          port numbers.<br>
        </blockquote>
      </div>
    </blockquote>
    that's a sentence fragment I can't parse, so I hope you revise that
    before submission. If there's a comma before "the &lt;fmt&gt;
    sub-fields" it parses.<br>
    <blockquote cite="mid:50B9E3ED.6010604@ericsson.com" type="cite">
      <div class="moz-cite-prefix">
        <blockquote> &nbsp;&nbsp; When a list of port number identifiers is given,
          this implies that<br>
          &nbsp;&nbsp; all of these associations MUST run on top of the DTLS
          connection.<br>
          &nbsp;&nbsp; For the payload type assignments the "a=sctpmap:" attribute
          <br>
          &nbsp;&nbsp; SHOULD be used to map from a port number to a media<br>
          &nbsp;&nbsp; encoding name that identifies the payload format
          transported by the<br>
          &nbsp;&nbsp; association or the actual application protocol running on
          top of it.<br>
          <br>
          <br>
          &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; m=application 54111 DTLS/SCTP 5000 5001 5002<br>
          &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; c=IN IP4 79.97.215.79<br>
          &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=sctpmap:5000
          protocol=webrtc-DataChannel;streams=16<br>
          &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=sctpmap:5001 protocol=XXX;streams=2<br>
          &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=sctpmap:5002 protocol=YYY;streams=1<br>
        </blockquote>
        &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <br>
        best regards<br>
        Salvatore<br>
        <br>
        <br>
        <br>
        On 11/2/12 12:39 PM, Harald Alvestrand wrote:<br>
      </div>
      <blockquote cite="mid:5093A2C9.9040001@alvestrand.no" type="cite">As

        currently specified, the DTLS/SCTP protocol is carefully
        specified in such a way that it can ride on the same 5-tuple as
        an SRTP session. I'd like to keep that property when we
        SDP-negotiate it. <br>
        <br>
        The proposed syntax for DTLS/SCTP seems to be <br>
        <br>
        m=&lt;something, likely application&gt; &lt;port number&gt;
        DTLS/SCTP &lt;fmt data, mostly meaningless&gt; <br>
        <br>
        I see two ways to define this as a multiplex: <br>
        <br>
        - Give the same port number as another m= line in the same SDP
        session <br>
        - Give a meaningless port number, and use the a=group:BUNDLE (of
        whatever flavor) to give which port number it's actually
        multiplexed together with. <br>
        <br>
        The best way to do this may be obvious after the BUNDLE
        discussion, but in case there are other considerations .... are
        there thoughts? <br>
        <br>
        &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald <br>
        <br>
        _______________________________________________ <br>
        mmusic mailing list <br>
        <a moz-do-not-send="true" class="moz-txt-link-abbreviated"
          href="mailto:mmusic@ietf.org">mmusic@ietf.org</a> <br>
        <a moz-do-not-send="true" class="moz-txt-link-freetext"
          href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a>
        <br>
        <br>
      </blockquote>
      <br>
      <br>
      <pre class="moz-signature" cols="72">-- 
Salvatore Loreto, PhD
<a moz-do-not-send="true" class="moz-txt-link-abbreviated" href="http://www.sloreto.com">www.sloreto.com</a></pre>
      <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 harald@alvestrand.no  Sat Dec  1 07:00:38 2012
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Date: Sat, 01 Dec 2012 16:00:32 +0100
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Subject: Re: [MMUSIC] M-lines: Wikifying it
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On 11/28/2012 09:13 PM, Paul Kyzivat wrote:
> Inline
>
> On 11/28/12 9:50 AM, Harald Alvestrand wrote:
>> Perhaps we can do some real work rather than just disagreeing on 
>> principles.
>>
>> Perhaps the most contentious issue now is "what should an M-line be".
>> As far as I can tell, M-lines and their parameters describe 3 things:
>>
>> - Properties of the transport flow / RTP session (Transport, or T for 
>> short)
>> - Properties that make sense to apply to an individual media stream, and
>> may need to be different for different media streams in the same
>> combined RTP session (Flow, or F for short)
>> - Properties that make sense to apply to all streams of a given type
>> (Media, or M for short)
>>
>> The properties that can be represented in M-lines are found in two IANA
>> registries, both on
>> http://www.iana.org/assignments/sdp-parameters/sdp-parameters.xml:
>>
>> - att-field (both session and media level)
>> - att-field (media level only)
>>
>> I make it 179 entries in these categories, combined.
>>
>> I see the same kind of work being necessary no matter which way we go:
>>
>> - If we go with M-lines describing a whole RTP session (the MMT
>> proposal), each property of the "F" class that we stil need needs to be
>> copied onto a property in the "att-field (source level)" registry.
>>
>> - If we go with M-lines describing groups of media streams, degenerating
>> to a single media stream in the case of using "a=inactive" for stream
>> muting (the BUNDLE and TOGETHER proposals), each property of the "T"
>> class needs to be described as "MUST have consistent values for all the
>> combined M-lines", "MUST be taken only from the first M-line in the
>> bundle argument", or "must get special treatment".
>>
>> - In both cases, the "M"-class lines don't need any special treatment.
>> They just need to be present in all the M-lines where they're relevant.
>>
>> I've created a spreadsheet here:
>>
>> https://docs.google.com/spreadsheet/ccc?key=0Av-aF69sk5C6dDhPblpRV2pkbXVGNV9lUFI3cHlIRFE 
>>
>>
>> with 2 extra columns: "Class" (M, F or T) and "Importance" (1 for
>> "necessary" and 0 for "will never matter").
>>
>> Anyone should be able to write it.
>> If a few of us do a quick job of markup on it, I think we should be able
>> to have a more realistic assessment in a few days.
>>
>> I've started, but much of this is unknown to me. Can others help?
>
> Don't forget the media formats listed in the m-line itself. The 
> meaning of these depends on the <proto> param. The focus of this 
> discussion is on those <proto> values that correspond to various 
> flavors of RTP. There are a bunch of <proto> values that don't 
> correspond to RTP, and that assign different meanings to the format 
> parameters.
I'm happy to ignore anything but RTP. BUNDLE isn't defined for anything 
else (while, in the parallel discussion, we may get DTLS/SCTP within its 
scope too, the RTP parameter set is our current problem).

>
> AFAIK, of the currently defined proto values only the RTP-related ones 
> have the notion of multiple flows within the session. But the sdp 
> extensions for SCTP media are currently being defined. That certainly 
> has the potential to be considered as multiple flows, and should be 
> considered in this discussion.
>
> For RTP transports, the formats are payload type numbers, and these 
> are mapped to actual formats (codecs) via a=rtpmap. I see that you 
> have proposed that rtpmap be per-media, not transport or flow. I guess 
> this means that the same payload type number could mean one thing for 
> all audio flows over the transport, and something different for all 
> video flows over the same transport.
Absolutely not; that would violate both common sense and the BUNDLE 
draft as specified. The BUNDLE drafts explicitly say that a payload type 
number can only be used for one thing.
> But the payload numbers to be used (specified on the m-line) is 
> negotiated in O/A. How would that work for audio vs. video? ISTM it 
> would be preferred to negotiate payload numbers and types 
> independently per flow.
See above. They can't collide.
>
> Regarding the "importance" column in the table: I don't understand how 
> you have assigned values. In what frame of reference is importance to 
> be assessed?
My completely subjective opinion, for RTCWEB. If others disagree, please 
change and add a comment. If we all agree, it becomes consensus.
>
> Regarding a=label: This could be any of T/F/M, depending on 
> application use. I think we must assume T, and consider an alternative 
> for F. (And for M if we think it is needed.)
The claim that a=label can be used to mark flows makes it F. If the 
people making that claim are willing to let go of the claim, we can 
change it to T. But at the moment its proponents are clearly calling for F.
>
> (I'm not convinced we need to identify an "M" scope distinct from T 
> and F.)
Please consider the ones marked as such and tell me the argument for why 
it's obviously one or the other.

In particular, a=rtpmap:<number> is clearly not T, but when we have 
multiple flows on an RTP session utilizing the same codec, I'd hate to 
be forced to create multiple identical payload types just because it was 
classified as an F field.

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


From pkyzivat@alum.mit.edu  Sat Dec  1 10:09:47 2012
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Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
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I must say that I am getting confused about what is being multiplexed 
here. We must be very careful to be precise.

SCTP defines an *association*, which in simple cases is bound to a 
5-tuple. (For now lets not talk about the cases where each end can have 
more than one address.)

SCTP multiplexes multiple one-way *streams* over an association. (These 
are not RTP streams.)

RTCWEB binds pairs of sctp-streams together into *channels*.

A channel could be used like a TCP connection. Or like a UDP connection.

Where we currently use SDP to negotiate the application level use of a 
UDP or TCP connection, we might (probably will) also like to use SDP to 
negotiate the application level use of an SCTP stream or channel.

For instance, we might want to negotiate one channel for use with the 
CLUE protocol and another for the BFCP channel. (We might want that in a 
pure sip environment - no rtcweb.) But we might also want do the same 
thing when one or both ends rtcweb clients. And in that case they might 
also want to dynamically assign other channels for rtcweb-specific use.

It could also be useful to define an RTP session over a channel. And in 
fact, define multiple RTP sessions, each over a distinct channel. For 
*each* such RTP session we would want to negotiate payload numbers and 
the mappings of those numbers to codec attributes, etc.

This starts to look like a fractal structure, which could present some 
problems for describing in SDP. Here are some examples of different ways 
multiplexing the same set of media belonging to a single sip 
session/call or rtcweb PeerConnection. (I'm intentionally not using SDP 
to describe this, in order to show the similarities between the 
alternatives and to avoid arguments of hypothetical SDP.)

For instance, we could have a sip session/call that has the following media:

- audio RTP session
   - a set of payload numbers for use in this audio RTP session
   - mappings for each payload number to codec parameters
   - several audio RTP streams within the audio RTP session
- video RTP session
   - a set of payload numbers for use in this audio RTP session
   - mappings for each payload number to codec parameters
   - several video RTP streams within the video RTP session
- a BFCP over TCP connection
- an SCTP association over DTLS over UDP
   - an SCTP channel used for the CLUE protocol

But we could also multiplex the same stuff differently:

- a single RTP session for audio/video
   - a set of payload numbers, some for audio and some for video
   - mappings for each payload number to codec a/v codec parameters
   - several audio RTP streams
   - several video RTP streams
- an SCTP association over DTLS over UDP
   - an SCTP channel used for BFCP
   - an SCTP channel used for the CLUE protocol

Or in principle the same stuff could be multiplexed as:

- an SCTP association over DTLS over UDP
   - an SCTP channel used for BFCP
   - an SCTP channel used for the CLUE protocol
   - an SCTP channel used for audio RTP
     - a set of payload numbers for use in this audio RTP session
     - mappings for each payload number to codec parameters
     - several audio RTP streams within the audio RTP session
   - an SCTP channel used for video RTP
     - a set of payload numbers for use in this video RTP session
     - mappings for each payload number to codec parameters
     - several video RTP streams within the video RTP session

Or:

- an SCTP association over DTLS over UDP
   - an SCTP channel used for BFCP
   - an SCTP channel used for the CLUE protocol
   - an SCTP channel used for a single audio session & stream
     - a set of payload numbers for use in this RTP session/stream
     - mappings for each payload number to codec parameters
   - an SCTP channel used for a single audio session & stream
     - a set of payload numbers for use in this RTP session/stream
     - mappings for each payload number to codec parameters
   - an SCTP channel used for a single audio session & stream
     - a set of payload numbers for use in this RTP session/stream
     - mappings for each payload number to codec parameters
   - an SCTP channel used for a single video session & stream
     - a set of payload numbers for use in this RTP session/stream
     - mappings for each payload number to codec parameters
   - an SCTP channel used for a single video session & stream
     - a set of payload numbers for use in this RTP session/stream
     - mappings for each payload number to codec parameters
   - an SCTP channel used for a single video session & stream
     - a set of payload numbers for use in this RTP session/stream
     - mappings for each payload number to codec parameters

Note that multiplexing multiple RTP sessions over SCTP channels is 
topologically equivalent to Magnus' proposal for a shim layer to 
multiplex multiple RTP sessions over a single UDP 5-tuple. So I could 
sketch out some added alternatives based on that. But I won't.

Of course not all of these are currently possible because we don't have 
all the necessary mappings defined. But they could be. For instance, I 
think we will *need* BFCP over a channel if we want to support an rtcweb 
client that has full functionality in a conference. It may be harder to 
motivate a need for RTP over an sctp channel.

My point in bringing this up is that I think we must be very clear about 
which of these various forms of multiplexing we are trying to represent 
*now*, and which we at least want to leave room to support in the 
future. Without at least thinking about it we are likely to come up with 
something that won't extend to a case we might later wish to support.

	Thanks,
	Paul

On 12/1/12 9:53 AM, Harald Alvestrand wrote:
> On 12/01/2012 12:03 PM, Salvatore Loreto wrote:
>> Hi Harald,
>>
>> sorry to be late on this
>> and thanks a lot for raising the question
>>
>> IMO how to multiplex DTLS/SCTP traffic with te rest of the "WebRTC"
>> traffic is something that belongs
>> to the BUNDLE discussion, and there won't be anything about how to
>> bundle in this draft.
>> To answer your question at present I don't think there should be
>> anything special for DTLS/SCTP
> 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 originally thought this, but after some discussions in Atlanta I am 
not sure about this. IIUC, SCTP has its own port concept. In a kernel 
implementation that would be used to demux multiple SCTP associations to 
the processes that own them, etc. But in a user mode implementation over 
another transport this should be available to the application to demux 
multiple SCTP associations.

So I think you might indeed be able to have multiple SCTP associations 
over the same DTLS/SCTP instance.

> - None of the ordinary BUNDLE considerations about colliding parameters
> apply
>
> Since the ability to multiplex was thought an important factor for
> selecting this stacking, we'd better make sure we specify how it's being
> done.
>
>>
>> However, what I have in my working copy on the draft (I will publish
>> it at some point next week)
>> and it can be interesting to you is the possibility to have multiple
>> SCTP associations on top of the same
>> DTLS connection (for future WebRTC scenarios, or non WebRTC scenarios):
>>
>>        Running SCTP over DTLS make possible to have multiple SCTP
>>        associations on top of the same DTLS connection; each SCTP
>>        association make use of a distinct port number that is mainly
>>     used to
>>        demultiplex the associations.  If the <proto> sub-field is 'DTLS/
>>        SCTP' the <fmt> sub-fields contain SCTP association port numbers.
>>
> that's a sentence fragment I can't parse, so I hope you revise that
> before submission. If there's a comma before "the <fmt> sub-fields" it
> parses.
>>
>>        When a list of port number identifiers is given, this implies that
>>        all of these associations MUST run on top of the DTLS connection.
>>        For the payload type assignments the "a=sctpmap:" attribute
>>        SHOULD be used to map from a port number to a media
>>        encoding name that identifies the payload format transported by the
>>        association or the actual application protocol running on top
>>     of it.
>>
>>
>>                   m=application 54111 DTLS/SCTP 5000 5001 5002
>>                   c=IN IP4 79.97.215.79
>>                   a=sctpmap:5000 protocol=webrtc-DataChannel;streams=16
>>                   a=sctpmap:5001 protocol=XXX;streams=2
>>                   a=sctpmap:5002 protocol=YYY;streams=1
>>
>>
>> best regards
>> Salvatore
>>
>>
>>
>> On 11/2/12 12:39 PM, Harald Alvestrand wrote:
>>> As currently specified, the DTLS/SCTP protocol is carefully specified
>>> in such a way that it can ride on the same 5-tuple as an SRTP
>>> session. I'd like to keep that property when we SDP-negotiate it.
>>>
>>> The proposed syntax for DTLS/SCTP seems to be
>>>
>>> m=<something, likely application> <port number> DTLS/SCTP <fmt data,
>>> mostly meaningless>
>>>
>>> I see two ways to define this as a multiplex:
>>>
>>> - Give the same port number as another m= line in the same SDP session
>>> - Give a meaningless port number, and use the a=group:BUNDLE (of
>>> whatever flavor) to give which port number it's actually multiplexed
>>> together with.
>>>
>>> The best way to do this may be obvious after the BUNDLE discussion,
>>> but in case there are other considerations .... are there thoughts?
>>>
>>>                Harald
>>>
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>
>>
>>
>> --
>> Salvatore Loreto, PhD
>> www.sloreto.com
>>
>>
>> _______________________________________________
>> 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
>


From pkyzivat@alum.mit.edu  Sat Dec  1 11:54:17 2012
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Subject: Re: [MMUSIC] M-lines: Wikifying it
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inline

On 12/1/12 10:00 AM, Harald Alvestrand wrote:
> On 11/28/2012 09:13 PM, Paul Kyzivat wrote:
>> Inline
>>
>> On 11/28/12 9:50 AM, Harald Alvestrand wrote:
>>> Perhaps we can do some real work rather than just disagreeing on
>>> principles.
>>>
>>> Perhaps the most contentious issue now is "what should an M-line be".
>>> As far as I can tell, M-lines and their parameters describe 3 things:
>>>
>>> - Properties of the transport flow / RTP session (Transport, or T for
>>> short)
>>> - Properties that make sense to apply to an individual media stream, and
>>> may need to be different for different media streams in the same
>>> combined RTP session (Flow, or F for short)
>>> - Properties that make sense to apply to all streams of a given type
>>> (Media, or M for short)
>>>
>>> The properties that can be represented in M-lines are found in two IANA
>>> registries, both on
>>> http://www.iana.org/assignments/sdp-parameters/sdp-parameters.xml:
>>>
>>> - att-field (both session and media level)
>>> - att-field (media level only)
>>>
>>> I make it 179 entries in these categories, combined.
>>>
>>> I see the same kind of work being necessary no matter which way we go:
>>>
>>> - If we go with M-lines describing a whole RTP session (the MMT
>>> proposal), each property of the "F" class that we stil need needs to be
>>> copied onto a property in the "att-field (source level)" registry.
>>>
>>> - If we go with M-lines describing groups of media streams, degenerating
>>> to a single media stream in the case of using "a=inactive" for stream
>>> muting (the BUNDLE and TOGETHER proposals), each property of the "T"
>>> class needs to be described as "MUST have consistent values for all the
>>> combined M-lines", "MUST be taken only from the first M-line in the
>>> bundle argument", or "must get special treatment".
>>>
>>> - In both cases, the "M"-class lines don't need any special treatment.
>>> They just need to be present in all the M-lines where they're relevant.
>>>
>>> I've created a spreadsheet here:
>>>
>>> https://docs.google.com/spreadsheet/ccc?key=0Av-aF69sk5C6dDhPblpRV2pkbXVGNV9lUFI3cHlIRFE
>>>
>>>
>>> with 2 extra columns: "Class" (M, F or T) and "Importance" (1 for
>>> "necessary" and 0 for "will never matter").
>>>
>>> Anyone should be able to write it.
>>> If a few of us do a quick job of markup on it, I think we should be able
>>> to have a more realistic assessment in a few days.
>>>
>>> I've started, but much of this is unknown to me. Can others help?
>>
>> Don't forget the media formats listed in the m-line itself. The
>> meaning of these depends on the <proto> param. The focus of this
>> discussion is on those <proto> values that correspond to various
>> flavors of RTP. There are a bunch of <proto> values that don't
>> correspond to RTP, and that assign different meanings to the format
>> parameters.
> I'm happy to ignore anything but RTP. BUNDLE isn't defined for anything
> else (while, in the parallel discussion, we may get DTLS/SCTP within its
> scope too, the RTP parameter set is our current problem).

Hmm. OK, I went looking for something that says that. Section 7.1 does 
say that, though I can find no text that talks about which m-line 
<proto> values may be bundled together.

>> AFAIK, of the currently defined proto values only the RTP-related ones
>> have the notion of multiple flows within the session. But the sdp
>> extensions for SCTP media are currently being defined. That certainly
>> has the potential to be considered as multiple flows, and should be
>> considered in this discussion.
>>
>> For RTP transports, the formats are payload type numbers, and these
>> are mapped to actual formats (codecs) via a=rtpmap. I see that you
>> have proposed that rtpmap be per-media, not transport or flow. I guess
>> this means that the same payload type number could mean one thing for
>> all audio flows over the transport, and something different for all
>> video flows over the same transport.
> Absolutely not; that would violate both common sense and the BUNDLE
> draft as specified. The BUNDLE drafts explicitly say that a payload type
> number can only be used for one thing.
>> But the payload numbers to be used (specified on the m-line) is
>> negotiated in O/A. How would that work for audio vs. video? ISTM it
>> would be preferred to negotiate payload numbers and types
>> independently per flow.
> See above. They can't collide.
>>
>> Regarding the "importance" column in the table: I don't understand how
>> you have assigned values. In what frame of reference is importance to
>> be assessed?
> My completely subjective opinion, for RTCWEB. If others disagree, please
> change and add a comment. If we all agree, it becomes consensus.

Frankly I don't even know what most of the parameters in the list are.
But I assume that most of them were at one time important to somebody - 
at least those who defined it. Perhaps some are now obsolete, but most 
likely not all of them.

It isn't helpful if we all agree on what is important and what is not if 
we don't agree on the criterion for importance.

A couple of meanings that could be useful here:

- those that might be used in an rtcweb context vs. those that nobody
   thinks make any sense to use with rtcweb

- those that make sense to use in a bundle/mmt vs. those that may only
   be used outside a bundle/mmt.

I guess I would prefer the latter. It has the benefit that those nobody 
understands can be put in the unimportant category, and they won't be 
broken by bundle/mmt.

>> Regarding a=label: This could be any of T/F/M, depending on
>> application use. I think we must assume T, and consider an alternative
>> for F. (And for M if we think it is needed.)

I may have misunderstood your intended meaning for M. I assume you meant 
all RTP streams of the same media type (audio or video) within a single 
5-tuple.

> The claim that a=label can be used to mark flows makes it F.

a=label is a media-level attribute, so it is bound to an m-line.

In the absence of bundle, regardless of whether the corresponding RTP 
session has one or several flows. The m-line has a particular media 
type, so I guess it could be viewed as either T or M.

Some applications of a=label may assume or require that the labeled 
m-line contain only one RTP stream. If so, then perhaps the label could 
be considered to be F. But we can't really know that.

> If the
> people making that claim are willing to let go of the claim, we can
> change it to T. But at the moment its proponents are clearly calling for F.

Is your point that you know of cases where there is a desire to label an 
individual RTP-stream, and you would like to have a way to do that when 
RTP-streams are multiplexed in an RTP session?

I'm ok with that.

>> (I'm not convinced we need to identify an "M" scope distinct from T
>> and F.)
> Please consider the ones marked as such and tell me the argument for why
> it's obviously one or the other.
>
> In particular, a=rtpmap:<number> is clearly not T, but when we have
> multiple flows on an RTP session utilizing the same codec, I'd hate to
> be forced to create multiple identical payload types just because it was
> classified as an F field.

a=rtpmap is the *only* one currently marked M. And ISTM that it clearly 
is a T.

I say this because, as you have said elsewhere, each payload type number 
is tied to a particular codec of a particular codec, whether audio or 
video. It isn't possible to have payload type number 100 mean one thing 
for audio and a different thing for video.

OTOH, when multiplexing audio and video on the same 5-tuple it is 
necessary to know whether a particular a=rtpmap is intended to be audio 
or video. Normally it inherits that from the m-line it belongs to. That 
still works for bundle. It doesn't work for mmt.

So, I still think there is no need for M. But a=rtpmap is "special" in 
that it requires an extra datum. Unless there are other attributes that 
have the same issue I think this is just a special case.

	Thanks,
	Paul


From christer.holmberg@ericsson.com  Sun Dec  2 10:37:33 2012
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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] M-lines: Wikifying it - importance column
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Subject: Re: [MMUSIC] M-lines: Wikifying it - importance column
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Hi,

>> Regarding the "importance" column in the table: I don't understand how
>> you have assigned values. In what frame of reference is importance to
>> be assessed?
> My completely subjective opinion, for RTCWEB. If others disagree, please
> change and add a comment. If we all agree, it becomes consensus.

Maybe we should not spend too much time on the importance column at this po=
int.

Keep in mind that, as we do this in MMUSIC, RTCWEB is technically seen as o=
nly one customer among others.

The impotance becomes important if we run into trouble, and can't find a wa=
y how to deal with a specific parameter. But, let's have that discussion on=
ce/if we get there.

Regards,

Christer


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Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
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On 12/01/2012 07:09 PM, Paul Kyzivat wrote:
> I must say that I am getting confused about what is being multiplexed 
> here. We must be very careful to be precise.
>
> SCTP defines an *association*, which in simple cases is bound to a 
> 5-tuple. (For now lets not talk about the cases where each end can 
> have more than one address.)
>
> SCTP multiplexes multiple one-way *streams* over an association. 
> (These are not RTP streams.)
>
> RTCWEB binds pairs of sctp-streams together into *channels*.

I'm not talking about any of these levels. I'm talking about the reuse 
of the RTP session's 5-tuple for the DTLS/SRTP 5-tuple, as described in 
RFC 5764 section 5.1.2.

I said the same thing earlier on this thread (Nov 5), when you first 
responded (thanks for being the first responder!)

You do make the point that the term "multiplex" is so over-used, it 
should probably be abandoned altogether - but that section talks about 
(de)multiplexing, so changing terminology is probably too late for that 
particular usage.

               Harald


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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] SCTP question: Where does it multiplex?
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Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
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Hi,

I guess it would be nice to hear from the browser/endpoint people whether t=
hey see a need for covering non-RTP in BUNDLE.

If NOT, it would mean a minimum of two 5-tuples, instead of one (one for th=
e bundled RTP media, and one for the data channel).

Regards,

Christer




-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 Harald Alvestrand
Sent: 3. joulukuuta 2012 9:54
To: mmusic@ietf.org
Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?

On 12/01/2012 07:09 PM, Paul Kyzivat wrote:
> I must say that I am getting confused about what is being multiplexed=20
> here. We must be very careful to be precise.
>
> SCTP defines an *association*, which in simple cases is bound to a=20
> 5-tuple. (For now lets not talk about the cases where each end can=20
> have more than one address.)
>
> SCTP multiplexes multiple one-way *streams* over an association.=20
> (These are not RTP streams.)
>
> RTCWEB binds pairs of sctp-streams together into *channels*.

I'm not talking about any of these levels. I'm talking about the reuse of t=
he RTP session's 5-tuple for the DTLS/SRTP 5-tuple, as described in RFC 576=
4 section 5.1.2.

I said the same thing earlier on this thread (Nov 5), when you first respon=
ded (thanks for being the first responder!)

You do make the point that the term "multiplex" is so over-used, it should =
probably be abandoned altogether - but that section talks about (de)multipl=
exing, so changing terminology is probably too late for that particular usa=
ge.

               Harald

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

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Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
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On 12/03/2012 09:00 AM, Christer Holmberg wrote:
> Hi,
>
> I guess it would be nice to hear from the browser/endpoint people whether they see a need for covering non-RTP in BUNDLE.
>
> If NOT, it would mean a minimum of two 5-tuples, instead of one (one for the bundled RTP media, and one for the data channel).

Speaking with my browser implementor hat on, I want to avoid adding an 
extra 5-tuple for data channels.

Our stats on improvement from the introduction of rtcp-mux were pretty 
compelling that the issue is real.

>
> Regards,
>
> Christer
>
>
>
>
> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of Harald Alvestrand
> Sent: 3. joulukuuta 2012 9:54
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
>
> On 12/01/2012 07:09 PM, Paul Kyzivat wrote:
>> I must say that I am getting confused about what is being multiplexed
>> here. We must be very careful to be precise.
>>
>> SCTP defines an *association*, which in simple cases is bound to a
>> 5-tuple. (For now lets not talk about the cases where each end can
>> have more than one address.)
>>
>> SCTP multiplexes multiple one-way *streams* over an association.
>> (These are not RTP streams.)
>>
>> RTCWEB binds pairs of sctp-streams together into *channels*.
> I'm not talking about any of these levels. I'm talking about the reuse of the RTP session's 5-tuple for the DTLS/SRTP 5-tuple, as described in RFC 5764 section 5.1.2.
>
> I said the same thing earlier on this thread (Nov 5), when you first responded (thanks for being the first responder!)
>
> You do make the point that the term "multiplex" is so over-used, it should probably be abandoned altogether - but that section talks about (de)multiplexing, so changing terminology is probably too late for that particular usage.
>
>                 Harald
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From Markus.Isomaki@nokia.com  Mon Dec  3 01:03:24 2012
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From: <Markus.Isomaki@nokia.com>
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Thread-Topic: [MMUSIC] SCTP question: Where does it multiplex?
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Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
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Hi,

I agree that it would be useful to be able to run data channel on the same =
5-tuple as RTP/RTCP, and even making it the default and must-implement way =
for the browsers.

QoS/prioritization comes to mind as the only potential drawback. The endpoi=
nt can naturally still give different priorities to the different component=
s (audio, video, "data") it is sending. I assume SCTP also has TCP-style re=
ceiver window that can be used to throttle reception if needed. And presuma=
bly RMCAT will find solutions to congestion control in genenral. But from t=
he network perspective there is only one flow, and the only(?) way to diffe=
rentiate the components is to use DSCP. I don't think our QoS (or API) draf=
t yet clearly explains how DSCP's are learned and set, or whether that is e=
ven going to be a realistic solution.=20

Markus

>-----Original Message-----
>From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
>Behalf Of ext Harald Alvestrand
>Sent: 03 December, 2012 10:11
>To: Christer Holmberg
>Cc: mmusic@ietf.org
>Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
>
>On 12/03/2012 09:00 AM, Christer Holmberg wrote:
>> Hi,
>>
>> I guess it would be nice to hear from the browser/endpoint people whethe=
r
>they see a need for covering non-RTP in BUNDLE.
>>
>> If NOT, it would mean a minimum of two 5-tuples, instead of one (one for
>the bundled RTP media, and one for the data channel).
>
>Speaking with my browser implementor hat on, I want to avoid adding an
>extra 5-tuple for data channels.
>
>Our stats on improvement from the introduction of rtcp-mux were pretty
>compelling that the issue is real.
>
>>
>> Regards,
>>
>> Christer
>>
>>
>>
>>
>> -----Original Message-----
>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
>> Behalf Of Harald Alvestrand
>> Sent: 3. joulukuuta 2012 9:54
>> To: mmusic@ietf.org
>> Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
>>
>> On 12/01/2012 07:09 PM, Paul Kyzivat wrote:
>>> I must say that I am getting confused about what is being multiplexed
>>> here. We must be very careful to be precise.
>>>
>>> SCTP defines an *association*, which in simple cases is bound to a
>>> 5-tuple. (For now lets not talk about the cases where each end can
>>> have more than one address.)
>>>
>>> SCTP multiplexes multiple one-way *streams* over an association.
>>> (These are not RTP streams.)
>>>
>>> RTCWEB binds pairs of sctp-streams together into *channels*.
>> I'm not talking about any of these levels. I'm talking about the reuse o=
f the
>RTP session's 5-tuple for the DTLS/SRTP 5-tuple, as described in RFC 5764
>section 5.1.2.
>>
>> I said the same thing earlier on this thread (Nov 5), when you first
>> responded (thanks for being the first responder!)
>>
>> You do make the point that the term "multiplex" is so over-used, it shou=
ld
>probably be abandoned altogether - but that section talks about
>(de)multiplexing, so changing terminology is probably too late for that
>particular usage.
>>
>>                 Harald
>>
>> _______________________________________________
>> 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

From christer.holmberg@ericsson.com  Mon Dec  3 02:23:34 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>, mmusic <mmusic@ietf.org>
Thread-Topic: [MMUSIC] M-lines: Wikifying it
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Subject: Re: [MMUSIC] M-lines: Wikifying it
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Hi,

A question for clarification:

When talking about the "F" class, what do you mean by "stream"? Do you mean=
 individual sources associated with the m- line? Or, do you mean all source=
s associated with the m- line?

I ASSUME you mean all sources associated with the m- line, because otherwis=
e you will need the a=3Dssrc attribute.

Regards,

Christer

From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 Harald Alvestrand
Sent: 28. marraskuuta 2012 16:51
To: mmusic
Subject: [MMUSIC] M-lines: Wikifying it

Perhaps we can do some real work rather than just disagreeing on principles=
.

Perhaps the most contentious issue now is "what should an M-line be".
As far as I can tell, M-lines and their parameters describe 3 things:

- Properties of the transport flow / RTP session (Transport, or T for short=
)
- Properties that make sense to apply to an individual media stream, and ma=
y need to be different for different media streams in the same combined RTP=
 session (Flow, or F for short)
- Properties that make sense to apply to all streams of a given type (Media=
, or M for short)

The properties that can be represented in M-lines are found in two IANA reg=
istries, both on
http://www.iana.org/assignments/sdp-parameters/sdp-parameters.xml:

- att-field (both session and media level)
- att-field (media level only)

I make it 179 entries in these categories, combined.

I see the same kind of work being necessary no matter which way we go:

- If we go with M-lines describing a whole RTP session (the MMT proposal), =
each property of the "F" class that we stil need needs to be copied onto a =
property in the "att-field (source level)" registry.

- If we go with M-lines describing groups of media streams, degenerating to=
 a single media stream in the case of using "a=3Dinactive" for stream mutin=
g (the BUNDLE and TOGETHER proposals), each property of the "T" class needs=
 to be described as "MUST have consistent values for all the combined M-lin=
es", "MUST be taken only from the first M-line in the bundle argument", or =
"must get special treatment".

- In both cases, the "M"-class lines don't need any special treatment. They=
 just need to be present in all the M-lines where they're relevant.

I've created a spreadsheet here:

https://docs.google.com/spreadsheet/ccc?key=3D0Av-aF69sk5C6dDhPblpRV2pkbXVG=
NV9lUFI3cHlIRFE

with 2 extra columns: "Class" (M, F or T) and "Importance" (1 for "necessar=
y" and 0 for "will never matter").

Anyone should be able to write it.
If a few of us do a quick job of markup on it, I think we should be able to=
 have a more realistic assessment in a few days.

I've started, but much of this is unknown to me. Can others help?

                      Harald





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<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">A question for clarificat=
ion:<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">When talking about the &#=
8220;F&#8221; class, what do you mean by &#8220;stream&#8221;? Do you mean =
individual sources associated with the m- line? Or, do you mean all sources=
 associated
 with the m- line?<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">I ASSUME you mean all sou=
rces associated with the m- line, because otherwise you will need the a=3Ds=
src attribute.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Regards,<o:p></o:p></span=
></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Christer<o:p></o:p></span=
></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">From:</span></b><spa=
n style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif=
&quot;;color:windowtext"> mmusic-bounces@ietf.org [mailto:mmusic-bounces@ie=
tf.org]
<b>On Behalf Of </b>Harald Alvestrand<br>
<b>Sent:</b> 28. marraskuuta 2012 16:51<br>
<b>To:</b> mmusic<br>
<b>Subject:</b> [MMUSIC] M-lines: Wikifying it<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">Perhaps we can do som=
e real work rather than just disagreeing on principles.<br>
<br>
Perhaps the most contentious issue now is &quot;what should an M-line be&qu=
ot;.<br>
As far as I can tell, M-lines and their parameters describe 3 things:<br>
<br>
- Properties of the transport flow / RTP session (Transport, or T for short=
)<br>
- Properties that make sense to apply to an individual media stream, and ma=
y need to be different for different media streams in the same combined RTP=
 session (Flow, or F for short)<br>
- Properties that make sense to apply to all streams of a given type (Media=
, or M for short)<br>
<br>
The properties that can be represented in M-lines are found in two IANA reg=
istries, both on<br>
<a href=3D"http://www.iana.org/assignments/sdp-parameters/sdp-parameters.xm=
l">http://www.iana.org/assignments/sdp-parameters/sdp-parameters.xml</a>:<b=
r>
<br>
- att-field (both session and media level)<br>
- att-field (media level only)<br>
<br>
I make it 179 entries in these categories, combined.<br>
<br>
I see the same kind of work being necessary no matter which way we go:<br>
<br>
- If we go with M-lines describing a whole RTP session (the MMT proposal), =
each property of the &quot;F&quot; class that we stil need needs to be copi=
ed onto a property in the &quot;att-field (source level)&quot; registry.<br=
>
<br>
- If we go with M-lines describing groups of media streams, degenerating to=
 a single media stream in the case of using &quot;a=3Dinactive&quot; for st=
ream muting (the BUNDLE and TOGETHER proposals), each property of the &quot=
;T&quot; class needs to be described as &quot;MUST have consistent
 values for all the combined M-lines&quot;, &quot;MUST be taken only from t=
he first M-line in the bundle argument&quot;, or &quot;must get special tre=
atment&quot;.<br>
<br>
- In both cases, the &quot;M&quot;-class lines don't need any special treat=
ment. They just need to be present in all the M-lines where they're relevan=
t.<br>
<br>
I've created a spreadsheet here:<br>
<br>
<a href=3D"https://docs.google.com/spreadsheet/ccc?key=3D0Av-aF69sk5C6dDhPb=
lpRV2pkbXVGNV9lUFI3cHlIRFE">https://docs.google.com/spreadsheet/ccc?key=3D0=
Av-aF69sk5C6dDhPblpRV2pkbXVGNV9lUFI3cHlIRFE</a><br>
<br>
with 2 extra columns: &quot;Class&quot; (M, F or T) and &quot;Importance&qu=
ot; (1 for &quot;necessary&quot; and 0 for &quot;will never matter&quot;).<=
br>
<br>
Anyone should be able to write it.<br>
If a few of us do a quick job of markup on it, I think we should be able to=
 have a more realistic assessment in a few days.<br>
<br>
I've started, but much of this is unknown to me. Can others help?<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
<br>
<br>
<br>
<br>
<o:p></o:p></p>
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Cc: mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] M-lines: Wikifying it
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On 12/03/2012 11:23 AM, Christer Holmberg wrote:
>
> Hi,
>
> A question for clarification:
>
> When talking about the "F" class, what do you mean by "stream"? Do you 
> mean individual sources associated with the m- line? Or, do you mean 
> all sources associated with the m- line?
>
> I ASSUME you mean all sources associated with the m- line, because 
> otherwise you will need the a=ssrc attribute.
>

The purpose of the exercise is to figure out what attributes we need to 
redefine if we define the M-line to be an RTP session (the MMT and the 
all-videos-under-one-M-line approach) as opposed to what attributes we 
need to redefine if we define the M-line to be a group of sources that, 
in turn, may be associated with an RTP session (the BUNDLE approach).

F-lines need redefinition under the first approach, T-lines need 
redefinition (or adjustment) under the second, M-lines may be able to 
work in both cases.

> Regards,
>
> Christer
>
> *From:*mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] *On 
> Behalf Of *Harald Alvestrand
> *Sent:* 28. marraskuuta 2012 16:51
> *To:* mmusic
> *Subject:* [MMUSIC] M-lines: Wikifying it
>
> Perhaps we can do some real work rather than just disagreeing on 
> principles.
>
> Perhaps the most contentious issue now is "what should an M-line be".
> As far as I can tell, M-lines and their parameters describe 3 things:
>
> - Properties of the transport flow / RTP session (Transport, or T for 
> short)
> - Properties that make sense to apply to an individual media stream, 
> and may need to be different for different media streams in the same 
> combined RTP session (Flow, or F for short)
> - Properties that make sense to apply to all streams of a given type 
> (Media, or M for short)
>
> The properties that can be represented in M-lines are found in two 
> IANA registries, both on
> http://www.iana.org/assignments/sdp-parameters/sdp-parameters.xml:
>
> - att-field (both session and media level)
> - att-field (media level only)
>
> I make it 179 entries in these categories, combined.
>
> I see the same kind of work being necessary no matter which way we go:
>
> - If we go with M-lines describing a whole RTP session (the MMT 
> proposal), each property of the "F" class that we stil need needs to 
> be copied onto a property in the "att-field (source level)" registry.
>
> - If we go with M-lines describing groups of media streams, 
> degenerating to a single media stream in the case of using 
> "a=inactive" for stream muting (the BUNDLE and TOGETHER proposals), 
> each property of the "T" class needs to be described as "MUST have 
> consistent values for all the combined M-lines", "MUST be taken only 
> from the first M-line in the bundle argument", or "must get special 
> treatment".
>
> - In both cases, the "M"-class lines don't need any special treatment. 
> They just need to be present in all the M-lines where they're relevant.
>
> I've created a spreadsheet here:
>
> https://docs.google.com/spreadsheet/ccc?key=0Av-aF69sk5C6dDhPblpRV2pkbXVGNV9lUFI3cHlIRFE
>
> with 2 extra columns: "Class" (M, F or T) and "Importance" (1 for 
> "necessary" and 0 for "will never matter").
>
> Anyone should be able to write it.
> If a few of us do a quick job of markup on it, I think we should be 
> able to have a more realistic assessment in a few days.
>
> I've started, but much of this is unknown to me. Can others help?
>
>                       Harald
>
>
>
>


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    <div class="moz-cite-prefix">On 12/03/2012 11:23 AM, Christer
      Holmberg wrote:<br>
    </div>
    <blockquote
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      <div class="WordSection1">
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi,<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">A
            question for clarification:<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">When
            talking about the &#8220;F&#8221; class, what do you mean by &#8220;stream&#8221;?
            Do you mean individual sources associated with the m- line?
            Or, do you mean all sources associated with the m- line?<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">I
            ASSUME you mean all sources associated with the m- line,
            because otherwise you will need the a=ssrc attribute.</span></p>
      </div>
    </blockquote>
    <br>
    The purpose of the exercise is to figure out what attributes we need
    to redefine if we define the M-line to be an RTP session (the MMT
    and the all-videos-under-one-M-line approach) as opposed to what
    attributes we need to redefine if we define the M-line to be a group
    of sources that, in turn, may be associated with an RTP session (the
    BUNDLE approach).<br>
    <br>
    F-lines need redefinition under the first approach, T-lines need
    redefinition (or adjustment) under the second, M-lines may be able
    to work in both cases.<br>
    <br>
    <blockquote
      cite="mid:7594FB04B1934943A5C02806D1A2204B04D1FE@ESESSMB209.ericsson.se"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Regards,<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Christer<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <div>
          <div style="border:none;border-top:solid #B5C4DF
            1.0pt;padding:3.0pt 0cm 0cm 0cm">
            <p class="MsoNormal"><b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">From:</span></b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">
                <a class="moz-txt-link-abbreviated" href="mailto:mmusic-bounces@ietf.org">mmusic-bounces@ietf.org</a> [<a class="moz-txt-link-freetext" href="mailto:mmusic-bounces@ietf.org">mailto:mmusic-bounces@ietf.org</a>]
                <b>On Behalf Of </b>Harald Alvestrand<br>
                <b>Sent:</b> 28. marraskuuta 2012 16:51<br>
                <b>To:</b> mmusic<br>
                <b>Subject:</b> [MMUSIC] M-lines: Wikifying it<o:p></o:p></span></p>
          </div>
        </div>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        <p class="MsoNormal" style="margin-bottom:12.0pt">Perhaps we can
          do some real work rather than just disagreeing on principles.<br>
          <br>
          Perhaps the most contentious issue now is "what should an
          M-line be".<br>
          As far as I can tell, M-lines and their parameters describe 3
          things:<br>
          <br>
          - Properties of the transport flow / RTP session (Transport,
          or T for short)<br>
          - Properties that make sense to apply to an individual media
          stream, and may need to be different for different media
          streams in the same combined RTP session (Flow, or F for
          short)<br>
          - Properties that make sense to apply to all streams of a
          given type (Media, or M for short)<br>
          <br>
          The properties that can be represented in M-lines are found in
          two IANA registries, both on<br>
          <a moz-do-not-send="true"
            href="http://www.iana.org/assignments/sdp-parameters/sdp-parameters.xml">http://www.iana.org/assignments/sdp-parameters/sdp-parameters.xml</a>:<br>
          <br>
          - att-field (both session and media level)<br>
          - att-field (media level only)<br>
          <br>
          I make it 179 entries in these categories, combined.<br>
          <br>
          I see the same kind of work being necessary no matter which
          way we go:<br>
          <br>
          - If we go with M-lines describing a whole RTP session (the
          MMT proposal), each property of the "F" class that we stil
          need needs to be copied onto a property in the "att-field
          (source level)" registry.<br>
          <br>
          - If we go with M-lines describing groups of media streams,
          degenerating to a single media stream in the case of using
          "a=inactive" for stream muting (the BUNDLE and TOGETHER
          proposals), each property of the "T" class needs to be
          described as "MUST have consistent values for all the combined
          M-lines", "MUST be taken only from the first M-line in the
          bundle argument", or "must get special treatment".<br>
          <br>
          - In both cases, the "M"-class lines don't need any special
          treatment. They just need to be present in all the M-lines
          where they're relevant.<br>
          <br>
          I've created a spreadsheet here:<br>
          <br>
          <a moz-do-not-send="true"
href="https://docs.google.com/spreadsheet/ccc?key=0Av-aF69sk5C6dDhPblpRV2pkbXVGNV9lUFI3cHlIRFE">https://docs.google.com/spreadsheet/ccc?key=0Av-aF69sk5C6dDhPblpRV2pkbXVGNV9lUFI3cHlIRFE</a><br>
          <br>
          with 2 extra columns: "Class" (M, F or T) and "Importance" (1
          for "necessary" and 0 for "will never matter").<br>
          <br>
          Anyone should be able to write it.<br>
          If a few of us do a quick job of markup on it, I think we
          should be able to have a more realistic assessment in a few
          days.<br>
          <br>
          I've started, but much of this is unknown to me. Can others
          help?<br>
          <br>
          &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
          <br>
          <br>
          <br>
          <br>
          <o:p></o:p></p>
      </div>
    </blockquote>
    <br>
  </body>
</html>

--------------080609090808090403020406--

From christer.holmberg@ericsson.com  Mon Dec  3 03:06:05 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: [MMUSIC] M-lines: Wikifying it
Thread-Index: AQHNzXfLf68XUJVyMEaaKo/W5rPZGpgG5AmA///5IgCAABQFMA==
Date: Mon, 3 Dec 2012 11:05:49 +0000
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References: <50B624CC.4010501@alvestrand.no> <7594FB04B1934943A5C02806D1A2204B04D1FE@ESESSMB209.ericsson.se> <50BC84AC.4040201@alvestrand.no>
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Cc: mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] M-lines: Wikifying it
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--_000_7594FB04B1934943A5C02806D1A2204B04D2C6ESESSMB209ericsso_
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Hi,

So, the exercise is NOT about how to apply attributes to individual sources=
?

Regards,

Christer

From: Harald Alvestrand [mailto:harald@alvestrand.no]
Sent: 3. joulukuuta 2012 12:54
To: Christer Holmberg
Cc: mmusic
Subject: Re: [MMUSIC] M-lines: Wikifying it

On 12/03/2012 11:23 AM, Christer Holmberg wrote:
Hi,

A question for clarification:

When talking about the "F" class, what do you mean by "stream"? Do you mean=
 individual sources associated with the m- line? Or, do you mean all source=
s associated with the m- line?

I ASSUME you mean all sources associated with the m- line, because otherwis=
e you will need the a=3Dssrc attribute.

The purpose of the exercise is to figure out what attributes we need to red=
efine if we define the M-line to be an RTP session (the MMT and the all-vid=
eos-under-one-M-line approach) as opposed to what attributes we need to red=
efine if we define the M-line to be a group of sources that, in turn, may b=
e associated with an RTP session (the BUNDLE approach).

F-lines need redefinition under the first approach, T-lines need redefiniti=
on (or adjustment) under the second, M-lines may be able to work in both ca=
ses.



Regards,

Christer

From: mmusic-bounces@ietf.org<mailto:mmusic-bounces@ietf.org> [mailto:mmusi=
c-bounces@ietf.org] On Behalf Of Harald Alvestrand
Sent: 28. marraskuuta 2012 16:51
To: mmusic
Subject: [MMUSIC] M-lines: Wikifying it

Perhaps we can do some real work rather than just disagreeing on principles=
.

Perhaps the most contentious issue now is "what should an M-line be".
As far as I can tell, M-lines and their parameters describe 3 things:

- Properties of the transport flow / RTP session (Transport, or T for short=
)
- Properties that make sense to apply to an individual media stream, and ma=
y need to be different for different media streams in the same combined RTP=
 session (Flow, or F for short)
- Properties that make sense to apply to all streams of a given type (Media=
, or M for short)

The properties that can be represented in M-lines are found in two IANA reg=
istries, both on
http://www.iana.org/assignments/sdp-parameters/sdp-parameters.xml:

- att-field (both session and media level)
- att-field (media level only)

I make it 179 entries in these categories, combined.

I see the same kind of work being necessary no matter which way we go:

- If we go with M-lines describing a whole RTP session (the MMT proposal), =
each property of the "F" class that we stil need needs to be copied onto a =
property in the "att-field (source level)" registry.

- If we go with M-lines describing groups of media streams, degenerating to=
 a single media stream in the case of using "a=3Dinactive" for stream mutin=
g (the BUNDLE and TOGETHER proposals), each property of the "T" class needs=
 to be described as "MUST have consistent values for all the combined M-lin=
es", "MUST be taken only from the first M-line in the bundle argument", or =
"must get special treatment".

- In both cases, the "M"-class lines don't need any special treatment. They=
 just need to be present in all the M-lines where they're relevant.

I've created a spreadsheet here:

https://docs.google.com/spreadsheet/ccc?key=3D0Av-aF69sk5C6dDhPblpRV2pkbXVG=
NV9lUFI3cHlIRFE

with 2 extra columns: "Class" (M, F or T) and "Importance" (1 for "necessar=
y" and 0 for "will never matter").

Anyone should be able to write it.
If a few of us do a quick job of markup on it, I think we should be able to=
 have a more realistic assessment in a few days.

I've started, but much of this is unknown to me. Can others help?

                      Harald







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<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">So, the exercise is NOT a=
bout how to apply attributes to individual sources?<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Regards,<o:p></o:p></span=
></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Christer
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">From:</span></b><spa=
n style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif=
&quot;;color:windowtext"> Harald Alvestrand [mailto:harald@alvestrand.no]
<br>
<b>Sent:</b> 3. joulukuuta 2012 12:54<br>
<b>To:</b> Christer Holmberg<br>
<b>Cc:</b> mmusic<br>
<b>Subject:</b> Re: [MMUSIC] M-lines: Wikifying it<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On 12/03/2012 11:23 AM, Christer Holmberg wrote:<o:p=
></o:p></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi,</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">A question for clarificat=
ion:</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">When talking about the &#=
8220;F&#8221; class, what do you mean by &#8220;stream&#8221;? Do you mean =
individual sources associated with the m- line? Or, do you mean all sources=
 associated
 with the m- line?</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">I ASSUME you mean all sou=
rces associated with the m- line, because otherwise you will need the a=3Ds=
src attribute.</span><o:p></o:p></p>
</blockquote>
<p class=3D"MsoNormal"><br>
The purpose of the exercise is to figure out what attributes we need to red=
efine if we define the M-line to be an RTP session (the MMT and the all-vid=
eos-under-one-M-line approach) as opposed to what attributes we need to red=
efine if we define the M-line to
 be a group of sources that, in turn, may be associated with an RTP session=
 (the BUNDLE approach).<br>
<br>
F-lines need redefinition under the first approach, T-lines need redefiniti=
on (or adjustment) under the second, M-lines may be able to work in both ca=
ses.<br>
<br>
<br>
<o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Regards,</span><o:p></o:p=
></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Christer</span><o:p></o:p=
></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p><=
/p>
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">From:</span></b><spa=
n style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif=
&quot;;color:windowtext">
<a href=3D"mailto:mmusic-bounces@ietf.org">mmusic-bounces@ietf.org</a> [<a =
href=3D"mailto:mmusic-bounces@ietf.org">mailto:mmusic-bounces@ietf.org</a>]
<b>On Behalf Of </b>Harald Alvestrand<br>
<b>Sent:</b> 28. marraskuuta 2012 16:51<br>
<b>To:</b> mmusic<br>
<b>Subject:</b> [MMUSIC] M-lines: Wikifying it</span><o:p></o:p></p>
</div>
</div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">Perhaps we can do som=
e real work rather than just disagreeing on principles.<br>
<br>
Perhaps the most contentious issue now is &quot;what should an M-line be&qu=
ot;.<br>
As far as I can tell, M-lines and their parameters describe 3 things:<br>
<br>
- Properties of the transport flow / RTP session (Transport, or T for short=
)<br>
- Properties that make sense to apply to an individual media stream, and ma=
y need to be different for different media streams in the same combined RTP=
 session (Flow, or F for short)<br>
- Properties that make sense to apply to all streams of a given type (Media=
, or M for short)<br>
<br>
The properties that can be represented in M-lines are found in two IANA reg=
istries, both on<br>
<a href=3D"http://www.iana.org/assignments/sdp-parameters/sdp-parameters.xm=
l">http://www.iana.org/assignments/sdp-parameters/sdp-parameters.xml</a>:<b=
r>
<br>
- att-field (both session and media level)<br>
- att-field (media level only)<br>
<br>
I make it 179 entries in these categories, combined.<br>
<br>
I see the same kind of work being necessary no matter which way we go:<br>
<br>
- If we go with M-lines describing a whole RTP session (the MMT proposal), =
each property of the &quot;F&quot; class that we stil need needs to be copi=
ed onto a property in the &quot;att-field (source level)&quot; registry.<br=
>
<br>
- If we go with M-lines describing groups of media streams, degenerating to=
 a single media stream in the case of using &quot;a=3Dinactive&quot; for st=
ream muting (the BUNDLE and TOGETHER proposals), each property of the &quot=
;T&quot; class needs to be described as &quot;MUST have consistent
 values for all the combined M-lines&quot;, &quot;MUST be taken only from t=
he first M-line in the bundle argument&quot;, or &quot;must get special tre=
atment&quot;.<br>
<br>
- In both cases, the &quot;M&quot;-class lines don't need any special treat=
ment. They just need to be present in all the M-lines where they're relevan=
t.<br>
<br>
I've created a spreadsheet here:<br>
<br>
<a href=3D"https://docs.google.com/spreadsheet/ccc?key=3D0Av-aF69sk5C6dDhPb=
lpRV2pkbXVGNV9lUFI3cHlIRFE">https://docs.google.com/spreadsheet/ccc?key=3D0=
Av-aF69sk5C6dDhPblpRV2pkbXVGNV9lUFI3cHlIRFE</a><br>
<br>
with 2 extra columns: &quot;Class&quot; (M, F or T) and &quot;Importance&qu=
ot; (1 for &quot;necessary&quot; and 0 for &quot;will never matter&quot;).<=
br>
<br>
Anyone should be able to write it.<br>
If a few of us do a quick job of markup on it, I think we should be able to=
 have a more realistic assessment in a few days.<br>
<br>
I've started, but much of this is unknown to me. Can others help?<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
<br>
<br>
<br>
<br>
<br>
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B04D2C6ESESSMB209ericsso_--

From harald@alvestrand.no  Mon Dec  3 03:06:55 2012
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Date: Mon, 03 Dec 2012 12:06:46 +0100
From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [MMUSIC] M-lines: Wikifying it
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On 12/01/2012 08:54 PM, Paul Kyzivat wrote:
> inline
>
> On 12/1/12 10:00 AM, Harald Alvestrand wrote:
>> On 11/28/2012 09:13 PM, Paul Kyzivat wrote:
>>> Inline
>>>
>>> On 11/28/12 9:50 AM, Harald Alvestrand wrote:
>>>> Perhaps we can do some real work rather than just disagreeing on
>>>> principles.
>>>>
>>>> Perhaps the most contentious issue now is "what should an M-line be".
>>>> As far as I can tell, M-lines and their parameters describe 3 things:
>>>>
>>>> - Properties of the transport flow / RTP session (Transport, or T for
>>>> short)
>>>> - Properties that make sense to apply to an individual media 
>>>> stream, and
>>>> may need to be different for different media streams in the same
>>>> combined RTP session (Flow, or F for short)
>>>> - Properties that make sense to apply to all streams of a given type
>>>> (Media, or M for short)
>>>>
>>>> The properties that can be represented in M-lines are found in two 
>>>> IANA
>>>> registries, both on
>>>> http://www.iana.org/assignments/sdp-parameters/sdp-parameters.xml:
>>>>
>>>> - att-field (both session and media level)
>>>> - att-field (media level only)
>>>>
>>>> I make it 179 entries in these categories, combined.
>>>>
>>>> I see the same kind of work being necessary no matter which way we go:
>>>>
>>>> - If we go with M-lines describing a whole RTP session (the MMT
>>>> proposal), each property of the "F" class that we stil need needs 
>>>> to be
>>>> copied onto a property in the "att-field (source level)" registry.
>>>>
>>>> - If we go with M-lines describing groups of media streams, 
>>>> degenerating
>>>> to a single media stream in the case of using "a=inactive" for stream
>>>> muting (the BUNDLE and TOGETHER proposals), each property of the "T"
>>>> class needs to be described as "MUST have consistent values for all 
>>>> the
>>>> combined M-lines", "MUST be taken only from the first M-line in the
>>>> bundle argument", or "must get special treatment".
>>>>
>>>> - In both cases, the "M"-class lines don't need any special treatment.
>>>> They just need to be present in all the M-lines where they're 
>>>> relevant.
>>>>
>>>> I've created a spreadsheet here:
>>>>
>>>> https://docs.google.com/spreadsheet/ccc?key=0Av-aF69sk5C6dDhPblpRV2pkbXVGNV9lUFI3cHlIRFE 
>>>>
>>>>
>>>>
>>>> with 2 extra columns: "Class" (M, F or T) and "Importance" (1 for
>>>> "necessary" and 0 for "will never matter").
>>>>
>>>> Anyone should be able to write it.
>>>> If a few of us do a quick job of markup on it, I think we should be 
>>>> able
>>>> to have a more realistic assessment in a few days.
>>>>
>>>> I've started, but much of this is unknown to me. Can others help?
>>>
>>> Don't forget the media formats listed in the m-line itself. The
>>> meaning of these depends on the <proto> param. The focus of this
>>> discussion is on those <proto> values that correspond to various
>>> flavors of RTP. There are a bunch of <proto> values that don't
>>> correspond to RTP, and that assign different meanings to the format
>>> parameters.
>> I'm happy to ignore anything but RTP. BUNDLE isn't defined for anything
>> else (while, in the parallel discussion, we may get DTLS/SCTP within its
>> scope too, the RTP parameter set is our current problem).
>
> Hmm. OK, I went looking for something that says that. Section 7.1 does 
> say that, though I can find no text that talks about which m-line 
> <proto> values may be bundled together.
Section 7.2 of draft-ietf-mmusic-sdp-bundle says that:

    o  - The "proto" value in each "m=" line MUST be identical (e.g.
       RTP/AVPF).

>
>>> AFAIK, of the currently defined proto values only the RTP-related ones
>>> have the notion of multiple flows within the session. But the sdp
>>> extensions for SCTP media are currently being defined. That certainly
>>> has the potential to be considered as multiple flows, and should be
>>> considered in this discussion.
>>>
>>> For RTP transports, the formats are payload type numbers, and these
>>> are mapped to actual formats (codecs) via a=rtpmap. I see that you
>>> have proposed that rtpmap be per-media, not transport or flow. I guess
>>> this means that the same payload type number could mean one thing for
>>> all audio flows over the transport, and something different for all
>>> video flows over the same transport.
>> Absolutely not; that would violate both common sense and the BUNDLE
>> draft as specified. The BUNDLE drafts explicitly say that a payload type
>> number can only be used for one thing.
>>> But the payload numbers to be used (specified on the m-line) is
>>> negotiated in O/A. How would that work for audio vs. video? ISTM it
>>> would be preferred to negotiate payload numbers and types
>>> independently per flow.
>> See above. They can't collide.
>>>
>>> Regarding the "importance" column in the table: I don't understand how
>>> you have assigned values. In what frame of reference is importance to
>>> be assessed?
>> My completely subjective opinion, for RTCWEB. If others disagree, please
>> change and add a comment. If we all agree, it becomes consensus.
>
> Frankly I don't even know what most of the parameters in the list are.
> But I assume that most of them were at one time important to somebody 
> - at least those who defined it. Perhaps some are now obsolete, but 
> most likely not all of them.
>
> It isn't helpful if we all agree on what is important and what is not 
> if we don't agree on the criterion for importance.
>
> A couple of meanings that could be useful here:
>
> - those that might be used in an rtcweb context vs. those that nobody
>   thinks make any sense to use with rtcweb
>
> - those that make sense to use in a bundle/mmt vs. those that may only
>   be used outside a bundle/mmt.
>
> I guess I would prefer the latter. It has the benefit that those 
> nobody understands can be put in the unimportant category, and they 
> won't be broken by bundle/mmt.
I changed the column title to "Importance to RTCWEB". Feel free to add 
new columns if you feel they are useful.
>
>>> Regarding a=label: This could be any of T/F/M, depending on
>>> application use. I think we must assume T, and consider an alternative
>>> for F. (And for M if we think it is needed.)
>
> I may have misunderstood your intended meaning for M. I assume you 
> meant all RTP streams of the same media type (audio or video) within a 
> single 5-tuple.
I was thinking "same payload type", since the payload type is the 
distinguisher used for groups of flows that should have the same 
parameters in many cases.
>
>> The claim that a=label can be used to mark flows makes it F.
>
> a=label is a media-level attribute, so it is bound to an m-line.
>
> In the absence of bundle, regardless of whether the corresponding RTP 
> session has one or several flows. The m-line has a particular media 
> type, so I guess it could be viewed as either T or M.
The reason why we're having the discussion is that we have two competing 
definitons of what an M-line should be, and need to choose one.
So arguing from whether or not something associates with an M-line is 
not useful.
>
> Some applications of a=label may assume or require that the labeled 
> m-line contain only one RTP stream. If so, then perhaps the label 
> could be considered to be F. But we can't really know that.
What applications of a=label do you know about?
>
>> If the
>> people making that claim are willing to let go of the claim, we can
>> change it to T. But at the moment its proponents are clearly calling 
>> for F.
>
> Is your point that you know of cases where there is a desire to label 
> an individual RTP-stream, and you would like to have a way to do that 
> when RTP-streams are multiplexed in an RTP session?

Yes, that's how I've interpreted statements made (mostly from Cullen) 
about a=label.

>
> I'm ok with that.
>
>>> (I'm not convinced we need to identify an "M" scope distinct from T
>>> and F.)
>> Please consider the ones marked as such and tell me the argument for why
>> it's obviously one or the other.
>>
>> In particular, a=rtpmap:<number> is clearly not T, but when we have
>> multiple flows on an RTP session utilizing the same codec, I'd hate to
>> be forced to create multiple identical payload types just because it was
>> classified as an F field.
>
> a=rtpmap is the *only* one currently marked M. And ISTM that it 
> clearly is a T.
The a=imageattr attribute is also indexed by payload type. I just 
haven't gotten to it yet.

>
> I say this because, as you have said elsewhere, each payload type 
> number is tied to a particular codec of a particular codec, whether 
> audio or video. It isn't possible to have payload type number 100 mean 
> one thing for audio and a different thing for video.
>
> OTOH, when multiplexing audio and video on the same 5-tuple it is 
> necessary to know whether a particular a=rtpmap is intended to be 
> audio or video. Normally it inherits that from the m-line it belongs 
> to. That still works for bundle. It doesn't work for mmt.
>
> So, I still think there is no need for M. But a=rtpmap is "special" in 
> that it requires an extra datum. Unless there are other attributes 
> that have the same issue I think this is just a special case.
>
>     Thanks,
>     Paul
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
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On 12/03/2012 10:02 AM, Markus.Isomaki@nokia.com wrote:
> Hi,
>
> I agree that it would be useful to be able to run data channel on the same 5-tuple as RTP/RTCP, and even making it the default and must-implement way for the browsers.
>
> QoS/prioritization comes to mind as the only potential drawback. The endpoint can naturally still give different priorities to the different components (audio, video, "data") it is sending. I assume SCTP also has TCP-style receiver window that can be used to throttle reception if needed. And presumably RMCAT will find solutions to congestion control in genenral. But from the network perspective there is only one flow, and the only(?) way to differentiate the components is to use DSCP. I don't think our QoS (or API) draft yet clearly explains how DSCP's are learned and set, or whether that is even going to be a realistic solution.

That is indeed the only reason I can think of to specify multiple 
m-lines for DTLS/SCTP (and we don't need to bundle them together).

At the moment, the MMUSIC API doesn't define a way to control which 
m-line a DTLS assocation gets mapped onto.

This may be more complexity than is really needed, though.

>
> Markus
>
>> -----Original Message-----
>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
>> Behalf Of ext Harald Alvestrand
>> Sent: 03 December, 2012 10:11
>> To: Christer Holmberg
>> Cc: mmusic@ietf.org
>> Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
>>
>> On 12/03/2012 09:00 AM, Christer Holmberg wrote:
>>> Hi,
>>>
>>> I guess it would be nice to hear from the browser/endpoint people whether
>> they see a need for covering non-RTP in BUNDLE.
>>> If NOT, it would mean a minimum of two 5-tuples, instead of one (one for
>> the bundled RTP media, and one for the data channel).
>>
>> Speaking with my browser implementor hat on, I want to avoid adding an
>> extra 5-tuple for data channels.
>>
>> Our stats on improvement from the introduction of rtcp-mux were pretty
>> compelling that the issue is real.
>>
>>> Regards,
>>>
>>> Christer
>>>
>>>
>>>
>>>
>>> -----Original Message-----
>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
>>> Behalf Of Harald Alvestrand
>>> Sent: 3. joulukuuta 2012 9:54
>>> To: mmusic@ietf.org
>>> Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
>>>
>>> On 12/01/2012 07:09 PM, Paul Kyzivat wrote:
>>>> I must say that I am getting confused about what is being multiplexed
>>>> here. We must be very careful to be precise.
>>>>
>>>> SCTP defines an *association*, which in simple cases is bound to a
>>>> 5-tuple. (For now lets not talk about the cases where each end can
>>>> have more than one address.)
>>>>
>>>> SCTP multiplexes multiple one-way *streams* over an association.
>>>> (These are not RTP streams.)
>>>>
>>>> RTCWEB binds pairs of sctp-streams together into *channels*.
>>> I'm not talking about any of these levels. I'm talking about the reuse of the
>> RTP session's 5-tuple for the DTLS/SRTP 5-tuple, as described in RFC 5764
>> section 5.1.2.
>>> I said the same thing earlier on this thread (Nov 5), when you first
>>> responded (thanks for being the first responder!)
>>>
>>> You do make the point that the term "multiplex" is so over-used, it should
>> probably be abandoned altogether - but that section talks about
>> (de)multiplexing, so changing terminology is probably too late for that
>> particular usage.
>>>                  Harald
>>>
>>> _______________________________________________
>>> 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


From harald@alvestrand.no  Mon Dec  3 03:14:46 2012
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Cc: mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] M-lines: Wikifying it
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On 12/03/2012 12:05 PM, Christer Holmberg wrote:
>
> Hi,
>
> So, the exercise is NOT about how to apply attributes to individual 
> sources?
>

Only in the sense that it is about figuring out which attributes need to 
be applied to individual sources.

> Regards,
>
> Christer
>
> *From:*Harald Alvestrand [mailto:harald@alvestrand.no]
> *Sent:* 3. joulukuuta 2012 12:54
> *To:* Christer Holmberg
> *Cc:* mmusic
> *Subject:* Re: [MMUSIC] M-lines: Wikifying it
>
> On 12/03/2012 11:23 AM, Christer Holmberg wrote:
>
>     Hi,
>
>     A question for clarification:
>
>     When talking about the "F" class, what do you mean by "stream"? Do
>     you mean individual sources associated with the m- line? Or, do
>     you mean all sources associated with the m- line?
>
>     I ASSUME you mean all sources associated with the m- line, because
>     otherwise you will need the a=ssrc attribute.
>
>
> The purpose of the exercise is to figure out what attributes we need 
> to redefine if we define the M-line to be an RTP session (the MMT and 
> the all-videos-under-one-M-line approach) as opposed to what 
> attributes we need to redefine if we define the M-line to be a group 
> of sources that, in turn, may be associated with an RTP session (the 
> BUNDLE approach).
>
> F-lines need redefinition under the first approach, T-lines need 
> redefinition (or adjustment) under the second, M-lines may be able to 
> work in both cases.
>
>
> Regards,
>
> Christer
>
> *From:*mmusic-bounces@ietf.org <mailto:mmusic-bounces@ietf.org> 
> [mailto:mmusic-bounces@ietf.org] *On Behalf Of *Harald Alvestrand
> *Sent:* 28. marraskuuta 2012 16:51
> *To:* mmusic
> *Subject:* [MMUSIC] M-lines: Wikifying it
>
> Perhaps we can do some real work rather than just disagreeing on 
> principles.
>
> Perhaps the most contentious issue now is "what should an M-line be".
> As far as I can tell, M-lines and their parameters describe 3 things:
>
> - Properties of the transport flow / RTP session (Transport, or T for 
> short)
> - Properties that make sense to apply to an individual media stream, 
> and may need to be different for different media streams in the same 
> combined RTP session (Flow, or F for short)
> - Properties that make sense to apply to all streams of a given type 
> (Media, or M for short)
>
> The properties that can be represented in M-lines are found in two 
> IANA registries, both on
> http://www.iana.org/assignments/sdp-parameters/sdp-parameters.xml:
>
> - att-field (both session and media level)
> - att-field (media level only)
>
> I make it 179 entries in these categories, combined.
>
> I see the same kind of work being necessary no matter which way we go:
>
> - If we go with M-lines describing a whole RTP session (the MMT 
> proposal), each property of the "F" class that we stil need needs to 
> be copied onto a property in the "att-field (source level)" registry.
>
> - If we go with M-lines describing groups of media streams, 
> degenerating to a single media stream in the case of using 
> "a=inactive" for stream muting (the BUNDLE and TOGETHER proposals), 
> each property of the "T" class needs to be described as "MUST have 
> consistent values for all the combined M-lines", "MUST be taken only 
> from the first M-line in the bundle argument", or "must get special 
> treatment".
>
> - In both cases, the "M"-class lines don't need any special treatment. 
> They just need to be present in all the M-lines where they're relevant.
>
> I've created a spreadsheet here:
>
> https://docs.google.com/spreadsheet/ccc?key=0Av-aF69sk5C6dDhPblpRV2pkbXVGNV9lUFI3cHlIRFE
>
> with 2 extra columns: "Class" (M, F or T) and "Importance" (1 for 
> "necessary" and 0 for "will never matter").
>
> Anyone should be able to write it.
> If a few of us do a quick job of markup on it, I think we should be 
> able to have a more realistic assessment in a few days.
>
> I've started, but much of this is unknown to me. Can others help?
>
>                       Harald
>
>
>
>
>


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    <div class="moz-cite-prefix">On 12/03/2012 12:05 PM, Christer
      Holmberg wrote:<br>
    </div>
    <blockquote
      cite="mid:7594FB04B1934943A5C02806D1A2204B04D2C6@ESESSMB209.ericsson.se"
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      <div class="WordSection1">
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi,<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">So,
            the exercise is NOT about how to apply attributes to
            individual sources?</span></p>
      </div>
    </blockquote>
    <br>
    Only in the sense that it is about figuring out which attributes
    need to be applied to individual sources.<br>
    <br>
    <blockquote
      cite="mid:7594FB04B1934943A5C02806D1A2204B04D2C6@ESESSMB209.ericsson.se"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Regards,<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Christer
            <o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <div>
          <div style="border:none;border-top:solid #B5C4DF
            1.0pt;padding:3.0pt 0cm 0cm 0cm">
            <p class="MsoNormal"><b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">From:</span></b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">
                Harald Alvestrand [<a class="moz-txt-link-freetext" href="mailto:harald@alvestrand.no">mailto:harald@alvestrand.no</a>]
                <br>
                <b>Sent:</b> 3. joulukuuta 2012 12:54<br>
                <b>To:</b> Christer Holmberg<br>
                <b>Cc:</b> mmusic<br>
                <b>Subject:</b> Re: [MMUSIC] M-lines: Wikifying it<o:p></o:p></span></p>
          </div>
        </div>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        <div>
          <p class="MsoNormal">On 12/03/2012 11:23 AM, Christer Holmberg
            wrote:<o:p></o:p></p>
        </div>
        <blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
          <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi,</span><o:p></o:p></p>
          <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p></p>
          <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">A
              question for clarification:</span><o:p></o:p></p>
          <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p></p>
          <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">When
              talking about the &#8220;F&#8221; class, what do you mean by &#8220;stream&#8221;?
              Do you mean individual sources associated with the m-
              line? Or, do you mean all sources associated with the m-
              line?</span><o:p></o:p></p>
          <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p></p>
          <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">I
              ASSUME you mean all sources associated with the m- line,
              because otherwise you will need the a=ssrc attribute.</span><o:p></o:p></p>
        </blockquote>
        <p class="MsoNormal"><br>
          The purpose of the exercise is to figure out what attributes
          we need to redefine if we define the M-line to be an RTP
          session (the MMT and the all-videos-under-one-M-line approach)
          as opposed to what attributes we need to redefine if we define
          the M-line to be a group of sources that, in turn, may be
          associated with an RTP session (the BUNDLE approach).<br>
          <br>
          F-lines need redefinition under the first approach, T-lines
          need redefinition (or adjustment) under the second, M-lines
          may be able to work in both cases.<br>
          <br>
          <br>
          <o:p></o:p></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Regards,</span><o:p></o:p></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Christer</span><o:p></o:p></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p></p>
        <div>
          <div style="border:none;border-top:solid #B5C4DF
            1.0pt;padding:3.0pt 0cm 0cm 0cm">
            <p class="MsoNormal"><b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">From:</span></b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">
                <a moz-do-not-send="true"
                  href="mailto:mmusic-bounces@ietf.org">mmusic-bounces@ietf.org</a>
                [<a moz-do-not-send="true"
                  href="mailto:mmusic-bounces@ietf.org">mailto:mmusic-bounces@ietf.org</a>]
                <b>On Behalf Of </b>Harald Alvestrand<br>
                <b>Sent:</b> 28. marraskuuta 2012 16:51<br>
                <b>To:</b> mmusic<br>
                <b>Subject:</b> [MMUSIC] M-lines: Wikifying it</span><o:p></o:p></p>
          </div>
        </div>
        <p class="MsoNormal">&nbsp;<o:p></o:p></p>
        <p class="MsoNormal" style="margin-bottom:12.0pt">Perhaps we can
          do some real work rather than just disagreeing on principles.<br>
          <br>
          Perhaps the most contentious issue now is "what should an
          M-line be".<br>
          As far as I can tell, M-lines and their parameters describe 3
          things:<br>
          <br>
          - Properties of the transport flow / RTP session (Transport,
          or T for short)<br>
          - Properties that make sense to apply to an individual media
          stream, and may need to be different for different media
          streams in the same combined RTP session (Flow, or F for
          short)<br>
          - Properties that make sense to apply to all streams of a
          given type (Media, or M for short)<br>
          <br>
          The properties that can be represented in M-lines are found in
          two IANA registries, both on<br>
          <a moz-do-not-send="true"
            href="http://www.iana.org/assignments/sdp-parameters/sdp-parameters.xml">http://www.iana.org/assignments/sdp-parameters/sdp-parameters.xml</a>:<br>
          <br>
          - att-field (both session and media level)<br>
          - att-field (media level only)<br>
          <br>
          I make it 179 entries in these categories, combined.<br>
          <br>
          I see the same kind of work being necessary no matter which
          way we go:<br>
          <br>
          - If we go with M-lines describing a whole RTP session (the
          MMT proposal), each property of the "F" class that we stil
          need needs to be copied onto a property in the "att-field
          (source level)" registry.<br>
          <br>
          - If we go with M-lines describing groups of media streams,
          degenerating to a single media stream in the case of using
          "a=inactive" for stream muting (the BUNDLE and TOGETHER
          proposals), each property of the "T" class needs to be
          described as "MUST have consistent values for all the combined
          M-lines", "MUST be taken only from the first M-line in the
          bundle argument", or "must get special treatment".<br>
          <br>
          - In both cases, the "M"-class lines don't need any special
          treatment. They just need to be present in all the M-lines
          where they're relevant.<br>
          <br>
          I've created a spreadsheet here:<br>
          <br>
          <a moz-do-not-send="true"
href="https://docs.google.com/spreadsheet/ccc?key=0Av-aF69sk5C6dDhPblpRV2pkbXVGNV9lUFI3cHlIRFE">https://docs.google.com/spreadsheet/ccc?key=0Av-aF69sk5C6dDhPblpRV2pkbXVGNV9lUFI3cHlIRFE</a><br>
          <br>
          with 2 extra columns: "Class" (M, F or T) and "Importance" (1
          for "necessary" and 0 for "will never matter").<br>
          <br>
          Anyone should be able to write it.<br>
          If a few of us do a quick job of markup on it, I think we
          should be able to have a more realistic assessment in a few
          days.<br>
          <br>
          I've started, but much of this is unknown to me. Can others
          help?<br>
          <br>
          &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
          <br>
          <br>
          <br>
          <br>
          <br>
          <o:p></o:p></p>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
      </div>
    </blockquote>
    <br>
  </body>
</html>

--------------030806030205050208080509--

From christer.holmberg@ericsson.com  Mon Dec  3 03:16:45 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: [MMUSIC] M-lines: Wikifying it
Thread-Index: AQHNzXfLf68XUJVyMEaaKo/W5rPZGpgG5AmA///5IgCAABQFMP//8eMAgAAREeA=
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Cc: mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] M-lines: Wikifying it
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Hi,
=A0
>>So, the exercise is NOT about how to apply attributes to individual sourc=
es?
>
>Only in the sense that it is about figuring out which attributes need to b=
e applied to individual sources.

Well, I believe that was my original question :)

If you want to apply an attribute to an individual source, you must use the=
 ssrc attribute.=20
=A0
Regards,
=A0
Christer=20
=A0
From: Harald Alvestrand [mailto:harald@alvestrand.no]=20
Sent: 3. joulukuuta 2012 12:54
To: Christer Holmberg
Cc: mmusic
Subject: Re: [MMUSIC] M-lines: Wikifying it
=A0
On 12/03/2012 11:23 AM, Christer Holmberg wrote:
Hi,
=A0
A question for clarification:
=A0
When talking about the "F" class, what do you mean by "stream"? Do you mean=
 individual sources associated with the m- line? Or, do you mean all source=
s associated with the m- line?
=A0
I ASSUME you mean all sources associated with the m- line, because otherwis=
e you will need the a=3Dssrc attribute.

The purpose of the exercise is to figure out what attributes we need to red=
efine if we define the M-line to be an RTP session (the MMT and the all-vid=
eos-under-one-M-line approach) as opposed to what attributes we need to red=
efine if we define the M-line to be a group of sources that, in turn, may b=
e associated with an RTP session (the BUNDLE approach).

F-lines need redefinition under the first approach, T-lines need redefiniti=
on (or adjustment) under the second, M-lines may be able to work in both ca=
ses.



=A0
Regards,
=A0
Christer
=A0
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 Harald Alvestrand
Sent: 28. marraskuuta 2012 16:51
To: mmusic
Subject: [MMUSIC] M-lines: Wikifying it
=A0
Perhaps we can do some real work rather than just disagreeing on principles=
.

Perhaps the most contentious issue now is "what should an M-line be".
As far as I can tell, M-lines and their parameters describe 3 things:

- Properties of the transport flow / RTP session (Transport, or T for short=
)
- Properties that make sense to apply to an individual media stream, and ma=
y need to be different for different media streams in the same combined RTP=
 session (Flow, or F for short)
- Properties that make sense to apply to all streams of a given type (Media=
, or M for short)

The properties that can be represented in M-lines are found in two IANA reg=
istries, both on
http://www.iana.org/assignments/sdp-parameters/sdp-parameters.xml:

- att-field (both session and media level)
- att-field (media level only)

I make it 179 entries in these categories, combined.

I see the same kind of work being necessary no matter which way we go:

- If we go with M-lines describing a whole RTP session (the MMT proposal), =
each property of the "F" class that we stil need needs to be copied onto a =
property in the "att-field (source level)" registry.

- If we go with M-lines describing groups of media streams, degenerating to=
 a single media stream in the case of using "a=3Dinactive" for stream mutin=
g (the BUNDLE and TOGETHER proposals), each property of the "T" class needs=
 to be described as "MUST have consistent values for all the combined M-lin=
es", "MUST be taken only from the first M-line in the bundle argument", or =
"must get special treatment".

- In both cases, the "M"-class lines don't need any special treatment. They=
 just need to be present in all the M-lines where they're relevant.

I've created a spreadsheet here:

https://docs.google.com/spreadsheet/ccc?key=3D0Av-aF69sk5C6dDhPblpRV2pkbXVG=
NV9lUFI3cHlIRFE

with 2 extra columns: "Class" (M, F or T) and "Importance" (1 for "necessar=
y" and 0 for "will never matter").

Anyone should be able to write it.
If a few of us do a quick job of markup on it, I think we should be able to=
 have a more realistic assessment in a few days.

I've started, but much of this is unknown to me. Can others help?

=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 Harald






=A0


From harald@alvestrand.no  Mon Dec  3 07:48:57 2012
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Cc: mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] M-lines: Wikifying it
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On 12/03/2012 12:16 PM, Christer Holmberg wrote:
> Hi,
>   
>>> So, the exercise is NOT about how to apply attributes to individual sources?
>> Only in the sense that it is about figuring out which attributes need to be applied to individual sources.
> Well, I believe that was my original question :)
>
> If you want to apply an attribute to an individual source, you must use the ssrc attribute.

If you are in the camp that believes that there should be more than one 
independent source per M-line.
That camp seemed to me to be in the minority in Atlanta; it was 
absolutely not the consensus opinion.


From christer.holmberg@ericsson.com  Mon Dec  3 08:12:37 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: [MMUSIC] M-lines: Wikifying it
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Subject: Re: [MMUSIC] M-lines: Wikifying it
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Hi,

>>>> So, the exercise is NOT about how to apply attributes to individual so=
urces?
>>> Only in the sense that it is about figuring out which attributes need t=
o be applied to individual sources.
>> Well, I believe that was my original question :)
>>
>> If you want to apply an attribute to an individual source, you must use =
the ssrc attribute.
>
> If you are in the camp that believes that there should be more than onein=
dependent source per M-line.
>
> That camp seemed to me to be in the minority in Atlanta; it was absolutel=
y not the consensus opinion.

It's not about in which camp someone is, it's about the assumptions we make=
 in the wiki study. If the assumption is one source per m- line, than I thi=
nk it should be explicitly stated.

Regards,

Christer




From harald@alvestrand.no  Mon Dec  3 08:38:28 2012
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Cc: mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] M-lines: Wikifying it
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On 12/03/2012 05:12 PM, Christer Holmberg wrote:
> Hi,
>
>>>>> So, the exercise is NOT about how to apply attributes to individual sources?
>>>> Only in the sense that it is about figuring out which attributes need to be applied to individual sources.
>>> Well, I believe that was my original question :)
>>>
>>> If you want to apply an attribute to an individual source, you must use the ssrc attribute.
>> If you are in the camp that believes that there should be more than oneindependent source per M-line.
>>
>> That camp seemed to me to be in the minority in Atlanta; it was absolutely not the consensus opinion.
> It's not about in which camp someone is, it's about the assumptions we make in the wiki study. If the assumption is one source per m- line, than I think it should be explicitly stated.
Trying again, since I'm obviously not making myself clear:

The two alternative approaches are:

- Everything that needs to be controlled separately is in its own 
M-line. Everything that needs to go on the same RTP session uses BUNDLE 
to get there. Transport parameters get figured out somehow.

- Everything that goes on the same RTP session goes on the same M-line. 
Due to backwards compatible SDP, we may choose to use BUNDLE to lump 
video and audio together - or we may choose MMT if that works for 
people. Anything that needs separate control uses a=ssrc.

In Atlanta, the room was clearly split between the two. I'm trying to 
get real data on how many things need to be redefined for each case.

>
> Regards,
>
> Christer
>
>
>


From christer.holmberg@ericsson.com  Mon Dec  3 08:45:00 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Harald Alvestrand' <harald@alvestrand.no>
Thread-Topic: [MMUSIC] M-lines: Wikifying it
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Cc: mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] M-lines: Wikifying it
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Hi,=20

>>>>>> So, the exercise is NOT about how to apply attributes to individual =
sources?
>>>>> Only in the sense that it is about figuring out which attributes need=
 to be applied to individual sources.
>>>> Well, I believe that was my original question :)
>>>>
>>>> If you want to apply an attribute to an individual source, you must us=
e the ssrc attribute.
>>> If you are in the camp that believes that there should be more than one=
independent source per M-line.
>>>
>>> That camp seemed to me to be in the minority in Atlanta; it was absolut=
ely not the consensus opinion.
>> It's not about in which camp someone is, it's about the assumptions we m=
ake in the wiki study. If the assumption is one source per m- line, than I =
think it should be explicitly stated.
>> Trying again, since I'm obviously not making myself clear:
>
> The two alternative approaches are:
>
> - Everything that needs to be controlled separately is in its own M-line.=
 Everything that needs to go on the same RTP session uses BUNDLE to get the=
re. Transport parameters get figured out somehow.
>
> - Everything that goes on the same RTP session goes on the same M-line.=20
> Due to backwards compatible SDP, we may choose to use BUNDLE to lump vide=
o and audio together - or we may choose MMT if that works for people. Anyth=
ing that needs separate control uses a=3Dssrc.

Exactly!

So, which (if any) approach do you assume in the wiki? :)

Regards,

Christer



> In Atlanta, the room was clearly split between the two. I'm trying to get=
 real data on how many things need to be redefined for each case.






From harald@alvestrand.no  Mon Dec  3 08:59:18 2012
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Subject: Re: [MMUSIC] M-lines: Wikifying it
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On 12/03/2012 05:44 PM, Christer Holmberg wrote:
> Hi,
>
>>>>>>> So, the exercise is NOT about how to apply attributes to individual sources?
>>>>>> Only in the sense that it is about figuring out which attributes need to be applied to individual sources.
>>>>> Well, I believe that was my original question :)
>>>>>
>>>>> If you want to apply an attribute to an individual source, you must use the ssrc attribute.
>>>> If you are in the camp that believes that there should be more than oneindependent source per M-line.
>>>>
>>>> That camp seemed to me to be in the minority in Atlanta; it was absolutely not the consensus opinion.
>>> It's not about in which camp someone is, it's about the assumptions we make in the wiki study. If the assumption is one source per m- line, than I think it should be explicitly stated.
>>> Trying again, since I'm obviously not making myself clear:
>> The two alternative approaches are:
>>
>> - Everything that needs to be controlled separately is in its own M-line. Everything that needs to go on the same RTP session uses BUNDLE to get there. Transport parameters get figured out somehow.
>>
>> - Everything that goes on the same RTP session goes on the same M-line.
>> Due to backwards compatible SDP, we may choose to use BUNDLE to lump video and audio together - or we may choose MMT if that works for people. Anything that needs separate control uses a=ssrc.
> Exactly!
>
> So, which (if any) approach do you assume in the wiki? :)
For the 5th time: I'm NOT assuming one or the other approach.
I'm trying to gather information that people can use to decide which 
approach to follow.
That REQUIRES that the wiki does NOT assume one approach.

Do you hear me now?


From christer.holmberg@ericsson.com  Mon Dec  3 09:18:49 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Harald Alvestrand' <harald@alvestrand.no>
Thread-Topic: [MMUSIC] M-lines: Wikifying it
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Hi,=20

>>>>>>>> So, the exercise is NOT about how to apply attributes to individua=
l sources?
>>>>>>> Only in the sense that it is about figuring out which attributes ne=
ed to be applied to individual sources.
>>>>>> Well, I believe that was my original question :)
>>>>>>
>>>>>> If you want to apply an attribute to an individual source, you must =
use the ssrc attribute.
>>>>> If you are in the camp that believes that there should be more than o=
neindependent source per M-line.
>>>>>
>>>>> That camp seemed to me to be in the minority in Atlanta; it was absol=
utely not the consensus opinion.
>>>> It's not about in which camp someone is, it's about the assumptions we=
 make in the wiki study. If the assumption is one source per m- line, than =
I think it should be explicitly stated.
>>>> Trying again, since I'm obviously not making myself clear:
>>> The two alternative approaches are:
>>>
>>> - Everything that needs to be controlled separately is in its own M-lin=
e. Everything that needs to go on the same RTP session uses BUNDLE to get t=
here. Transport parameters get figured out somehow.
>>>
>>> - Everything that goes on the same RTP session goes on the same M-line.
>>> Due to backwards compatible SDP, we may choose to use BUNDLE to lump vi=
deo and audio together - or we may choose MMT if that works for people. Any=
thing that needs separate control uses a=3Dssrc.
>> Exactly!
>>
>> So, which (if any) approach do you assume in the wiki? :)
> For the 5th time: I'm NOT assuming one or the other approach.
> I'm trying to gather information that people can use to decide which appr=
oach to follow.
> That REQUIRES that the wiki does NOT assume one approach.
>
> Do you hear me now?

So, coming back to my original question: For "F", a *stream* refers to an i=
ndividual source (and is not related to whether one m- line will or will no=
t contain multiple sources). Correct?

Regards,

Christer


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Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
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On 12/3/12 2:53 AM, Harald Alvestrand wrote:
> On 12/01/2012 07:09 PM, Paul Kyzivat wrote:
>> I must say that I am getting confused about what is being multiplexed
>> here. We must be very careful to be precise.
>>
>> SCTP defines an *association*, which in simple cases is bound to a
>> 5-tuple. (For now lets not talk about the cases where each end can
>> have more than one address.)
>>
>> SCTP multiplexes multiple one-way *streams* over an association.
>> (These are not RTP streams.)
>>
>> RTCWEB binds pairs of sctp-streams together into *channels*.
>
> I'm not talking about any of these levels. I'm talking about the reuse
> of the RTP session's 5-tuple for the DTLS/SRTP 5-tuple, as described in
> RFC 5764 section 5.1.2.

These are all related. We are making even more of a mess of SDP than it 
already is by patching on a new hack for each sort of multiplexing 
somebody comes up with. But focusing on just this:

How do you imagine this multiplexing will be signaled in SDP?
The obvious way would be to bundle the DTLS/SCTP m-line with the RTP 
m-lines. But that will violate the rule that all the bundled m-lines 
have the same <proto>. (And there is an even worst fit between this and 
mmt.)

	Thanks,
	Paul


From pkyzivat@alum.mit.edu  Mon Dec  3 12:00:21 2012
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Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
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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. 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.

	Thanks,
	Paul


From pkyzivat@alum.mit.edu  Mon Dec  3 12:13:53 2012
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Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
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On 12/3/12 6:13 AM, Harald Alvestrand wrote:
> On 12/03/2012 10:02 AM, Markus.Isomaki@nokia.com wrote:
>> Hi,
>>
>> I agree that it would be useful to be able to run data channel on the
>> same 5-tuple as RTP/RTCP, and even making it the default and
>> must-implement way for the browsers.
>>
>> QoS/prioritization comes to mind as the only potential drawback. The
>> endpoint can naturally still give different priorities to the
>> different components (audio, video, "data") it is sending. I assume
>> SCTP also has TCP-style receiver window that can be used to throttle
>> reception if needed. And presumably RMCAT will find solutions to
>> congestion control in genenral. But from the network perspective there
>> is only one flow, and the only(?) way to differentiate the components
>> is to use DSCP. I don't think our QoS (or API) draft yet clearly
>> explains how DSCP's are learned and set, or whether that is even going
>> to be a realistic solution.
>
> That is indeed the only reason I can think of to specify multiple
> m-lines for DTLS/SCTP (and we don't need to bundle them together).
>
> At the moment, the MMUSIC API doesn't define a way to control which
> m-line a DTLS assocation gets mapped onto.

MMUSIC *API*???

Do you mean the SDP signaling for this? (draft-ietf-mmusic-sctp-sdp-02)

It specifies a new m-line format for this.

	Thanks,
	Paul


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Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
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Hi,

Paul Kyzivat wrote:
>On 12/3/12 6:13 AM, Harald Alvestrand wrote:
>> On 12/03/2012 10:02 AM, Markus.Isomaki@nokia.com wrote:
>>> Hi,
>>>I don't think our QoS (or API) draft yet
>>> clearly explains how DSCP's are learned and set, or whether that is
>>> even going to be a realistic solution.
>>
>> That is indeed the only reason I can think of to specify multiple
>> m-lines for DTLS/SCTP (and we don't need to bundle them together).
>>
>> At the moment, the MMUSIC API doesn't define a way to control which
>> m-line a DTLS assocation gets mapped onto.
>
>MMUSIC *API*???
>
>Do you mean the SDP signaling for this? (draft-ietf-mmusic-sctp-sdp-02)
>
>It specifies a new m-line format for this.
>

Sorry about starting the confusion. I thought this was on rtcweb mailing li=
st and with API I was in my reply actually referring to the QoS/prioritizat=
ion related APIs in W3C related to WebRTC. Don't know which API Harald is r=
eferring to, though :-)

Markus

From pkyzivat@alum.mit.edu  Mon Dec  3 12:29:32 2012
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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: [MMUSIC] Some issues with draft-ietf-mmusic-sctp-sdp-02
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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.

	Thanks,
	Paul

From pkyzivat@alum.mit.edu  Mon Dec  3 13:37:54 2012
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Subject: Re: [MMUSIC] M-lines: Wikifying it - a=label
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On 12/3/12 6:06 AM, Harald Alvestrand wrote:

>>>> Regarding a=label: This could be any of T/F/M, depending on
>>>> application use. I think we must assume T, and consider an alternative
>>>> for F. (And for M if we think it is needed.)
>>
>> I may have misunderstood your intended meaning for M. I assume you
>> meant all RTP streams of the same media type (audio or video) within a
>> single 5-tuple.
> I was thinking "same payload type", since the payload type is the
> distinguisher used for groups of flows that should have the same
> parameters in many cases.
>>
>>> The claim that a=label can be used to mark flows makes it F.
>>
>> a=label is a media-level attribute, so it is bound to an m-line.
>>
>> In the absence of bundle, regardless of whether the corresponding RTP
>> session has one or several flows. The m-line has a particular media
>> type, so I guess it could be viewed as either T or M.
> The reason why we're having the discussion is that we have two competing
> definitons of what an M-line should be, and need to choose one.
> So arguing from whether or not something associates with an M-line is
> not useful.
>>
>> Some applications of a=label may assume or require that the labeled
>> m-line contain only one RTP stream. If so, then perhaps the label
>> could be considered to be F. But we can't really know that.
> What applications of a=label do you know about?

I know of two: BFCP and SIPREC.

In BFCP the label is used to associate a bfcp floor with an m-line. I 
don't think it matters whether that m-line carries one or several RTP 
streams.

We can speculate how we would want this to work in the case that several 
media m-lines were bundled together. I don't know - too many hypotheticals.

In SIPREC, the label is used to associate an m-line with certain 
metadata that describes the source of the media being sent via that 
m-line. The siprec group has intentionally kept the question of whether 
the content in that m-line is a single RTP stream or if it can be 
several multiplexed together. There is nothing to forbid multiplexing, 
but no specification of what it means either.

SIPREC *might* have specified use of bundle, or explicitly specified 
multiplexing of RTP streams over a single m-line, if those things were 
better defined. But how it might eventually use bundle or mmt is again a 
hypothetical right now.

A problem for siprec is that it is recording media from existing sip 
sessions, without any knowledge of how they use SDP and RTP. We could 
find it recording a call initiated with RTCWEB that uses bundle or mmt. 
Most likely siprec will need to be extended to support that, though I 
wish that wasn't so.

	Thanks,
	Paul


From pkyzivat@alum.mit.edu  Mon Dec  3 13:44:36 2012
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Subject: Re: [MMUSIC] M-lines: Wikifying it
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On 12/3/12 10:48 AM, Harald Alvestrand wrote:
> On 12/03/2012 12:16 PM, Christer Holmberg wrote:
>> Hi,
>>>> So, the exercise is NOT about how to apply attributes to individual
>>>> sources?
>>> Only in the sense that it is about figuring out which attributes need
>>> to be applied to individual sources.
>> Well, I believe that was my original question :)
>>
>> If you want to apply an attribute to an individual source, you must
>> use the ssrc attribute.
>
> If you are in the camp that believes that there should be more than one
> independent source per M-line.
> That camp seemed to me to be in the minority in Atlanta; it was
> absolutely not the consensus opinion.

There were certainly competing camps supporting bundle and mmt.
It was not my understanding that supporting bundle means believing that 
each source gets its own m-line.

(I was going to check the mmusic minutes, but they aren't posted yet.)

	Thanks,
	Paul


From christer.holmberg@ericsson.com  Tue Dec  4 00:08:46 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] M-lines: Wikifying it
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Subject: Re: [MMUSIC] M-lines: Wikifying it
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Hi,

>>>>> So, the exercise is NOT about how to apply attributes to individual=20
>>>>> sources?
>>>> Only in the sense that it is about figuring out which attributes=20
>>>> need to be applied to individual sources.
>>> Well, I believe that was my original question :)
>>>
>>> If you want to apply an attribute to an individual source, you must=20
>>> use the ssrc attribute.
>>
>> If you are in the camp that believes that there should be more than=20
>> one independent source per M-line.
>> That camp seemed to me to be in the minority in Atlanta; it was=20
>> absolutely not the consensus opinion.
>
> There were certainly competing camps supporting bundle and mmt.
>
> It was not my understanding that supporting bundle means believing that e=
ach source gets its own m-line.

Correct.

However, I think the "BUNDLE vs MMT" issue is separate from the "single- vs=
 multiple source per m- line" issue, which I guess is what Harald was refer=
ring to.

Regards,

Christer


From harald@alvestrand.no  Tue Dec  4 01:22:53 2012
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] M-lines: Wikifying it
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On 12/03/2012 06:18 PM, Christer Holmberg wrote:
> Hi,
>
>>>>>>>>> So, the exercise is NOT about how to apply attributes to individual sources?
>>>>>>>> Only in the sense that it is about figuring out which attributes need to be applied to individual sources.
>>>>>>> Well, I believe that was my original question :)
>>>>>>>
>>>>>>> If you want to apply an attribute to an individual source, you must use the ssrc attribute.
>>>>>> If you are in the camp that believes that there should be more than oneindependent source per M-line.
>>>>>>
>>>>>> That camp seemed to me to be in the minority in Atlanta; it was absolutely not the consensus opinion.
>>>>> It's not about in which camp someone is, it's about the assumptions we make in the wiki study. If the assumption is one source per m- line, than I think it should be explicitly stated.
>>>>> Trying again, since I'm obviously not making myself clear:
>>>> The two alternative approaches are:
>>>>
>>>> - Everything that needs to be controlled separately is in its own M-line. Everything that needs to go on the same RTP session uses BUNDLE to get there. Transport parameters get figured out somehow.
>>>>
>>>> - Everything that goes on the same RTP session goes on the same M-line.
>>>> Due to backwards compatible SDP, we may choose to use BUNDLE to lump video and audio together - or we may choose MMT if that works for people. Anything that needs separate control uses a=ssrc.
>>> Exactly!
>>>
>>> So, which (if any) approach do you assume in the wiki? :)
>> For the 5th time: I'm NOT assuming one or the other approach.
>> I'm trying to gather information that people can use to decide which approach to follow.
>> That REQUIRES that the wiki does NOT assume one approach.
>>
>> Do you hear me now?
> So, coming back to my original question: For "F", a *stream* refers to an individual source (and is not related to whether one m- line will or will not contain multiple sources). Correct?
I don't want to mix in the "source" word (CLUE has talked about sources 
that originate multiple streams enough to confuse that term), but if you 
want to think in terms of sources, yes.

>
> Regards,
>
> Christer
>


From adam@nostrum.com  Tue Dec  4 07:47:51 2012
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Subject: [MMUSIC] SDP Directorate Feedback for draft-ietf-xrblock-rtcp-xr-burst-gap-loss
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I have been asked to perform the SDP Directorate review for 
draft-ietf-xrblock-rtcp-xr-burst-gap-loss. I reviewed the -05 version of 
the document.

In general, the syntax looks fine -- this document makes use of a 
defined extension point in RFC 3611. I find no mismatch between the 
proposed semantics of this document and what appears to be intended by 
xrblock usage.

I found a handful of tiny nits in the IANA section:

 1. The first line of section 6.1 says "NDEL" where it means "NBGL." I
    would imagine this gets cleaned up during IANA review, but there's
    no point leaving it confusing.

 2. Section 6.2 refers to the "RTCP XR SDP Parameters Registry," whereas
    IANA has titled the registry "RTP Control Protocol Extended Reports
    (RTCP XR) Session Description Protocol (SDP) Parameters Registry"
    (cf.
    http://www.iana.org/assignments/rtcp-xr-sdp-parameters/rtcp-xr-sdp-parameters.xml).
    It would help readers of this document who wish to locate the
    registry if the precise IANA name were used.

 3. Section 6.1 has a similar mismatch, using "RTCP XR Block Type
    Registry" rather than "RTP Control Protocol Extended Reports (RTCP
    XR) Block Type Registry"


Thanks.

/a

--------------060805080809080809080509
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<html>
  <head>

    <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    I have been asked to perform the SDP Directorate review for
    draft-ietf-xrblock-rtcp-xr-burst-gap-loss. I reviewed the -05
    version of the document.<br>
    <br>
    In general, the syntax looks fine -- this document makes use of a
    defined extension point in RFC 3611. I find no mismatch between the
    proposed semantics of this document and what appears to be intended
    by xrblock usage.<br>
    <br>
    I found a handful of tiny nits in the IANA section:<br>
    <br>
    <ol>
      <li>The first line of section 6.1 says "NDEL" where it means
        "NBGL." I would imagine this gets cleaned up during IANA review,
        but there's no point leaving it confusing.<br>
        <br>
      </li>
      <li>Section 6.2 refers to the "RTCP XR SDP Parameters Registry,"
        whereas IANA has titled the registry "RTP Control Protocol
        Extended Reports (RTCP XR) Session Description Protocol (SDP)
        Parameters Registry" (cf. <a class="moz-txt-link-freetext"
href="http://www.iana.org/assignments/rtcp-xr-sdp-parameters/rtcp-xr-sdp-parameters.xml">http://www.iana.org/assignments/rtcp-xr-sdp-parameters/rtcp-xr-sdp-parameters.xml</a>).&nbsp;

        It would help readers of this document who wish to locate the
        registry if the precise IANA name were used.<br>
        <br>
      </li>
      <li>Section 6.1 has a similar mismatch, using "RTCP XR Block Type
        Registry" rather than "RTP Control Protocol Extended Reports
        (RTCP XR) Block Type Registry"</li>
    </ol>
    <br>
    Thanks.<br>
    <br>
    /a<br>
  </body>
</html>

--------------060805080809080809080509--

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From ari.keranen@nomadiclab.com  Wed Dec  5 13:22:05 2012
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From: Ari Keranen <ari.keranen@nomadiclab.com>
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Subject: [MMUSIC] draft-ietf-mmusic-rtsp-nat-14 review
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Hi,

I reviewed the draft-ietf-mmusic-rtsp-nat-14 and I think the document is 
in pretty good shape and ready to move forward. Some (mostly editorial) 
comments to consider:


3.  Solution Overview

         The client then installs the STUN servers on each of
         the local candidates transport addresses it has gathered

I would rather say "The client then prepares to act as a STUN server on 
each [...]"


         If the server has a public
         IP address, then a single candidate per address family

In some cases (e.g., certain ISPs, I've heard) even "public" IP address 
does not mean you're not behind a NAT. Maybe it's better to say here "If 
the server is not behind a NAT". Same issue in step 8.


         If the server has a public IP address with a single candidate
         per media stream, component and address family, then the server
         may be configured to not initiate connectivity checks.

This is the "ICE-RTSP", right? Maybe one (small) subsection defining 
this "feature" would make sense. Now it's a bit trickled around in this 
section, sec 4.8.1, 5.4, etc. For someone not familiar with ICE-RTSP, it 
looks like you're talking about ICE lite.


    the NAT need to be refreshed by the client to server traffic provided
    by the STUN keep-alive.

s/keep-alive/keep-alive mechanism/ (or messages).
Also in section 5.11.


5.7.  Client to Server ICE Connectivity Check

    The client receives the SETUP response and learns the candidate
    address to use for the connectivity checks.

s/address/addresses/ (right?)


    Aggressive nomination SHALL be used with RTSP.  This doesn't have the
    negative impact that it has in offer/answer as media playing only
    starts after issuing a PLAY request.

But it does have the downside of possibly selecting a lower-priority 
pair due to single dropped (e.g., due to congestion) connectivity check 
message for a higher priority pair. That said, probably SHALL is 
appropriate here, just wanted to make sure the implications are understood.


5.12.  Re-SETUP

    If the client decides to change any parameters related to the media
    stream setup

Could clarify (again) here that we talk about ICE parameters.


Cheers,
Ari

From harald@alvestrand.no  Thu Dec  6 05:02:43 2012
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Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
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On 12/03/2012 09:13 PM, Paul Kyzivat wrote:
> On 12/3/12 6:13 AM, Harald Alvestrand wrote:
>> On 12/03/2012 10:02 AM, Markus.Isomaki@nokia.com wrote:
>>> Hi,
>>>
>>> I agree that it would be useful to be able to run data channel on the
>>> same 5-tuple as RTP/RTCP, and even making it the default and
>>> must-implement way for the browsers.
>>>
>>> QoS/prioritization comes to mind as the only potential drawback. The
>>> endpoint can naturally still give different priorities to the
>>> different components (audio, video, "data") it is sending. I assume
>>> SCTP also has TCP-style receiver window that can be used to throttle
>>> reception if needed. And presumably RMCAT will find solutions to
>>> congestion control in genenral. But from the network perspective there
>>> is only one flow, and the only(?) way to differentiate the components
>>> is to use DSCP. I don't think our QoS (or API) draft yet clearly
>>> explains how DSCP's are learned and set, or whether that is even going
>>> to be a realistic solution.
>>
>> That is indeed the only reason I can think of to specify multiple
>> m-lines for DTLS/SCTP (and we don't need to bundle them together).
>>
>> At the moment, the MMUSIC API doesn't define a way to control which
>> m-line a DTLS assocation gets mapped onto.
>
> MMUSIC *API*???

Sorry, I was writing at a non-awake time.
I meant the WebRTC API (Javascript, specified in the W3C WG).

I

From magnus.westerlund@ericsson.com  Thu Dec  6 07:03:14 2012
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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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To: Ari Keranen <ari.keranen@nomadiclab.com>
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Cc: draft-ietf-mmusic-rtsp-nat@tools.ietf.org, mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] draft-ietf-mmusic-rtsp-nat-14 review
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Hi Ari,

Thanks for the review. I feel much more confident now that we haven't
significantly screwed up any ICE details.

On 2012-12-05 22:22, Ari Keranen wrote:
> Hi,
> 
> I reviewed the draft-ietf-mmusic-rtsp-nat-14 and I think the document is
> in pretty good shape and ready to move forward. Some (mostly editorial)
> comments to consider:
> 
> 
> 3.  Solution Overview
> 
>         The client then installs the STUN servers on each of
>         the local candidates transport addresses it has gathered
> 
> I would rather say "The client then prepares to act as a STUN server on
> each [...]"

Sure, maybe "runs STUN servers" is better than both "installs" and
"act". They are STUN server, but with limited utility. Act as STUN
servers sounds like it is impersonating STUN, rather than actually being
STUN servers.

> 
> 
>         If the server has a public
>         IP address, then a single candidate per address family
> 
> In some cases (e.g., certain ISPs, I've heard) even "public" IP address
> does not mean you're not behind a NAT. Maybe it's better to say here "If
> the server is not behind a NAT". Same issue in step 8.

Sure, maybe we need to define what we mean with not being behind a NAT.
It is a matter of being reachable from the top anchorin address domain
in regards to the clients.

> 
> 
>         If the server has a public IP address with a single candidate
>         per media stream, component and address family, then the server
>         may be configured to not initiate connectivity checks.
> 
> This is the "ICE-RTSP", right? Maybe one (small) subsection defining
> this "feature" would make sense. Now it's a bit trickled around in this
> section, sec 4.8.1, 5.4, etc. For someone not familiar with ICE-RTSP, it
> looks like you're talking about ICE lite.

Yes, this is definitely ICE-RTSP. It is still doing triggered checks.

> 
> 
>    the NAT need to be refreshed by the client to server traffic provided
>    by the STUN keep-alive.
> 
> s/keep-alive/keep-alive mechanism/ (or messages).
> Also in section 5.11.

Will address.

> 
> 
> 5.7.  Client to Server ICE Connectivity Check
> 
>    The client receives the SETUP response and learns the candidate
>    address to use for the connectivity checks.
> 
> s/address/addresses/ (right?)
> 
> 
>    Aggressive nomination SHALL be used with RTSP.  This doesn't have the
>    negative impact that it has in offer/answer as media playing only
>    starts after issuing a PLAY request.
> 
> But it does have the downside of possibly selecting a lower-priority
> pair due to single dropped (e.g., due to congestion) connectivity check
> message for a higher priority pair. That said, probably SHALL is
> appropriate here, just wanted to make sure the implications are understood.

Yes, but with ICE-RTSP you will retransmit the higher priority also,
thus if that works and you get a response that alternative is nominated
and will be used. But we probably should clarify what it means when
multiple candidate pairs are nominated, and in worse case the server
will use different paths for different media streams.

> 
> 
> 5.12.  Re-SETUP
> 
>    If the client decides to change any parameters related to the media
>    stream setup
> 
> Could clarify (again) here that we talk about ICE parameters.

Yes, definitely. I actually had to think a long time if this was only
ICE parameters or also RTSP parameters. But, it is so unnecessary to
restart ICE for most RTSP parameter changes.


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 pkyzivat@alum.mit.edu  Thu Dec  6 09:04:03 2012
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Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
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Harald - comment at end.

On 12/6/12 8:02 AM, Harald Alvestrand wrote:
> On 12/03/2012 09:13 PM, Paul Kyzivat wrote:
>> On 12/3/12 6:13 AM, Harald Alvestrand wrote:
>>> On 12/03/2012 10:02 AM, Markus.Isomaki@nokia.com wrote:
>>>> Hi,
>>>>
>>>> I agree that it would be useful to be able to run data channel on the
>>>> same 5-tuple as RTP/RTCP, and even making it the default and
>>>> must-implement way for the browsers.
>>>>
>>>> QoS/prioritization comes to mind as the only potential drawback. The
>>>> endpoint can naturally still give different priorities to the
>>>> different components (audio, video, "data") it is sending. I assume
>>>> SCTP also has TCP-style receiver window that can be used to throttle
>>>> reception if needed. And presumably RMCAT will find solutions to
>>>> congestion control in genenral. But from the network perspective there
>>>> is only one flow, and the only(?) way to differentiate the components
>>>> is to use DSCP. I don't think our QoS (or API) draft yet clearly
>>>> explains how DSCP's are learned and set, or whether that is even going
>>>> to be a realistic solution.
>>>
>>> That is indeed the only reason I can think of to specify multiple
>>> m-lines for DTLS/SCTP (and we don't need to bundle them together).
>>>
>>> At the moment, the MMUSIC API doesn't define a way to control which
>>> m-line a DTLS assocation gets mapped onto.
>>
>> MMUSIC *API*???
>
> Sorry, I was writing at a non-awake time.
> I meant the WebRTC API (Javascript, specified in the W3C WG).

OK. That makes more sense.

Well, there are two solutions to that:

- enhance the API to provide that capability, perhaps with a default
   to make it simple in common cases.

- limit WebRTC to follow a convention for which only supports a
   subset of what is possible in SDP.

IMO the first is the wiser choice.

In any case we are not defining this in SDP solely for WebRTC. We need 
to specify this in a way that works for everybody that wants to use it.

One way or another the SDP must define the SCTP association, and if it 
is to share a 5-tuple then the grouping to do that needs to be defined.
IIUC, the current bundle draft isn't capable of doing that.

	Thanks,
	Paul


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] SCTP question: Where does it multiplex?
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Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
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Hi,

>One way or another the SDP must define the SCTP association, and if it
>is to share a 5-tuple then the grouping to do that needs to be defined.
>IIUC, the current bundle draft isn't capable of doing that.

The current bundle draft needs additional work, mainly regarding the handli=
ng of SDP attributes etc, even to share a 5-tuple for RTP.

But, in general, what problems do you see with using bundle also for SCTP?

Regards,

Christer

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Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
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On 12/6/12 1:01 PM, Christer Holmberg wrote:
> Hi,
>
>> One way or another the SDP must define the SCTP association, and if it
>> is to share a 5-tuple then the grouping to do that needs to be defined.
>> IIUC, the current bundle draft isn't capable of doing that.
>
> The current bundle draft needs additional work, mainly regarding the handling of SDP attributes etc, even to share a 5-tuple for RTP.

The current draft requires the <proto> fields of all the bundled m-lines 
to be the same. It also strongly implies that all of them must be 
RTP-related. To include the SCTP m-line for support of the data 
channels, it will be necessary dissimilar <proto> fields to be bundled. 
But not all combinations are valid.

So it needs some additional text to explain which other cases may be 
bundled. The simple way out would be to simply enumerate the different 
combinations of <proto>s that may be bundled and any other constraints 
that may apply. Better would be to define some more general criteria for 
those that may be bundled, so that it might survive future extensions.

An obvious constraint is that there must be some way of demuxing them.

I am still trying to understand the details of how the DTLS/SRTP is 
multiplexed with RTP (actually I guess it will be UDP/TLS/RTP/SAVP(F)). 
Is there some place that explains that? RFC 5764 explains how RTP, DTLS, 
and STUN are demuxed for DTLS/SRTP. And 
draft-tuexen-tsvwg-sctp-dtls-encaps explains how to run SCTP over DTLS. 
But those two mappings don't seem to be compatible.

Has anyone written down a proposal for how that will work?

	Thanks,
	Paul

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From: "Rosen, Brian" <Brian.Rosen@neustar.biz>
To: "Klatsky, Carl" <Carl_Klatsky@cable.comcast.com>
Date: Thu, 6 Dec 2012 20:50:47 -0500
Thread-Topic: [MMUSIC] Comments on draft-ietf-mmusic-media-loopback-24
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Subject: Re: [MMUSIC] Comments on draft-ietf-mmusic-media-loopback-24
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--_000_6C6819F46CBD4B40B6C765918F179D52neustarbiz_
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No one replied to you.

I believe there is no restriction, any codec could be used in theory.  We j=
ust had a conversation about if any media streams were looped, did all of t=
hem have to be, and the consensus on that was no.  Not sure any device woul=
d choose to allow loopback of telephone-events, but it could I suppose.

I don't think it could reasonably be extended to hack caller id in the way =
you describe.

Brian


On Nov 12, 2012, at 3:46 PM, "Klatsky, Carl" <Carl_Klatsky@cable.comcast.co=
m<mailto:Carl_Klatsky@cable.comcast.com>> wrote:

Hi,

A colleague of mine forwarded me this draft and I had two comments if someo=
ne on the distro has input:

1)      When setting up the media loopback, are there any restrictions on t=
ype of codec used?  Could even telephone-events codec be specified?
2)      Although this draft is centered around media, can it be extended to=
 for use as means to test the caller ID set connected to an analog telephon=
y interface?  The approach outlined in this draft seems to imply no interac=
tion with the end subscriber is needed.  If the network operator wanted to =
silently send a test pattern to the caller ID set, perhaps this same loopba=
ck mechanism can be used with either an additional SDP attribute or some ch=
ange in the SIP header itself.  A related application would be to advance t=
he timestamp on the caller ID set following Daylight Savings Time transitio=
n.  Typically that is not updated until the next inbound call following the=
 DST change.  A loopback call could signal to the device to update the call=
er ID set but without ringing / alerting the end subscriber without the nee=
d to wait for an inbound call.


Regards,

Carl Klatsky
Comcast
_______________________________________________
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--_000_6C6819F46CBD4B40B6C765918F179D52neustarbiz_
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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html charset=
=3Dus-ascii"><base href=3D"x-msg://1632/"></head><body style=3D"word-wrap: =
break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space=
; ">No one replied to you.<div><br></div><div>I believe there is no restric=
tion, any codec could be used in theory. &nbsp;We just had a conversation a=
bout if any media streams were looped, did all of them have to be, and the =
consensus on that was no. &nbsp;Not sure any device would choose to allow l=
oopback of telephone-events, but it could I suppose.</div><div><br></div><d=
iv>I don't think it could reasonably be extended to hack caller id in the w=
ay you describe. &nbsp;</div><div><br></div><div>Brian</div><div><br></div>=
<div><br><div><div>On Nov 12, 2012, at 3:46 PM, "Klatsky, Carl" &lt;<a href=
=3D"mailto:Carl_Klatsky@cable.comcast.com">Carl_Klatsky@cable.comcast.com</=
a>&gt; wrote:</div><br class=3D"Apple-interchange-newline"><blockquote type=
=3D"cite"><div lang=3D"EN-US" link=3D"blue" vlink=3D"purple" style=3D"font-=
family: Helvetica; font-size: medium; font-style: normal; font-variant: nor=
mal; font-weight: normal; letter-spacing: normal; line-height: normal; orph=
ans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: none; w=
hite-space: normal; widows: 2; word-spacing: 0px; -webkit-text-size-adjust:=
 auto; -webkit-text-stroke-width: 0px; "><div class=3D"Section1" style=3D"p=
age: Section1; "><div style=3D"margin: 0in 0in 0.0001pt; font-size: 11pt; f=
ont-family: Calibri, sans-serif; ">Hi,<o:p></o:p></div><div style=3D"margin=
: 0in 0in 0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif; "><o=
:p>&nbsp;</o:p></div><div style=3D"margin: 0in 0in 0.0001pt; font-size: 11p=
t; font-family: Calibri, sans-serif; ">A colleague of mine forwarded me thi=
s draft and I had two comments if someone on the distro has input:<o:p></o:=
p></div><div style=3D"margin: 0in 0in 0.0001pt; font-size: 11pt; font-famil=
y: Calibri, sans-serif; "><o:p>&nbsp;</o:p></div><div style=3D"margin: 0in =
0in 0.0001pt 0.5in; font-size: 11pt; font-family: Calibri, sans-serif; text=
-indent: -0.25in; "><span>1)<span style=3D"font-style: normal; font-variant=
: normal; font-weight: normal; font-size: 7pt; line-height: normal; font-fa=
mily: 'Times New Roman'; ">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<span class=3D"App=
le-converted-space">&nbsp;</span></span></span>When setting up the media lo=
opback, are there any restrictions on type of codec used?&nbsp; Could even =
telephone-events codec be specified?<o:p></o:p></div><div style=3D"margin: =
0in 0in 0.0001pt 0.5in; font-size: 11pt; font-family: Calibri, sans-serif; =
text-indent: -0.25in; "><span>2)<span style=3D"font-style: normal; font-var=
iant: normal; font-weight: normal; font-size: 7pt; line-height: normal; fon=
t-family: 'Times New Roman'; ">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<span class=3D=
"Apple-converted-space">&nbsp;</span></span></span>Although this draft is c=
entered around media, can it be extended to for use as means to test the ca=
ller ID set connected to an analog telephony interface?&nbsp; The approach =
outlined in this draft seems to imply no interaction with the end subscribe=
r is needed.&nbsp; If the network operator wanted to silently send a test p=
attern to the caller ID set, perhaps this same loopback mechanism can be us=
ed with either an additional SDP attribute or some change in the SIP header=
 itself.&nbsp; A related application would be to advance the timestamp on t=
he caller ID set following Daylight Savings Time transition.&nbsp; Typicall=
y that is not updated until the next inbound call following the DST change.=
&nbsp; A loopback call could signal to the device to update the caller ID s=
et but without ringing / alerting the end subscriber without the need to wa=
it for an inbound call.<o:p></o:p></div><div style=3D"margin: 0in 0in 0.000=
1pt; font-size: 11pt; font-family: Calibri, sans-serif; "><o:p>&nbsp;</o:p>=
</div><div style=3D"margin: 0in 0in 0.0001pt; font-size: 11pt; font-family:=
 Calibri, sans-serif; "><o:p>&nbsp;</o:p></div><div style=3D"margin: 0in 0i=
n 0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif; "><span styl=
e=3D"font-size: 10pt; font-family: Arial, sans-serif; ">Regards,</span><o:p=
></o:p></div><div style=3D"margin: 0in 0in 0.0001pt; font-size: 11pt; font-=
family: Calibri, sans-serif; ">&nbsp;<o:p></o:p></div><div style=3D"margin:=
 0in 0in 0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif; "><sp=
an style=3D"font-size: 10pt; font-family: Arial, sans-serif; ">Carl Klatsky=
<o:p></o:p></span></div><div style=3D"margin: 0in 0in 0.0001pt; font-size: =
11pt; font-family: Calibri, sans-serif; "><span style=3D"font-size: 10pt; f=
ont-family: Arial, sans-serif; ">Comcast<o:p></o:p></span></div></div>_____=
__________________________________________<br>mmusic mailing list<br><a hre=
f=3D"mailto:mmusic@ietf.org" style=3D"color: purple; text-decoration: under=
line; ">mmusic@ietf.org</a><br><a href=3D"https://www.ietf.org/mailman/list=
info/mmusic" style=3D"color: purple; text-decoration: underline; ">https://=
www.ietf.org/mailman/listinfo/mmusic</a><br></div></blockquote></div><br></=
div></body></html>=

--_000_6C6819F46CBD4B40B6C765918F179D52neustarbiz_--

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Subject: [MMUSIC] Draft Minutes of MMUSIC WG meeting at IETF 85, Atlanta
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Hi

The draft minutes of the MMUSIC WG meeting at IETF 85 (Atlanta) have 
been posted at:

     http://www.ietf.org/proceedings/85/minutes/minutes-85-mmusic

Please review the minutes and let us know of any comments or corrections.

Thanks

-- Miguel, Paul & Flemming

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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: mmusic WG <mmusic@ietf.org>
Thread-Topic: SDP work needed for WebRTC stuff
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Subject: [MMUSIC] SDP work needed for WebRTC stuff
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I was looking over everything that needs to be completed to finish a fist c=
ut of the WebRTC related work. There are a handful of big SDP problems that=
 are currently blocking some of the WebRTC work and I'd like to figure out =
how to make some progress on them.=20

Let me loosely characterize them as=20

1) If we have several video streams, how do theses map up to 1 or more m li=
nes.=20

2) if we are doing port multiplexing, what does the SDP look like (the bund=
le problem)=20

3) How do we map the RTCWeb track and stream label concepts to identifiers =
in SDP

3) SDP for application running over SCTP/DTLS=20

I don't want to speak for all the various chairs but I am under the impress=
ion that most of chairs of related groups in W3C and IETF believe these are=
 issues that need to be resolved primarily in the MMUSIC WG and that they i=
mpact both WebRTC and CLUE as well as the general long term use of SDP in S=
IP and other protocols.=20

I'd like to get some discussion going on how we can make some progress on t=
hese. I don't think we are going to solve these in 20 minutes of discussion=
 at an IETF meeting so I think we probably need some interim (virtual or fa=
ce to face) to sort this out.=20

Thoughts?

Cullen <in my individual contributor role>=20






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From: Eric Rescorla <ekr@rtfm.com>
Date: Sat, 8 Dec 2012 15:21:06 -0800
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Cc: rtcweb@ietf.org, mmusic WG <mmusic@ietf.org>, public-webrtc@w3.org
Subject: Re: [MMUSIC] SDP work needed for WebRTC stuff
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--14dae939996984aaee04d05f98b6
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+rtcweb

On Sat, Dec 8, 2012 at 2:41 PM, Cullen Jennings (fluffy)
<fluffy@cisco.com>wrote:

>
> I was looking over everything that needs to be completed to finish a fist
> cut of the WebRTC related work. There are a handful of big SDP problems
> that are currently blocking some of the WebRTC work and I'd like to figure
> out how to make some progress on them.
>
> Let me loosely characterize them as
>
> 1) If we have several video streams, how do theses map up to 1 or more m
> lines.
>
> 2) if we are doing port multiplexing, what does the SDP look like (the
> bundle problem)
>
> 3) How do we map the RTCWeb track and stream label concepts to identifiers
> in SDP
>
> 3) SDP for application running over SCTP/DTLS
>

I don't want to speak for all the various chairs but I am under the
> impression that most of chairs of related groups in W3C and IETF believe
> these are issues that need to be resolved primarily in the MMUSIC WG and
> that they impact both WebRTC and CLUE as well as the general long term use
> of SDP in SIP and other protocols.
>
> I'd like to get some discussion going on how we can make some progress on
> these. I don't think we are going to solve these in 20 minutes of
> discussion at an IETF meeting so I think we probably need some interim
> (virtual or face to face) to sort this out.
>
> Thoughts?
>

Wow, I'm totally confused here.

I had assumed that the SDP-related issues were going to be the main
topics at the WebRTC/RTCWEB interim in January. Is that not the case?

IMO the lack of clarity around how to encode various media
configurations into SDP is the major thing blocking progress here. In
particular, Firefox has opted not to implement multiplexing of media
streams over the same transport flow (whether of the bundle or
multiple m-line variety) until the SDP for it is well-defined. The
same thing applies to the question of how to map multiple m-lines to
incoming MediaStreams/Tracks.

We really need to cover these issues in the interim.

-Ekr

--14dae939996984aaee04d05f98b6
Content-Type: text/html; charset=ISO-8859-1
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+rtcweb<div><br></div><div>On Sat, Dec 8, 2012 at 2:41 PM, 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_q=
uote"><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-l=
eft:1px #ccc solid;padding-left:1ex">

<br>
I was looking over everything that needs to be completed to finish a fist c=
ut of the WebRTC related work. There are a handful of big SDP problems that=
 are currently blocking some of the WebRTC work and I&#39;d like to figure =
out how to make some progress on them.<br>


<br>
Let me loosely characterize them as<br>
<br>
1) If we have several video streams, how do theses map up to 1 or more m li=
nes.<br>
<br>
2) if we are doing port multiplexing, what does the SDP look like (the bund=
le problem)<br>
<br>
3) How do we map the RTCWeb track and stream label concepts to identifiers =
in SDP<br>
<br>
3) SDP for application running over SCTP/DTLS<br></blockquote><div><br></di=
v><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:=
1px #ccc solid;padding-left:1ex">
I don&#39;t want to speak for all the various chairs but I am under the imp=
ression that most of chairs of related groups in W3C and IETF believe these=
 are issues that need to be resolved primarily in the MMUSIC WG and that th=
ey impact both WebRTC and CLUE as well as the general long term use of SDP =
in SIP and other protocols.<br>


<br>
I&#39;d like to get some discussion going on how we can make some progress =
on these. I don&#39;t think we are going to solve these in 20 minutes of di=
scussion at an IETF meeting so I think we probably need some interim (virtu=
al or face to face) to sort this out.<br>


<br>
Thoughts?<br></blockquote><div><br></div><div class=3D"gmail_quote">Wow, I&=
#39;m totally confused here.</div><div class=3D"gmail_quote"><br></div><div=
 class=3D"gmail_quote">I had assumed that the SDP-related issues were going=
 to be the main</div>

<div class=3D"gmail_quote">topics at the WebRTC/RTCWEB interim in January. =
Is that not the case?</div><div class=3D"gmail_quote"><br></div><div class=
=3D"gmail_quote">IMO the lack of clarity around how to encode various media=
</div>

<div class=3D"gmail_quote">configurations into SDP is the major thing block=
ing progress here. In</div><div class=3D"gmail_quote">particular, Firefox h=
as opted not to implement multiplexing of media</div><div class=3D"gmail_qu=
ote">

streams over the same transport flow (whether of the bundle or</div><div cl=
ass=3D"gmail_quote">multiple m-line variety) until the SDP for it is well-d=
efined. The</div><div class=3D"gmail_quote">same thing applies to the quest=
ion of how to map multiple m-lines to</div>

<div class=3D"gmail_quote">incoming MediaStreams/Tracks.</div><div class=3D=
"gmail_quote"><br></div><div class=3D"gmail_quote">We really need to cover =
these issues in the interim.</div><div class=3D"gmail_quote"><br></div><div=
 class=3D"gmail_quote">

-Ekr</div><div><br></div><div><br></div></div></div>

--14dae939996984aaee04d05f98b6--

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To: Eric Rescorla <ekr@rtfm.com>
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Cc: "Cullen Jennings \(fluffy\)" <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, mmusic WG <mmusic@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Subject: Re: [MMUSIC] [rtcweb]  SDP work needed for WebRTC stuff
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On 8 December 2012 15:21, Eric Rescorla <ekr@rtfm.com> wrote:
> We really need to cover these issues in the interim.

+1.  I would have said *resolve* rather than cover, but I'm learning
not to be too optimistic in these matters.  Still, an outcome of one
form or other would be nice.

From fluffy@cisco.com  Sat Dec  8 16:19:49 2012
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Eric Rescorla <ekr@rtfm.com>
Thread-Topic: [MMUSIC] SDP work needed for WebRTC stuff
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Subject: Re: [MMUSIC] SDP work needed for WebRTC stuff
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On Dec 8, 2012, at 4:21 PM, Eric Rescorla <ekr@rtfm.com> wrote:

> +rtcweb
>=20
> On Sat, Dec 8, 2012 at 2:41 PM, Cullen Jennings (fluffy) <fluffy@cisco.co=
m> wrote:
>=20
> I was looking over everything that needs to be completed to finish a fist=
 cut of the WebRTC related work. There are a handful of big SDP problems th=
at are currently blocking some of the WebRTC work and I'd like to figure ou=
t how to make some progress on them.
>=20
> Let me loosely characterize them as
>=20
> 1) If we have several video streams, how do theses map up to 1 or more m =
lines.
>=20
> 2) if we are doing port multiplexing, what does the SDP look like (the bu=
ndle problem)
>=20
> 3) How do we map the RTCWeb track and stream label concepts to identifier=
s in SDP
>=20
> 3) SDP for application running over SCTP/DTLS
>=20
> I don't want to speak for all the various chairs but I am under the impre=
ssion that most of chairs of related groups in W3C and IETF believe these a=
re issues that need to be resolved primarily in the MMUSIC WG and that they=
 impact both WebRTC and CLUE as well as the general long term use of SDP in=
 SIP and other protocols.
>=20
> I'd like to get some discussion going on how we can make some progress on=
 these. I don't think we are going to solve these in 20 minutes of discussi=
on at an IETF meeting so I think we probably need some interim (virtual or =
face to face) to sort this out.
>=20
> Thoughts?
>=20
> Wow, I'm totally confused here.
>=20
> I had assumed that the SDP-related issues were going to be the main
> topics at the WebRTC/RTCWEB interim in January. Is that not the case?

So far the interim was only talking about being a WebRTC & RTCWeb so this S=
DP stuff would be out of scope. Perhaps it would be better to have some of =
the time for mmusic topics?=20


>=20
> IMO the lack of clarity around how to encode various media
> configurations into SDP is the major thing blocking progress here. In
> particular, Firefox has opted not to implement multiplexing of media
> streams over the same transport flow (whether of the bundle or
> multiple m-line variety) until the SDP for it is well-defined. The
> same thing applies to the question of how to map multiple m-lines to
> incoming MediaStreams/Tracks.
>=20
> We really need to cover these issues in the interim.
>=20
> -Ekr
>=20
>=20


From ron.even.tlv@gmail.com  Sat Dec  8 23:10:46 2012
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References: <C5E08FE080ACFD4DAE31E4BDBF944EB1132A380B@xmb-aln-x02.cisco.com> <CABcZeBPfhKoDvHceNLo_Ab9rRDD0V-mMok4H4y_FitYcBZ7vTA@mail.gmail.com> <C5E08FE080ACFD4DAE31E4BDBF944EB1132A5D53@xmb-aln-x02.cisco.com>
Date: Sun, 9 Dec 2012 09:10:45 +0200
Message-ID: <CAHy0fzDFeNWUwbv0XYhicMVJ7uD0b8ceQmEbhbmNDWk7D92KwA@mail.gmail.com>
From: Ron Even <ron.even.tlv@gmail.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>, mmusic WG <mmusic@ietf.org>, "<public-webrtc@w3.org>" <public-webrtc@w3.org>
Subject: Re: [MMUSIC] SDP work needed for WebRTC stuff
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--14dae93b62a2be3d2d04d0662536
Content-Type: text/plain; charset=ISO-8859-1

Hi Cullen,
I agree that we need to put more effort into this work and am willing to be
part of it.
You mentioned CLUE and we have the mapping of RTP streams to CLUE media
captures as a topic, discussed in CLUE.
 We started some work during the Atlanta meeting and it is supposed to be
discussed on the RAI mailing list since it involves multiple RAI WGs but
this is just the formality
Roni Even

On Sunday, December 9, 2012, Cullen Jennings (fluffy) wrote:

>
> On Dec 8, 2012, at 4:21 PM, Eric Rescorla <ekr@rtfm.com <javascript:;>>
> wrote:
>
> > +rtcweb
> >
> > On Sat, Dec 8, 2012 at 2:41 PM, Cullen Jennings (fluffy) <
> fluffy@cisco.com <javascript:;>> wrote:
> >
> > I was looking over everything that needs to be completed to finish a
> fist cut of the WebRTC related work. There are a handful of big SDP
> problems that are currently blocking some of the WebRTC work and I'd like
> to figure out how to make some progress on them.
> >
> > Let me loosely characterize them as
> >
> > 1) If we have several video streams, how do theses map up to 1 or more m
> lines.
> >
> > 2) if we are doing port multiplexing, what does the SDP look like (the
> bundle problem)
> >
> > 3) How do we map the RTCWeb track and stream label concepts to
> identifiers in SDP
> >
> > 3) SDP for application running over SCTP/DTLS
> >
> > I don't want to speak for all the various chairs but I am under the
> impression that most of chairs of related groups in W3C and IETF believe
> these are issues that need to be resolved primarily in the MMUSIC WG and
> that they impact both WebRTC and CLUE as well as the general long term use
> of SDP in SIP and other protocols.
> >
> > I'd like to get some discussion going on how we can make some progress
> on these. I don't think we are going to solve these in 20 minutes of
> discussion at an IETF meeting so I think we probably need some interim
> (virtual or face to face) to sort this out.
> >
> > Thoughts?
> >
> > Wow, I'm totally confused here.
> >
> > I had assumed that the SDP-related issues were going to be the main
> > topics at the WebRTC/RTCWEB interim in January. Is that not the case?
>
> So far the interim was only talking about being a WebRTC & RTCWeb so this
> SDP stuff would be out of scope. Perhaps it would be better to have some of
> the time for mmusic topics?
>
>
> >
> > IMO the lack of clarity around how to encode various media
> > configurations into SDP is the major thing blocking progress here. In
> > particular, Firefox has opted not to implement multiplexing of media
> > streams over the same transport flow (whether of the bundle or
> > multiple m-line variety) until the SDP for it is well-defined. The
> > same thing applies to the question of how to map multiple m-lines to
> > incoming MediaStreams/Tracks.
> >
> > We really need to cover these issues in the interim.
> >
> > -Ekr
> >
> >
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org <javascript:;>
> https://www.ietf.org/mailman/listinfo/mmusic
>

--14dae93b62a2be3d2d04d0662536
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Hi Cullen,<div>I agree that we need to put more effort into this work and a=
m willing to be part of it.</div><div>You mentioned CLUE and we have the ma=
pping of RTP streams to CLUE media captures as a topic, discussed in CLUE.<=
span></span></div>
<div>=A0We started some work during the Atlanta meeting and it is supposed =
to be discussed on the RAI mailing list since it involves multiple RAI WGs =
but this is just the formality</div><div>Roni Even<br><br>On Sunday, Decemb=
er 9, 2012, Cullen Jennings (fluffy)  wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><br>
On Dec 8, 2012, at 4:21 PM, Eric Rescorla &lt;<a href=3D"javascript:;" oncl=
ick=3D"_e(event, &#39;cvml&#39;, &#39;ekr@rtfm.com&#39;)">ekr@rtfm.com</a>&=
gt; wrote:<br>
<br>
&gt; +rtcweb<br>
&gt;<br>
&gt; On Sat, Dec 8, 2012 at 2:41 PM, Cullen Jennings (fluffy) &lt;<a href=
=3D"javascript:;" onclick=3D"_e(event, &#39;cvml&#39;, &#39;fluffy@cisco.co=
m&#39;)">fluffy@cisco.com</a>&gt; wrote:<br>
&gt;<br>
&gt; I was looking over everything that needs to be completed to finish a f=
ist cut of the WebRTC related work. There are a handful of big SDP problems=
 that are currently blocking some of the WebRTC work and I&#39;d like to fi=
gure out how to make some progress on them.<br>

&gt;<br>
&gt; Let me loosely characterize them as<br>
&gt;<br>
&gt; 1) If we have several video streams, how do theses map up to 1 or more=
 m lines.<br>
&gt;<br>
&gt; 2) if we are doing port multiplexing, what does the SDP look like (the=
 bundle problem)<br>
&gt;<br>
&gt; 3) How do we map the RTCWeb track and stream label concepts to identif=
iers in SDP<br>
&gt;<br>
&gt; 3) SDP for application running over SCTP/DTLS<br>
&gt;<br>
&gt; I don&#39;t want to speak for all the various chairs but I am under th=
e impression that most of chairs of related groups in W3C and IETF believe =
these are issues that need to be resolved primarily in the MMUSIC WG and th=
at they impact both WebRTC and CLUE as well as the general long term use of=
 SDP in SIP and other protocols.<br>

&gt;<br>
&gt; I&#39;d like to get some discussion going on how we can make some prog=
ress on these. I don&#39;t think we are going to solve these in 20 minutes =
of discussion at an IETF meeting so I think we probably need some interim (=
virtual or face to face) to sort this out.<br>

&gt;<br>
&gt; Thoughts?<br>
&gt;<br>
&gt; Wow, I&#39;m totally confused here.<br>
&gt;<br>
&gt; I had assumed that the SDP-related issues were going to be the main<br=
>
&gt; topics at the WebRTC/RTCWEB interim in January. Is that not the case?<=
br>
<br>
So far the interim was only talking about being a WebRTC &amp; RTCWeb so th=
is SDP stuff would be out of scope. Perhaps it would be better to have some=
 of the time for mmusic topics?<br>
<br>
<br>
&gt;<br>
&gt; IMO the lack of clarity around how to encode various media<br>
&gt; configurations into SDP is the major thing blocking progress here. In<=
br>
&gt; particular, Firefox has opted not to implement multiplexing of media<b=
r>
&gt; streams over the same transport flow (whether of the bundle or<br>
&gt; multiple m-line variety) until the SDP for it is well-defined. The<br>
&gt; same thing applies to the question of how to map multiple m-lines to<b=
r>
&gt; incoming MediaStreams/Tracks.<br>
&gt;<br>
&gt; We really need to cover these issues in the interim.<br>
&gt;<br>
&gt; -Ekr<br>
&gt;<br>
&gt;<br>
<br>
_______________________________________________<br>
mmusic mailing list<br>
<a href=3D"javascript:;" onclick=3D"_e(event, &#39;cvml&#39;, &#39;mmusic@i=
etf.org&#39;)">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>

--14dae93b62a2be3d2d04d0662536--

From harald@alvestrand.no  Sun Dec  9 23:57:47 2012
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Date: Mon, 10 Dec 2012 08:57:42 +0100
From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [MMUSIC] SDP work needed for WebRTC stuff
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I agree that these are the important topics that need to be settled, and 
agree that it's urgent to settle them.
I'm willing to spend time going to interims (virtual or physical, 
although I have a preference for virtual) if there's any sign of them 
being useful to progress.

I would like to see us move on these issues before the RTCWEB interim in 
early February, if possible.

On 12/08/2012 11:41 PM, Cullen Jennings (fluffy) wrote:
> I was looking over everything that needs to be completed to finish a fist cut of the WebRTC related work. There are a handful of big SDP problems that are currently blocking some of the WebRTC work and I'd like to figure out how to make some progress on them.
>
> Let me loosely characterize them as
>
> 1) If we have several video streams, how do theses map up to 1 or more m lines.
>
> 2) if we are doing port multiplexing, what does the SDP look like (the bundle problem)
>
> 3) How do we map the RTCWeb track and stream label concepts to identifiers in SDP
>
> 3) SDP for application running over SCTP/DTLS
>
> I don't want to speak for all the various chairs but I am under the impression that most of chairs of related groups in W3C and IETF believe these are issues that need to be resolved primarily in the MMUSIC WG and that they impact both WebRTC and CLUE as well as the general long term use of SDP in SIP and other protocols.
>
> I'd like to get some discussion going on how we can make some progress on these. I don't think we are going to solve these in 20 minutes of discussion at an IETF meeting so I think we probably need some interim (virtual or face to face) to sort this out.
>
> Thoughts?
>
> Cullen <in my individual contributor role>
>
>
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From bernard_aboba@hotmail.com  Mon Dec 10 00:31:51 2012
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Subject: Re: [MMUSIC] SDP work needed for WebRTC stuff
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I agree that accelerated progress in MMUSIC should be a goal. If there are e=
nough cycles to spare, monthly virtual interims might do the trick.  This is=
 a painful way to go about it, but it has worked before (MARTINI) and concen=
trated pain now is preferable to years of chronic aches.

On Dec 9, 2012, at 23:57, "Harald Alvestrand" <harald@alvestrand.no> wrote:
> I agree that these are the important topics that need to be settled, and a=
gree that it's urgent to settle them.
> I'm willing to spend time going to interims (virtual or physical, although=
 I have a preference for is virtual) if there's any sign of them being usefu=
l to progress.
>=20
> I would like to see us move on these issues before the RTCWEB interim in e=
arly February, if possible.
>=20
> On 12/08/2012 11:41 PM, Cullen Jennings (fluffy) wrote:
>> I was looking over everything that needs to be completed to finish a fist=
 cut of the WebRTC related work. There are a handful of big SDP problems tha=
t are currently blocking some of the WebRTC work and I'd like to figure out h=
ow to make some progress on them.
>>=20
>> Let me loosely characterize them as
>>=20
>> 1) If we have several video streams, how do theses map up to 1 or more m l=
ines.
>>=20
>> 2) if we are doing port multiplexing, what does the SDP look like (the bu=
ndle problem)
>>=20
>> 3) How do we map the RTCWeb track and stream label concepts to identifier=
s in SDP
>>=20
>> 3) SDP for application running over SCTP/DTLS
>>=20
>> I don't want to speak for all the various chairs but I am under the impre=
ssion that most of chairs of related groups in W3C and IETF believe these ar=
e issues that need to be resolved primarily in the MMUSIC WG and that they i=
mpact both WebRTC and CLUE as well as the general long term use of SDP in SI=
P and other protocols.
>>=20
>> I'd like to get some discussion going on how we can make some progress on=
 these. I don't think we are going to solve these in 20 minutes of discussio=
n at an IETF meeting so I think we probably need some interim (virtual or fa=
ce to face) to sort this out.
>>=20
>> Thoughts?
>>=20
>> Cullen <in my individual contributor role>
>>=20
>>=20
>>=20
>>=20
>>=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

From keith.drage@alcatel-lucent.com  Mon Dec 10 05:16:42 2012
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>, Eric Rescorla <ekr@rtfm.com>
Date: Mon, 10 Dec 2012 14:16:27 +0100
Thread-Topic: [MMUSIC] SDP work needed for WebRTC stuff
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>, mmusic WG <mmusic@ietf.org>, "<public-webrtc@w3.org>" <public-webrtc@w3.org>
Subject: Re: [MMUSIC] SDP work needed for WebRTC stuff
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This is not just MMUSIC, there are bits of AVTCORE and AVTEXT responsibilit=
y in there as well.

And I don't see how you can suddenly extend the interim of RTCWEB to the sc=
ope of other working groups, without even having a discussion with all the =
relevant chairs.

I'd further note that Jonathan Lennox has an action (with the support of qu=
ite a number of other people) to produce an internet draft on trying to sor=
t out the terminology concents that exist within RTP at the moment. This wo=
uld be the basis for progressing a number of these issues. That would proba=
bly have to be an RAI wide draft.

There has been some progress on this - maybe Jonathan can report on where t=
his now is. I know he was waiting on volunteers for some of the sections an=
d on input from others where volunteers already existed.

Keith

> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf
> Of Cullen Jennings (fluffy)
> Sent: 09 December 2012 00:20
> To: Eric Rescorla
> Cc: <rtcweb@ietf.org>; mmusic WG; <public-webrtc@w3.org>
> Subject: Re: [MMUSIC] SDP work needed for WebRTC stuff
>=20
>=20
> On Dec 8, 2012, at 4:21 PM, Eric Rescorla <ekr@rtfm.com> wrote:
>=20
> > +rtcweb
> >
> > On Sat, Dec 8, 2012 at 2:41 PM, Cullen Jennings (fluffy)
> <fluffy@cisco.com> wrote:
> >
> > I was looking over everything that needs to be completed to finish a
> fist cut of the WebRTC related work. There are a handful of big SDP
> problems that are currently blocking some of the WebRTC work and I'd like
> to figure out how to make some progress on them.
> >
> > Let me loosely characterize them as
> >
> > 1) If we have several video streams, how do theses map up to 1 or more =
m
> lines.
> >
> > 2) if we are doing port multiplexing, what does the SDP look like (the
> bundle problem)
> >
> > 3) How do we map the RTCWeb track and stream label concepts to
> identifiers in SDP
> >
> > 3) SDP for application running over SCTP/DTLS
> >
> > I don't want to speak for all the various chairs but I am under the
> impression that most of chairs of related groups in W3C and IETF believe
> these are issues that need to be resolved primarily in the MMUSIC WG and
> that they impact both WebRTC and CLUE as well as the general long term us=
e
> of SDP in SIP and other protocols.
> >
> > I'd like to get some discussion going on how we can make some progress
> on these. I don't think we are going to solve these in 20 minutes of
> discussion at an IETF meeting so I think we probably need some interim
> (virtual or face to face) to sort this out.
> >
> > Thoughts?
> >
> > Wow, I'm totally confused here.
> >
> > I had assumed that the SDP-related issues were going to be the main
> > topics at the WebRTC/RTCWEB interim in January. Is that not the case?
>=20
> So far the interim was only talking about being a WebRTC & RTCWeb so this
> SDP stuff would be out of scope. Perhaps it would be better to have some
> of the time for mmusic topics?
>=20
>=20
> >
> > IMO the lack of clarity around how to encode various media
> > configurations into SDP is the major thing blocking progress here. In
> > particular, Firefox has opted not to implement multiplexing of media
> > streams over the same transport flow (whether of the bundle or
> > multiple m-line variety) until the SDP for it is well-defined. The
> > same thing applies to the question of how to map multiple m-lines to
> > incoming MediaStreams/Tracks.
> >
> > We really need to cover these issues in the interim.
> >
> > -Ekr
> >
> >
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

From ted.ietf@gmail.com  Mon Dec 10 09:36:29 2012
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Date: Mon, 10 Dec 2012 09:36:27 -0800
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From: Ted Hardie <ted.ietf@gmail.com>
To: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
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Cc: "Cullen Jennings \(fluffy\)" <fluffy@cisco.com>, "<rtcweb@ietf.org>" <rtcweb@ietf.org>, mmusic WG <mmusic@ietf.org>, "<public-webrtc@w3.org>" <public-webrtc@w3.org>
Subject: Re: [MMUSIC] SDP work needed for WebRTC stuff
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--20cf30780ca44d069a04d08301c7
Content-Type: text/plain; charset=ISO-8859-1

On Mon, Dec 10, 2012 at 5:16 AM, DRAGE, Keith (Keith) <
keith.drage@alcatel-lucent.com> wrote:

> This is not just MMUSIC, there are bits of AVTCORE and AVTEXT
> responsibility in there as well.
>
> And I don't see how you can suddenly extend the interim of RTCWEB to the
> scope of other working groups, without even having a discussion with all
> the relevant chairs.
>
>
That's not how I read the overall sentiment--people are simply saying that
they need this work done and are willing to put in the effort to get it
done.  It would, in fact, be ideal if some of the MMUSIC questions were
resolve *before* an RTCWEB/WEBRTC interim, as we can then work through what
is decided.

Put another way, I take it as encouragement to MMUSIC folks that they would
find a January virtual interim very well received/attended.

Ted



> I'd further note that Jonathan Lennox has an action (with the support of
> quite a number of other people) to produce an internet draft on trying to
> sort out the terminology concents that exist within RTP at the moment. This
> would be the basis for progressing a number of these issues. That would
> probably have to be an RAI wide draft.
>
> There has been some progress on this - maybe Jonathan can report on where
> this now is. I know he was waiting on volunteers for some of the sections
> and on input from others where volunteers already existed.
>
> Keith
>
> > -----Original Message-----
> > From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf
> > Of Cullen Jennings (fluffy)
> > Sent: 09 December 2012 00:20
> > To: Eric Rescorla
> > Cc: <rtcweb@ietf.org>; mmusic WG; <public-webrtc@w3.org>
> > Subject: Re: [MMUSIC] SDP work needed for WebRTC stuff
> >
> >
> > On Dec 8, 2012, at 4:21 PM, Eric Rescorla <ekr@rtfm.com> wrote:
> >
> > > +rtcweb
> > >
> > > On Sat, Dec 8, 2012 at 2:41 PM, Cullen Jennings (fluffy)
> > <fluffy@cisco.com> wrote:
> > >
> > > I was looking over everything that needs to be completed to finish a
> > fist cut of the WebRTC related work. There are a handful of big SDP
> > problems that are currently blocking some of the WebRTC work and I'd like
> > to figure out how to make some progress on them.
> > >
> > > Let me loosely characterize them as
> > >
> > > 1) If we have several video streams, how do theses map up to 1 or more
> m
> > lines.
> > >
> > > 2) if we are doing port multiplexing, what does the SDP look like (the
> > bundle problem)
> > >
> > > 3) How do we map the RTCWeb track and stream label concepts to
> > identifiers in SDP
> > >
> > > 3) SDP for application running over SCTP/DTLS
> > >
> > > I don't want to speak for all the various chairs but I am under the
> > impression that most of chairs of related groups in W3C and IETF believe
> > these are issues that need to be resolved primarily in the MMUSIC WG and
> > that they impact both WebRTC and CLUE as well as the general long term
> use
> > of SDP in SIP and other protocols.
> > >
> > > I'd like to get some discussion going on how we can make some progress
> > on these. I don't think we are going to solve these in 20 minutes of
> > discussion at an IETF meeting so I think we probably need some interim
> > (virtual or face to face) to sort this out.
> > >
> > > Thoughts?
> > >
> > > Wow, I'm totally confused here.
> > >
> > > I had assumed that the SDP-related issues were going to be the main
> > > topics at the WebRTC/RTCWEB interim in January. Is that not the case?
> >
> > So far the interim was only talking about being a WebRTC & RTCWeb so this
> > SDP stuff would be out of scope. Perhaps it would be better to have some
> > of the time for mmusic topics?
> >
> >
> > >
> > > IMO the lack of clarity around how to encode various media
> > > configurations into SDP is the major thing blocking progress here. In
> > > particular, Firefox has opted not to implement multiplexing of media
> > > streams over the same transport flow (whether of the bundle or
> > > multiple m-line variety) until the SDP for it is well-defined. The
> > > same thing applies to the question of how to map multiple m-lines to
> > > incoming MediaStreams/Tracks.
> > >
> > > We really need to cover these issues in the interim.
> > >
> > > -Ekr
> > >
> > >
> >
> > _______________________________________________
> > 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
>

--20cf30780ca44d069a04d08301c7
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

On Mon, Dec 10, 2012 at 5:16 AM, DRAGE, Keith (Keith) <span dir=3D"ltr">&lt=
;<a href=3D"mailto:keith.drage@alcatel-lucent.com" target=3D"_blank">keith.=
drage@alcatel-lucent.com</a>&gt;</span> wrote:<br><div class=3D"gmail_quote=
"><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:=
1px #ccc solid;padding-left:1ex">
This is not just MMUSIC, there are bits of AVTCORE and AVTEXT responsibilit=
y in there as well.<br>
<br>
And I don&#39;t see how you can suddenly extend the interim of RTCWEB to th=
e scope of other working groups, without even having a discussion with all =
the relevant chairs.<br>
<br></blockquote><div><br>That&#39;s not how I read the overall sentiment--=
people are simply saying that they need this work done and are willing to p=
ut in the effort to get it done.=A0 It would, in fact, be ideal if some of =
the MMUSIC questions were resolve *before* an RTCWEB/WEBRTC interim, as we =
can then work through what is decided. <br>
<br>Put another way, I take it as encouragement to MMUSIC folks that they w=
ould find a January virtual interim very well received/attended.<br><br>Ted=
<br><br>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8=
ex;border-left:1px #ccc solid;padding-left:1ex">

I&#39;d further note that Jonathan Lennox has an action (with the support o=
f quite a number of other people) to produce an internet draft on trying to=
 sort out the terminology concents that exist within RTP at the moment. Thi=
s would be the basis for progressing a number of these issues. That would p=
robably have to be an RAI wide draft.<br>

<br>
There has been some progress on this - maybe Jonathan can report on where t=
his now is. I know he was waiting on volunteers for some of the sections an=
d on input from others where volunteers already existed.<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
Keith<br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
&gt; -----Original Message-----<br>
&gt; From: <a href=3D"mailto:mmusic-bounces@ietf.org">mmusic-bounces@ietf.o=
rg</a> [mailto:<a href=3D"mailto:mmusic-bounces@ietf.org">mmusic-bounces@ie=
tf.org</a>] On Behalf<br>
&gt; Of Cullen Jennings (fluffy)<br>
&gt; Sent: 09 December 2012 00:20<br>
&gt; To: Eric Rescorla<br>
&gt; Cc: &lt;<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;; mm=
usic WG; &lt;<a href=3D"mailto:public-webrtc@w3.org">public-webrtc@w3.org</=
a>&gt;<br>
&gt; Subject: Re: [MMUSIC] SDP work needed for WebRTC stuff<br>
&gt;<br>
&gt;<br>
&gt; On Dec 8, 2012, at 4:21 PM, Eric Rescorla &lt;<a href=3D"mailto:ekr@rt=
fm.com">ekr@rtfm.com</a>&gt; wrote:<br>
&gt;<br>
&gt; &gt; +rtcweb<br>
&gt; &gt;<br>
&gt; &gt; On Sat, Dec 8, 2012 at 2:41 PM, Cullen Jennings (fluffy)<br>
&gt; &lt;<a href=3D"mailto:fluffy@cisco.com">fluffy@cisco.com</a>&gt; wrote=
:<br>
&gt; &gt;<br>
&gt; &gt; I was looking over everything that needs to be completed to finis=
h a<br>
&gt; fist cut of the WebRTC related work. There are a handful of big SDP<br=
>
&gt; problems that are currently blocking some of the WebRTC work and I&#39=
;d like<br>
&gt; to figure out how to make some progress on them.<br>
&gt; &gt;<br>
&gt; &gt; Let me loosely characterize them as<br>
&gt; &gt;<br>
&gt; &gt; 1) If we have several video streams, how do theses map up to 1 or=
 more m<br>
&gt; lines.<br>
&gt; &gt;<br>
&gt; &gt; 2) if we are doing port multiplexing, what does the SDP look like=
 (the<br>
&gt; bundle problem)<br>
&gt; &gt;<br>
&gt; &gt; 3) How do we map the RTCWeb track and stream label concepts to<br=
>
&gt; identifiers in SDP<br>
&gt; &gt;<br>
&gt; &gt; 3) SDP for application running over SCTP/DTLS<br>
&gt; &gt;<br>
&gt; &gt; I don&#39;t want to speak for all the various chairs but I am und=
er the<br>
&gt; impression that most of chairs of related groups in W3C and IETF belie=
ve<br>
&gt; these are issues that need to be resolved primarily in the MMUSIC WG a=
nd<br>
&gt; that they impact both WebRTC and CLUE as well as the general long term=
 use<br>
&gt; of SDP in SIP and other protocols.<br>
&gt; &gt;<br>
&gt; &gt; I&#39;d like to get some discussion going on how we can make some=
 progress<br>
&gt; on these. I don&#39;t think we are going to solve these in 20 minutes =
of<br>
&gt; discussion at an IETF meeting so I think we probably need some interim=
<br>
&gt; (virtual or face to face) to sort this out.<br>
&gt; &gt;<br>
&gt; &gt; Thoughts?<br>
&gt; &gt;<br>
&gt; &gt; Wow, I&#39;m totally confused here.<br>
&gt; &gt;<br>
&gt; &gt; I had assumed that the SDP-related issues were going to be the ma=
in<br>
&gt; &gt; topics at the WebRTC/RTCWEB interim in January. Is that not the c=
ase?<br>
&gt;<br>
&gt; So far the interim was only talking about being a WebRTC &amp; RTCWeb =
so this<br>
&gt; SDP stuff would be out of scope. Perhaps it would be better to have so=
me<br>
&gt; of the time for mmusic topics?<br>
&gt;<br>
&gt;<br>
&gt; &gt;<br>
&gt; &gt; IMO the lack of clarity around how to encode various media<br>
&gt; &gt; configurations into SDP is the major thing blocking progress here=
. In<br>
&gt; &gt; particular, Firefox has opted not to implement multiplexing of me=
dia<br>
&gt; &gt; streams over the same transport flow (whether of the bundle or<br=
>
&gt; &gt; multiple m-line variety) until the SDP for it is well-defined. Th=
e<br>
&gt; &gt; same thing applies to the question of how to map multiple m-lines=
 to<br>
&gt; &gt; incoming MediaStreams/Tracks.<br>
&gt; &gt;<br>
&gt; &gt; We really need to cover these issues in the interim.<br>
&gt; &gt;<br>
&gt; &gt; -Ekr<br>
&gt; &gt;<br>
&gt; &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" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/mmusic</a><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>
</div></div></blockquote></div><br>

--20cf30780ca44d069a04d08301c7--

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Subject: Re: [MMUSIC] SDP work needed for WebRTC stuff
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On Dec 10, 2012, at 5:16 , "DRAGE, Keith (Keith)" =
<keith.drage@alcatel-lucent.com> wrote:

> This is not just MMUSIC, there are bits of AVTCORE and AVTEXT =
responsibility in there as well.

can you add a bit on what is needed in these WG just so I can keep track =
of it all

>=20
> And I don't see how you can suddenly extend the interim of RTCWEB to =
the scope of other working groups, without even having a discussion with =
all the relevant chairs.

I did not mean to do any such thing - obviously the MMUSIC WG would need =
to decide what to how it was going to do the work. All I was trying to =
say is there are some things that I think needs to be done in MMUSIC and =
I view it as out of scope for rtcweb. This clearly came as a surprise to =
some people who thought that was the main thing the interim meeting was =
going to be about. As an individual contributor for to the MMUSIC wg, I =
think that mmusic needs to figure out a substantial block of face to =
face time to sort out these SDP issues. I look forward to proposals on =
how to do that.=20



From ekr@rtfm.com  Mon Dec 10 17:18:59 2012
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From: Eric Rescorla <ekr@rtfm.com>
Date: Mon, 10 Dec 2012 17:18:17 -0800
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Subject: Re: [MMUSIC] SDP work needed for WebRTC stuff
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--e89a8f64355658ba5404d0897766
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I feel like the priorities here are kind of confused.

It seems to me that the purpose of having a physical interim is to resolve
issues that are hard to resolve on the list or teleconferences. As I
mentioned
earlier, the SDP issues are the most important to resolve now and have
also proven to be among the most intractable. How does it make sense
to try to resolve those at a "virtual" interim while having people fly
to a "physical" interim for three days?

I would strongly encourage the chairs to figure out how to have at least
one of the days of this interim devoted solely to resolving the SDP
issues (really, my preference would be to take these issues one at
a time and do nothing else until they are resolved!). I'm not expert
enough at the RAI WG mechanics to know what's required, so perhaps
this requires making this also an MMUSIC interim or a mini-RAI interim,
but if so that's what we should do, even if it means we need to reschedule
or move it.

Best,
-Ekr

On Mon, Dec 10, 2012 at 9:36 AM, Ted Hardie <ted.ietf@gmail.com> wrote:

> On Mon, Dec 10, 2012 at 5:16 AM, DRAGE, Keith (Keith) <
> keith.drage@alcatel-lucent.com> wrote:
>
>> This is not just MMUSIC, there are bits of AVTCORE and AVTEXT
>> responsibility in there as well.
>>
>> And I don't see how you can suddenly extend the interim of RTCWEB to the
>> scope of other working groups, without even having a discussion with all
>> the relevant chairs.
>>
>>
> That's not how I read the overall sentiment--people are simply saying that
> they need this work done and are willing to put in the effort to get it
> done.  It would, in fact, be ideal if some of the MMUSIC questions were
> resolve *before* an RTCWEB/WEBRTC interim, as we can then work through what
> is decided.
>
> Put another way, I take it as encouragement to MMUSIC folks that they
> would find a January virtual interim very well received/attended.
>
> Ted
>
>
>
>> I'd further note that Jonathan Lennox has an action (with the support of
>> quite a number of other people) to produce an internet draft on trying to
>> sort out the terminology concents that exist within RTP at the moment. This
>> would be the basis for progressing a number of these issues. That would
>> probably have to be an RAI wide draft.
>>
>> There has been some progress on this - maybe Jonathan can report on where
>> this now is. I know he was waiting on volunteers for some of the sections
>> and on input from others where volunteers already existed.
>>
>> Keith
>>
>> > -----Original Message-----
>> > From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
>> Behalf
>> > Of Cullen Jennings (fluffy)
>> > Sent: 09 December 2012 00:20
>> > To: Eric Rescorla
>> > Cc: <rtcweb@ietf.org>; mmusic WG; <public-webrtc@w3.org>
>> > Subject: Re: [MMUSIC] SDP work needed for WebRTC stuff
>> >
>> >
>> > On Dec 8, 2012, at 4:21 PM, Eric Rescorla <ekr@rtfm.com> wrote:
>> >
>> > > +rtcweb
>> > >
>> > > On Sat, Dec 8, 2012 at 2:41 PM, Cullen Jennings (fluffy)
>> > <fluffy@cisco.com> wrote:
>> > >
>> > > I was looking over everything that needs to be completed to finish a
>> > fist cut of the WebRTC related work. There are a handful of big SDP
>> > problems that are currently blocking some of the WebRTC work and I'd
>> like
>> > to figure out how to make some progress on them.
>> > >
>> > > Let me loosely characterize them as
>> > >
>> > > 1) If we have several video streams, how do theses map up to 1 or
>> more m
>> > lines.
>> > >
>> > > 2) if we are doing port multiplexing, what does the SDP look like (the
>> > bundle problem)
>> > >
>> > > 3) How do we map the RTCWeb track and stream label concepts to
>> > identifiers in SDP
>> > >
>> > > 3) SDP for application running over SCTP/DTLS
>> > >
>> > > I don't want to speak for all the various chairs but I am under the
>> > impression that most of chairs of related groups in W3C and IETF believe
>> > these are issues that need to be resolved primarily in the MMUSIC WG and
>> > that they impact both WebRTC and CLUE as well as the general long term
>> use
>> > of SDP in SIP and other protocols.
>> > >
>> > > I'd like to get some discussion going on how we can make some progress
>> > on these. I don't think we are going to solve these in 20 minutes of
>> > discussion at an IETF meeting so I think we probably need some interim
>> > (virtual or face to face) to sort this out.
>> > >
>> > > Thoughts?
>> > >
>> > > Wow, I'm totally confused here.
>> > >
>> > > I had assumed that the SDP-related issues were going to be the main
>> > > topics at the WebRTC/RTCWEB interim in January. Is that not the case?
>> >
>> > So far the interim was only talking about being a WebRTC & RTCWeb so
>> this
>> > SDP stuff would be out of scope. Perhaps it would be better to have some
>> > of the time for mmusic topics?
>> >
>> >
>> > >
>> > > IMO the lack of clarity around how to encode various media
>> > > configurations into SDP is the major thing blocking progress here. In
>> > > particular, Firefox has opted not to implement multiplexing of media
>> > > streams over the same transport flow (whether of the bundle or
>> > > multiple m-line variety) until the SDP for it is well-defined. The
>> > > same thing applies to the question of how to map multiple m-lines to
>> > > incoming MediaStreams/Tracks.
>> > >
>> > > We really need to cover these issues in the interim.
>> > >
>> > > -Ekr
>> > >
>> > >
>> >
>> > _______________________________________________
>> > 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
>>
>
>

--e89a8f64355658ba5404d0897766
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

I feel like the priorities here are kind of confused.<div><br></div><div>It=
 seems to me that the purpose of having a physical interim is to resolve</d=
iv><div>issues that are hard to resolve on the list or teleconferences. As =
I mentioned</div>


<div>earlier, the SDP issues are the most important to resolve now and have=
</div><div>also proven to be among the most intractable. How does it make s=
ense</div><div>to try to resolve those at a &quot;virtual&quot; interim whi=
le having people fly</div>


<div>to a &quot;physical&quot; interim for three days?</div><div><br></div>=
<div>I would strongly encourage the chairs to figure out how to have at lea=
st</div><div>one of the days of this interim devoted solely to resolving th=
e SDP=A0</div>


<div>issues (really, my preference would be to take these issues one at</di=
v><div>a time and do nothing else until they are resolved!). I&#39;m not ex=
pert</div><div>enough at the RAI WG mechanics to know what&#39;s required, =
so perhaps</div>


<div>this requires making this also an MMUSIC interim or a mini-RAI interim=
,</div><div>but if so that&#39;s what we should do, even if it means we nee=
d to reschedule</div><div>or move it.</div><div><br></div><div>Best,</div>


<div>-Ekr</div><div><br></div><div><div class=3D"gmail_quote">On Mon, Dec 1=
0, 2012 at 9:36 AM, Ted Hardie <span dir=3D"ltr">&lt;<a href=3D"mailto:ted.=
ietf@gmail.com" target=3D"_blank">ted.ietf@gmail.com</a>&gt;</span> wrote:<=
br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div>On Mon, Dec 10, 2012 at 5:16 AM, DRAGE, Keith (Keith) <span dir=3D"ltr=
">&lt;<a href=3D"mailto:keith.drage@alcatel-lucent.com" target=3D"_blank">k=
eith.drage@alcatel-lucent.com</a>&gt;</span> wrote:<br></div><div class=3D"=
gmail_quote">


<div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-le=
ft:1px #ccc solid;padding-left:1ex">
This is not just MMUSIC, there are bits of AVTCORE and AVTEXT responsibilit=
y in there as well.<br>
<br>
And I don&#39;t see how you can suddenly extend the interim of RTCWEB to th=
e scope of other working groups, without even having a discussion with all =
the relevant chairs.<br>
<br></blockquote></div><div><br>That&#39;s not how I read the overall senti=
ment--people are simply saying that they need this work done and are willin=
g to put in the effort to get it done.=A0 It would, in fact, be ideal if so=
me of the MMUSIC questions were resolve *before* an RTCWEB/WEBRTC interim, =
as we can then work through what is decided. <br>



<br>Put another way, I take it as encouragement to MMUSIC folks that they w=
ould find a January virtual interim very well received/attended.<span><font=
 color=3D"#888888"><br><br>Ted<br><br>=A0</font></span></div>
<div><div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bord=
er-left:1px #ccc solid;padding-left:1ex">

I&#39;d further note that Jonathan Lennox has an action (with the support o=
f quite a number of other people) to produce an internet draft on trying to=
 sort out the terminology concents that exist within RTP at the moment. Thi=
s would be the basis for progressing a number of these issues. That would p=
robably have to be an RAI wide draft.<br>




<br>
There has been some progress on this - maybe Jonathan can report on where t=
his now is. I know he was waiting on volunteers for some of the sections an=
d on input from others where volunteers already existed.<br>
<span><font color=3D"#888888"><br>
Keith<br>
</font></span><div><div><br>
&gt; -----Original Message-----<br>
&gt; From: <a href=3D"mailto:mmusic-bounces@ietf.org" target=3D"_blank">mmu=
sic-bounces@ietf.org</a> [mailto:<a href=3D"mailto:mmusic-bounces@ietf.org"=
 target=3D"_blank">mmusic-bounces@ietf.org</a>] On Behalf<br>
&gt; Of Cullen Jennings (fluffy)<br>
&gt; Sent: 09 December 2012 00:20<br>
&gt; To: Eric Rescorla<br>
&gt; Cc: &lt;<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ie=
tf.org</a>&gt;; mmusic WG; &lt;<a href=3D"mailto:public-webrtc@w3.org" targ=
et=3D"_blank">public-webrtc@w3.org</a>&gt;<br>
&gt; Subject: Re: [MMUSIC] SDP work needed for WebRTC stuff<br>
&gt;<br>
&gt;<br>
&gt; On Dec 8, 2012, at 4:21 PM, Eric Rescorla &lt;<a href=3D"mailto:ekr@rt=
fm.com" target=3D"_blank">ekr@rtfm.com</a>&gt; wrote:<br>
&gt;<br>
&gt; &gt; +rtcweb<br>
&gt; &gt;<br>
&gt; &gt; On Sat, Dec 8, 2012 at 2:41 PM, Cullen Jennings (fluffy)<br>
&gt; &lt;<a href=3D"mailto:fluffy@cisco.com" target=3D"_blank">fluffy@cisco=
.com</a>&gt; wrote:<br>
&gt; &gt;<br>
&gt; &gt; I was looking over everything that needs to be completed to finis=
h a<br>
&gt; fist cut of the WebRTC related work. There are a handful of big SDP<br=
>
&gt; problems that are currently blocking some of the WebRTC work and I&#39=
;d like<br>
&gt; to figure out how to make some progress on them.<br>
&gt; &gt;<br>
&gt; &gt; Let me loosely characterize them as<br>
&gt; &gt;<br>
&gt; &gt; 1) If we have several video streams, how do theses map up to 1 or=
 more m<br>
&gt; lines.<br>
&gt; &gt;<br>
&gt; &gt; 2) if we are doing port multiplexing, what does the SDP look like=
 (the<br>
&gt; bundle problem)<br>
&gt; &gt;<br>
&gt; &gt; 3) How do we map the RTCWeb track and stream label concepts to<br=
>
&gt; identifiers in SDP<br>
&gt; &gt;<br>
&gt; &gt; 3) SDP for application running over SCTP/DTLS<br>
&gt; &gt;<br>
&gt; &gt; I don&#39;t want to speak for all the various chairs but I am und=
er the<br>
&gt; impression that most of chairs of related groups in W3C and IETF belie=
ve<br>
&gt; these are issues that need to be resolved primarily in the MMUSIC WG a=
nd<br>
&gt; that they impact both WebRTC and CLUE as well as the general long term=
 use<br>
&gt; of SDP in SIP and other protocols.<br>
&gt; &gt;<br>
&gt; &gt; I&#39;d like to get some discussion going on how we can make some=
 progress<br>
&gt; on these. I don&#39;t think we are going to solve these in 20 minutes =
of<br>
&gt; discussion at an IETF meeting so I think we probably need some interim=
<br>
&gt; (virtual or face to face) to sort this out.<br>
&gt; &gt;<br>
&gt; &gt; Thoughts?<br>
&gt; &gt;<br>
&gt; &gt; Wow, I&#39;m totally confused here.<br>
&gt; &gt;<br>
&gt; &gt; I had assumed that the SDP-related issues were going to be the ma=
in<br>
&gt; &gt; topics at the WebRTC/RTCWEB interim in January. Is that not the c=
ase?<br>
&gt;<br>
&gt; So far the interim was only talking about being a WebRTC &amp; RTCWeb =
so this<br>
&gt; SDP stuff would be out of scope. Perhaps it would be better to have so=
me<br>
&gt; of the time for mmusic topics?<br>
&gt;<br>
&gt;<br>
&gt; &gt;<br>
&gt; &gt; IMO the lack of clarity around how to encode various media<br>
&gt; &gt; configurations into SDP is the major thing blocking progress here=
. In<br>
&gt; &gt; particular, Firefox has opted not to implement multiplexing of me=
dia<br>
&gt; &gt; streams over the same transport flow (whether of the bundle or<br=
>
&gt; &gt; multiple m-line variety) until the SDP for it is well-defined. Th=
e<br>
&gt; &gt; same thing applies to the question of how to map multiple m-lines=
 to<br>
&gt; &gt; incoming MediaStreams/Tracks.<br>
&gt; &gt;<br>
&gt; &gt; We really need to cover these issues in the interim.<br>
&gt; &gt;<br>
&gt; &gt; -Ekr<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; mmusic mailing list<br>
&gt; <a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">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>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">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>
</div></div></blockquote></div></div></div><br>
</blockquote></div><br></div>

--e89a8f64355658ba5404d0897766--

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Date: Mon, 10 Dec 2012 17:29:26 -0800
Message-ID: <CABkgnnXyyX9jbjaExVh9J9SLnRE=soqNfvfwX+N6bj4-5ikW8g@mail.gmail.com>
From: Martin Thomson <martin.thomson@gmail.com>
To: Eric Rescorla <ekr@rtfm.com>
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Cc: "Cullen Jennings \(fluffy\)" <fluffy@cisco.com>, mmusic@ietf.org, "DRAGE, Keith \(Keith\)" <keith.drage@alcatel-lucent.com>, "&lt, rtcweb@ietf.org&gt, " <rtcweb@ietf.org>, "&lt, public-webrtc@w3.org&gt, " <public-webrtc@w3.org>
Subject: Re: [MMUSIC] SDP work needed for WebRTC stuff
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If there is a way to make this happen, that would be good.  If not, then I
think the only realistic option available to rtcweb would be to drop
bundle, etc... That wouldn't be popular.
On Dec 10, 2012 5:19 PM, "Eric Rescorla" <ekr@rtfm.com> wrote:

> I feel like the priorities here are kind of confused.
>
> It seems to me that the purpose of having a physical interim is to resolve
> issues that are hard to resolve on the list or teleconferences. As I
> mentioned
> earlier, the SDP issues are the most important to resolve now and have
> also proven to be among the most intractable. How does it make sense
> to try to resolve those at a "virtual" interim while having people fly
> to a "physical" interim for three days?
>
> I would strongly encourage the chairs to figure out how to have at least
> one of the days of this interim devoted solely to resolving the SDP
> issues (really, my preference would be to take these issues one at
> a time and do nothing else until they are resolved!). I'm not expert
> enough at the RAI WG mechanics to know what's required, so perhaps
> this requires making this also an MMUSIC interim or a mini-RAI interim,
> but if so that's what we should do, even if it means we need to reschedule
> or move it.
>
> Best,
> -Ekr
>
> On Mon, Dec 10, 2012 at 9:36 AM, Ted Hardie <ted.ietf@gmail.com> wrote:
>
>> On Mon, Dec 10, 2012 at 5:16 AM, DRAGE, Keith (Keith) <
>> keith.drage@alcatel-lucent.com> wrote:
>>
>>> This is not just MMUSIC, there are bits of AVTCORE and AVTEXT
>>> responsibility in there as well.
>>>
>>> And I don't see how you can suddenly extend the interim of RTCWEB to the
>>> scope of other working groups, without even having a discussion with all
>>> the relevant chairs.
>>>
>>>
>> That's not how I read the overall sentiment--people are simply saying
>> that they need this work done and are willing to put in the effort to get
>> it done.  It would, in fact, be ideal if some of the MMUSIC questions were
>> resolve *before* an RTCWEB/WEBRTC interim, as we can then work through what
>> is decided.
>>
>> Put another way, I take it as encouragement to MMUSIC folks that they
>> would find a January virtual interim very well received/attended.
>>
>> Ted
>>
>>
>>
>>> I'd further note that Jonathan Lennox has an action (with the support of
>>> quite a number of other people) to produce an internet draft on trying to
>>> sort out the terminology concents that exist within RTP at the moment. This
>>> would be the basis for progressing a number of these issues. That would
>>> probably have to be an RAI wide draft.
>>>
>>> There has been some progress on this - maybe Jonathan can report on
>>> where this now is. I know he was waiting on volunteers for some of the
>>> sections and on input from others where volunteers already existed.
>>>
>>> Keith
>>>
>>> > -----Original Message-----
>>> > From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
>>> Behalf
>>> > Of Cullen Jennings (fluffy)
>>> > Sent: 09 December 2012 00:20
>>> > To: Eric Rescorla
>>> > Cc: <rtcweb@ietf.org>; mmusic WG; <public-webrtc@w3.org>
>>> > Subject: Re: [MMUSIC] SDP work needed for WebRTC stuff
>>> >
>>> >
>>> > On Dec 8, 2012, at 4:21 PM, Eric Rescorla <ekr@rtfm.com> wrote:
>>> >
>>> > > +rtcweb
>>> > >
>>> > > On Sat, Dec 8, 2012 at 2:41 PM, Cullen Jennings (fluffy)
>>> > <fluffy@cisco.com> wrote:
>>> > >
>>> > > I was looking over everything that needs to be completed to finish a
>>> > fist cut of the WebRTC related work. There are a handful of big SDP
>>> > problems that are currently blocking some of the WebRTC work and I'd
>>> like
>>> > to figure out how to make some progress on them.
>>> > >
>>> > > Let me loosely characterize them as
>>> > >
>>> > > 1) If we have several video streams, how do theses map up to 1 or
>>> more m
>>> > lines.
>>> > >
>>> > > 2) if we are doing port multiplexing, what does the SDP look like
>>> (the
>>> > bundle problem)
>>> > >
>>> > > 3) How do we map the RTCWeb track and stream label concepts to
>>> > identifiers in SDP
>>> > >
>>> > > 3) SDP for application running over SCTP/DTLS
>>> > >
>>> > > I don't want to speak for all the various chairs but I am under the
>>> > impression that most of chairs of related groups in W3C and IETF
>>> believe
>>> > these are issues that need to be resolved primarily in the MMUSIC WG
>>> and
>>> > that they impact both WebRTC and CLUE as well as the general long term
>>> use
>>> > of SDP in SIP and other protocols.
>>> > >
>>> > > I'd like to get some discussion going on how we can make some
>>> progress
>>> > on these. I don't think we are going to solve these in 20 minutes of
>>> > discussion at an IETF meeting so I think we probably need some interim
>>> > (virtual or face to face) to sort this out.
>>> > >
>>> > > Thoughts?
>>> > >
>>> > > Wow, I'm totally confused here.
>>> > >
>>> > > I had assumed that the SDP-related issues were going to be the main
>>> > > topics at the WebRTC/RTCWEB interim in January. Is that not the case?
>>> >
>>> > So far the interim was only talking about being a WebRTC & RTCWeb so
>>> this
>>> > SDP stuff would be out of scope. Perhaps it would be better to have
>>> some
>>> > of the time for mmusic topics?
>>> >
>>> >
>>> > >
>>> > > IMO the lack of clarity around how to encode various media
>>> > > configurations into SDP is the major thing blocking progress here. In
>>> > > particular, Firefox has opted not to implement multiplexing of media
>>> > > streams over the same transport flow (whether of the bundle or
>>> > > multiple m-line variety) until the SDP for it is well-defined. The
>>> > > same thing applies to the question of how to map multiple m-lines to
>>> > > incoming MediaStreams/Tracks.
>>> > >
>>> > > We really need to cover these issues in the interim.
>>> > >
>>> > > -Ekr
>>> > >
>>> > >
>>> >
>>> > _______________________________________________
>>> > 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
>
>

--f46d0420a695cf5d7e04d0899c81
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<p>If there is a way to make this happen, that would be good.=C2=A0 If not,=
 then I think the only realistic option available to rtcweb would be to dro=
p bundle, etc... That wouldn&#39;t be popular.</p>
<div class=3D"gmail_quote">On Dec 10, 2012 5:19 PM, &quot;Eric Rescorla&quo=
t; &lt;<a href=3D"mailto:ekr@rtfm.com">ekr@rtfm.com</a>&gt; wrote:<br type=
=3D"attribution"><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8=
ex;border-left:1px #ccc solid;padding-left:1ex">
I feel like the priorities here are kind of confused.<div><br></div><div>It=
 seems to me that the purpose of having a physical interim is to resolve</d=
iv><div>issues that are hard to resolve on the list or teleconferences. As =
I mentioned</div>



<div>earlier, the SDP issues are the most important to resolve now and have=
</div><div>also proven to be among the most intractable. How does it make s=
ense</div><div>to try to resolve those at a &quot;virtual&quot; interim whi=
le having people fly</div>



<div>to a &quot;physical&quot; interim for three days?</div><div><br></div>=
<div>I would strongly encourage the chairs to figure out how to have at lea=
st</div><div>one of the days of this interim devoted solely to resolving th=
e SDP=C2=A0</div>



<div>issues (really, my preference would be to take these issues one at</di=
v><div>a time and do nothing else until they are resolved!). I&#39;m not ex=
pert</div><div>enough at the RAI WG mechanics to know what&#39;s required, =
so perhaps</div>



<div>this requires making this also an MMUSIC interim or a mini-RAI interim=
,</div><div>but if so that&#39;s what we should do, even if it means we nee=
d to reschedule</div><div>or move it.</div><div><br></div><div>Best,</div>



<div>-Ekr</div><div><br></div><div><div class=3D"gmail_quote">On Mon, Dec 1=
0, 2012 at 9:36 AM, Ted Hardie <span dir=3D"ltr">&lt;<a href=3D"mailto:ted.=
ietf@gmail.com" target=3D"_blank">ted.ietf@gmail.com</a>&gt;</span> wrote:<=
br>


<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div>On Mon, Dec 10, 2012 at 5:16 AM, DRAGE, Keith (Keith) <span dir=3D"ltr=
">&lt;<a href=3D"mailto:keith.drage@alcatel-lucent.com" target=3D"_blank">k=
eith.drage@alcatel-lucent.com</a>&gt;</span> wrote:<br></div><div class=3D"=
gmail_quote">



<div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-le=
ft:1px #ccc solid;padding-left:1ex">
This is not just MMUSIC, there are bits of AVTCORE and AVTEXT responsibilit=
y in there as well.<br>
<br>
And I don&#39;t see how you can suddenly extend the interim of RTCWEB to th=
e scope of other working groups, without even having a discussion with all =
the relevant chairs.<br>
<br></blockquote></div><div><br>That&#39;s not how I read the overall senti=
ment--people are simply saying that they need this work done and are willin=
g to put in the effort to get it done.=C2=A0 It would, in fact, be ideal if=
 some of the MMUSIC questions were resolve *before* an RTCWEB/WEBRTC interi=
m, as we can then work through what is decided. <br>




<br>Put another way, I take it as encouragement to MMUSIC folks that they w=
ould find a January virtual interim very well received/attended.<span><font=
 color=3D"#888888"><br><br>Ted<br><br>=C2=A0</font></span></div>
<div><div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bord=
er-left:1px #ccc solid;padding-left:1ex">

I&#39;d further note that Jonathan Lennox has an action (with the support o=
f quite a number of other people) to produce an internet draft on trying to=
 sort out the terminology concents that exist within RTP at the moment. Thi=
s would be the basis for progressing a number of these issues. That would p=
robably have to be an RAI wide draft.<br>





<br>
There has been some progress on this - maybe Jonathan can report on where t=
his now is. I know he was waiting on volunteers for some of the sections an=
d on input from others where volunteers already existed.<br>
<span><font color=3D"#888888"><br>
Keith<br>
</font></span><div><div><br>
&gt; -----Original Message-----<br>
&gt; From: <a href=3D"mailto:mmusic-bounces@ietf.org" target=3D"_blank">mmu=
sic-bounces@ietf.org</a> [mailto:<a href=3D"mailto:mmusic-bounces@ietf.org"=
 target=3D"_blank">mmusic-bounces@ietf.org</a>] On Behalf<br>
&gt; Of Cullen Jennings (fluffy)<br>
&gt; Sent: 09 December 2012 00:20<br>
&gt; To: Eric Rescorla<br>
&gt; Cc: &lt;<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ie=
tf.org</a>&gt;; mmusic WG; &lt;<a href=3D"mailto:public-webrtc@w3.org" targ=
et=3D"_blank">public-webrtc@w3.org</a>&gt;<br>
&gt; Subject: Re: [MMUSIC] SDP work needed for WebRTC stuff<br>
&gt;<br>
&gt;<br>
&gt; On Dec 8, 2012, at 4:21 PM, Eric Rescorla &lt;<a href=3D"mailto:ekr@rt=
fm.com" target=3D"_blank">ekr@rtfm.com</a>&gt; wrote:<br>
&gt;<br>
&gt; &gt; +rtcweb<br>
&gt; &gt;<br>
&gt; &gt; On Sat, Dec 8, 2012 at 2:41 PM, Cullen Jennings (fluffy)<br>
&gt; &lt;<a href=3D"mailto:fluffy@cisco.com" target=3D"_blank">fluffy@cisco=
.com</a>&gt; wrote:<br>
&gt; &gt;<br>
&gt; &gt; I was looking over everything that needs to be completed to finis=
h a<br>
&gt; fist cut of the WebRTC related work. There are a handful of big SDP<br=
>
&gt; problems that are currently blocking some of the WebRTC work and I&#39=
;d like<br>
&gt; to figure out how to make some progress on them.<br>
&gt; &gt;<br>
&gt; &gt; Let me loosely characterize them as<br>
&gt; &gt;<br>
&gt; &gt; 1) If we have several video streams, how do theses map up to 1 or=
 more m<br>
&gt; lines.<br>
&gt; &gt;<br>
&gt; &gt; 2) if we are doing port multiplexing, what does the SDP look like=
 (the<br>
&gt; bundle problem)<br>
&gt; &gt;<br>
&gt; &gt; 3) How do we map the RTCWeb track and stream label concepts to<br=
>
&gt; identifiers in SDP<br>
&gt; &gt;<br>
&gt; &gt; 3) SDP for application running over SCTP/DTLS<br>
&gt; &gt;<br>
&gt; &gt; I don&#39;t want to speak for all the various chairs but I am und=
er the<br>
&gt; impression that most of chairs of related groups in W3C and IETF belie=
ve<br>
&gt; these are issues that need to be resolved primarily in the MMUSIC WG a=
nd<br>
&gt; that they impact both WebRTC and CLUE as well as the general long term=
 use<br>
&gt; of SDP in SIP and other protocols.<br>
&gt; &gt;<br>
&gt; &gt; I&#39;d like to get some discussion going on how we can make some=
 progress<br>
&gt; on these. I don&#39;t think we are going to solve these in 20 minutes =
of<br>
&gt; discussion at an IETF meeting so I think we probably need some interim=
<br>
&gt; (virtual or face to face) to sort this out.<br>
&gt; &gt;<br>
&gt; &gt; Thoughts?<br>
&gt; &gt;<br>
&gt; &gt; Wow, I&#39;m totally confused here.<br>
&gt; &gt;<br>
&gt; &gt; I had assumed that the SDP-related issues were going to be the ma=
in<br>
&gt; &gt; topics at the WebRTC/RTCWEB interim in January. Is that not the c=
ase?<br>
&gt;<br>
&gt; So far the interim was only talking about being a WebRTC &amp; RTCWeb =
so this<br>
&gt; SDP stuff would be out of scope. Perhaps it would be better to have so=
me<br>
&gt; of the time for mmusic topics?<br>
&gt;<br>
&gt;<br>
&gt; &gt;<br>
&gt; &gt; IMO the lack of clarity around how to encode various media<br>
&gt; &gt; configurations into SDP is the major thing blocking progress here=
. In<br>
&gt; &gt; particular, Firefox has opted not to implement multiplexing of me=
dia<br>
&gt; &gt; streams over the same transport flow (whether of the bundle or<br=
>
&gt; &gt; multiple m-line variety) until the SDP for it is well-defined. Th=
e<br>
&gt; &gt; same thing applies to the question of how to map multiple m-lines=
 to<br>
&gt; &gt; incoming MediaStreams/Tracks.<br>
&gt; &gt;<br>
&gt; &gt; We really need to cover these issues in the interim.<br>
&gt; &gt;<br>
&gt; &gt; -Ekr<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; mmusic mailing list<br>
&gt; <a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">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>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">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>
</div></div></blockquote></div></div></div><br>
</blockquote></div><br></div>
<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>
<br></blockquote></div>

--f46d0420a695cf5d7e04d0899c81--

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Cc: "Cullen Jennings \(fluffy\)" <fluffy@cisco.com>, "<rtcweb@ietf.org>" <rtcweb@ietf.org>, mmusic WG <mmusic@ietf.org>, "<public-webrtc@w3.org>" <public-webrtc@w3.org>
Subject: Re: [MMUSIC] [rtcweb]  SDP work needed for WebRTC stuff
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I agree with Ekr on all points.  We have put off resolving the SDP 
issues as long as we can, and the lack of resolution has started to 
impact our WebRTC implementation (in Firefox) already.  I feel we need 
to maximize our chances of resolving the SDP issues as soon as we can.

To me that means meeting in person as soon as possible to hammer this 
stuff out.

Please let's find a way to discuss this in person.  I think it's a big 
mistake if we rely on a virtual meeting only.

Thanks.
-Maire


On 12/10/2012 8:18 PM, Eric Rescorla wrote:
> I feel like the priorities here are kind of confused.
>
> It seems to me that the purpose of having a physical interim is to resolve
> issues that are hard to resolve on the list or teleconferences. As I 
> mentioned
> earlier, the SDP issues are the most important to resolve now and have
> also proven to be among the most intractable. How does it make sense
> to try to resolve those at a "virtual" interim while having people fly
> to a "physical" interim for three days?
>
> I would strongly encourage the chairs to figure out how to have at least
> one of the days of this interim devoted solely to resolving the SDP
> issues (really, my preference would be to take these issues one at
> a time and do nothing else until they are resolved!). I'm not expert
> enough at the RAI WG mechanics to know what's required, so perhaps
> this requires making this also an MMUSIC interim or a mini-RAI interim,
> but if so that's what we should do, even if it means we need to reschedule
> or move it.
>
> Best,
> -Ekr
>
> On Mon, Dec 10, 2012 at 9:36 AM, Ted Hardie <ted.ietf@gmail.com 
> <mailto:ted.ietf@gmail.com>> wrote:
>
>     On Mon, Dec 10, 2012 at 5:16 AM, DRAGE, Keith (Keith)
>     <keith.drage@alcatel-lucent.com
>     <mailto:keith.drage@alcatel-lucent.com>> wrote:
>
>         This is not just MMUSIC, there are bits of AVTCORE and AVTEXT
>         responsibility in there as well.
>
>         And I don't see how you can suddenly extend the interim of
>         RTCWEB to the scope of other working groups, without even
>         having a discussion with all the relevant chairs.
>
>
>     That's not how I read the overall sentiment--people are simply
>     saying that they need this work done and are willing to put in the
>     effort to get it done.  It would, in fact, be ideal if some of the
>     MMUSIC questions were resolve *before* an RTCWEB/WEBRTC interim,
>     as we can then work through what is decided.
>
>     Put another way, I take it as encouragement to MMUSIC folks that
>     they would find a January virtual interim very well received/attended.
>
>     Ted
>
>         I'd further note that Jonathan Lennox has an action (with the
>         support of quite a number of other people) to produce an
>         internet draft on trying to sort out the terminology concents
>         that exist within RTP at the moment. This would be the basis
>         for progressing a number of these issues. That would probably
>         have to be an RAI wide draft.
>
>         There has been some progress on this - maybe Jonathan can
>         report on where this now is. I know he was waiting on
>         volunteers for some of the sections and on input from others
>         where volunteers already existed.
>
>         Keith
>
>         > -----Original Message-----
>         > From: mmusic-bounces@ietf.org
>         <mailto:mmusic-bounces@ietf.org>
>         [mailto:mmusic-bounces@ietf.org
>         <mailto:mmusic-bounces@ietf.org>] On Behalf
>         > Of Cullen Jennings (fluffy)
>         > Sent: 09 December 2012 00:20
>         > To: Eric Rescorla
>         > Cc: <rtcweb@ietf.org <mailto:rtcweb@ietf.org>>; mmusic WG;
>         <public-webrtc@w3.org <mailto:public-webrtc@w3.org>>
>         > Subject: Re: [MMUSIC] SDP work needed for WebRTC stuff
>         >
>         >
>         > On Dec 8, 2012, at 4:21 PM, Eric Rescorla <ekr@rtfm.com
>         <mailto:ekr@rtfm.com>> wrote:
>         >
>         > > +rtcweb
>         > >
>         > > On Sat, Dec 8, 2012 at 2:41 PM, Cullen Jennings (fluffy)
>         > <fluffy@cisco.com <mailto:fluffy@cisco.com>> wrote:
>         > >
>         > > I was looking over everything that needs to be completed
>         to finish a
>         > fist cut of the WebRTC related work. There are a handful of
>         big SDP
>         > problems that are currently blocking some of the WebRTC work
>         and I'd like
>         > to figure out how to make some progress on them.
>         > >
>         > > Let me loosely characterize them as
>         > >
>         > > 1) If we have several video streams, how do theses map up
>         to 1 or more m
>         > lines.
>         > >
>         > > 2) if we are doing port multiplexing, what does the SDP
>         look like (the
>         > bundle problem)
>         > >
>         > > 3) How do we map the RTCWeb track and stream label concepts to
>         > identifiers in SDP
>         > >
>         > > 3) SDP for application running over SCTP/DTLS
>         > >
>         > > I don't want to speak for all the various chairs but I am
>         under the
>         > impression that most of chairs of related groups in W3C and
>         IETF believe
>         > these are issues that need to be resolved primarily in the
>         MMUSIC WG and
>         > that they impact both WebRTC and CLUE as well as the general
>         long term use
>         > of SDP in SIP and other protocols.
>         > >
>         > > I'd like to get some discussion going on how we can make
>         some progress
>         > on these. I don't think we are going to solve these in 20
>         minutes of
>         > discussion at an IETF meeting so I think we probably need
>         some interim
>         > (virtual or face to face) to sort this out.
>         > >
>         > > Thoughts?
>         > >
>         > > Wow, I'm totally confused here.
>         > >
>         > > I had assumed that the SDP-related issues were going to be
>         the main
>         > > topics at the WebRTC/RTCWEB interim in January. Is that
>         not the case?
>         >
>         > So far the interim was only talking about being a WebRTC &
>         RTCWeb so this
>         > SDP stuff would be out of scope. Perhaps it would be better
>         to have some
>         > of the time for mmusic topics?
>         >
>         >
>         > >
>         > > IMO the lack of clarity around how to encode various media
>         > > configurations into SDP is the major thing blocking
>         progress here. In
>         > > particular, Firefox has opted not to implement
>         multiplexing of media
>         > > streams over the same transport flow (whether of the bundle or
>         > > multiple m-line variety) until the SDP for it is
>         well-defined. The
>         > > same thing applies to the question of how to map multiple
>         m-lines to
>         > > incoming MediaStreams/Tracks.
>         > >
>         > > We really need to cover these issues in the interim.
>         > >
>         > > -Ekr
>         > >
>         > >
>         >
>         > _______________________________________________
>         > mmusic mailing list
>         > mmusic@ietf.org <mailto:mmusic@ietf.org>
>         > https://www.ietf.org/mailman/listinfo/mmusic
>         _______________________________________________
>         mmusic mailing list
>         mmusic@ietf.org <mailto:mmusic@ietf.org>
>         https://www.ietf.org/mailman/listinfo/mmusic
>
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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    <div class="moz-cite-prefix">I agree with Ekr on all points.&nbsp; We
      have put off resolving the SDP issues as long as we can, and the
      lack of resolution has started to impact our WebRTC implementation
      (in Firefox) already.&nbsp; I feel we need to maximize our chances of
      resolving the SDP issues as soon as we can.&nbsp; <br>
      <br>
      To me that means meeting in person as soon as possible to hammer
      this stuff out.&nbsp; <br>
      <br>
      Please let's find a way to discuss this in person.&nbsp; I think it's a
      big mistake if we rely on a virtual meeting only.<br>
      <br>
      Thanks.<br>
      -Maire<br>
      <br>
      <br>
      On 12/10/2012 8:18 PM, Eric Rescorla wrote:<br>
    </div>
    <blockquote
cite="mid:CABcZeBMjREPNVAg3UvEdink89qk+bwmbARK431wLLwv=gK6yhg@mail.gmail.com"
      type="cite">I feel like the priorities here are kind of confused.
      <div><br>
      </div>
      <div>It seems to me that the purpose of having a physical interim
        is to resolve</div>
      <div>issues that are hard to resolve on the list or
        teleconferences. As I mentioned</div>
      <div>earlier, the SDP issues are the most important to resolve now
        and have</div>
      <div>also proven to be among the most intractable. How does it
        make sense</div>
      <div>to try to resolve those at a "virtual" interim while having
        people fly</div>
      <div>to a "physical" interim for three days?</div>
      <div><br>
      </div>
      <div>I would strongly encourage the chairs to figure out how to
        have at least</div>
      <div>one of the days of this interim devoted solely to resolving
        the SDP&nbsp;</div>
      <div>issues (really, my preference would be to take these issues
        one at</div>
      <div>a time and do nothing else until they are resolved!). I'm not
        expert</div>
      <div>enough at the RAI WG mechanics to know what's required, so
        perhaps</div>
      <div>this requires making this also an MMUSIC interim or a
        mini-RAI interim,</div>
      <div>but if so that's what we should do, even if it means we need
        to reschedule</div>
      <div>or move it.</div>
      <div><br>
      </div>
      <div>Best,</div>
      <div>-Ekr</div>
      <div><br>
      </div>
      <div>
        <div class="gmail_quote">On Mon, Dec 10, 2012 at 9:36 AM, Ted
          Hardie <span dir="ltr">&lt;<a moz-do-not-send="true"
              href="mailto:ted.ietf@gmail.com" target="_blank">ted.ietf@gmail.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>On Mon, Dec 10, 2012 at 5:16 AM, DRAGE, Keith (Keith) <span
                dir="ltr">&lt;<a moz-do-not-send="true"
                  href="mailto:keith.drage@alcatel-lucent.com"
                  target="_blank">keith.drage@alcatel-lucent.com</a>&gt;</span>
              wrote:<br>
            </div>
            <div class="gmail_quote">
              <div>
                <blockquote class="gmail_quote" style="margin:0 0 0
                  .8ex;border-left:1px #ccc solid;padding-left:1ex">
                  This is not just MMUSIC, there are bits of AVTCORE and
                  AVTEXT responsibility in there as well.<br>
                  <br>
                  And I don't see how you can suddenly extend the
                  interim of RTCWEB to the scope of other working
                  groups, without even having a discussion with all the
                  relevant chairs.<br>
                  <br>
                </blockquote>
              </div>
              <div><br>
                That's not how I read the overall sentiment--people are
                simply saying that they need this work done and are
                willing to put in the effort to get it done.&nbsp; It would,
                in fact, be ideal if some of the MMUSIC questions were
                resolve *before* an RTCWEB/WEBRTC interim, as we can
                then work through what is decided. <br>
                <br>
                Put another way, I take it as encouragement to MMUSIC
                folks that they would find a January virtual interim
                very well received/attended.<span><font color="#888888"><br>
                    <br>
                    Ted<br>
                    <br>
                    &nbsp;</font></span></div>
              <div>
                <div>
                  <blockquote class="gmail_quote" style="margin:0 0 0
                    .8ex;border-left:1px #ccc solid;padding-left:1ex">
                    I'd further note that Jonathan Lennox has an action
                    (with the support of quite a number of other people)
                    to produce an internet draft on trying to sort out
                    the terminology concents that exist within RTP at
                    the moment. This would be the basis for progressing
                    a number of these issues. That would probably have
                    to be an RAI wide draft.<br>
                    <br>
                    There has been some progress on this - maybe
                    Jonathan can report on where this now is. I know he
                    was waiting on volunteers for some of the sections
                    and on input from others where volunteers already
                    existed.<br>
                    <span><font color="#888888"><br>
                        Keith<br>
                      </font></span>
                    <div>
                      <div><br>
                        &gt; -----Original Message-----<br>
                        &gt; From: <a moz-do-not-send="true"
                          href="mailto:mmusic-bounces@ietf.org"
                          target="_blank">mmusic-bounces@ietf.org</a>
                        [mailto:<a moz-do-not-send="true"
                          href="mailto:mmusic-bounces@ietf.org"
                          target="_blank">mmusic-bounces@ietf.org</a>]
                        On Behalf<br>
                        &gt; Of Cullen Jennings (fluffy)<br>
                        &gt; Sent: 09 December 2012 00:20<br>
                        &gt; To: Eric Rescorla<br>
                        &gt; Cc: &lt;<a moz-do-not-send="true"
                          href="mailto:rtcweb@ietf.org" target="_blank">rtcweb@ietf.org</a>&gt;;
                        mmusic WG; &lt;<a moz-do-not-send="true"
                          href="mailto:public-webrtc@w3.org"
                          target="_blank">public-webrtc@w3.org</a>&gt;<br>
                        &gt; Subject: Re: [MMUSIC] SDP work needed for
                        WebRTC stuff<br>
                        &gt;<br>
                        &gt;<br>
                        &gt; On Dec 8, 2012, at 4:21 PM, Eric Rescorla
                        &lt;<a moz-do-not-send="true"
                          href="mailto:ekr@rtfm.com" target="_blank">ekr@rtfm.com</a>&gt;
                        wrote:<br>
                        &gt;<br>
                        &gt; &gt; +rtcweb<br>
                        &gt; &gt;<br>
                        &gt; &gt; On Sat, Dec 8, 2012 at 2:41 PM, Cullen
                        Jennings (fluffy)<br>
                        &gt; &lt;<a moz-do-not-send="true"
                          href="mailto:fluffy@cisco.com" target="_blank">fluffy@cisco.com</a>&gt;
                        wrote:<br>
                        &gt; &gt;<br>
                        &gt; &gt; I was looking over everything that
                        needs to be completed to finish a<br>
                        &gt; fist cut of the WebRTC related work. There
                        are a handful of big SDP<br>
                        &gt; problems that are currently blocking some
                        of the WebRTC work and I'd like<br>
                        &gt; to figure out how to make some progress on
                        them.<br>
                        &gt; &gt;<br>
                        &gt; &gt; Let me loosely characterize them as<br>
                        &gt; &gt;<br>
                        &gt; &gt; 1) If we have several video streams,
                        how do theses map up to 1 or more m<br>
                        &gt; lines.<br>
                        &gt; &gt;<br>
                        &gt; &gt; 2) if we are doing port multiplexing,
                        what does the SDP look like (the<br>
                        &gt; bundle problem)<br>
                        &gt; &gt;<br>
                        &gt; &gt; 3) How do we map the RTCWeb track and
                        stream label concepts to<br>
                        &gt; identifiers in SDP<br>
                        &gt; &gt;<br>
                        &gt; &gt; 3) SDP for application running over
                        SCTP/DTLS<br>
                        &gt; &gt;<br>
                        &gt; &gt; I don't want to speak for all the
                        various chairs but I am under the<br>
                        &gt; impression that most of chairs of related
                        groups in W3C and IETF believe<br>
                        &gt; these are issues that need to be resolved
                        primarily in the MMUSIC WG and<br>
                        &gt; that they impact both WebRTC and CLUE as
                        well as the general long term use<br>
                        &gt; of SDP in SIP and other protocols.<br>
                        &gt; &gt;<br>
                        &gt; &gt; I'd like to get some discussion going
                        on how we can make some progress<br>
                        &gt; on these. I don't think we are going to
                        solve these in 20 minutes of<br>
                        &gt; discussion at an IETF meeting so I think we
                        probably need some interim<br>
                        &gt; (virtual or face to face) to sort this out.<br>
                        &gt; &gt;<br>
                        &gt; &gt; Thoughts?<br>
                        &gt; &gt;<br>
                        &gt; &gt; Wow, I'm totally confused here.<br>
                        &gt; &gt;<br>
                        &gt; &gt; I had assumed that the SDP-related
                        issues were going to be the main<br>
                        &gt; &gt; topics at the WebRTC/RTCWEB interim in
                        January. Is that not the case?<br>
                        &gt;<br>
                        &gt; So far the interim was only talking about
                        being a WebRTC &amp; RTCWeb so this<br>
                        &gt; SDP stuff would be out of scope. Perhaps it
                        would be better to have some<br>
                        &gt; of the time for mmusic topics?<br>
                        &gt;<br>
                        &gt;<br>
                        &gt; &gt;<br>
                        &gt; &gt; IMO the lack of clarity around how to
                        encode various media<br>
                        &gt; &gt; configurations into SDP is the major
                        thing blocking progress here. In<br>
                        &gt; &gt; particular, Firefox has opted not to
                        implement multiplexing of media<br>
                        &gt; &gt; streams over the same transport flow
                        (whether of the bundle or<br>
                        &gt; &gt; multiple m-line variety) until the SDP
                        for it is well-defined. The<br>
                        &gt; &gt; same thing applies to the question of
                        how to map multiple m-lines to<br>
                        &gt; &gt; incoming MediaStreams/Tracks.<br>
                        &gt; &gt;<br>
                        &gt; &gt; We really need to cover these issues
                        in the interim.<br>
                        &gt; &gt;<br>
                        &gt; &gt; -Ekr<br>
                        &gt; &gt;<br>
                        &gt; &gt;<br>
                        &gt;<br>
                        &gt;
                        _______________________________________________<br>
                        &gt; mmusic mailing list<br>
                        &gt; <a moz-do-not-send="true"
                          href="mailto:mmusic@ietf.org" target="_blank">mmusic@ietf.org</a><br>
                        &gt; <a moz-do-not-send="true"
                          href="https://www.ietf.org/mailman/listinfo/mmusic"
                          target="_blank">https://www.ietf.org/mailman/listinfo/mmusic</a><br>
                        _______________________________________________<br>
                        mmusic mailing list<br>
                        <a moz-do-not-send="true"
                          href="mailto:mmusic@ietf.org" target="_blank">mmusic@ietf.org</a><br>
                        <a moz-do-not-send="true"
                          href="https://www.ietf.org/mailman/listinfo/mmusic"
                          target="_blank">https://www.ietf.org/mailman/listinfo/mmusic</a><br>
                      </div>
                    </div>
                  </blockquote>
                </div>
              </div>
            </div>
            <br>
          </blockquote>
        </div>
        <br>
      </div>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
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Subject: Re: [MMUSIC] [rtcweb]  SDP work needed for WebRTC stuff
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I do not think that a single meeting, virtual or actual, will suffice to get
things back on track.  We need a fundamental change of approach. 

 

The basic problem is that WebRTC work items have been dispersed between a
number of WGs other than RTCWEB.  Within these WGs, WebRTC is only a
sideline. 

 


As a result, work items such as BUNDLE (which require non-trivial analysis
and discussion) have found themselves a needle in a haystack of other items,
getting a fraction of the attention that they need to move forward.  

 

While allocating some substantial time at an in-person RTCWEB/MMUSIC interim
can help, for that time to be well spent will require substantial
preparation.  As an example, I could see a virtual AVTEXT interim being
scheduled for mid-January at which Jonathan Lennox's hierarchy draft could
be given substantial time and an MMUSIC virtual interim in the same time
frame which could be devoted to one or two important topics (e.g. BUNDLE,
SCTP).

 

If that is done, then the issues requiring additional intensive work might
be narrowed to the point where an in-person meeting in early February might
be able to finish some of them off, with others handled another round of
virtual interims two weeks later. 

 

Bernard "MARTINI was hell - but it only lasted a year" Aboba

 



On 12/10/2012 8:18 PM, Eric Rescorla wrote:

I'm not expert enough at the RAI WG mechanics to know what's required, so
perhaps

this requires making this also an MMUSIC interim or a mini-RAI interim,

but if so that's what we should do, even if it means we need to reschedule

or move it.

 

 


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lang=3DEN-US link=3Dblue vlink=3Dpurple><div class=3DWordSection1><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>I do not think that a =
single meeting, virtual or actual, will suffice to get things back on =
track.&nbsp; We need a fundamental change of approach. =
<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>The basic problem is =
that WebRTC work items have been dispersed between a number of WGs other =
than RTCWEB.&nbsp; Within these WGs, WebRTC is only a sideline. =
<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
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bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
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p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <o:p></o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>As a result, work items =
such as BUNDLE (which require non-trivial analysis and discussion) have =
found themselves a needle in a haystack of other items, getting a =
fraction of the attention that they need to move forward. =
&nbsp;<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>While allocating some =
substantial time at an in-person RTCWEB/MMUSIC interim can help, for =
that time to be well spent will require substantial preparation. =
&nbsp;As an example, I could see a virtual AVTEXT interim being =
scheduled for mid-January at which Jonathan Lennox&#8217;s hierarchy =
draft could be given substantial time and an MMUSIC virtual interim in =
the same time frame which could be devoted to one or two important =
topics (e.g. BUNDLE, SCTP).<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>If that is done, then =
the issues requiring additional intensive work might be narrowed to the =
point where an in-person meeting in early February might be able to =
finish some of them off, with others handled another round of virtual =
interims two weeks later. <o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>Bernard &#8220;MARTINI =
was hell &#8211; but it only lasted a year&#8221; =
Aboba<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><br><br>On 12/10/2012 8:18 PM, Eric Rescorla =
wrote:<o:p></o:p></p><blockquote =
style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><div><p =
class=3DMsoNormal>I'm not expert<span style=3D'color:#1F497D'> =
</span>enough at the RAI WG mechanics to know what's required, so =
perhaps<span style=3D'color:#1F497D'><o:p></o:p></span></p></div><div><p =
class=3DMsoNormal>this requires making this also an MMUSIC interim or a =
mini-RAI interim,<o:p></o:p></p></div><div><p class=3DMsoNormal>but if =
so that's what we should do, even if it means we need to =
reschedule<o:p></o:p></p></div><div><p class=3DMsoNormal>or move =
it.<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></blockquote><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></body></html>
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Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
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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.
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.

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.


> 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


Salvatore

-- 
Salvatore Loreto, PhD
www.sloreto.com


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Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
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Hi Sal,

Salvatore Loreto wrote:
>
>based on the current stack the SCTP traffic is the only traffic that runs =
directly
>over the DTLS stack.
>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.
>
>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 fu=
ture.
>

It's a bit unclear to me what the benefit of this would be. SCTP itself all=
ows multiple streams to be transported in parallel within a single SCTP ass=
ociation. So, in what type of situations would we need multiple parallel SC=
TP associations? I thought the main point of SCTP is that it provides the p=
arallelism and multiplexing by itself.

In other words: I do understand the desire to multiplex RTP and SCTP within=
 the same UDP flow, so we can for instance reduce the number of needed NAT/=
FW bindings. I also understand why multiple independent non-HOL blocking st=
reams within SCTP are useful for applications. But I don't yet understand t=
he additional benefit of multiple parallel SCTP associations. No doubt it c=
an be technically done, but to what purposes?

Markus=20

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Subject: Re: [MMUSIC] [rtcweb]  SDP work needed for WebRTC stuff
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On 12/11/2012 06:01 AM, Bernard Aboba wrote:
> I do not think that a single meeting, virtual or actual, will suffice to
> get things back on track.  We need a fundamental change of approach.
>
> The basic problem is that WebRTC work items have been dispersed between
> a number of WGs other than RTCWEB.  Within these WGs, WebRTC is only a
> sideline.
>
> As a result, work items such as BUNDLE (which require non-trivial
> analysis and discussion) have found themselves a needle in a haystack of
> other items, getting a fraction of the attention that they need to move
> forward.

I'd say that Bundle is something that comes on top of more fundamental 
things (such as the representation of multiple streams and transports in 
SDP) needing resolution. If I have to prioritize I would do the 
fundamentals first and worry about Bundle later.

>
> While allocating some substantial time at an in-person RTCWEB/MMUSIC
> interim can help, for that time to be well spent will require
> substantial preparation.  As an example, I could see a virtual AVTEXT
> interim being scheduled for mid-January at which Jonathan Lennoxs
> hierarchy draft could be given substantial time and an MMUSIC virtual
> interim in the same time frame which could be devoted to one or two
> important topics (e.g. BUNDLE, SCTP).

I think this is a great idea!

Stefan


From pkyzivat@alum.mit.edu  Tue Dec 11 07:43:10 2012
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Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
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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.

That of course depends on having a signaling mechanism to set it up.

> 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.

	Thanks,
	Paul


From harald@alvestrand.no  Tue Dec 11 07:58:36 2012
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Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
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On 12/11/2012 04:43 PM, Paul Kyzivat wrote:
> 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.)
Yes, it's SRTP traffic that the DTLS/SCTP traffic will be multiplexed 
with (the DTLS part of DTLS/SRTP is just a key negotiation mechanism, 
unlike the DTLS part of DTLS/SCTP, which is an actual encapsulation).


From harald@alvestrand.no  Tue Dec 11 08:51:51 2012
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Subject: Re: [MMUSIC] [rtcweb]  SDP work needed for WebRTC stuff
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On 12/11/2012 06:01 AM, Bernard Aboba wrote:
>
> I do not think that a single meeting, virtual or actual, will suffice 
> to get things back on track.  We need a fundamental change of approach.
>
I see the main purpose of meetings to serve as stopping points for 
discussions.
In order to be effective, the proposals for solutions have to be created 
before the meeting, and the meeting serves to accept, reject or modify 
the proposals.
>
> The basic problem is that WebRTC work items have been dispersed 
> between a number of WGs other than RTCWEB.  Within these WGs, WebRTC 
> is only a sideline.
>
> As a result, work items such as BUNDLE (which require non-trivial 
> analysis and discussion) have found themselves a needle in a haystack 
> of other items, getting a fraction of the attention that they need to 
> move forward.
>
Indeed.
>
> While allocating some substantial time at an in-person RTCWEB/MMUSIC 
> interim can help, for that time to be well spent will require 
> substantial preparation.  As an example, I could see a virtual AVTEXT 
> interim being scheduled for mid-January at which Jonathan Lennox's 
> hierarchy draft could be given substantial time and an MMUSIC virtual 
> interim in the same time frame which could be devoted to one or two 
> important topics (e.g. BUNDLE, SCTP).
>
> If that is done, then the issues requiring additional intensive work 
> might be narrowed to the point where an in-person meeting in early 
> February might be able to finish some of them off, with others handled 
> another round of virtual interims two weeks later.
>
> Bernard "MARTINI was hell -- but it only lasted a year" Aboba
>
>
>
> On 12/10/2012 8:18 PM, Eric Rescorla wrote:
>
>     I'm not expertenough at the RAI WG mechanics to know what's
>     required, so perhaps
>
>     this requires making this also an MMUSIC interim or a mini-RAI
>     interim,
>
>     but if so that's what we should do, even if it means we need to
>     reschedule
>
>     or move it.
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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    <div class="moz-cite-prefix">On 12/11/2012 06:01 AM, Bernard Aboba
      wrote:<br>
    </div>
    <blockquote
      cite="mid:BLU405-EAS2665AFEC5A0D3FBF6070B6393480@phx.gbl"
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      <div class="WordSection1">
        <p class="MsoNormal"><span style="color:#1F497D">I do not think
            that a single meeting, virtual or actual, will suffice to
            get things back on track.&nbsp; We need a fundamental change of
            approach. </span></p>
      </div>
    </blockquote>
    I see the main purpose of meetings to serve as stopping points for
    discussions.<br>
    In order to be effective, the proposals for solutions have to be
    created before the meeting, and the meeting serves to accept, reject
    or modify the proposals.<br>
    <blockquote
      cite="mid:BLU405-EAS2665AFEC5A0D3FBF6070B6393480@phx.gbl"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal"><span style="color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D">The basic
            problem is that WebRTC work items have been dispersed
            between a number of WGs other than RTCWEB.&nbsp; Within these
            WGs, WebRTC is only a sideline. <o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
            <o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D">As a result,
            work items such as BUNDLE (which require non-trivial
            analysis and discussion) have found themselves a needle in a
            haystack of other items, getting a fraction of the attention
            that they need to move forward.&nbsp; <br>
          </span></p>
      </div>
    </blockquote>
    Indeed.<br>
    <blockquote
      cite="mid:BLU405-EAS2665AFEC5A0D3FBF6070B6393480@phx.gbl"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal"><span style="color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D">While
            allocating some substantial time at an in-person
            RTCWEB/MMUSIC interim can help, for that time to be well
            spent will require substantial preparation. &nbsp;As an example,
            I could see a virtual AVTEXT interim being scheduled for
            mid-January at which Jonathan Lennox&#8217;s hierarchy draft could
            be given substantial time and an MMUSIC virtual interim in
            the same time frame which could be devoted to one or two
            important topics (e.g. BUNDLE, SCTP).<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D">If that is
            done, then the issues requiring additional intensive work
            might be narrowed to the point where an in-person meeting in
            early February might be able to finish some of them off,
            with others handled another round of virtual interims two
            weeks later. <o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D">Bernard
            &#8220;MARTINI was hell &#8211; but it only lasted a year&#8221; Aboba<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><br>
          <br>
          On 12/10/2012 8:18 PM, Eric Rescorla wrote:<o:p></o:p></p>
        <blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
          <div>
            <p class="MsoNormal">I'm not expert<span
                style="color:#1F497D"> </span>enough at the RAI WG
              mechanics to know what's required, so perhaps<span
                style="color:#1F497D"><o:p></o:p></span></p>
          </div>
          <div>
            <p class="MsoNormal">this requires making this also an
              MMUSIC interim or a mini-RAI interim,<o:p></o:p></p>
          </div>
          <div>
            <p class="MsoNormal">but if so that's what we should do,
              even if it means we need to reschedule<o:p></o:p></p>
          </div>
          <div>
            <p class="MsoNormal">or move it.<o:p></o:p></p>
          </div>
          <div>
            <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
          </div>
        </blockquote>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
      </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>
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From pkyzivat@alum.mit.edu  Tue Dec 11 09:05:53 2012
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Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
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On 12/11/12 5:34 AM, Markus.Isomaki@nokia.com wrote:
> Hi Sal,
>
> Salvatore Loreto wrote:
>>
>> based on the current stack the SCTP traffic is the only traffic that runs directly
>> over the DTLS stack.
>> 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.
>>
>> 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.
>>
>
> It's a bit unclear to me what the benefit of this would be. SCTP itself allows multiple streams to be transported in parallel within a single SCTP association. So, in what type of situations would we need multiple parallel SCTP associations? I thought the main point of SCTP is that it provides the parallelism and multiplexing by itself.

It may not provide any benefit for WebRTC. To provide some benefit it 
would need to be exposed in the WebRTC API.

If every stream/channel within the SCTP session is always negotiated in 
SDP then there is less likely to be benefit in general.

But if (as planned, I think) the streams/channels within the SCTP 
association can be managed dynamically, then having two associations 
means you could have two separate pools of streams to dynamically 
manage. This could be helpful if you have multiple components within 
your application that want their own control over dynamic data streams.

Whether the multiple SCTP associations are over the same DTLS connection 
or different ones is an independent question.

> In other words: I do understand the desire to multiplex RTP and SCTP within the same UDP flow, so we can for instance reduce the number of needed NAT/FW bindings. I also understand why multiple independent non-HOL blocking streams within SCTP are useful for applications. But I don't yet understand the additional benefit of multiple parallel SCTP associations. No doubt it can be technically done, but to what purposes?

IIUC this is a mechanism that is already latent in the SCTP. It is a 
matter of whether to expose the capability. Not exposing it just means 
that the latent capability isn't being used - everyone will be operating 
on a default SCTP port.

Exposing it needs to occur at two levels, and the decision whether to do 
so or not can be made separately:

IIUC the SCTP *protocol* already has this capability.

The proposed SDP currently provides no way to specify the SCTP port. If 
it is to continue that way then it minimally needs to specify which port 
value is to be used in the protocol. (Otherwise we can have 
interoperability failures.)

It should be very simple to provide an attribute to explicitly specify 
the SCTP port, while still specifying a default value if not specified. 
By itself this would not be very useful if you could only specify one.

More complex would be to specify use of multiple SCTP ports over the 
same DTLS connection. That starts to look like a bundle problem again 
because they share the same address and UDP port, while having a number 
of attributes to specify about each.

If we want to keep this simple for now, then I suggest that 
draft-ietf-mmusic-sctp-sdp simply specify that implementations of this 
draft always use SCTP port zero, and that use of other ports is out of 
scope.

	Thanks,
	Paul


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Subject: Re: [MMUSIC] [rtcweb] SDP work needed for WebRTC stuff
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On 11 December 2012 08:51, Harald Alvestrand <harald@alvestrand.no> wrote:
> In order to be effective, the proposals for solutions have to be created
> before the meeting, and the meeting serves to accept, reject or modify the
> proposals.

That's true, but it's also true that scheduled meetings tend to act as
an effective form of stimulus.  I really don't want more meetings, but
this could be a contributing factor.  I don't think that it's the main
one.

At the current time, the main reason for lack of progress is the
impotence of the rtcweb working group.

From dwing@cisco.com  Tue Dec 11 10:53:32 2012
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From: "Dan Wing" <dwing@cisco.com>
To: "'Paul Kyzivat'" <pkyzivat@alum.mit.edu>, <mmusic@ietf.org>
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>
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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 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.

> 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).

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



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From: "Dan Wing" <dwing@cisco.com>
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References: <5093A2C9.9040001@alvestrand.no> <50B9E3ED.6010604@ericsson.com>	<50BA19F9.4040701@alvestrand.no> <50BD04D2.7090207@alum.mit.edu>	<50C6F800.1080500@ericsson.com> <E44893DD4E290745BB608EB23FDDB76232C004@008-AM1MPN1-041.mgdnok.nokia.com>
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Cc: randell@jesup.org
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 Markus.Isomaki@nokia.com
> Sent: Tuesday, December 11, 2012 2:34 AM
> To: salvatore.loreto@ericsson.com; mmusic@ietf.org
> Cc: randell@jesup.org
> Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
> 
> Hi Sal,
> 
> Salvatore Loreto wrote:
> >
> >based on the current stack the SCTP traffic is the only traffic that
> >runs directly over the DTLS stack.
> >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.
> >
> >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.
> >
> 
> It's a bit unclear to me what the benefit of this would be. SCTP itself
> allows multiple streams to be transported in parallel within a single
> SCTP association. So, in what type of situations would we need multiple
> parallel SCTP associations? I thought the main point of SCTP is that it
> provides the parallelism and multiplexing by itself.
> 
> In other words: I do understand the desire to multiplex RTP and SCTP
> within the same UDP flow, so we can for instance reduce the number of
> needed NAT/FW bindings.

Sending bulk data (SCTP) and interactive audio/video over the same port
will break prioritization of the audio/video flow above the bulk data.

> I also understand why multiple independent non-
> HOL blocking streams within SCTP are useful for applications. But I
> don't yet understand the additional benefit of multiple parallel SCTP
> associations. No doubt it can be technically done, but to what purposes?

One example:  transferring a big file while, at the same time, doing 
screen sharing or an instant message chat.

-d



From pkyzivat@alum.mit.edu  Tue Dec 11 11:17:38 2012
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Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
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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.)

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.

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?

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
>
>
>


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From: "Dan Wing" <dwing@cisco.com>
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Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
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> -----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.

> 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.

-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
> >
> >
> >


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Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
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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 Markus.Isomaki@nokia.com
>> Sent: Tuesday, December 11, 2012 2:34 AM
>> To: salvatore.loreto@ericsson.com; mmusic@ietf.org
>> Cc: randell@jesup.org
>> Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
>>
>> Hi Sal,
>>
>> Salvatore Loreto wrote:
>>>
>>> based on the current stack the SCTP traffic is the only traffic that
>>> runs directly over the DTLS stack.
>>> 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.
>>>
>>> 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.
>>>
>>
>> It's a bit unclear to me what the benefit of this would be. SCTP itself
>> allows multiple streams to be transported in parallel within a single
>> SCTP association. So, in what type of situations would we need multiple
>> parallel SCTP associations? I thought the main point of SCTP is that it
>> provides the parallelism and multiplexing by itself.
>>
>> In other words: I do understand the desire to multiplex RTP and SCTP
>> within the same UDP flow, so we can for instance reduce the number of
>> needed NAT/FW bindings.
>
> Sending bulk data (SCTP) and interactive audio/video over the same port
> will break prioritization of the audio/video flow above the bulk data.

Yeah, probably. In theory you could use DSCP to mark the packets 
differently.

This indeed might be a reason to *not* multiplex the SCTP association 
for data channels with the media. Assuming we find a way to do that 
multiplexing, this would be an argument for making that multiplexing 
*optional*.

>> I also understand why multiple independent non-
>> HOL blocking streams within SCTP are useful for applications. But I
>> don't yet understand the additional benefit of multiple parallel SCTP
>> associations. No doubt it can be technically done, but to what purposes?
>
> One example:  transferring a big file while, at the same time, doing
> screen sharing or an instant message chat.

When using DTLS/SCTP this is an argument for using SCTP associations 
over separate DTLS connections. Two SCTP associations (differing by SCTP 
port) over the same DTLS connection wouldn't solve this problem.

	Thanks,
	Paul


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Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
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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.

>> 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.

	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
>>>
>>>
>>>
>
>


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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> <50C79573.1000007@alum.mit.edu>
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Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
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> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
> 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

Agreed.

> 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 should work fine with
user-mode SCTP over UDP.  It causes the SCTP stack to send that
additional (meta) data to the remote peer.

> >> 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.

Yes.

But the greater question is how does the industry want such mobility
to function when the mobility can be instantiated or disabled at
different layers of this deep X-over-Y stack of tunneled protocols.

That decision has a pretty significant impact on SDP, especially
if we want to follow DTLS-SRTP's lead of using "UDP/TLS/RTP/SAVP"
of enumerating every layer in the tunnels.

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


From randell-ietf@jesup.org  Tue Dec 11 14:50:01 2012
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Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
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On 12/11/2012 1:53 PM, Dan Wing wrote:
>>
>> 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.

Right.  We're defining SDP for SCTP here, not *just* for WebRTC. (and 
direct use of rtcweb without WebRTC could allow for multiple 
associations in theory).

>> It's a bit unclear to me what the benefit of this would be. SCTP itself
>> allows multiple streams to be transported in parallel within a single
>> SCTP association. So, in what type of situations would we need multiple
>> parallel SCTP associations? I thought the main point of SCTP is that it
>> provides the parallelism and multiplexing by itself.
>>
>> In other words: I do understand the desire to multiplex RTP and SCTP
>> within the same UDP flow, so we can for instance reduce the number of
>> needed NAT/FW bindings.
> Sending bulk data (SCTP) and interactive audio/video over the same port
> will break prioritization of the audio/video flow above the bulk data.

For the general case, you could provide different DSCP markings, etc, 
per Dan's suggestion.

And also you could run different protocols on top of SCTP, each in an 
association.  One could run DataChannels, another could run BFCP, etc.  
That way they don't conflict (in how they allocate streams, etc).

>> I also understand why multiple independent non-
>> HOL blocking streams within SCTP are useful for applications. But I
>> don't yet understand the additional benefit of multiple parallel SCTP
>> associations. No doubt it can be technically done, but to what purposes?
> One example:  transferring a big file while, at the same time, doing
> screen sharing or an instant message chat.

Right - until we create a new draft, there are buffer-hogging issues 
with large transfers (transfer a 270MB file, like I did at IETF, and all 
other streams are starved until the transfer finishes). Separate 
associations would also deal with the case without needing draft.  
(We'll want the draft for rtcweb).  They might use different congestion 
priorities, etc.


-- 
Randell Jesup
randell-ietf@jesup.org


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Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
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On 12/11/2012 10:53 AM, Dan Wing wrote:

> Sending bulk data (SCTP) and interactive audio/video over the same port
> will break prioritization of the audio/video flow above the bulk data.

In Lyon, we made the assertion that a browser network stack MUST be able 
to maintain a common congestion state that is shared between the various 
streams (1 or more RTP, at least 1 SCTP, etc.) and which is used to 
prioritize the dispatching of packets onto the network appropriate 
across at least 3 (I recommended 4) priority levels. This is *in 
addition* to the idea that these 3 (or 4) levels are also diffserv 
marked in a way that the network can apply different treatments to them.

(I recommended at least 4 levels to cover 
data-that-is-more-important-than-anything, audio, video, and 
data-that-is-less-important-than-anything)

>
>> I also understand why multiple independent non-
>> HOL blocking streams within SCTP are useful for applications. But I
>> don't yet understand the additional benefit of multiple parallel SCTP
>> associations. No doubt it can be technically done, but to what purposes?
> One example:  transferring a big file while, at the same time, doing
> screen sharing or an instant message chat.

The fact that SCTP also suffers from a head-of-line blocking issue 
because its flow control is per-association not per-flow (ref: 
http://www.ietf.org/mail-archive/web/rtcweb/current/msg02787.html ) does 
suggest that until that is resolved one might want to use >1 SCTP 
association between peers as well.

Matthew Kaufman

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Subject: Re: [MMUSIC] SDP work needed for WebRTC stuff
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On 12/8/2012 3:21 PM, Eric Rescorla wrote:
> +rtcweb
>
> On Sat, Dec 8, 2012 at 2:41 PM, Cullen Jennings (fluffy) 
> <fluffy@cisco.com <mailto:fluffy@cisco.com>> wrote:
>
>
>     I was looking over everything that needs to be completed to finish
>     a fist cut of the WebRTC related work. There are a handful of big
>     SDP problems that are currently blocking some of the WebRTC work
>     and I'd like to figure out how to make some progress on them.
>
>     Let me loosely characterize them as
>
>     1) If we have several video streams, how do theses map up to 1 or
>     more m lines.
>
>     2) if we are doing port multiplexing, what does the SDP look like
>     (the bundle problem)
>
>     3) How do we map the RTCWeb track and stream label concepts to
>     identifiers in SDP
>
>     3) SDP for application running over SCTP/DTLS
>
>
>     I don't want to speak for all the various chairs but I am under
>     the impression that most of chairs of related groups in W3C and
>     IETF believe these are issues that need to be resolved primarily
>     in the MMUSIC WG and that they impact both WebRTC and CLUE as well
>     as the general long term use of SDP in SIP and other protocols.
>
>     I'd like to get some discussion going on how we can make some
>     progress on these. I don't think we are going to solve these in 20
>     minutes of discussion at an IETF meeting so I think we probably
>     need some interim (virtual or face to face) to sort this out.
>
>     Thoughts?
>
>
> Wow, I'm totally confused here.
>
> I had assumed that the SDP-related issues were going to be the main
> topics at the WebRTC/RTCWEB interim in January. Is that not the case?
>
> IMO the lack of clarity around how to encode various media
> configurations into SDP is the major thing blocking progress here. In
> particular, Firefox has opted not to implement multiplexing of media
> streams over the same transport flow (whether of the bundle or
> multiple m-line variety) until the SDP for it is well-defined. The
> same thing applies to the question of how to map multiple m-lines to
> incoming MediaStreams/Tracks.
>
> We really need to cover these issues in the interim.
>
> -Ekr
>
>

We only need to cover these issues soon if RTCWEB continues to insist 
that SDP is a good idea for an API *and* that changes to SDP are 
necessary in order for RTCWEB to then be successful.

I would posit that if SDP as used in (and referenced from) RFC3264 was 
the appropriate API surface for RTCWEB then NO ADDITIONAL WORK in mmusic 
would be required in order to ship RTCWEB 1.0.

Matthew Kaufman


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    <div class="moz-cite-prefix">On 12/8/2012 3:21 PM, Eric Rescorla
      wrote:<br>
    </div>
    <blockquote
cite="mid:CABcZeBPfhKoDvHceNLo_Ab9rRDD0V-mMok4H4y_FitYcBZ7vTA@mail.gmail.com"
      type="cite">+rtcweb
      <div><br>
      </div>
      <div>On Sat, Dec 8, 2012 at 2:41 PM, Cullen Jennings (fluffy) <span
          dir="ltr">&lt;<a moz-do-not-send="true"
            href="mailto:fluffy@cisco.com" target="_blank">fluffy@cisco.com</a>&gt;</span>
        wrote:<br>
        <div class="gmail_quote">
          <blockquote class="gmail_quote" style="margin:0 0 0
            .8ex;border-left:1px #ccc solid;padding-left:1ex">
            <br>
            I was looking over everything that needs to be completed to
            finish a fist cut of the WebRTC related work. There are a
            handful of big SDP problems that are currently blocking some
            of the WebRTC work and I'd like to figure out how to make
            some progress on them.<br>
            <br>
            Let me loosely characterize them as<br>
            <br>
            1) If we have several video streams, how do theses map up to
            1 or more m lines.<br>
            <br>
            2) if we are doing port multiplexing, what does the SDP look
            like (the bundle problem)<br>
            <br>
            3) How do we map the RTCWeb track and stream label concepts
            to identifiers in SDP<br>
            <br>
            3) SDP for application running over SCTP/DTLS<br>
          </blockquote>
          <div><br>
          </div>
          <blockquote class="gmail_quote" style="margin:0 0 0
            .8ex;border-left:1px #ccc solid;padding-left:1ex">
            I don't want to speak for all the various chairs but I am
            under the impression that most of chairs of related groups
            in W3C and IETF believe these are issues that need to be
            resolved primarily in the MMUSIC WG and that they impact
            both WebRTC and CLUE as well as the general long term use of
            SDP in SIP and other protocols.<br>
            <br>
            I'd like to get some discussion going on how we can make
            some progress on these. I don't think we are going to solve
            these in 20 minutes of discussion at an IETF meeting so I
            think we probably need some interim (virtual or face to
            face) to sort this out.<br>
            <br>
            Thoughts?<br>
          </blockquote>
          <div><br>
          </div>
          <div class="gmail_quote">Wow, I'm totally confused here.</div>
          <div class="gmail_quote"><br>
          </div>
          <div class="gmail_quote">I had assumed that the SDP-related
            issues were going to be the main</div>
          <div class="gmail_quote">topics at the WebRTC/RTCWEB interim
            in January. Is that not the case?</div>
          <div class="gmail_quote"><br>
          </div>
          <div class="gmail_quote">IMO the lack of clarity around how to
            encode various media</div>
          <div class="gmail_quote">configurations into SDP is the major
            thing blocking progress here. In</div>
          <div class="gmail_quote">particular, Firefox has opted not to
            implement multiplexing of media</div>
          <div class="gmail_quote">
            streams over the same transport flow (whether of the bundle
            or</div>
          <div class="gmail_quote">multiple m-line variety) until the
            SDP for it is well-defined. The</div>
          <div class="gmail_quote">same thing applies to the question of
            how to map multiple m-lines to</div>
          <div class="gmail_quote">incoming MediaStreams/Tracks.</div>
          <div class="gmail_quote"><br>
          </div>
          <div class="gmail_quote">We really need to cover these issues
            in the interim.</div>
          <div class="gmail_quote"><br>
          </div>
          <div class="gmail_quote">
            -Ekr</div>
          <div><br>
          </div>
          <br>
        </div>
      </div>
    </blockquote>
    <br>
    We only need to cover these issues soon if RTCWEB continues to
    insist that SDP is a good idea for an API *and* that changes to SDP
    are necessary in order for RTCWEB to then be successful.<br>
    <br>
    I would posit that if SDP as used in (and referenced from) RFC3264
    was the appropriate API surface for RTCWEB then NO ADDITIONAL WORK
    in mmusic would be required in order to ship RTCWEB 1.0.<br>
    <br>
    Matthew Kaufman<br>
    <br>
  </body>
</html>

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Subject: Re: [MMUSIC] BUNDLE: Session bandwidth (Re: comments on	draft-ietf-mmusic-sdp-bundle-negotiation-01)
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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.

Cheers

Magnus Westerlund

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From magnus.westerlund@ericsson.com  Wed Dec 12 02:37:28 2012
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Date: Wed, 12 Dec 2012 11:37:24 +0100
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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References: <C5E08FE080ACFD4DAE31E4BDBF944EB1132A380B@xmb-aln-x02.cisco.com>	<CABcZeBPfhKoDvHceNLo_Ab9rRDD0V-mMok4H4y_FitYcBZ7vTA@mail.gmail.com>	<C5E08FE080ACFD4DAE31E4BDBF944EB1132A5D53@xmb-aln-x02.cisco.com>	<EDC0A1AE77C57744B664A310A0B23AE20D74247E9C@FRMRSSXCHMBSC3.dc-m.alcatel-lucent.com>	<CA+9kkMASrAGvAEY7aU5H1CjVqBj0bQfkXr3XdBmO8M-FQNGS0Q@mail.gmail.com>	<CABcZeBMjREPNVAg3UvEdink89qk+bwmbARK431wLLwv=gK6yhg@mail.gmail.com> <50C6A3A5.3060009@reavy.org> <BLU405-EAS2665AFEC5A0D3FBF6070B6393480@phx.gbl>
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Cc: "'Cullen Jennings \(fluffy\)'" <fluffy@cisco.com>, rtcweb@ietf.org, 'mmusic WG' <mmusic@ietf.org>
Subject: Re: [MMUSIC] [rtcweb]  SDP work needed for WebRTC stuff
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Hi,

My Personal view on this topics.

On 2012-12-11 06:01, Bernard Aboba wrote:
> I do not think that a single meeting, virtual or actual, will suffice to
> get things back on track.  We need a fundamental change of approach.
> 

Fully agreed. One of the issues, is that not a single individual of use
across 5-6 WG have a complete picture of the issues we are stepping into
with the fundamental issues that are in the center of this. This will
require multiple individuals having knowledge on different aspects to
fill in details and educate the others on the diverse issues.


>  
> 
> The basic problem is that WebRTC work items have been dispersed between
> a number of WGs other than RTCWEB.  Within these WGs, WebRTC is only a
> sideline.

I am not certain that is the fundamental issue here. I think the very
basic problem is that we are attempting for fork lift SDP into a clear
specification supporting multi-stream RTP sessions and dealing with the
diverse landscape of relationships that exists in multi-media
applications between a large number of terminologies that include
concepts such as:

- Multi-party
- Multiple media sources, i.e. camera microphones
- Media Mixes or Compositions
- Synchronization Contexts
- End-point concepts, virtual, distributed, etc.
- Diverse signalling models
- Simulcast
- Layered Encoding
- Transport robustification, i.e RTP Retransmission, Duplications, FEC.
- Different transports, aggregated over single unicast, multiple unicast
flows, multicast
- Different security mechanisms

This coupled with that the people putting in requirements on this are
working on different type of systems or applications, RTCWEB/WebRTC and
CLUE just being two. We also have existing users of RTP that made
different assumptions in extensions and signalling.

It was very evident in RTCWEB and AVTEXT that we are lacking common
terminology to even discuss the above and what it means in different
contexts.

Then we haven't even discussed what happens the minute you drag the word
"Legacy" into any of the above debates for actual solutions. But now I
am getting ahead of myself. I think the most important is to build a
good description of the different functionalities and their
relationships that we today believe are necessary to deal with currently
and in the near future.

> 
>                                                                                                                            
> 
> 
> As a result, work items such as BUNDLE (which require non-trivial
> analysis and discussion) have found themselves a needle in a haystack of
> other items, getting a fraction of the attention that they need to move
> forward.  

Yes, bundle is something that covers parts of the above and clearly
related. I, however think more close to the core of the issues is the
multiple stream issues and how we deal with relationships. However, I
think failing to take the related topics into account and think that
this will be quickly resolved will only result in generating even more
mess.

I also believe that in the end we are going to have to take some though
decisions when it comes how to deal with existing definitions. Either
replace them or basically deprecate some behaviors.

> 
> While allocating some substantial time at an in-person RTCWEB/MMUSIC
> interim can help, for that time to be well spent will require
> substantial preparation.  As an example, I could see a virtual AVTEXT
> interim being scheduled for mid-January at which Jonathan Lennoxs
> hierarchy draft could be given substantial time and an MMUSIC virtual
> interim in the same time frame which could be devoted to one or two
> important topics (e.g. BUNDLE, SCTP).
> 

Fully agreed. This takes preparations and hard work.

Cheers

Magnus Westerlund

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Multimedia Technologies, Ericsson Research EAB/TVM
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From gonzalo.camarillo@ericsson.com  Wed Dec 12 03:29:45 2012
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Date: Wed, 12 Dec 2012 13:29:42 +0200
From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: Hadriel Kaplan <HKaplan@acmepacket.com>
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Cc: "mmusic@ietf.org \(E-mail\)" <mmusic@ietf.org>, "draft-ietf-mmusic-media-loopback@tools.ietf.org" <draft-ietf-mmusic-media-loopback@tools.ietf.org>
Subject: Re: [MMUSIC] NEED WG CONSENSUS: media-loopback is all-or-nothing
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Hi Hadriel,

the seems to be consensus around the "NO" path. Could you please revise
the draft accordingly so that we can also clear the last discuss?

Thanks,

Gonzalo

On 20/11/2012 8:16 PM, Hadriel Kaplan wrote:
> Howdy,
> during IESG review of the media-loopback draft, a discussion arose regarding the language around indicating media-loopback support per media description (i.e., per m-line), in sections 3.1 and 3.2.
> 
> My interpretation/understanding was we were requiring ALL m-lines to indicate media-loopback; in other words it's an all-or-nothing in both the offer+answer, and either every media stream is looped back, or none are.  However the text currently doesn't make this clear, and it could be argued either way, with pro's/con's either way too; so I'd like the WG's consensus on whether it should in fact be an all-or-none approach, or not.
> 
> Please respond to this email by **Tuesday, December 4th** with a "YES" if the draft should mandate ALL media be looped-back, or "NO" if it should allow a hybrid approach of some streams being looped while others are not.
> 
> For reference, the draft is here:
> http://tools.ietf.org/html/draft-ietf-mmusic-media-loopback-24
> 
> -hadriel
> 
> 


From internet-drafts@ietf.org  Wed Dec 12 14:06:32 2012
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-media-capabilities-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) Media Capabilities Ne=
gotiation
	Author(s)       : Robert R Gilman
                          Roni Even
                          Flemming Andreasen
	Filename        : draft-ietf-mmusic-sdp-media-capabilities-16.txt
	Pages           : 64
	Date            : 2012-12-12

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-16

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


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


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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-miscellaneous-caps-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           : Miscellaneous Capabilities Negotiation in the Session De=
scription Protocol (SDP)
	Author(s)       : Miguel A. Garcia-Martin
                          Simo Veikkolainen
                          Robert R. Gilman
	Filename        : draft-ietf-mmusic-sdp-miscellaneous-caps-03.txt
	Pages           : 20
	Date            : 2012-12-12

Abstract:
   SDP has been extended with a capability negotiation mechanism
   framework that allows the endpoints to negotiate transport protocols
   and attributes.  This framework has been extended with a media
   capabilities negotiation mechanism that allows endpoints to negotiate
   additional media-related capabilities.  This negotiation is embedded
   into the widely-used SDP offer/answer procedures.

   This memo extends the SDP capability negotiation framework to allow
   endpoints to negotiate three additional SDP capabilities.  In
   particular, this memo provides a mechanism to negotiate bandwidth
   ('b=3D' line), connection data ('c=3D' line), and titles ('i=3D' line for
   each session or media).


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

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http://tools.ietf.org/html/draft-ietf-mmusic-sdp-miscellaneous-caps-03

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-03


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From dushyant2u@gmail.com  Thu Dec 13 02:42:44 2012
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From: Dushyant Joshi <dushyant2u@gmail.com>
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Subject: [MMUSIC] RTSP "alt-ports" are not mentioned in the RFC.
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Hello team,

I could see RTSP alt-ports are not defined in either RFC 2326 or RFC 6064

if you see IANA

http://www.iana.org/assignments/service-names-port-numbers/service-names-port-numbers.xml

we have defined port 8554 as the alternate RTSP port

rtsp            554/tcp    Real Time Stream Control Protocol
rtsp            554/udp    Real Time Stream Control Protocol
rtsp-alt        8554/tcp   RTSP Alternate (see port 554)
rtsp-alt        8554/udp   RTSP Alternate (see port 554)

The RTSP 2.0 specification is still an Internet Draft and could add text.
 can we add this as an informative mentioning that there do exist an
alternative registered port.

Regards,
Dushyant Joshi
91-9886886757

From magnus.westerlund@ericsson.com  Fri Dec 14 01:13:21 2012
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Cc: mmusic <mmusic@ietf.org>, draft-ietf-mmusic-rtsp-nat@tools.ietf.org
Subject: Re: [MMUSIC] draft-ietf-mmusic-rtsp-nat-14 review
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Hi,

Follow up on comments as I have edited proposals to resolve the issues.
I have the general set of changes on review with my co-authors. Hope to
be able to submit next week.


On 2012-12-06 16:03, Magnus Westerlund wrote:
> 
> On 2012-12-05 22:22, Ari Keranen wrote:

>>
>>
>>         If the server has a public IP address with a single candidate
>>         per media stream, component and address family, then the server
>>         may be configured to not initiate connectivity checks.
>>
>> This is the "ICE-RTSP", right? Maybe one (small) subsection defining
>> this "feature" would make sense. Now it's a bit trickled around in this
>> section, sec 4.8.1, 5.4, etc. For someone not familiar with ICE-RTSP, it
>> looks like you're talking about ICE lite.
> 
> Yes, this is definitely ICE-RTSP. It is still doing triggered checks.

However, I think you might have misinterpreted what ICE-RTSP really is.
It is the full usage of ICE as defined by this specification. The high
reachability configuration of the RTSP server is a variant of the normal
ICE-RTSP behavior.

I have tried to clarify this and created a special section for
discussing the differences, this include the text in the old 4.8.

5.  ICE-RTSP

   This Section discusses differences to the regular ICE usage defined
   in [RFC5245].  The basic for the modifications in the general
   procedures are in the clearer client/server roles that RTSP provides
   and how the RTSP Session establishment signalling occurs within RTSP
   compared to SIP/SDP Offer/Answer.

5.1.  ICE Features Not Required

   A number of ICE signalling features are not needed with RTSP and are
   discussed below.

5.1.1.  ICE-Lite

   The ICE-Lite attribute shall not be used in the context of RTSP.  The
   ICE specification describes two implementations of ICE: Full and
   Lite, where hosts that are not behind a NAT are allowed to implement
   only Lite.  For RTSP, the Lite implementation is insufficient because
   it does not cause the media server to send a connectivity check,
   which is used to protect against making the RTSP server a denial of
   service tool.

5.1.2.  ICE-Mismatch

   The ice-mismatch parameter indicates that the offer arrived with a
   default destination for a media component that didn't have a
   corresponding candidate attribute.  This is not needed for RTSP as
   the ICE based lower layer transport specification either is supported
   or another alternative transport is used.  This is always explicitly
   indicated in the SETUP request and response.

5.1.3.  ICE Remote Candidate Transport Header Parameter

   The Remote candidate attribute is not needed for RTSP for the
   following reasons.  Each SETUP results in an independent ICE
   processing chain which either fails or results in promoting a single
   candidate pair to usage.  If a new SETUP request for the same media
   is sent, this needs to use a new username fragment and password to
   avoid any race conditions or uncertainty about which round of
   processing the STUN requests relate to.

5.2.  High-Reachability Configuration

   ICE-RTSP contains one variant for RTSP Servers that are not behind
   NATs, i.e. are highly reachable by the clients.  Similar to ICE-Lite
   this allows for some reductions in the servers burden.  However, due
   to the need to still verify that the client is actually present where
   it claims the server must also initiate binding requests and await
   binding responses.  The reduction for the high-reachability
   configuration of ICE-RTSP is that they don't need to initiate its own
   checks, and instead rely on triggered checks for verification.  This
   also removes a denial of service threat where a RTSP SETUP request
   will trigger large amount of STUN connectivity checks towards
   provided candidate addresses.



>>
>> 5.12.  Re-SETUP
>>
>>    If the client decides to change any parameters related to the media
>>    stream setup
>>
>> Could clarify (again) here that we talk about ICE parameters.
> 
> Yes, definitely. I actually had to think a long time if this was only
> ICE parameters or also RTSP parameters. But, it is so unnecessary to
> restart ICE for most RTSP parameter changes.

When looking at this again, I though it unclear enough that I have
attempted a bit of rewording. I propose the section should read like this:

6.12.  Re-SETUP

   A client that decides to change any parameters related to the media
   stream setup it will send a new SETUP request.  In this new SETUP
   request the client MAY include a new different username fragment and
   password to use in the ICE processing.  New username and password
   SHALL cause the ICE processing to start from the beginning again,
   i.e. an ICE restart (Section 9.1.1.1 of [RFC5245]).  The client SHALL
   in case of ICE restart gather candidates and include the candidates
   in the transport specification for D-ICE.

   ICE restarts may be triggered due to changes of clients or servers
   attachment to the network, i.e. changes to the media streams
   destination or source address or port.  Most RTSP parameter changes
   would not require an ICE restart, instead existing mechanisms in RTSP
   for indicating from where in the RTP stream they apply should be
   used.  These include performing a pause prior to the parameter change
   and then resume or if server supports in do SETUP during PLAY state,
   and thus use RTP-Info header (Section 18.43 of
   [I-D.ietf-mmusic-rfc2326bis]) to indicate from where in the media
   stream the change apply.

   The server SHALL support SETUP requests in PLAY state, as long as the
   SETUP changes only the ICE parameters, which are: ICE-Password, ICE-
   ufrag and the content of ICE candidates.

   If the RTSP session is in playing state at the time of sending the
   SETUP request requiring ICE restart, then the ICE connectivity checks
   SHALL use Regular nomination.  Any ongoing media delivery continues
   on the previously nominated candidate pairs until the new pairs have
   been nominated for the individual candidate.  Once the nomination of
   the new candidate pair has completed, all unused candidates may be
   released.

Cheers

Magnus Westerlund

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From magnus.westerlund@ericsson.com  Fri Dec 14 01:13:23 2012
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Subject: Re: [MMUSIC] WGLC for draft-ietf-mmusic-rtsp-nat-14
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Hi,

In the editing of Ari's comments I found the following issue that I have
attempted to address:

Section 4.2:
   <component-id>:  identifies the specific component of the media
      stream for which this is a candidate and is a positive integer
      between 1 and 256.  It MUST start at 1 and MUST increment by 1 for
      each component of a particular candidate.  For media streams based
      on RTP, candidates for the actual RTP media MUST have a component
      ID of 1, and candidates for RTCP MUST have a component ID of 2.

The last sentence MUST is not compatible with RTP/RTCP multiplexing
using RFC 5761 as defined by Section 7. Thus I have reworded it to:

For media streams based on RTP, candidates for the actual RTP media MUST
have a component ID of 1, and candidates for RTCP MUST have a component
ID of 2 unless RTP and RTCP Multiplexing (Section 7) is used when the
second component is omitted and RTP and RTCP is both transported over
the first component.


Feedback on this change would be appreciated.

Cheers

Magnus



On 2012-11-20 22:55, Flemming Andreasen wrote:
> This is to announce a 2 week Working Group Last Call for
> 
>     draft-ietf-mmusic-rtsp-nat-14
> 
> as Proposed Standard. Please review and provide any comments you may
> have on the document by December 5, 2012. Comments should be sent to the
> document authors and the MMUSIC WG list. If you review the document but
> do not have any comments, please send a note to that effect as well.
> 
> 
> Thanks
> 
>        Flemming
> 
> 
> Draft Info:
> 
>  This draft is a work item of the Multiparty Multimedia Session Control Working Group of the IETF.
> 
> 	Title           : A Network Address Translator (NAT) Traversal mechanism for media controlled by Real-Time Streaming Protocol (RTSP)
> 	Author(s)       : Jeff Goldberg
>                           Magnus Westerlund
>                           Thomas Zeng
> 	Filename        : draft-ietf-mmusic-rtsp-nat-14.txt
> 	Pages           : 31
> 	Date            : 2012-11-15
> 
> Abstract:
>    This document defines a solution for Network Address Translation
>    (NAT) traversal for datagram based media streams setup and controlled
>    with Real-time Streaming Protocol version 2 (RTSP 2.0).  It uses
>    Interactive Connectivity Establishment (ICE) adapted to use RTSP as a
>    signalling channel, defining the necessary extra RTSP extensions and
>    procedures.
> 
> 
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-mmusic-rtsp-nat
> 
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-mmusic-rtsp-nat-14
> 
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-rtsp-nat-14
> 
> 
> 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
> 


-- 

Magnus Westerlund

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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: [MMUSIC] BUNDLE: Session bandwidth (Re: comments on draft-ietf-mmusic-sdp-bundle-negotiation-01)
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Subject: Re: [MMUSIC] BUNDLE: Session bandwidth (Re: comments on	draft-ietf-mmusic-sdp-bundle-negotiation-01)
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> -----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)
>=20
> Hi,
>=20
> Going through old emails. One important comment:
>=20
> On 2012-11-05 21:04, Charles Eckel (eckelcu) wrote:
> >>
> >> Last time I looked b=3DCT was scoped to a whole conference ("supplied =
for
> >> the session"), while b=3DAS was scoped to a single RTP session. But I =
may
> >> be misinterpreting RFC 4566 section 5.8.
> >
> > I recommend using b=3DTIAS, 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=3DTIAS/AS values from the
> bundled media lines and using that as that bandwidth for the bundle even =
if
> it exceeds the b=3DTIAS/AS/CT value from the session level.
> >
>=20
> 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 will=
ing to receive. It does not say anything about what the declarer is going t=
o send. The same is true for AS and CT.=20
If we want something that indicates the maximum bandwidth the declarer is p=
lanning 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 vi=
olating limits declared by the corresponding b=3DTIAS/AS/CT attributes stil=
l applies.

Cheers,
Charles


> Cheers
>=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
> ----------------------------------------------------------------------


From dwing@cisco.com  Fri Dec 14 14:29:41 2012
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From: "Dan Wing" <dwing@cisco.com>
To: <mmusic@ietf.org>
Date: Fri, 14 Dec 2012 14:29:35 -0800
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Cc: jo@comnet.tkk.fi, draft-ietf-xrblock-rtcp-xr-discard-rle-metrics@tools.ietf.org, varun@comnet.tkk.fi
Subject: [MMUSIC] draft-ietf-xrblock-rtcp-xr-discard-rle-metrics-04
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This is my SDP Directorate review of
draft-ietf-xrblock-rtcp-xr-discard-rle-metrics-04.

Section 6 of draft-ietf-xrblock-rtcp-xr-discard-rle-metrics-04 left me
a bit puzzled.  It was not clear until the very end of that section
that including the defined SDP lines was entirely optional (MAY).
This does not align with RFC3611 (RTCP-XR), which says:
  "If a participant receives an SDP offer and understands the
   "rtcp-xr" attribute but does not wish to implement XR functionality
   offered, its answer SHOULD include an "rtcp-xr" attribute without
   parameters. By doing so, the party declares that, at a minimum, is
   capable of understanding the signaling."

I would like the authors to consider:

  * A shorter SDP section, something like
    http://tools.ietf.org/html/rfc5725#section-4

  * adding an explanation of the benefits (or harm) of including the
    new discard-rle and discard-bytes formats on the a=rtcp-xr line,
    so that implementers can weigh the benefit (or harm) of including
    one or both of those new formats on the a=rtcp-xr line.

  * align the document with RFC3611 (RTCP-XR), especially that the new
    discard-rle and discard-bytes are negotiated in SDP when doing
    Offer/Answer, or describe in the document why RFC3611's
    Offer/Answer model is inappropriate for discard-rle and for
    discard-bytes.

-d



From harald@alvestrand.no  Mon Dec 17 01:22:45 2012
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Subject: Re: [MMUSIC] RTSP "alt-ports" are not mentioned in the RFC.
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On 12/13/2012 11:42 AM, Dushyant Joshi wrote:
> Hello team,
>
> I could see RTSP alt-ports are not defined in either RFC 2326 or RFC 6064
What is the point of defining an RTSP alt-port?

Usually, in port numbers, two is worse than one; one should either have 
a single port or have protocol/uri syntax to describe using any port.

>
> if you see IANA
>
> http://www.iana.org/assignments/service-names-port-numbers/service-names-port-numbers.xml
>
> we have defined port 8554 as the alternate RTSP port
>
> rtsp            554/tcp    Real Time Stream Control Protocol
> rtsp            554/udp    Real Time Stream Control Protocol
> rtsp-alt        8554/tcp   RTSP Alternate (see port 554)
> rtsp-alt        8554/udp   RTSP Alternate (see port 554)
>
> The RTSP 2.0 specification is still an Internet Draft and could add text.
>   can we add this as an informative mentioning that there do exist an
> alternative registered port.
>
> Regards,
> Dushyant Joshi
> 91-9886886757
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From magnus.westerlund@ericsson.com  Mon Dec 17 01:31:33 2012
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Cc: "dushyant2u@gmail.com >> Dushyant Joshi" <dushyant2u@gmail.com>, mmusic@ietf.org
Subject: Re: [MMUSIC] RTSP "alt-ports" are not mentioned in the RFC.
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On 2012-12-17 10:22, Harald Alvestrand wrote:
> On 12/13/2012 11:42 AM, Dushyant Joshi wrote:
>> Hello team,
>>
>> I could see RTSP alt-ports are not defined in either RFC 2326 or RFC 6064
> What is the point of defining an RTSP alt-port?

It appears to be old legacy. I assume the motivation was that it was
difficult to deploy on the <1024 port number and wanted an alternative
that still was prevented to be used by other services.

> 
> Usually, in port numbers, two is worse than one; one should either have
> a single port or have protocol/uri syntax to describe using any port.

The question is if this port number has any benefit at all. Any port
number for RTSP that isn't the default needs to be explicitly called out
in the URI anyway, thus I see little different between 8554 and 7892 for
example. If it is a matter of deploying security policies I would
believe that you need to coordinate that anyway. Or is this for the
cases of outbound policies where you want to say, general TCP isn't ok,
but RTSP is, but only to port 554 and 8554?

Still I think this is strange and I do question if even a informative
reference in the RTSP 2.0 spec is of benefit.

Cheers

Magnus

> 
>>
>> if you see IANA
>>
>> http://www.iana.org/assignments/service-names-port-numbers/service-names-port-numbers.xml
>>
>>
>> we have defined port 8554 as the alternate RTSP port
>>
>> rtsp            554/tcp    Real Time Stream Control Protocol
>> rtsp            554/udp    Real Time Stream Control Protocol
>> rtsp-alt        8554/tcp   RTSP Alternate (see port 554)
>> rtsp-alt        8554/udp   RTSP Alternate (see port 554)
>>
>> The RTSP 2.0 specification is still an Internet Draft and could add text.
>>   can we add this as an informative mentioning that there do exist an
>> alternative registered port.
>>
>> Regards,
>> Dushyant Joshi
>> 91-9886886757
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
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-- 

Magnus Westerlund

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Multimedia Technologies, Ericsson Research EAB/TVM
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Ericsson AB                | Phone  +46 10 7148287
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SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
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From ari.keranen@nomadiclab.com  Mon Dec 17 07:36:38 2012
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Cc: draft-ietf-mmusic-rtsp-nat@tools.ietf.org, mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] draft-ietf-mmusic-rtsp-nat-14 review
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Hi Magnus,

On 12/6/12 5:03 PM, Magnus Westerlund wrote:
> On 2012-12-05 22:22, Ari Keranen wrote:
>> 3.  Solution Overview
>>
>>          The client then installs the STUN servers on each of
>>          the local candidates transport addresses it has gathered
>>
>> I would rather say "The client then prepares to act as a STUN server on
>> each [...]"
>
> Sure, maybe "runs STUN servers" is better than both "installs" and
> "act". They are STUN server, but with limited utility. Act as STUN
> servers sounds like it is impersonating STUN, rather than actually being
> STUN servers.

"runs" sounds good to me.

>>
>>
>>          If the server has a public
>>          IP address, then a single candidate per address family
>>
>> In some cases (e.g., certain ISPs, I've heard) even "public" IP address
>> does not mean you're not behind a NAT. Maybe it's better to say here "If
>> the server is not behind a NAT". Same issue in step 8.
>
> Sure, maybe we need to define what we mean with not being behind a NAT.
> It is a matter of being reachable from the top anchorin address domain
> in regards to the clients.

Yes, that would be good.

[...]
>>     Aggressive nomination SHALL be used with RTSP.  This doesn't have the
>>     negative impact that it has in offer/answer as media playing only
>>     starts after issuing a PLAY request.
>>
>> But it does have the downside of possibly selecting a lower-priority
>> pair due to single dropped (e.g., due to congestion) connectivity check
>> message for a higher priority pair. That said, probably SHALL is
>> appropriate here, just wanted to make sure the implications are understood.
>
> Yes, but with ICE-RTSP you will retransmit the higher priority also,
> thus if that works and you get a response that alternative is nominated
> and will be used. But we probably should clarify what it means when
> multiple candidate pairs are nominated, and in worse case the server
> will use different paths for different media streams.

According to ICE RFC, if multiple pairs end up nominated due to 
aggressive nomination, the highest priority one will be used (you get 
some transient behavior).

But do you mean that you will keep retransmitting on a higher priority 
pair even if a lower priority pair has been (aggressively) selected and 
all check lists are in Completed state?


Cheers,
Ari

From ari.keranen@nomadiclab.com  Mon Dec 17 07:38:22 2012
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Cc: draft-ietf-mmusic-rtsp-nat@tools.ietf.org, mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] draft-ietf-mmusic-rtsp-nat-14 review
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Hi Magnus,

These changes look good to me. And please add forward-reference(s) to 
the ICE-RTSP section where applicable. Couple of nits inline.


Cheers,
Ari

On 12/14/12 11:13 AM, Magnus Westerlund wrote:
> On 2012-12-06 16:03, Magnus Westerlund wrote:
>> On 2012-12-05 22:22, Ari Keranen wrote:
>>>
>>>          If the server has a public IP address with a single candidate
>>>          per media stream, component and address family, then the server
>>>          may be configured to not initiate connectivity checks.
>>>
>>> This is the "ICE-RTSP", right? Maybe one (small) subsection defining
>>> this "feature" would make sense. Now it's a bit trickled around in this
>>> section, sec 4.8.1, 5.4, etc. For someone not familiar with ICE-RTSP, it
>>> looks like you're talking about ICE lite.
>>
>> Yes, this is definitely ICE-RTSP. It is still doing triggered checks.
>
> However, I think you might have misinterpreted what ICE-RTSP really is.
> It is the full usage of ICE as defined by this specification. The high
> reachability configuration of the RTSP server is a variant of the normal
> ICE-RTSP behavior.
>
> I have tried to clarify this and created a special section for
> discussing the differences, this include the text in the old 4.8.

> 5.  ICE-RTSP
>
>     This Section discusses differences to the regular ICE usage defined
>     in [RFC5245].  The basic for the modifications in the general
>     procedures are in the clearer client/server roles that RTSP provides
>     and how the RTSP Session establishment signalling occurs within RTSP
>     compared to SIP/SDP Offer/Answer.
>
> 5.1.  ICE Features Not Required
>
>     A number of ICE signalling features are not needed with RTSP and are
>     discussed below.
>
> 5.1.1.  ICE-Lite
>
>     The ICE-Lite attribute shall not be used in the context of RTSP.  The
>     ICE specification describes two implementations of ICE: Full and
>     Lite, where hosts that are not behind a NAT are allowed to implement
>     only Lite.  For RTSP, the Lite implementation is insufficient because
>     it does not cause the media server to send a connectivity check,
>     which is used to protect against making the RTSP server a denial of
>     service tool.
>
> 5.1.2.  ICE-Mismatch
>
>     The ice-mismatch parameter indicates that the offer arrived with a
>     default destination for a media component that didn't have a
>     corresponding candidate attribute.  This is not needed for RTSP as
>     the ICE based lower layer transport specification either is supported
>     or another alternative transport is used.  This is always explicitly
>     indicated in the SETUP request and response.
>
> 5.1.3.  ICE Remote Candidate Transport Header Parameter
>
>     The Remote candidate attribute is not needed for RTSP for the
>     following reasons.  Each SETUP results in an independent ICE
>     processing chain which either fails or results in promoting a single
>     candidate pair to usage.  If a new SETUP request for the same media
>     is sent, this needs to use a new username fragment and password to
>     avoid any race conditions or uncertainty about which round of
>     processing the STUN requests relate to.
>
> 5.2.  High-Reachability Configuration
>
>     ICE-RTSP contains one variant for RTSP Servers that are not behind
>     NATs, i.e. are highly reachable by the clients.  Similar to ICE-Lite
>     this allows for some reductions in the servers burden.  However, due
>     to the need to still verify that the client is actually present where
>     it claims the server must also initiate binding requests and await
>     binding responses.  The reduction for the high-reachability
>     configuration of ICE-RTSP is that they don't need to initiate its own
>     checks, and instead rely on triggered checks for verification.  This
>     also removes a denial of service threat where a RTSP SETUP request
>     will trigger large amount of STUN connectivity checks towards
>     provided candidate addresses.
>
>
>
>>>
>>> 5.12.  Re-SETUP
>>>
>>>     If the client decides to change any parameters related to the media
>>>     stream setup
>>>
>>> Could clarify (again) here that we talk about ICE parameters.
>>
>> Yes, definitely. I actually had to think a long time if this was only
>> ICE parameters or also RTSP parameters. But, it is so unnecessary to
>> restart ICE for most RTSP parameter changes.
>
> When looking at this again, I though it unclear enough that I have
> attempted a bit of rewording. I propose the section should read like this:
>
> 6.12.  Re-SETUP
>
>     A client that decides to change any parameters related to the media
>     stream setup it will send a new SETUP request.  In this new SETUP

(remove "it")

>     request the client MAY include a new different username fragment and
>     password to use in the ICE processing.  New username and password
>     SHALL cause the ICE processing to start from the beginning again,
>     i.e. an ICE restart (Section 9.1.1.1 of [RFC5245]).  The client SHALL
>     in case of ICE restart gather candidates and include the candidates
>     in the transport specification for D-ICE.
>
>     ICE restarts may be triggered due to changes of clients or servers
>     attachment to the network, i.e. changes to the media streams
>     destination or source address or port.  Most RTSP parameter changes
>     would not require an ICE restart, instead existing mechanisms in RTSP
>     for indicating from where in the RTP stream they apply should be
>     used.  These include performing a pause prior to the parameter change
>     and then resume or if server supports in do SETUP during PLAY state,

"in do SETUP" ?

>     and thus use RTP-Info header (Section 18.43 of
>     [I-D.ietf-mmusic-rfc2326bis]) to indicate from where in the media
>     stream the change apply.
>
>     The server SHALL support SETUP requests in PLAY state, as long as the
>     SETUP changes only the ICE parameters, which are: ICE-Password, ICE-
>     ufrag and the content of ICE candidates.
>
>     If the RTSP session is in playing state at the time of sending the
>     SETUP request requiring ICE restart, then the ICE connectivity checks
>     SHALL use Regular nomination.  Any ongoing media delivery continues
>     on the previously nominated candidate pairs until the new pairs have
>     been nominated for the individual candidate.  Once the nomination of
>     the new candidate pair has completed, all unused candidates may be
>     released.


From jonathan@vidyo.com  Mon Dec 17 07:50:01 2012
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From: Jonathan Lennox <jonathan@vidyo.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>, "draft-ietf-xrblock-rtcp-xr-discard@tools.ietf.org" <draft-ietf-xrblock-rtcp-xr-discard@tools.ietf.org>
Date: Mon, 17 Dec 2012 10:49:41 -0500
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Subject: [MMUSIC] SDP Directorate review of draft-ietf-xrblock-rtcp-xr-discard
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--_000_C3759687E4991243A1A0BD44EAC823034DFB915B07BE235maillan_
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I have been asked to perform the SDP Directorate review for draft-ietf-xrbl=
ock-rtcp-xr-discard. I reviewed the -10 version of the document.

Syntactically, the SDP usage in this document seems fine.  It is a simple a=
nd correct usage of an extension point defined in RFC 3711.

However, the statement in Section 4 that "XR blocks MAY be used without exp=
licit signaling" is confusing, as it implies that the entire SDP section is=
 entirely optional.  This document should at least reference Section 5 of R=
FC 3711, which gives guidance as to when SDP signaling of the use of XR blo=
cks is recommended.

In the IANA considerations section, I echo Adam Roach's comments on the bur=
st-gap-lost draft: the full and proper names of the IANA registries should =
be spelled out.  These are the "RTP Control Protocol Extended Reports (RTCP=
 XR) Block Type Registry" in Section 5.1, and the "RTP Control Protocol Ext=
ended Reports (RTCP XR) Session Description Protocol (SDP) Parameters Regis=
try" in Section 5.2.

Additionally, the last sentence of section 4.2 is not correct English.  The=
 sentence

   For detailed usage in Offer/Answer for unilateral parameter, refer to se=
ction 5.2 of [RFC3611].

should be

   For detailed usage of Offer/Answer for unilateral parameters, refer to s=
ection 5.2 of [RFC3611].

--
Jonathan Lennox
jonathan@vidyo.com


--_000_C3759687E4991243A1A0BD44EAC823034DFB915B07BE235maillan_
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</o:shapelayout></xml><![endif]--></head><body lang=3DEN-US link=3Dblue vli=
nk=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal>I have been aske=
d to perform the SDP Directorate review for draft-ietf-xrblock-rtcp-xr-disc=
ard. I reviewed the -10 version of the document.<o:p></o:p></p><p class=3DM=
soNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Syntactically, the SDP u=
sage in this document seems fine.&nbsp; It is a simple and correct usage of=
 an extension point defined in RFC 3711.<o:p></o:p></p><p class=3DMsoNormal=
><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>However, the statement in Sectio=
n 4 that &quot;XR blocks MAY be used without explicit signaling&quot; is co=
nfusing, as it implies that the entire SDP section is entirely optional.&nb=
sp; This document should at least reference Section 5 of RFC 3711, which gi=
ves guidance as to when SDP signaling of the use of XR blocks is recommende=
d.<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNo=
rmal>In the IANA considerations section, I echo Adam Roach's comments on th=
e burst-gap-lost draft: the full and proper names of the IANA registries sh=
ould be spelled out.&nbsp; These are the &quot;RTP Control Protocol Extende=
d Reports (RTCP XR) Block Type Registry&quot; in Section 5.1, and the &quot=
;RTP Control Protocol Extended Reports (RTCP XR) Session Description Protoc=
ol (SDP) Parameters Registry&quot; in Section 5.2.<o:p></o:p></p><p class=
=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Additionally, the la=
st sentence of section 4.2 is not correct English.&nbsp; The sentence<o:p><=
/o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>&nb=
sp;&nbsp; For detailed usage in Offer/Answer for unilateral parameter, refe=
r to section 5.2 of [RFC3611].<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbs=
p;</o:p></p><p class=3DMsoNormal>should be<o:p></o:p></p><p class=3DMsoNorm=
al><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>&nbsp;&nbsp; For detailed usag=
e of Offer/Answer for unilateral parameters, refer to section 5.2 of [RFC36=
11].<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMso=
Normal>-- <o:p></o:p></p><p class=3DMsoNormal>Jonathan Lennox<o:p></o:p></p=
><p class=3DMsoNormal>jonathan@vidyo.com<o:p></o:p></p><p class=3DMsoNormal=
><o:p>&nbsp;</o:p></p></div></body></html>=

--_000_C3759687E4991243A1A0BD44EAC823034DFB915B07BE235maillan_--

From magnus.westerlund@ericsson.com  Mon Dec 17 08:07:40 2012
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Subject: Re: [MMUSIC] draft-ietf-mmusic-rtsp-nat-14 review
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On 2012-12-17 16:36, Ari Keranen wrote:
> Hi Magnus,
> 
> On 12/6/12 5:03 PM, Magnus Westerlund wrote:
>> On 2012-12-05 22:22, Ari Keranen wrote:
>>> 3.  Solution Overview
>>>
>>>          The client then installs the STUN servers on each of
>>>          the local candidates transport addresses it has gathered
>>>
>>> I would rather say "The client then prepares to act as a STUN server on
>>> each [...]"
>>
>> Sure, maybe "runs STUN servers" is better than both "installs" and
>> "act". They are STUN server, but with limited utility. Act as STUN
>> servers sounds like it is impersonating STUN, rather than actually being
>> STUN servers.
> 
> "runs" sounds good to me.
> 
>>>
>>>
>>>          If the server has a public
>>>          IP address, then a single candidate per address family
>>>
>>> In some cases (e.g., certain ISPs, I've heard) even "public" IP address
>>> does not mean you're not behind a NAT. Maybe it's better to say here "If
>>> the server is not behind a NAT". Same issue in step 8.
>>
>> Sure, maybe we need to define what we mean with not being behind a NAT.
>> It is a matter of being reachable from the top anchorin address domain
>> in regards to the clients.
> 
> Yes, that would be good.

OK, so far I haven't actually done such an addition to the modified
texts. But I will see what I can do before submitting the update.

> 
> [...]
>>>     Aggressive nomination SHALL be used with RTSP.  This doesn't have
>>> the
>>>     negative impact that it has in offer/answer as media playing only
>>>     starts after issuing a PLAY request.
>>>
>>> But it does have the downside of possibly selecting a lower-priority
>>> pair due to single dropped (e.g., due to congestion) connectivity check
>>> message for a higher priority pair. That said, probably SHALL is
>>> appropriate here, just wanted to make sure the implications are
>>> understood.
>>
>> Yes, but with ICE-RTSP you will retransmit the higher priority also,
>> thus if that works and you get a response that alternative is nominated
>> and will be used. But we probably should clarify what it means when
>> multiple candidate pairs are nominated, and in worse case the server
>> will use different paths for different media streams.
> 
> According to ICE RFC, if multiple pairs end up nominated due to
> aggressive nomination, the highest priority one will be used (you get
> some transient behavior).
> 
> But do you mean that you will keep retransmitting on a higher priority
> pair even if a lower priority pair has been (aggressively) selected and
> all check lists are in Completed state?

I guess this needs to be expanded a bit. What I mean is that if due to
some reason one get a lower prioritized pair to respond and then get an
answer from the more prioritized then it actually makes sense to unlock
the full checklist for the more prioritized and let it run to completion
prior to sending the PLAY request.

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 btv1==69844116b32==HKaplan@acmepacket.com  Mon Dec 17 09:04:38 2012
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "mmusic@ietf.org (E-mail)" <mmusic@ietf.org>
Thread-Topic: NEED WG CONSENSUS: media-loopback is all-or-nothing
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Cc: Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>, "draft-ietf-mmusic-media-loopback@tools.ietf.org" <draft-ietf-mmusic-media-loopback@tools.ietf.org>
Subject: Re: [MMUSIC] NEED WG CONSENSUS: media-loopback is all-or-nothing
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Howdy,
WG consensus is the "NO" path, and therefore section 3.1 of draft-24 will b=
e changed from this:
"An SDP offerer compliant to this specification and attempting to establish=
 a media session with media loopback MUST include "loopback" media attribut=
es for each individual media description in the offer message.  The offerer=
 will look for the "loopback" media attributes in the media description(s) =
of the response from the answer for confirmation that the request is accept=
ed."

To this in draft-25:
"An SDP offerer compliant to this specification and attempting to establish=
 a media session with media loopback will include "loopback" media attribut=
es for each individual media description in the offer message that it wishe=
s to have looped back.  Note that the offerer may choose to only request lo=
op back for some media descriptions/streams but not others.  For example it=
 might wish to request loopback for a video stream but not audio, or vice-v=
ersa.
=20
The offerer will look for the "loopback" media attributes in the media desc=
ription(s) of the response from the SDP answer for confirmation that the re=
quest is accepted."

-hadriel


On Dec 12, 2012, at 6:29 AM, Gonzalo Camarillo <gonzalo.camarillo@ericsson.=
com> wrote:

> Hi Hadriel,
>=20
> the seems to be consensus around the "NO" path. Could you please revise
> the draft accordingly so that we can also clear the last discuss?
>=20
> Thanks,
>=20
> Gonzalo
>=20
> On 20/11/2012 8:16 PM, Hadriel Kaplan wrote:
>> Howdy,
>> during IESG review of the media-loopback draft, a discussion arose regar=
ding the language around indicating media-loopback support per media descri=
ption (i.e., per m-line), in sections 3.1 and 3.2.
>>=20
>> My interpretation/understanding was we were requiring ALL m-lines to ind=
icate media-loopback; in other words it's an all-or-nothing in both the off=
er+answer, and either every media stream is looped back, or none are.  Howe=
ver the text currently doesn't make this clear, and it could be argued eith=
er way, with pro's/con's either way too; so I'd like the WG's consensus on =
whether it should in fact be an all-or-none approach, or not.
>>=20
>> Please respond to this email by **Tuesday, December 4th** with a "YES" i=
f the draft should mandate ALL media be looped-back, or "NO" if it should a=
llow a hybrid approach of some streams being looped while others are not.
>>=20
>> For reference, the draft is here:
>> http://tools.ietf.org/html/draft-ietf-mmusic-media-loopback-24
>>=20
>> -hadriel
>>=20
>>=20
>=20


From internet-drafts@ietf.org  Mon Dec 17 09:42:23 2012
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Cc: mmusic@ietf.org
Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-media-loopback-25.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-25.txt
	Pages           : 34
	Date            : 2012-12-17

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-25

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-media-loopback-25


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


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From finlayson@live555.com  Mon Dec 17 19:17:22 2012
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From: Ross Finlayson <finlayson@live555.com>
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References: <CA+TAfnm2p_TtF=x2_jm+YR_NVk94gBDYnKtQcy13307-YmiZQw@mail.gmail.com> <50CEE45A.3000502@alvestrand.no> <50CEE66F.8020902@ericsson.com>
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Subject: Re: [MMUSIC] RTSP "alt-ports" are not mentioned in the RFC.
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(Grumple... I wish this mailing list were set up so that replies went to =
the mailing list by default.  Resending to the whole mailing list =
instead of just Magnus.)

> The question is if this port number has any benefit at all.

Yes it does.  If a RTSP server cannot use port 554 (most likely because =
it is running on a Unix system, but not as 'root'), then it can try =
using port 8554 instead.  (FWIW, our RTSP server implementation - widely =
used on the Internet - does this.)  As you noted, if a server uses port =
8554 (or any port other than 554), then this port number will need to be =
included in the "rtsp://" URLs that clients use.  But it's beneficial to =
know - e.g., if you're managing a network's 'security' policy - that =
RTSP will likely be used on either port 554 or port 8554 (rather than =
either port 554 or some unspecified other port number).  Similarly, it's =
useful for network monitoring tools to know that traffic on port 8554 is =
likely to be RTSP.

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/


--Apple-Mail=_B26309B0-3548-4B0B-A6E0-5C1B6F56DDC1
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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dus-ascii"><meta http-equiv=3D"Content-Type" content=3D"text/html=
 charset=3Dus-ascii"><meta http-equiv=3D"Content-Type" =
content=3D"text/html charset=3Diso-8859-1"></head><body =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><div>(Grumple... I wish this =
mailing list were set up so that replies went to the mailing list by =
default. &nbsp;Resending to the whole mailing list instead of just =
Magnus.)</div><div><br><blockquote type=3D"cite">The question is if this =
port number has any benefit at all.</blockquote><div><br></div>Yes it =
does. &nbsp;If a RTSP server cannot use port 554 (most likely because it =
is running on a Unix system, but not as 'root'), then it can try using =
port 8554 instead. &nbsp;(FWIW, our RTSP server implementation - widely =
used on the Internet - does this.) &nbsp;As you noted, if a server uses =
port 8554 (or any port other than 554), then this port number will need =
to be included in the "rtsp://" URLs that clients use. &nbsp;But it's =
beneficial to know - e.g., if you're managing a network's 'security' =
policy - that RTSP will likely be used on either port 554 or port 8554 =
(rather than either port 554 or some unspecified other port number). =
&nbsp;Similarly, it's useful for network monitoring tools to know that =
traffic on port 8554 is likely to be RTSP.</div><div><br></div><div =
apple-content-edited=3D"true">
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; =
color: rgb(0, 0, 0); font-family: Helvetica; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: =
0px; text-transform: none; white-space: normal; widows: 2; word-spacing: =
0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; ">Ross =
Finlayson<br>Live Networks, Inc.<br><a =
href=3D"http://www.live555.com/">http://www.live555.com/</a></span>
</div>
<br></body></html>=

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From gwz@net-zen.net  Mon Dec 17 17:50:56 2012
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Cc: "draft-ietf-xrblock-rtcp-xr-discard@tools.ietf.org" <draft-ietf-xrblock-rtcp-xr-discard@tools.ietf.org>, "mmusic@ietf.org" <mmusic@ietf.org>, gwz@net-zen.net
Subject: Re: [MMUSIC] SDP Directorate review of draft-ietf-xrblock-rtcp-xr-discard
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On 12/17/2012 10:49 PM, Jonathan Lennox wrote:

> I have been asked to perform  the SDP Directorate review for
 > draft-ietf-xrblock-rtcp-xr-discard. I reviewed the -10 version of the
 > document.
 >
 >
 >
 > Syntactically, the SDP usage in this document seems fine. It is a
 > simple and correct usage of an extension point defined in RFC 3711.
 >
 >
 >
 > However, the statement in Section 4 that "XR blocks MAY be used
 > without explicit signaling" is confusing, as it implies that the
 > entire SDP section is entirely optional. This document should at
 > least reference Section 5 of RFC 3711, which gives guidance as to
 > when SDP signaling of the use of XR blocks is recommended.
 >

I'm not sure why this is confusing (aside from fact that RFC 3711 is 
itself confusing: it says "although the use of SDP signaling for XR 
blocks may be optional, if used, it MUST be used as defined here". /May/ 
be optional?  Optionality seems pretty black and white to me ;-).  That 
said, however, it does seem to be relatively clear that the usage of SDP 
/is/ optional & along with it the section on SDP in this and other 
similar drafts.  What am I missing?

...


From gwz@net-zen.net  Mon Dec 17 19:03:32 2012
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Subject: Re: [MMUSIC] SDP Directorate review of draft-ietf-xrblock-rtcp-xr-discard
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On 12/18/2012 09:20 AM, Qin Wu wrote:

> I guess the reference "RFC3711  section 5" Jonathan mentioned should
 > be corrected as "RFC3611 section 5". If my understanding is correct,
 > what Jonathan suggested is to have the following change: OLD TEXT: "
 > RFC 3611 defines the use of SDP (Session Description Protocol)
 > [RFC4566] for signaling the use of RTCP XR blocks. XR blocks MAY be
 > used without prior signaling.
 >
 > " NEW TEXT: " RFC 3611 defines the use of SDP (Session Description
 > Protocol) [RFC4566] for signaling the use of RTCP XR blocks. However
 > XR blocks MAY be used without prior signaling (see section 5 of
 > RFC3611). " Please correct me if I am wrong.

My comment still stands: the existing text is not confusing & needn't be 
changed.

...

From ari.keranen@nomadiclab.com  Tue Dec 18 04:43:12 2012
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Cc: draft-ietf-mmusic-rtsp-nat@tools.ietf.org, mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] draft-ietf-mmusic-rtsp-nat-14 review
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On 12/17/12 6:07 PM, Magnus Westerlund wrote:
> On 2012-12-17 16:36, Ari Keranen wrote:
>> On 12/6/12 5:03 PM, Magnus Westerlund wrote:
>>> On 2012-12-05 22:22, Ari Keranen wrote:
[...]
>>>>      Aggressive nomination SHALL be used with RTSP.  This doesn't have
>>>> the
>>>>      negative impact that it has in offer/answer as media playing only
>>>>      starts after issuing a PLAY request.
>>>>
>>>> But it does have the downside of possibly selecting a lower-priority
>>>> pair due to single dropped (e.g., due to congestion) connectivity check
>>>> message for a higher priority pair. That said, probably SHALL is
>>>> appropriate here, just wanted to make sure the implications are
>>>> understood.
>>>
>>> Yes, but with ICE-RTSP you will retransmit the higher priority also,
>>> thus if that works and you get a response that alternative is nominated
>>> and will be used. But we probably should clarify what it means when
>>> multiple candidate pairs are nominated, and in worse case the server
>>> will use different paths for different media streams.
>>
>> According to ICE RFC, if multiple pairs end up nominated due to
>> aggressive nomination, the highest priority one will be used (you get
>> some transient behavior).
>>
>> But do you mean that you will keep retransmitting on a higher priority
>> pair even if a lower priority pair has been (aggressively) selected and
>> all check lists are in Completed state?
>
> I guess this needs to be expanded a bit. What I mean is that if due to
> some reason one get a lower prioritized pair to respond and then get an
> answer from the more prioritized then it actually makes sense to unlock
> the full checklist for the more prioritized and let it run to completion
> prior to sending the PLAY request.

By unlocking checklist, do you mean moving it from Completed back to 
Running state and continuing all running checks (until when)? And in the 
case you mention above, both candidates are in the same check list, right?

Anyway, it would be good to clarify this.


Cheers,
Ari


From internet-drafts@ietf.org  Tue Dec 18 05:49:11 2012
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-15.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-15.txt
	Pages           : 37
	Date            : 2012-12-18

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-15

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sdp-cs-15


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


From Simo.Veikkolainen@nokia.com  Tue Dec 18 06:01:39 2012
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From: <Simo.Veikkolainen@nokia.com>
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Subject: [MMUSIC] FW:  I-D Action: draft-ietf-mmusic-sdp-cs-15.txt
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Hello,

We have submitted a new version of the draft on SDP for circuit-switched be=
arers. This now addresses all comments that were raised during review for p=
ublication. There is one technical change in this document: instead of refe=
rring to 'global-number' of RFC3966 we now refer to 'global-number-digits'.=
 We did not have any clear usage for the additional parameters allowed by '=
global-number', and referring to the digits only makes the format simpler.

Additionally, some typos were corrected.

On behalf of the authors,
Simo

-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 ext internet-drafts@ietf.org
Sent: 18. joulukuuta 2012 15:49
To: i-d-announce@ietf.org
Cc: mmusic@ietf.org
Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-15.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-15.txt
	Pages           : 37
	Date            : 2012-12-18

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-15

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sdp-cs-15


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 jonathan@vidyo.com  Tue Dec 18 07:18:23 2012
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From: Jonathan Lennox <jonathan@vidyo.com>
To: Qin Wu <bill.wu@huawei.com>, Glen Zorn <gwz@net-zen.net>
Date: Tue, 18 Dec 2012 10:18:12 -0500
Thread-Topic: SDP Directorate review of draft-ietf-xrblock-rtcp-xr-discard
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Subject: Re: [MMUSIC] SDP Directorate review of draft-ietf-xrblock-rtcp-xr-discard
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Yes, that text addition seems reasonable to me.

For people experienced with XRBlocks, Glen's point that this isn't confusin=
g is taken; but I think this addition would be helpful for those (like me) =
who are coming to this somewhat cold.

Sorry for the typo in the RFC number.

-----Original Message-----
From: Qin Wu [mailto:bill.wu@huawei.com]=20
Sent: Monday, December 17, 2012 9:20 PM
To: Glen Zorn; Jonathan Lennox
Cc: mmusic@ietf.org; draft-ietf-xrblock-rtcp-xr-discard@tools.ietf.org; gwz=
@net-zen.net; xrblock@ietf.org
Subject: Re: SDP Directorate review of draft-ietf-xrblock-rtcp-xr-discard

I guess the reference "RFC3711 section 5" Jonathan mentioned should be corr=
ected as "RFC3611 section 5".
If my understanding is correct, what Jonathan suggested is to have the foll=
owing change:
OLD TEXT:
"
   RFC 3611 defines the use of SDP (Session Description Protocol)
   [RFC4566] for signaling the use of RTCP XR blocks.  XR blocks MAY be
   used without prior signaling.

"
NEW TEXT:
"
   RFC 3611 defines the use of SDP (Session Description Protocol)
   [RFC4566] for signaling the use of RTCP XR blocks.  However XR blocks MA=
Y be
   used without prior signaling (see section 5 of RFC3611).
"
Please correct me if I am wrong.

Regards!
-Qin
----- Original Message -----=20
From: "Glen Zorn" <gwz@net-zen.net>
To: "Jonathan Lennox" <jonathan@vidyo.com>
Cc: <mmusic@ietf.org>; <draft-ietf-xrblock-rtcp-xr-discard@tools.ietf.org>;=
 <gwz@net-zen.net>
Sent: Tuesday, December 18, 2012 9:50 AM
Subject: Re: SDP Directorate review of draft-ietf-xrblock-rtcp-xr-discard


> On 12/17/2012 10:49 PM, Jonathan Lennox wrote:
>=20
>> I have been asked to perform  the SDP Directorate review for
> > draft-ietf-xrblock-rtcp-xr-discard. I reviewed the -10 version of the
> > document.
> >
> >
> >
> > Syntactically, the SDP usage in this document seems fine. It is a
> > simple and correct usage of an extension point defined in RFC 3711.
> >
> >
> >
> > However, the statement in Section 4 that "XR blocks MAY be used
> > without explicit signaling" is confusing, as it implies that the
> > entire SDP section is entirely optional. This document should at
> > least reference Section 5 of RFC 3711, which gives guidance as to
> > when SDP signaling of the use of XR blocks is recommended.
> >
>=20
> I'm not sure why this is confusing (aside from fact that RFC 3711 is=20
> itself confusing: it says "although the use of SDP signaling for XR=20
> blocks may be optional, if used, it MUST be used as defined here". /May/=
=20
> be optional?  Optionality seems pretty black and white to me ;-).  That=20
> said, however, it does seem to be relatively clear that the usage of SDP=
=20
> /is/ optional & along with it the section on SDP in this and other=20
> similar drafts.  What am I missing?
>=20
> ...
>

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Cc: draft-ietf-xrblock-rtcp-xr-discard@tools.ietf.org, mmusic@ietf.org
Subject: Re: [MMUSIC] SDP Directorate review of	draft-ietf-xrblock-rtcp-xr-discard
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> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf
> Of Glen Zorn
> Sent: Monday, December 17, 2012 5:51 PM
> To: Jonathan Lennox
> Cc: draft-ietf-xrblock-rtcp-xr-discard@tools.ietf.org; mmusic@ietf.org;
> gwz@net-zen.net
> Subject: Re: [MMUSIC] SDP Directorate review of draft-ietf-xrblock-rtcp-
> xr-discard
> 
> On 12/17/2012 10:49 PM, Jonathan Lennox wrote:
> 
> > I have been asked to perform  the SDP Directorate review for
>  > draft-ietf-xrblock-rtcp-xr-discard. I reviewed the -10 version of the
> > document.
>  >
>  >
>  >
>  > Syntactically, the SDP usage in this document seems fine. It is a  >
> simple and correct usage of an extension point defined in RFC 3711.
>  >
>  >
>  >
>  > However, the statement in Section 4 that "XR blocks MAY be used  >
> without explicit signaling" is confusing, as it implies that the  >
> entire SDP section is entirely optional. This document should at  >
> least reference Section 5 of RFC 3711, which gives guidance as to  >
> when SDP signaling of the use of XR blocks is recommended.
>  >
> 
> I'm not sure why this is confusing (aside from fact that RFC 3711 is
> itself confusing: it says "although the use of SDP signaling for XR
> blocks may be optional, if used, it MUST be used as defined here". /May/
> be optional?  Optionality seems pretty black and white to me ;-).  That
> said, however, it does seem to be relatively clear that the usage of SDP
> /is/ optional & along with it the section on SDP in this and other
> similar drafts.  What am I missing?

(RFC3611, you both meant to write.  RFC3711 is SRTP, RFC3611 is RTCP-XR).

SDP lines are supposed to be declarative or negotiated.  RTCP-XR seems to
have done neither:  SDP is just there as a check-box.

What is missing is an explanation of when and why would an endpoint would 
include the SDP, and when and why would it omit the SDP.

But, if the endpoints are going to send RTCP-XR whenever it wants,
no matter if the remote end cares to receive RTCP-XR, why do we have
SDP defined at all?  Just send RTCP-XR and let's save SDP implementation
effort and remove the SDP definitions.

-d





From magnus.westerlund@ericsson.com  Wed Dec 19 05:10:00 2012
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To: Ari Keranen <ari.keranen@nomadiclab.com>
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Cc: draft-ietf-mmusic-rtsp-nat@tools.ietf.org, mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] draft-ietf-mmusic-rtsp-nat-14 review
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Ari,

Thanks for keep following up on this. Please see below.

On 2012-12-18 13:43, Ari Keranen wrote:
> On 12/17/12 6:07 PM, Magnus Westerlund wrote:
>> On 2012-12-17 16:36, Ari Keranen wrote:
>>> On 12/6/12 5:03 PM, Magnus Westerlund wrote:
>>>> On 2012-12-05 22:22, Ari Keranen wrote:
> [...]
>>>>>      Aggressive nomination SHALL be used with RTSP.  This doesn't have
>>>>> the
>>>>>      negative impact that it has in offer/answer as media playing only
>>>>>      starts after issuing a PLAY request.
>>>>>
>>>>> But it does have the downside of possibly selecting a lower-priority
>>>>> pair due to single dropped (e.g., due to congestion) connectivity
>>>>> check
>>>>> message for a higher priority pair. That said, probably SHALL is
>>>>> appropriate here, just wanted to make sure the implications are
>>>>> understood.
>>>>
>>>> Yes, but with ICE-RTSP you will retransmit the higher priority also,
>>>> thus if that works and you get a response that alternative is nominated
>>>> and will be used. But we probably should clarify what it means when
>>>> multiple candidate pairs are nominated, and in worse case the server
>>>> will use different paths for different media streams.
>>>
>>> According to ICE RFC, if multiple pairs end up nominated due to
>>> aggressive nomination, the highest priority one will be used (you get
>>> some transient behavior).
>>>
>>> But do you mean that you will keep retransmitting on a higher priority
>>> pair even if a lower priority pair has been (aggressively) selected and
>>> all check lists are in Completed state?
>>
>> I guess this needs to be expanded a bit. What I mean is that if due to
>> some reason one get a lower prioritized pair to respond and then get an
>> answer from the more prioritized then it actually makes sense to unlock
>> the full checklist for the more prioritized and let it run to completion
>> prior to sending the PLAY request.
> 
> By unlocking checklist, do you mean moving it from Completed back to
> Running state and continuing all running checks (until when)? And in the
> case you mention above, both candidates are in the same check list, right?
> 
> Anyway, it would be good to clarify this.

I have produced a bit of new text. However, having thought about this I
don't want to make any change to the ICE procedures as that could easily
introduce issues and makes the changes for ICE-RTSP even bigger. Instead
I propose that we change the SHALL to a SHOULD so that client
implementations that see issues with using aggressive nomination can
select to do regular nomination instead and always accept a bit more
delay instead of possibly ending up with different network paths between
different media resources SETUP in the RTSP session.

Thus I have written the following text, please review and comment.

6.7.  Client to Server ICE Connectivity Check

   The client receives the SETUP response and learns the candidate
   addresses to use for the connectivity checks.  The client SHALL
   initiate its connectivity check, following the procedures in Section
   6 of ICE [RFC5245].  The STUN transaction pacer SHALL be used across
   all media streams part of the same RTSP session.

   Aggressive nomination SHOULD be used with RTSP during initial SETUP
   for a resource.  This doesn't have all the negative impact that it
   has in offer/answer as media playing only starts after issuing a PLAY
   request.  Thus the issue with a change of the media path being used
   for delivery can be avoided by not issuing a PLAY request while STUN
   connectivity checks are still outstanding.  Aggressive nomination can
   result in multiple candidate pairs having their nominated flag set
   but according to Section 8.1.1.2 of ICE [RFC5245] when the PLAY
   request is sent the media will arrive on the pair with the highest
   priority.  Note, different media resources may still end up with
   different foundations.

   Note: The above does not change ICE and its handling of aggressive
   nomination.  Using aggressive nomination a given media stream can
   have a higher priority candidate pair with an outstanding
   connectivity check message when its check list completes, to also
   move that candidate pair into Succeeded state and set the Nominated
   flag.  Thus moving the used pair to this higher priority candidate
   pair.  To avoid this occurring during actual media transport, the
   RTSP client can add an additional checks when getting the ICE
   processing overall is completed to indicate if there is still higher
   priority connectivity checks outstanding.  If some check is still
   outstanding then some additional timeout or the completion of the
   outstanding checks before progressing with a PLAY request.  One
   alternative is to accept the risk for a path change during media
   delivery.

   For RTSP clients that want to ensure that each media resource uses
   the same path can use regular nomination where both the ICE
   processing completion criteria can be controlled in addition to which
   media streams being nominated for use.  This does not affect the RTSP
   server, as its role is the one of being controlled.






-- 

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 ari.keranen@nomadiclab.com  Wed Dec 19 09:50:17 2012
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Cc: draft-ietf-mmusic-rtsp-nat@tools.ietf.org, mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] draft-ietf-mmusic-rtsp-nat-14 review
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Hi Magnus,

On 12/19/12 3:09 PM, Magnus Westerlund wrote:
> Ari,
>
> Thanks for keep following up on this. Please see below.

No problem! Comments inline.

> On 2012-12-18 13:43, Ari Keranen wrote:
>> On 12/17/12 6:07 PM, Magnus Westerlund wrote:
>>> On 2012-12-17 16:36, Ari Keranen wrote:
>>>> On 12/6/12 5:03 PM, Magnus Westerlund wrote:
>>>>> On 2012-12-05 22:22, Ari Keranen wrote:
>> [...]
>>>>>>       Aggressive nomination SHALL be used with RTSP.  This doesn't have
>>>>>> the
>>>>>>       negative impact that it has in offer/answer as media playing only
>>>>>>       starts after issuing a PLAY request.
>>>>>>
>>>>>> But it does have the downside of possibly selecting a lower-priority
>>>>>> pair due to single dropped (e.g., due to congestion) connectivity
>>>>>> check
>>>>>> message for a higher priority pair. That said, probably SHALL is
>>>>>> appropriate here, just wanted to make sure the implications are
>>>>>> understood.
>>>>>
>>>>> Yes, but with ICE-RTSP you will retransmit the higher priority also,
>>>>> thus if that works and you get a response that alternative is nominated
>>>>> and will be used. But we probably should clarify what it means when
>>>>> multiple candidate pairs are nominated, and in worse case the server
>>>>> will use different paths for different media streams.
>>>>
>>>> According to ICE RFC, if multiple pairs end up nominated due to
>>>> aggressive nomination, the highest priority one will be used (you get
>>>> some transient behavior).
>>>>
>>>> But do you mean that you will keep retransmitting on a higher priority
>>>> pair even if a lower priority pair has been (aggressively) selected and
>>>> all check lists are in Completed state?
>>>
>>> I guess this needs to be expanded a bit. What I mean is that if due to
>>> some reason one get a lower prioritized pair to respond and then get an
>>> answer from the more prioritized then it actually makes sense to unlock
>>> the full checklist for the more prioritized and let it run to completion
>>> prior to sending the PLAY request.
>>
>> By unlocking checklist, do you mean moving it from Completed back to
>> Running state and continuing all running checks (until when)? And in the
>> case you mention above, both candidates are in the same check list, right?
>>
>> Anyway, it would be good to clarify this.
>
> I have produced a bit of new text. However, having thought about this I
> don't want to make any change to the ICE procedures as that could easily
> introduce issues and makes the changes for ICE-RTSP even bigger. Instead
> I propose that we change the SHALL to a SHOULD so that client
> implementations that see issues with using aggressive nomination can
> select to do regular nomination instead and always accept a bit more
> delay instead of possibly ending up with different network paths between
> different media resources SETUP in the RTSP session.

Makes sense. I also now remember another problem with aggressive 
nomination. I'll send a separate mail to the list about that.

> Thus I have written the following text, please review and comment.
>
> 6.7.  Client to Server ICE Connectivity Check
>
>     The client receives the SETUP response and learns the candidate
>     addresses to use for the connectivity checks.  The client SHALL
>     initiate its connectivity check, following the procedures in Section
>     6 of ICE [RFC5245].  The STUN transaction pacer SHALL be used across
>     all media streams part of the same RTSP session.

BTW, how is this (transaction pacer) different from regular ICE? If it's 
not, does it need to be mentioned?

Also "(that are) part of".

>     Aggressive nomination SHOULD be used with RTSP during initial SETUP
>     for a resource.  This doesn't have all the negative impact that it
>     has in offer/answer as media playing only starts after issuing a PLAY
>     request.  Thus the issue with a change of the media path being used
>     for delivery can be avoided by not issuing a PLAY request while STUN
>     connectivity checks are still outstanding.  Aggressive nomination can
>     result in multiple candidate pairs having their nominated flag set
>     but according to Section 8.1.1.2 of ICE [RFC5245] when the PLAY
>     request is sent the media will arrive on the pair with the highest
>     priority.  Note, different media resources may still end up with
>     different foundations.
 >
>     Note: The above does not change ICE and its handling of aggressive
>     nomination.  Using aggressive nomination a given media stream can
>     have a higher priority candidate pair with an outstanding
>     connectivity check message when its check list completes, to also
>     move that candidate pair into Succeeded state and set the Nominated

Can't parse this sentence. Something missing?

>     flag.  Thus moving the used pair to this higher priority candidate
>     pair.  To avoid this occurring during actual media transport, the
>     RTSP client can add an additional checks when getting the ICE

Remove "an" and "getting"

Also, I guess you mean "continue sending checks" (i.e., no "new" 
additional checks).

>     processing overall is completed to indicate if there is still higher
>     priority connectivity checks outstanding.  If some check is still
>     outstanding then some additional timeout or the completion of the

Some word missing from this sentence.

>     outstanding checks before progressing with a PLAY request.  One
>     alternative is to accept the risk for a path change during media
>     delivery.
>
>     For RTSP clients that want to ensure that each media resource uses

Remove "For"

>     the same path can use regular nomination where both the ICE
>     processing completion criteria can be controlled in addition to which
>     media streams being nominated for use.  This does not affect the RTSP
>     server, as its role is the one of being controlled.

But yes, overall the idea looks OK to me.


Cheers,
Ari

From ari.keranen@nomadiclab.com  Wed Dec 19 09:50:48 2012
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Hi all,

Going through the ICE spec reminded me of one old problem with 
aggressive nomination that I discussed with Philip Matthews back when 
ICE was published:

Two endpoints L and R. L is the offerer and controlling end. There are 
two possible paths, and just one component.

Since L is using aggressive nomination, it sets the USE-CANDIDATE flag 
in each Binding request. However, the Binding response for the first 
path does not make it back to L. When the second check succeeds, L 
immediately stops ICE processing and uses the second path. However, R 
thinks the first path is being used.

Initially, L will use path 2 for data, while R will use path 1. 
Eventually, both paths will fail, because L is only sending keep-alives 
on path 2, while R is only sending them on path 1.

L                                         R
  <--------- Bind req ---------------------  ]
  ---------- Bind resp ------------------->  ]  Path 1
  ---------- Bind req, USE-CAND ---------->  ]
       X <-- Bind resp --------------------  ]

  <--------- Bind req ---------------------  ]
  ---------- Bind resp ------------------->  ]  Path 2
  ---------- Bind req, USE-CAND ---------->  ]
  <--------- Bind resp --------------------  ]

---


Any good ideas how to fix this? Should the controlling agent just keep 
retransmitting even after successful nomination (for how long)?

The updated offer (if we make that MUST in all cases) will help to 
detect the problem though and this could also be added to the list of 
"reasons for not using aggressive nomination unless you really have to".


Cheers,
Ari

From fluffy@iii.ca  Mon Dec 24 06:18:10 2012
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seems there is something wrong with mailing list=85=20


Begin forwarded message:

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> Subject: Undelivered Mail Returned to Sender
> Date: December 24, 2012 12:35:57 AM MST
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>=20
> This is the mail system at host ietfa.amsl.com.
>=20
> I'm sorry to have to inform you that your message could not
> be delivered to one or more recipients. It's attached below.
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> For further assistance, please send mail to postmaster.
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> If you do so, please include this problem report. You can
> delete your own text from the attached returned message.
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>                   The mail system
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>=20
> Final-Recipient: rfc822; mmusic@ietfa.amsl.com
> Original-Recipient: rfc822;mmusic@ietf.org
> Action: failed
> Status: 5.4.6
> Diagnostic-Code: X-Postfix; mail forwarding loop for =
mmusic@ietfa.amsl.com
>=20
> From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
> Subject: [MMUSIC] SDP work needed for WebRTC stuff
> Date: December 8, 2012 3:41:49 PM MST
> To: mmusic WG <mmusic@ietf.org>
>=20
>=20
>=20
> I was looking over everything that needs to be completed to finish a =
fist cut of the WebRTC related work. There are a handful of big SDP =
problems that are currently blocking some of the WebRTC work and I'd =
like to figure out how to make some progress on them.=20
>=20
> Let me loosely characterize them as=20
>=20
> 1) If we have several video streams, how do theses map up to 1 or more =
m lines.=20
>=20
> 2) if we are doing port multiplexing, what does the SDP look like (the =
bundle problem)=20
>=20
> 3) How do we map the RTCWeb track and stream label concepts to =
identifiers in SDP=20
>=20
> 3) SDP for application running over SCTP/DTLS=20
>=20
> I don't want to speak for all the various chairs but I am under the =
impression that most of chairs of related groups in W3C and IETF believe =
these are issues that need to be resolved primarily in the MMUSIC WG and =
that they impact both WebRTC and CLUE as well as the general long term =
use of SDP in SIP and other protocols.=20
>=20
> I'd like to get some discussion going on how we can make some progress =
on these. I don't think we are going to solve these in 20 minutes of =
discussion at an IETF meeting so I think we probably need some interim =
(virtual or face to face) to sort this out.=20
>=20
> Thoughts?=20
>=20
> Cullen <in my individual contributor role>=20
>=20
>=20
>=20
>=20
>=20
> _______________________________________________=20
> mmusic mailing list=20
> mmusic@ietf.org=20
> https://www.ietf.org/mailman/listinfo/mmusic=20
>=20
>=20


--Apple-Mail=_FD734865-79D6-4C88-B5A5-A09EE2B7D318
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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
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"><div><br></div>seems there is something wrong with mailing =
list=85&nbsp;<div><br><div><br><div>Begin forwarded message:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><div =
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System &lt;<a =
href=3D"mailto:MAILER-DAEMON@ietfa.amsl.com">MAILER-DAEMON@ietfa.amsl.com<=
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0px; margin-bottom: 0px; margin-left: 0px;"><span =
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href=3D"mailto:fluffy@cisco.com">fluffy@cisco.com</a>&gt;<br></span></div>=
<br>This is the mail system at host <a =
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&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
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href=3D"mailto:mmusic@ietfa.amsl.com">mmusic@ietfa.amsl.com</a>&gt;: =
mail forwarding loop for <a =
href=3D"mailto:mmusic@ietfa.amsl.com">mmusic@ietfa.amsl.com</a><br>Reporti=
ng-MTA: dns; <a =
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746F321F8C0F<br>X-Postfix-Sender: rfc822; <a =
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Sun, 23 Dec 2012 23:35:57 -0800 (PST)<br><br>Final-Recipient: rfc822; <a =
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failed<br>Status: 5.4.6<br>Diagnostic-Code: X-Postfix; mail forwarding =
loop for <a =
href=3D"mailto:mmusic@ietfa.amsl.com">mmusic@ietfa.amsl.com</a><br><br><di=
v style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px;"><span style=3D"font-family:'Helvetica'; =
font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>From: =
</b></span><span style=3D"font-family:'Helvetica'; =
font-size:medium;">"Cullen Jennings (fluffy)" &lt;<a =
href=3D"mailto:fluffy@cisco.com">fluffy@cisco.com</a>&gt;<br></span></div>=
<div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px;"><span style=3D"font-family:'Helvetica'; =
font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>Subject: =
</b></span><span style=3D"font-family:'Helvetica'; =
font-size:medium;"><b>[MMUSIC] SDP work needed for WebRTC =
stuff</b><br></span></div><div style=3D"margin-top: 0px; margin-right: =
0px; margin-bottom: 0px; margin-left: 0px;"><span =
style=3D"font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, =
127, 1.0);"><b>Date: </b></span><span style=3D"font-family:'Helvetica'; =
font-size:medium;">December 8, 2012 3:41:49 PM MST<br></span></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px;"><span style=3D"font-family:'Helvetica'; =
font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>To: =
</b></span><span style=3D"font-family:'Helvetica'; =
font-size:medium;">mmusic WG &lt;<a =
href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a>&gt;<br></span></div><b=
r><br><meta http-equiv=3D"Content-Type" content=3D"text/html; =
charset=3Dus-ascii"><br>
I was looking over everything that needs to be completed to finish a =
fist cut of the WebRTC related work. There are a handful of big SDP =
problems that are currently blocking some of the WebRTC work and I'd =
like to figure out how to make some progress on them.=20
<br>

<br>
Let me loosely characterize them as=20
<br>

<br>
1) If we have several video streams, how do theses map up to 1 or more m =
lines.=20
<br>

<br>
2) if we are doing port multiplexing, what does the SDP look like (the =
bundle problem)=20
<br>

<br>
3) How do we map the RTCWeb track and stream label concepts to =
identifiers in SDP
<br>

<br>
3) SDP for application running over SCTP/DTLS=20
<br>

<br>
I don't want to speak for all the various chairs but I am under the =
impression that most of chairs of related groups in W3C and IETF believe =
these are issues that need to be resolved primarily in the MMUSIC WG and =
that they impact both WebRTC and CLUE as well as the general long term =
use of SDP in SIP and other protocols.=20
<br>

<br>
I'd like to get some discussion going on how we can make some progress =
on these. I don't think we are going to solve these in 20 minutes of =
discussion at an IETF meeting so I think we probably need some interim =
(virtual or face to face) to sort this out.=20
<br>

<br>
Thoughts?
<br>

<br>
Cullen &lt;in my individual contributor role&gt;=20
<br>

<br>

<br>

<br>

<br>

<br>
_______________________________________________
<br>
mmusic mailing list
<br>
<a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a>
<br>
<a =
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/mailman/listinfo/mmusic</a>
<br>
<br><br></blockquote></div><br></div></body></html>=

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From eckelcu@cisco.com  Wed Dec 26 09:50:08 2012
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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: Cullen Jennings <fluffy@iii.ca>, mmusic WG <mmusic@ietf.org>
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Yes, I can see you sent the original on Dec 8. It is captured in the archiv=
es correctly as having been sent only once and on Dec 8. The resending on D=
ec 23 is not shown in the archive. The same is true for a message Magnus se=
nt on Dec 12 that was resent on the 23rd as well.=20
http://www.ietf.org/mail-archive/web/mmusic/current/maillist.html

When I replied to Magnus's e-mail on Dec 14, it went out to the list correc=
tly; however, I did receive an error report a few days later saying "Mail D=
elivery System <Mailer-Daemon@sessmg10.ericsson.net>" was unable to send th=
e message to Magnus's address.

I have attached that e-mail in case it is useful in troubleshooting this is=
sue.

Cheers,
Charles


> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Cullen Jennings
> Sent: Monday, December 24, 2012 6:18 AM
> To: mmusic WG
> Subject: [MMUSIC] I sent this long ago =85.
>=20
>=20
> seems there is something wrong with mailing list=85
>=20
>=20
> Begin forwarded message:
>=20
>=20
> 	From: Mail Delivery System <MAILER-DAEMON@ietfa.amsl.com>
>=20
> 	Subject: Undelivered Mail Returned to Sender
>=20
> 	Date: December 24, 2012 12:35:57 AM MST
>=20
> 	To: <fluffy@cisco.com>
>=20
>=20
> 	This is the mail system at host ietfa.amsl.com.
>=20
> 	I'm sorry to have to inform you that your message could not
> 	be delivered to one or more recipients. It's attached below.
>=20
> 	For further assistance, please send mail to postmaster.
>=20
> 	If you do so, please include this problem report. You can
> 	delete your own text from the attached returned message.
>=20
> 	                  The mail system
>=20
> 	<mmusic@ietfa.amsl.com>: mail forwarding loop for
> mmusic@ietfa.amsl.com
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>=20
> 	Final-Recipient: rfc822; mmusic@ietfa.amsl.com
> 	Original-Recipient: rfc822;mmusic@ietf.org
> 	Action: failed
> 	Status: 5.4.6
> 	Diagnostic-Code: X-Postfix; mail forwarding loop for
> mmusic@ietfa.amsl.com
>=20
>=20
> 	From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
>=20
> 	Subject: [MMUSIC] SDP work needed for WebRTC stuff
>=20
> 	Date: December 8, 2012 3:41:49 PM MST
>=20
> 	To: mmusic WG <mmusic@ietf.org>
>=20
>=20
>=20
>=20
> 	I was looking over everything that needs to be completed to finish a
> fist cut of the WebRTC related work. There are a handful of big SDP
> problems that are currently blocking some of the WebRTC work and I'd like
> to figure out how to make some progress on them.
>=20
> 	Let me loosely characterize them as
>=20
> 	1) If we have several video streams, how do theses map up to 1 or
> more m lines.
>=20
> 	2) if we are doing port multiplexing, what does the SDP look like
> (the bundle problem)
>=20
> 	3) How do we map the RTCWeb track and stream label concepts to
> identifiers in SDP
>=20
> 	3) SDP for application running over SCTP/DTLS
>=20
> 	I don't want to speak for all the various chairs but I am under the
> impression that most of chairs of related groups in W3C and IETF believe
> these are issues that need to be resolved primarily in the MMUSIC WG and
> that they impact both WebRTC and CLUE as well as the general long term
> use of SDP in SIP and other protocols.
>=20
> 	I'd like to get some discussion going on how we can make some
> progress on these. I don't think we are going to solve these in 20 minute=
s of
> discussion at an IETF meeting so I think we probably need some interim
> (virtual or face to face) to sort this out.
>=20
> 	Thoughts?
>=20
> 	Cullen <in my individual contributor role>
>=20
>=20
>=20
>=20
>=20
> 	_______________________________________________
> 	mmusic mailing list
> 	mmusic@ietf.org
> 	https://www.ietf.org/mailman/listinfo/mmusic
>=20
>=20
>=20
>=20


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From abegen@cisco.com  Sat Dec 29 14:55:29 2012
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "bernard_aboba@hotmail.com" <bernard_aboba@hotmail.com>
Thread-Topic: [mmusic] #13: Section 5.14
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Subject: Re: [MMUSIC] [mmusic] #13: Section 5.14
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Hi Bernard,

Do you think this is necessary?
http://www.iana.org/assignments/sdp-parameters/sdp-parameters.xml#sdp-param
eters-2 lists bunch of other protos that we don't list in 5.14.

The text in 8.2.2 is clear saying that:

If other RTP profiles are defined in the future, their "proto" name
   SHOULD be specified in the same manner.  For example, an RTP profile
   whose short name is "XYZ" would be denoted by a "proto" field of
   "RTP/XYZ".


I don't think we need to clutter the text with more examples.

-acbegen

-----Original Message-----
From: mmusic issue tracker <trac+mmusic@trac.tools.ietf.org>
Date: Saturday, September 22, 2012 3:48 PM
To: "Ali C. Begen" <abegen@cisco.com>, Bernard Aboba
<bernard_aboba@hotmail.com>
Cc: "mmusic@ietf.org" <mmusic@ietf.org>
Subject: [mmusic] #13: Section 5.14

>#13: Section 5.14
>
> In Section 5.14 the <proto> description only mentions RTP/AVP and
> RTP/SAVP.  RTP/AVPF and RTP/SAVPF should also be mentioned.
>
>--=20
>--------------------------------+------------------------
> Reporter:  bernard_aboba@=8A     |      Owner:  abegen@=8A
>     Type:  defect              |     Status:  new
> Priority:  major               |  Milestone:  milestone1
>Component:  rfc4566bis          |    Version:  1.0
> Severity:  Active WG Document  |   Keywords:
>--------------------------------+------------------------
>
>Ticket URL: <http://tools.ietf.org/wg/mmusic/trac/ticket/13>
>mmusic <http://tools.ietf.org/mmusic/>
>


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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Draft-ietf-mmusic-rfc4566bis-05: session-level attributes
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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 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=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..."
>
>"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."
>
>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=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."
>>
>> Proposed text:
>>
>> "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 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=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."
>>>>>
>>>>> 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".
>>>
>>> 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").
>>>
>>> "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


From pkyzivat@alum.mit.edu  Sat Dec 29 16:36:46 2012
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Date: Sat, 29 Dec 2012 19:36:42 -0500
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [MMUSIC] Draft-ietf-mmusic-rfc4566bis-05: session-level attributes
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On 12/29/12 6:03 PM, Ali C. Begen (abegen) wrote:
> I donıt 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).

I found one: a=maxprate in RFC3890.

	Thanks,
	Paul

> 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>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Draft-ietf-mmusic-rfc4566bis-05: session-level attributes
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Subject: Re: [MMUSIC] Draft-ietf-mmusic-rfc4566bis-05: session-level attributes
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Subject: Re: [MMUSIC] Draft-ietf-mmusic-rfc4566bis-05: session-level attributes
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Inline.

On 12/29/12 7:45 PM, Ali C. Begen (abegen) wrote:
> 
> 
> -----Original Message-----
> From: Paul Kyzivat <pkyzivat@alum.mit.edu>
> Date: Saturday, December 29, 2012 7:36 PM
> To: "mmusic@ietf.org" <mmusic@ietf.org>
> Subject: Re: [MMUSIC] Draft-ietf-mmusic-rfc4566bis-05:
> session-level attributes
> 
>  >On 12/29/12 6:03 PM, Ali C. Begen (abegen) wrote:
>  >> I donİöt 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).
>  >
>  >I found one: a=maxprate in RFC3890.

> Well, I still don't see a problem in replacing the current text with the
> proposed text. Do you?

Yes - I do see a problem.

I wish I didn't. IMO it was intended to work this way.
(The text we are proposing to revise was present in RFC2327.)
But it is a little bit confusing when you start to talk about attributes
that are session-only, or are explicitly defined for both session and
media level, as maxprate is.

ISTM that one way to move forward while retaining backward compatibility
is to make clear that this only applies to attributes defined as valid
for both session and media level, and and that has no special syntax or
semantics defined for session level.

But even this is a bit of a risk for backward compatibility, because
some existing implementations may not work this way. I'm ambivalent
about whether to take the hit on this. We could argue that such
implementations have always been  broken.

I'll also note that of the attributes defined in this draft, the only
ones defined as both media and session level are
sendrecv/sendonly/recvonly/inactive, sdplang, and lang. So these are the
only ones eligible to be used in an example here. And all of these have
special issues that make them bad examples:

The sendrecv, sendonly, ... attributes override each other: a sendonly
at media level overrides sendrecv at session level. This requires
special per-attribute description of the interaction between session and
media level, so they aren't a good example for the general override
principle. (The descriptions of these in section 6 should be updated to
discuss this.)

And a=lang and a=sdplang may appear multiple times to specify multiple
languages. E.g. suppose I have:

      a=lang:en
      a=lang:fr
      m=audio 49170 RTP/AVP 0
      m=video 51372 RTP/AVP 99
      a=lang:de

Presumably this means the audio has both English and French.
But oes the "a=lang:de" *override* those?
Or does it add a third language for the video stream?

ISTM that the specifications of these in section 6 need to be beefed up
to make this clear.
And once they are, they probably preempt the default rule.

And once that is done, we have no attributes defined at both media and
session level that are good to use in an example of the general
principle of session level providing a default for media level.
(Certainly there are others, but they aren't defined in this document.)

Now we *could* say this session-level default rule also applies to
attributes that are defined as media-level-only attributes. But then I
think we would incur a significant backward compatibility issue. (E.g.
if I use a=rtpmap at session level.)

I had hoped to provide some alternative wording here, but now I don't
think I'm ready to do that.

I think this needs more discussion, and we need to work out improvements
to the description of a=sendrecv, ... a=lang, ...
I am willing to take a crack at proposing new language for them, but
lets have this discussion play out first.

	Thanks,
	Paul

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