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Date: Sat, 01 Feb 2014 06:29:33 -0800
Cc: mmusic@ietf.org
Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-udptl-dtls-04.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Multiparty Multimedia Session Control Working Group of the IETF.

        Title           : UDP Transport Layer (UDPTL) over Datagram Transport Layer Security (DTLS)
        Authors         : Christer Holmberg
                          Ivo Sedlacek
                          Gonzalo Salgueiro
	Filename        : draft-ietf-mmusic-udptl-dtls-04.txt
	Pages           : 21
	Date            : 2014-02-01

Abstract:
   This document specifies how the UDP Transport Layer (UDPTL) protocol,
   the predominant transport protocol for T.38 fax, can be transported
   over the Datagram Transport Layer Security (DTLS) protocol, how the
   usage of UDPTL over DTLS is indicated in the Session Description
   Protocol (SDP), and how UDPTL over DTLS is negotiated in a session
   established using the Session Initiation Protocol (SIP).


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-udptl-dtls-04

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-udptl-dtls-04


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

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


From christer.holmberg@ericsson.com  Sat Feb  1 06:31:44 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: Draft new version: draft-ietf-mmusic-udptl-dtls-04
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Date: Sat, 1 Feb 2014 14:31:37 +0000
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Subject: [MMUSIC] Draft new version: draft-ietf-mmusic-udptl-dtls-04
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Hi,

Based on the review comments by Adam G and Ari K, I've submitted a new vers=
ion (-04) of the udptl-dtls draft.

Regards,

Christer=

From palmarti@cisco.com  Mon Feb  3 05:59:35 2014
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From: "Pal Martinsen (palmarti)" <palmarti@cisco.com>
To: "Balint Menyhart (balmenyh)" <balmenyh@cisco.com>
Thread-Topic: Candidate priority in draft-reddy-mmusic-ice-happy-eyeballs-05
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Subject: Re: [MMUSIC] Candidate priority in draft-reddy-mmusic-ice-happy-eyeballs-05
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On 30 Jan 2014, at 14:30 pm, Balint Menyhart <balmenyh@cisco.com> wrote:

> Hi,
>=20
> This is a comment related to http://www.ietf.org/id/draft-reddy-mmusic-ic=
e-happy-eyeballs-05.txt. Section 4 states:
>=20
>   ICE [RFC5245] section 4.1.2.2 has guidelines for how the type
>   preference value should be chosen.  Instead of having a static value
>   for IPv4 and a static value for IPv6 type of addresses, it is
>   possible to choose this value dynamically in such a way that IPv4 and
>   IPv6 address candidate priorities ends up intermingled.
>=20
>   The local and remote agent can have different algorithms for choosing
>   the type preference value without any impact on coordination between
>   the local and remote check list.
>=20
> I believe the intention here was to reference the local preference, inste=
ad of the type preference.

Yes. You are absolutely right. This was a glitch from the authors. Thinking=
 to much about the IP addr type.


> The former is meant to distinguish between network interfaces from which =
the candidates have been gathered, while the later was intended to distingu=
ish the type of the candidate obtained. It sounds to me, that IPv4 / IPv6 d=
ifference falls therefore in the local preference category, and not the typ=
e preference.
>=20
> From a different perspective, applying the formula given above:
>=20
>        priority =3D (2^24)*(type preference) +
>                   (2^8)*(local preference) +
>                   (2^0)*(256 - component ID)
>=20
> it also makes sense to "play" with the less significant bits, which are t=
he local preference, as opposed to the more significant bits, which are the=
 type preference. This will indeed lead to a finer grained intermingling of=
 the IPv4 and IPv6 candidates.
>=20
You make a good point. I will take this as hint that we are moving in the r=
ight direction by playing with the _local preference_ bits.

Thanks for taking the time to review.

.-.
P=E5l-Erik


> Regards,
>=20
> Balint


From juberti@google.com  Mon Feb  3 12:28:26 2014
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From: Justin Uberti <juberti@google.com>
Date: Mon, 3 Feb 2014 12:28:03 -0800
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Cc: Cullen Jennings <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [MMUSIC] [rtcweb] Rejecting MediaStreamTracks in JSEP
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--001a11c1e7fc8718b604f1865d89
Content-Type: text/plain; charset=UTF-8

+mmusic (meant to include the first time around)


On Mon, Feb 3, 2014 at 12:25 PM, Justin Uberti <juberti@google.com> wrote:

>
>
>
> On Mon, Feb 3, 2014 at 12:19 PM, Adam Roach <adam@nostrum.com> wrote:
>
>>  On 2/3/14 13:27, Justin Uberti wrote:
>>
>> Therefore, I think we need some similar way for a receiver to do the same
>> for a remote track.
>>
>>
>> Yes.
>>
>>
>>  And fortunately, I think we have a surface that can be used for this,
>> namely the "recv-appId" attribute that has been discussed in the context of
>> unified-plan and explained in
>> http://tools.ietf.org/id/draft-even-mmusic-application-token-02.txt and
>> unified-plan. If an endpoint doesn't want to receive a particular track on
>> a m= line, it sets the m= line to a=sendonly and also removes any
>> a=recv-appId.
>>
>>
>> No.
>>
>> The problem is that you're defining semantics to a behavior that is
>> indistinguishable from the behavior exhibited by something that does not
>> support app-id: if the thing on the other end isn't using app-id, then an
>> attempt to put a stream on hold will be misinterpreted as an attempt to
>> reject or remove it.
>>
>
> See my response to Martin. Putting a stream on hold will never result in
> that being misinterpreted as a reject; on the contrary, without appid,
> there is no way to 'fully' reject half of a stream, and, as a problem
> confined to pre-unified-plan devices, I think that is probably OK.
>
>>
>> I agree that we want an SDP operation that differentiates between "don't
>> send me this now" and "don't send me this ever" [1]  -- but it needs to be
>> explicit. Something more like "a=i-am-rejecting-this-stream", but with a
>> less cumbersome name.
>>
>> It's not clear to me that adding a new attribute would solve any problem
> that appid doesn't solve; you still need to deal with the "other end not
> using it" issue.
>
>
>> ____
>> [1] Along with a Javascript control that allows us to do the same thing,
>> but that's a W3C issue.
>>
>
> Agree. But while we're on the subject, I am thinking that MST.stop() ==
> "not ever", and RtpReceiver.active = false causes "not now".
>
>

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Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr">+mmusic (meant to include the first time around)</div><div=
 class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Mon, Feb 3, 20=
14 at 12:25 PM, Justin Uberti <span dir=3D"ltr">&lt;<a href=3D"mailto:juber=
ti@google.com" target=3D"_blank">juberti@google.com</a>&gt;</span> wrote:<b=
r>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr"><br><div class=3D"gmail_ext=
ra"><br><br><div class=3D"gmail_quote"><div class=3D"im">On Mon, Feb 3, 201=
4 at 12:19 PM, Adam Roach <span dir=3D"ltr">&lt;<a href=3D"mailto:adam@nost=
rum.com" target=3D"_blank">adam@nostrum.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">
 =20
   =20
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000"><div>
    <div>On 2/3/14 13:27, Justin Uberti wrote:<br>
    </div>
    <blockquote type=3D"cite">
      <div style=3D"font-family:arial,sans-serif;font-size:13px">Therefore,
        I think we need some similar way for a receiver to do the same
        for a remote track.</div>
    </blockquote>
    <br></div>
    Yes.<div><br>
    <br>
    <blockquote type=3D"cite">
      <div style=3D"font-family:arial,sans-serif;font-size:13px">And
        fortunately, I think we have a surface that can be used for
        this, namely the &quot;recv-appId&quot; attribute that has been dis=
cussed
        in the context of unified-plan and explained in <a href=3D"http://t=
ools.ietf.org/id/draft-even-mmusic-application-token-02.txt" target=3D"_bla=
nk">http://tools.ietf.org/id/draft-even-mmusic-application-token-02.txt</a>
        and unified-plan. If an endpoint doesn&#39;t want to receive a
        particular track on a m=3D line, it sets the m=3D line to a=3Dsendo=
nly
        and also removes any a=3Drecv-appId.</div>
    </blockquote>
    <br></div>
    No.<br>
    <br>
    The problem is that you&#39;re defining semantics to a behavior that is
    indistinguishable from the behavior exhibited by something that does
    not support app-id: if the thing on the other end isn&#39;t using
    app-id, then an attempt to put a stream on hold will be
    misinterpreted as an attempt to reject or remove it.<br></div></blockqu=
ote><div><br></div></div><div>See my response to Martin. Putting a stream o=
n hold will never result in that being misinterpreted as a reject; on the c=
ontrary, without appid, there is no way to &#39;fully&#39; reject half of a=
 stream, and, as a problem confined to pre-unified-plan devices, I think th=
at is probably OK.=C2=A0</div>

<div class=3D"im">
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div bgcolor=3D"#FFFFFF" text=3D"#000000">
    <br>
    I agree that we want an SDP operation that differentiates between
    &quot;don&#39;t send me this now&quot; and &quot;don&#39;t send me this=
 ever&quot; [1]=C2=A0 -- but
    it needs to be explicit. Something more like
    &quot;a=3Di-am-rejecting-this-stream&quot;, but with a less cumbersome =
name.<br><br></div></blockquote></div><div>It&#39;s not clear to me that ad=
ding a new attribute would solve any problem that appid doesn&#39;t solve; =
you still need to deal with the &quot;other end not using it&quot; issue.</=
div>

<div class=3D"im">
<div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8=
ex;border-left:1px #ccc solid;padding-left:1ex"><div bgcolor=3D"#FFFFFF" te=
xt=3D"#000000">
    ____<br>
    [1] Along with a Javascript control that allows us to do the same
    thing, but that&#39;s a W3C issue.<br></div></blockquote><div><br></div=
></div><div>Agree. But while we&#39;re on the subject, I am thinking that M=
ST.stop() =3D=3D &quot;not ever&quot;, and RtpReceiver.active =3D false cau=
ses &quot;not now&quot;.=C2=A0</div>


</div><br></div></div>
</blockquote></div><br></div>

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Date: Mon, 3 Feb 2014 13:03:22 -0800
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Cc: Cullen Jennings <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [MMUSIC] [rtcweb] Rejecting MediaStreamTracks in JSEP
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On Mon, Feb 3, 2014 at 12:42 PM, Adam Roach <adam@nostrum.com> wrote:

> On 2/3/14 14:25, Justin Uberti wrote:
>
>> It's not clear to me that adding a new attribute would solve any problem
>> that appid doesn't solve; you still need to deal with the "other end not
>> using it" issue.
>>
>
> Regardless of whether you think there's a problem in the legacy interop
> case, the big benefit here is that an explicit indicator by definition
> makes the operation explicit, and that in-and-of-itself is a good thing.
>
> The core issue is that you're trying to make the operation implicit; your
> proposal edges awfully close to the DWIM model of design, which we know
> causes problems when we try to add further extensions.


Good point, but since we are still defining exactly what appId does, I
think we have the opportunity to make it explicit.

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Mon, Feb 3, 2014 at 12:42 PM, Adam Roach <span dir=3D"ltr">&lt;<=
a href=3D"mailto:adam@nostrum.com" target=3D"_blank">adam@nostrum.com</a>&g=
t;</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 class=3D"im">On 2/3/14 14:25, Justin Ub=
erti wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
It&#39;s not clear to me that adding a new attribute would solve any proble=
m that appid doesn&#39;t solve; you still need to deal with the &quot;other=
 end not using it&quot; issue.<br>
</blockquote>
<br></div>
Regardless of whether you think there&#39;s a problem in the legacy interop=
 case, the big benefit here is that an explicit indicator by definition mak=
es the operation explicit, and that in-and-of-itself is a good thing.<br>


<br>
The core issue is that you&#39;re trying to make the operation implicit; yo=
ur proposal edges awfully close to the DWIM model of design, which we know =
causes problems when we try to add further extensions.</blockquote><div>

<br></div><div>Good point, but since we are still defining exactly what app=
Id does, I think we have the opportunity to make it explicit.=C2=A0</div></=
div><br></div></div>

--047d7b343296d8404004f186db09--

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Subject: Re: [MMUSIC] [rtcweb] Rejecting MediaStreamTracks in JSEP
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On 2/3/14 15:03, Justin Uberti wrote:
>
> On Mon, Feb 3, 2014 at 12:42 PM, Adam Roach <adam@nostrum.com 
> <mailto:adam@nostrum.com>> wrote:
>
>     On 2/3/14 14:25, Justin Uberti wrote:
>
>         It's not clear to me that adding a new attribute would solve
>         any problem that appid doesn't solve; you still need to deal
>         with the "other end not using it" issue.
>
>
>     Regardless of whether you think there's a problem in the legacy
>     interop case, the big benefit here is that an explicit indicator
>     by definition makes the operation explicit, and that
>     in-and-of-itself is a good thing.
>
>     The core issue is that you're trying to make the operation
>     implicit; your proposal edges awfully close to the DWIM model of
>     design, which we know causes problems when we try to add further
>     extensions.
>
>
> Good point, but since we are still defining exactly what appId does, I 
> think we have the opportunity to make it explicit.
>

I think I agree, but this isn't congruent with the proposal you made 
before. Omitting app-id as a means of rejecting a stream is an implicit 
operation, regardless of how well it's documented.

If you want to have something like "a=recv-appId:reject", that would be 
suitably explicit, but I really have a hard time imagining how something 
like that would be preferable to just having a new attribute.

/a

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    <div class="moz-cite-prefix">On 2/3/14 15:03, Justin Uberti wrote:<br>
    </div>
    <blockquote
cite="mid:CAOJ7v-1iLOuJ-MhHpGp99j1qNieQTP9=EmTpibCh21fj5-6hMA@mail.gmail.com"
      type="cite">
      <div dir="ltr"><br>
        <div class="gmail_extra">On Mon, Feb 3, 2014 at 12:42 PM, Adam
          Roach <span dir="ltr">&lt;<a moz-do-not-send="true"
              href="mailto:adam@nostrum.com" target="_blank">adam@nostrum.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">
              <div class="im">On 2/3/14 14:25, Justin Uberti wrote:<br>
                <blockquote class="gmail_quote" style="margin:0 0 0
                  .8ex;border-left:1px #ccc solid;padding-left:1ex">
                  It's not clear to me that adding a new attribute would
                  solve any problem that appid doesn't solve; you still
                  need to deal with the "other end not using it" issue.<br>
                </blockquote>
                <br>
              </div>
              Regardless of whether you think there's a problem in the
              legacy interop case, the big benefit here is that an
              explicit indicator by definition makes the operation
              explicit, and that in-and-of-itself is a good thing.<br>
              <br>
              The core issue is that you're trying to make the operation
              implicit; your proposal edges awfully close to the DWIM
              model of design, which we know causes problems when we try
              to add further extensions.</blockquote>
            <div>
              <br>
            </div>
            <div>Good point, but since we are still defining exactly
              what appId does, I think we have the opportunity to make
              it explicit.Â </div>
          </div>
          <br>
        </div>
      </div>
    </blockquote>
    <br>
    I think I agree, but this isn't congruent with the proposal you made
    before. Omitting app-id as a means of rejecting a stream is an
    implicit operation, regardless of how well it's documented.<br>
    <br>
    If you want to have something like "a=recv-appId:reject", that would
    be suitably explicit, but I really have a hard time imagining how
    something like that would be preferable to just having a new
    attribute.<br>
    <br>
    /a<br>
  </body>
</html>

--------------020506060903060308030504--

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From: Justin Uberti <juberti@google.com>
Date: Mon, 3 Feb 2014 13:22:00 -0800
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To: Adam Roach <adam@nostrum.com>
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Cc: Cullen Jennings <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] [rtcweb] Rejecting MediaStreamTracks in JSEP
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On Mon, Feb 3, 2014 at 1:10 PM, Adam Roach <adam@nostrum.com> wrote:

>  On 2/3/14 15:03, Justin Uberti wrote:
>
>
> On Mon, Feb 3, 2014 at 12:42 PM, Adam Roach <adam@nostrum.com> wrote:
>
>> On 2/3/14 14:25, Justin Uberti wrote:
>>
>>> It's not clear to me that adding a new attribute would solve any problem
>>> that appid doesn't solve; you still need to deal with the "other end not
>>> using it" issue.
>>>
>>
>>  Regardless of whether you think there's a problem in the legacy interop
>> case, the big benefit here is that an explicit indicator by definition
>> makes the operation explicit, and that in-and-of-itself is a good thing.
>>
>> The core issue is that you're trying to make the operation implicit; your
>> proposal edges awfully close to the DWIM model of design, which we know
>> causes problems when we try to add further extensions.
>
>
>  Good point, but since we are still defining exactly what appId does, I
> think we have the opportunity to make it explicit.
>
>
> I think I agree, but this isn't congruent with the proposal you made
> before. Omitting app-id as a means of rejecting a stream is an implicit
> operation, regardless of how well it's documented.
>
> If you want to have something like "a=recv-appId:reject", that would be
> suitably explicit, but I really have a hard time imagining how something
> like that would be preferable to just having a new attribute.
>

I was thinking a=recv-appId:0, which is both explicit, and has parallels
with the zero port form of rejecting a m= line.

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Mon, Feb 3, 2014 at 1:10 PM, Adam Roach <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:adam@nostrum.com" target=3D"_blank">adam@nostrum.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">
 =20
   =20
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000"><div><div class=3D"h5">
    <div>On 2/3/14 15:03, Justin Uberti wrote:<br>
    </div>
    <blockquote type=3D"cite">
      <div dir=3D"ltr"><br>
        <div class=3D"gmail_extra">On Mon, Feb 3, 2014 at 12:42 PM, Adam
          Roach <span dir=3D"ltr">&lt;<a href=3D"mailto:adam@nostrum.com" t=
arget=3D"_blank">adam@nostrum.com</a>&gt;</span>
          wrote:<br>
          <div class=3D"gmail_quote">
            <blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bo=
rder-left:1px #ccc solid;padding-left:1ex">
              <div>On 2/3/14 14:25, Justin Uberti wrote:<br>
                <blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8e=
x;border-left:1px #ccc solid;padding-left:1ex">
                  It&#39;s not clear to me that adding a new attribute woul=
d
                  solve any problem that appid doesn&#39;t solve; you still
                  need to deal with the &quot;other end not using it&quot; =
issue.<br>
                </blockquote>
                <br>
              </div>
              Regardless of whether you think there&#39;s a problem in the
              legacy interop case, the big benefit here is that an
              explicit indicator by definition makes the operation
              explicit, and that in-and-of-itself is a good thing.<br>
              <br>
              The core issue is that you&#39;re trying to make the operatio=
n
              implicit; your proposal edges awfully close to the DWIM
              model of design, which we know causes problems when we try
              to add further extensions.</blockquote>
            <div>
              <br>
            </div>
            <div>Good point, but since we are still defining exactly
              what appId does, I think we have the opportunity to make
              it explicit.=C2=A0</div>
          </div>
          <br>
        </div>
      </div>
    </blockquote>
    <br></div></div>
    I think I agree, but this isn&#39;t congruent with the proposal you mad=
e
    before. Omitting app-id as a means of rejecting a stream is an
    implicit operation, regardless of how well it&#39;s documented.<br>
    <br>
    If you want to have something like &quot;a=3Drecv-appId:reject&quot;, t=
hat would
    be suitably explicit, but I really have a hard time imagining how
    something like that would be preferable to just having a new
    attribute.</div></blockquote><div><br></div><div>I was thinking a=3Drec=
v-appId:0, which is both explicit, and has parallels with the zero port for=
m of rejecting a m=3D line.=C2=A0</div></div><br></div></div>

--001a1136b6107ac4b404f1871e5f--

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Subject: Re: [MMUSIC] [rtcweb] Rejecting MediaStreamTracks in JSEP
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On 2/3/14 15:22, Justin Uberti wrote:
>
> I was thinking a=recv-appId:0, which is both explicit, and has 
> parallels with the zero port form of rejecting a m= line.
>

I have no objection to such an approach.

/a

From jonathan@vidyo.com  Mon Feb  3 14:32:22 2014
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From: Jonathan Lennox <jonathan@vidyo.com>
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Subject: Re: [MMUSIC] [rtcweb] Rejecting MediaStreamTracks in JSEP
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On Feb 3, 2014, at 2:27 PM, Justin Uberti <juberti@google.com<mailto:jubert=
i@google.com>> wrote:

Also, this essentially makes recv-appId the converse of MSID; for simplicit=
y, I think we should consider unifying these concepts into a single documen=
t.

AppID (as I envision it, as the author) and MSID (as I understand it) have =
rather different scopes, which are visible once sources are sent using more=
 than one packet stream.

For example, an m-line which has both a primary encoded stream (of media) a=
nd a redundancy stream (of rtx packets) would have two appid values, but on=
ly one msid.  On the receive side, if you wanted to receive both a primary =
encoded stream and a redundancy stream, you would have two recv-appid value=
s.

As such, appid and msid are solving different (though related) problems, an=
d I don't think unifying them is a good idea.

Perhaps we need a recv-msid, explicitly?


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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; ">
<div>On Feb 3, 2014, at 2:27 PM, Justin Uberti &lt;<a href=3D"mailto:jubert=
i@google.com">juberti@google.com</a>&gt;&nbsp;wrote:</div>
<div><br>
<blockquote type=3D"cite">
<div dir=3D"ltr">
<div style=3D"font-family:arial,sans-serif;font-size:13px">Also, this essen=
tially makes recv-appId the converse of MSID; for simplicity, I think we sh=
ould consider unifying these concepts into a single document.</div>
</div>
</blockquote>
</div>
<div><br>
</div>
AppID (as I envision it, as the author) and MSID (as I understand it) have =
rather different scopes, which are visible once sources are sent using more=
 than one packet stream.
<div><br>
</div>
<div>For example, an m-line which has both a primary encoded stream (of med=
ia) and a redundancy stream (of rtx packets) would have two appid values, b=
ut only one msid. &nbsp;On the receive side, if you wanted to receive both =
a primary encoded stream and a redundancy
 stream, you would have two recv-appid values.</div>
<div><br>
</div>
<div>As such, appid and msid are solving different (though related) problem=
s, and I don't think unifying them is a good idea.</div>
<div><br>
</div>
<div>Perhaps we need a recv-msid, explicitly?<br>
<div><br>
</div>
</div>
</body>
</html>

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From: Justin Uberti <juberti@google.com>
Date: Mon, 3 Feb 2014 15:00:26 -0800
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To: Jonathan Lennox <jonathan@vidyo.com>
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Cc: Cullen Jennings <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] [rtcweb] Rejecting MediaStreamTracks in JSEP
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On Mon, Feb 3, 2014 at 2:32 PM, Jonathan Lennox <jonathan@vidyo.com> wrote:

>  On Feb 3, 2014, at 2:27 PM, Justin Uberti <juberti@google.com> wrote:
>
>  Also, this essentially makes recv-appId the converse of MSID; for
> simplicity, I think we should consider unifying these concepts into a
> single document.
>
>
>  AppID (as I envision it, as the author) and MSID (as I understand it)
> have rather different scopes, which are visible once sources are sent using
> more than one packet stream.
>
>  For example, an m-line which has both a primary encoded stream (of
> media) and a redundancy stream (of rtx packets) would have two appid
> values, but only one msid.  On the receive side, if you wanted to receive
> both a primary encoded stream and a redundancy stream, you would have two
> recv-appid values.
>
>  As such, appid and msid are solving different (though related) problems,
> and I don't think unifying them is a good idea.
>
>  Perhaps we need a recv-msid, explicitly?
>

That would be fine by me - it would allow a single invention (msid +
recv-msid) to focus on solving the three identified problems with MSTs in
unified-plan:
- correlation of a MediaStreamTrack to m= line in signaling (msid)
- correlation of packets from a MST to m= line ahead of signaling
(recv-msid)
- rejection of an individual MST (recv-msid)

Based on your description, appid seems to go one or two levels down from a
MST to an individual RTP flow, but we can let that develop on its own,
separate from the timetable of webrtc 1.0 and unified plan.

--047d7b6da02872d46104f1887ec4
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Mon, Feb 3, 2014 at 2:32 PM, Jonathan Lennox <span dir=3D"ltr">&=
lt;<a href=3D"mailto:jonathan@vidyo.com" target=3D"_blank">jonathan@vidyo.c=
om</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 style=3D"word-wrap:break-word">
<div>On Feb 3, 2014, at 2:27 PM, Justin Uberti &lt;<a href=3D"mailto:jubert=
i@google.com" target=3D"_blank">juberti@google.com</a>&gt;=C2=A0wrote:</div=
><div class=3D"im">
<div><br>
<blockquote type=3D"cite">
<div dir=3D"ltr">
<div style=3D"font-family:arial,sans-serif;font-size:13px">Also, this essen=
tially makes recv-appId the converse of MSID; for simplicity, I think we sh=
ould consider unifying these concepts into a single document.</div>
</div>
</blockquote>
</div>
<div><br>
</div></div>
AppID (as I envision it, as the author) and MSID (as I understand it) have =
rather different scopes, which are visible once sources are sent using more=
 than one packet stream.
<div><br>
</div>
<div>For example, an m-line which has both a primary encoded stream (of med=
ia) and a redundancy stream (of rtx packets) would have two appid values, b=
ut only one msid. =C2=A0On the receive side, if you wanted to receive both =
a primary encoded stream and a redundancy
 stream, you would have two recv-appid values.</div>
<div><br>
</div>
<div>As such, appid and msid are solving different (though related) problem=
s, and I don&#39;t think unifying them is a good idea.</div>
<div><br>
</div>
<div>Perhaps we need a recv-msid, explicitly?<br></div></div></blockquote><=
div><br></div><div>That would be fine by me - it would allow a single inven=
tion (msid + recv-msid) to focus on solving the three identified problems w=
ith MSTs in unified-plan:</div>

<div>- correlation of a MediaStreamTrack to m=3D line in signaling (msid)</=
div><div>- correlation of packets from a MST to m=3D line ahead of signalin=
g (recv-msid)</div><div>- rejection of an individual MST (recv-msid)</div>
<div>
<br></div><div>Based on your description, appid seems to go one or two leve=
ls down from a MST to an individual RTP flow, but we can let that develop o=
n its own, separate from the timetable of webrtc 1.0 and unified plan.</div=
>

<div><br></div></div><br></div></div>

--047d7b6da02872d46104f1887ec4--

From juberti@google.com  Mon Feb  3 15:03:59 2014
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From: Justin Uberti <juberti@google.com>
Date: Mon, 3 Feb 2014 15:03:36 -0800
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To: Jonathan Lennox <jonathan@vidyo.com>, Harald Alvestrand <hta@google.com>
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Cc: Cullen Jennings <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] [rtcweb] Rejecting MediaStreamTracks in JSEP
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--089e011844dad5304004f1888911
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+Harald as this affects the MSID doc


On Mon, Feb 3, 2014 at 3:00 PM, Justin Uberti <juberti@google.com> wrote:

>
>
>
> On Mon, Feb 3, 2014 at 2:32 PM, Jonathan Lennox <jonathan@vidyo.com>wrote:
>
>>  On Feb 3, 2014, at 2:27 PM, Justin Uberti <juberti@google.com> wrote:
>>
>>  Also, this essentially makes recv-appId the converse of MSID; for
>> simplicity, I think we should consider unifying these concepts into a
>> single document.
>>
>>
>>  AppID (as I envision it, as the author) and MSID (as I understand it)
>> have rather different scopes, which are visible once sources are sent using
>> more than one packet stream.
>>
>>  For example, an m-line which has both a primary encoded stream (of
>> media) and a redundancy stream (of rtx packets) would have two appid
>> values, but only one msid.  On the receive side, if you wanted to receive
>> both a primary encoded stream and a redundancy stream, you would have two
>> recv-appid values.
>>
>>  As such, appid and msid are solving different (though related)
>> problems, and I don't think unifying them is a good idea.
>>
>>  Perhaps we need a recv-msid, explicitly?
>>
>
> That would be fine by me - it would allow a single invention (msid +
> recv-msid) to focus on solving the three identified problems with MSTs in
> unified-plan:
> - correlation of a MediaStreamTrack to m= line in signaling (msid)
> - correlation of packets from a MST to m= line ahead of signaling
> (recv-msid)
> - rejection of an individual MST (recv-msid)
>
> Based on your description, appid seems to go one or two levels down from a
> MST to an individual RTP flow, but we can let that develop on its own,
> separate from the timetable of webrtc 1.0 and unified plan.
>
>
>

--089e011844dad5304004f1888911
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">+Harald as this affects the MSID doc</div><div class=3D"gm=
ail_extra"><br><br><div class=3D"gmail_quote">On Mon, Feb 3, 2014 at 3:00 P=
M, Justin Uberti <span dir=3D"ltr">&lt;<a href=3D"mailto:juberti@google.com=
" target=3D"_blank">juberti@google.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 dir=3D"ltr"><br><div class=3D"gmail_ext=
ra"><br><br><div class=3D"gmail_quote"><div><div class=3D"h5">On Mon, Feb 3=
, 2014 at 2:32 PM, Jonathan Lennox <span dir=3D"ltr">&lt;<a href=3D"mailto:=
jonathan@vidyo.com" target=3D"_blank">jonathan@vidyo.com</a>&gt;</span> wro=
te:<br>


<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">



<div style=3D"word-wrap:break-word">
<div>On Feb 3, 2014, at 2:27 PM, Justin Uberti &lt;<a href=3D"mailto:jubert=
i@google.com" target=3D"_blank">juberti@google.com</a>&gt;=C2=A0wrote:</div=
><div>
<div><br>
<blockquote type=3D"cite">
<div dir=3D"ltr">
<div style=3D"font-family:arial,sans-serif;font-size:13px">Also, this essen=
tially makes recv-appId the converse of MSID; for simplicity, I think we sh=
ould consider unifying these concepts into a single document.</div>
</div>
</blockquote>
</div>
<div><br>
</div></div>
AppID (as I envision it, as the author) and MSID (as I understand it) have =
rather different scopes, which are visible once sources are sent using more=
 than one packet stream.
<div><br>
</div>
<div>For example, an m-line which has both a primary encoded stream (of med=
ia) and a redundancy stream (of rtx packets) would have two appid values, b=
ut only one msid. =C2=A0On the receive side, if you wanted to receive both =
a primary encoded stream and a redundancy
 stream, you would have two recv-appid values.</div>
<div><br>
</div>
<div>As such, appid and msid are solving different (though related) problem=
s, and I don&#39;t think unifying them is a good idea.</div>
<div><br>
</div>
<div>Perhaps we need a recv-msid, explicitly?<br></div></div></blockquote><=
div><br></div></div></div><div>That would be fine by me - it would allow a =
single invention (msid + recv-msid) to focus on solving the three identifie=
d problems with MSTs in unified-plan:</div>


<div>- correlation of a MediaStreamTrack to m=3D line in signaling (msid)</=
div><div>- correlation of packets from a MST to m=3D line ahead of signalin=
g (recv-msid)</div><div>- rejection of an individual MST (recv-msid)</div>

<div>
<br></div><div>Based on your description, appid seems to go one or two leve=
ls down from a MST to an individual RTP flow, but we can let that develop o=
n its own, separate from the timetable of webrtc 1.0 and unified plan.</div=
>


<div><br></div></div><br></div></div>
</blockquote></div><br></div>

--089e011844dad5304004f1888911--

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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Adam Roach <adam@nostrum.com>
Thread-Topic: [rtcweb] Rejecting MediaStreamTracks in JSEP
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Date: Mon, 3 Feb 2014 23:04:07 +0000
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Subject: Re: [MMUSIC] [rtcweb] Rejecting MediaStreamTracks in JSEP
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On Feb 3, 2014, at 3:10 PM, Adam Roach <adam@nostrum.com> wrote:

> On 2/3/14 15:22, Justin Uberti wrote:
>>=20
>> I was thinking a=3Drecv-appId:0, which is both explicit, and has paralle=
ls with the zero port form of rejecting a m=3D line.
>>=20
>=20
> I have no objection to such an approach.
>=20
> /a

+1 with Adam on we need something explicitly somewhere.=20

If we do it with some combination msid / app-id - I just want to make sure =
the behavior is still reasonable with things that don=92t low how to signal=
 ssrc.=20

 =

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From: Jonathan Lennox <jonathan@vidyo.com>
To: Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] Rejecting MediaStreamTracks in JSEP
Thread-Index: AQHPIS46pADKsTFIzECMWWmLIWbna5qkgc6AgAAH3gCAAAMqAA==
Date: Mon, 3 Feb 2014 23:11:46 +0000
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Cc: Cullen Jennings <fluffy@cisco.com>, Harald Alvestrand <hta@google.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] [rtcweb] Rejecting MediaStreamTracks in JSEP
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On Feb 3, 2014, at 6:00 PM, Justin Uberti <juberti@google.com<mailto:jubert=
i@google.com>>
 wrote:

On Mon, Feb 3, 2014 at 2:32 PM, Jonathan Lennox <jonathan@vidyo.com<mailto:=
jonathan@vidyo.com>> wrote:
On Feb 3, 2014, at 2:27 PM, Justin Uberti <juberti@google.com<mailto:jubert=
i@google.com>> wrote:

Also, this essentially makes recv-appId the converse of MSID; for simplicit=
y, I think we should consider unifying these concepts into a single documen=
t.

AppID (as I envision it, as the author) and MSID (as I understand it) have =
rather different scopes, which are visible once sources are sent using more=
 than one packet stream.

For example, an m-line which has both a primary encoded stream (of media) a=
nd a redundancy stream (of rtx packets) would have two appid values, but on=
ly one msid.  On the receive side, if you wanted to receive both a primary =
encoded stream and a redundancy stream, you would have two recv-appid value=
s.

As such, appid and msid are solving different (though related) problems, an=
d I don't think unifying them is a good idea.

Perhaps we need a recv-msid, explicitly?

That would be fine by me - it would allow a single invention (msid + recv-m=
sid) to focus on solving the three identified problems with MSTs in unified=
-plan:
- correlation of a MediaStreamTrack to m=3D line in signaling (msid)
- correlation of packets from a MST to m=3D line ahead of signaling (recv-m=
sid)
- rejection of an individual MST (recv-msid)

Based on your description, appid seems to go one or two levels down from a =
MST to an individual RTP flow, but we can let that develop on its own, sepa=
rate from the timetable of webrtc 1.0 and unified plan.

Points 1 and 3 make sense to me, and are I think what msid is for; but I do=
n't think point 2 will work.

I don't think a single, m-line-wide, identifier will work for the correlati=
on-ahead-of-signaling problem, once you have multiple RTP flows in the m-li=
ne.  That's why we've designed appid the way we have.

I think appid is the right solution to point 2.  More specifically, the pro=
blem appid is solving is better expressed as the correlation of packets fro=
m an RTP flow to an m-line; the correlation from the m-line to a MST is a s=
eparate step, and is msid's job.



--_000_9E51C939CCFE4EA683BF9B871D64CDA0vidyocom_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; ">
<br>
<div>
<div>On Feb 3, 2014, at 6:00 PM, Justin Uberti &lt;<a href=3D"mailto:jubert=
i@google.com">juberti@google.com</a>&gt;</div>
<div>&nbsp;wrote:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite">
<div dir=3D"ltr">
<div class=3D"gmail_extra">On Mon, Feb 3, 2014 at 2:32 PM, Jonathan Lennox =
<span dir=3D"ltr">
&lt;<a href=3D"mailto:jonathan@vidyo.com" target=3D"_blank">jonathan@vidyo.=
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:1p=
x #ccc solid;padding-left:1ex">
<div style=3D"word-wrap:break-word">
<div>On Feb 3, 2014, at 2:27 PM, Justin Uberti &lt;<a href=3D"mailto:jubert=
i@google.com" target=3D"_blank">juberti@google.com</a>&gt;&nbsp;wrote:</div=
>
<div class=3D"im">
<div><br>
<blockquote type=3D"cite">
<div dir=3D"ltr">
<div style=3D"font-family:arial,sans-serif;font-size:13px">Also, this essen=
tially makes recv-appId the converse of MSID; for simplicity, I think we sh=
ould consider unifying these concepts into a single document.</div>
</div>
</blockquote>
</div>
<div><br>
</div>
</div>
AppID (as I envision it, as the author) and MSID (as I understand it) have =
rather different scopes, which are visible once sources are sent using more=
 than one packet stream.
<div><br>
</div>
<div>For example, an m-line which has both a primary encoded stream (of med=
ia) and a redundancy stream (of rtx packets) would have two appid values, b=
ut only one msid. &nbsp;On the receive side, if you wanted to receive both =
a primary encoded stream and a redundancy
 stream, you would have two recv-appid values.</div>
<div><br>
</div>
<div>As such, appid and msid are solving different (though related) problem=
s, and I don't think unifying them is a good idea.</div>
<div><br>
</div>
<div>Perhaps we need a recv-msid, explicitly?<br>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>That would be fine by me - it would allow a single invention (msid &#4=
3; recv-msid) to focus on solving the three identified problems with MSTs i=
n unified-plan:</div>
<div>- correlation of a MediaStreamTrack to m=3D line in signaling (msid)</=
div>
<div>- correlation of packets from a MST to m=3D line ahead of signaling (r=
ecv-msid)</div>
<div>- rejection of an individual MST (recv-msid)</div>
<div><br>
</div>
<div>Based on your description, appid seems to go one or two levels down fr=
om a MST to an individual RTP flow, but we can let that develop on its own,=
 separate from the timetable of webrtc 1.0 and unified plan.</div>
</div>
</div>
</div>
</blockquote>
<br>
</div>
<div>Points 1 and 3 make sense to me, and are I think what msid is for; but=
 I don't think point 2 will work.</div>
<div><br>
</div>
<div>I don't think a single, m-line-wide, identifier will work for the corr=
elation-ahead-of-signaling&nbsp;problem, once you have multiple RTP flows i=
n the m-line. &nbsp;That's why we've designed appid the way we have.</div>
<div><br>
</div>
<div>I think appid is the right solution to point 2. &nbsp;More specificall=
y, the problem appid is solving is better expressed as the correlation of p=
ackets from an RTP flow to an m-line; the correlation from the m-line to a =
MST is a separate step, and is msid's
 job.</div>
<div><br>
</div>
<br>
</body>
</html>

--_000_9E51C939CCFE4EA683BF9B871D64CDA0vidyocom_--

From juberti@google.com  Mon Feb  3 16:05:39 2014
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From: Justin Uberti <juberti@google.com>
Date: Mon, 3 Feb 2014 16:05:15 -0800
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To: Jonathan Lennox <jonathan@vidyo.com>
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Cc: Cullen Jennings <fluffy@cisco.com>, Harald Alvestrand <hta@google.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] [rtcweb] Rejecting MediaStreamTracks in JSEP
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On Mon, Feb 3, 2014 at 3:11 PM, Jonathan Lennox <jonathan@vidyo.com> wrote:

>
>  On Feb 3, 2014, at 6:00 PM, Justin Uberti <juberti@google.com>
>  wrote:
>
>  On Mon, Feb 3, 2014 at 2:32 PM, Jonathan Lennox <jonathan@vidyo.com>wrote:
>
>>  On Feb 3, 2014, at 2:27 PM, Justin Uberti <juberti@google.com> wrote:
>>
>>  Also, this essentially makes recv-appId the converse of MSID; for
>> simplicity, I think we should consider unifying these concepts into a
>> single document.
>>
>>
>>  AppID (as I envision it, as the author) and MSID (as I understand it)
>> have rather different scopes, which are visible once sources are sent using
>> more than one packet stream.
>>
>>  For example, an m-line which has both a primary encoded stream (of
>> media) and a redundancy stream (of rtx packets) would have two appid
>> values, but only one msid.  On the receive side, if you wanted to receive
>> both a primary encoded stream and a redundancy stream, you would have two
>> recv-appid values.
>>
>>  As such, appid and msid are solving different (though related)
>> problems, and I don't think unifying them is a good idea.
>>
>>  Perhaps we need a recv-msid, explicitly?
>>
>
>  That would be fine by me - it would allow a single invention (msid +
> recv-msid) to focus on solving the three identified problems with MSTs in
> unified-plan:
> - correlation of a MediaStreamTrack to m= line in signaling (msid)
> - correlation of packets from a MST to m= line ahead of signaling
> (recv-msid)
> - rejection of an individual MST (recv-msid)
>
>  Based on your description, appid seems to go one or two levels down from
> a MST to an individual RTP flow, but we can let that develop on its own,
> separate from the timetable of webrtc 1.0 and unified plan.
>
>
>  Points 1 and 3 make sense to me, and are I think what msid is for; but I
> don't think point 2 will work.
>
>  I don't think a single, m-line-wide, identifier will work for the
> correlation-ahead-of-signaling problem, once you have multiple RTP flows in
> the m-line.  That's why we've designed appid the way we have.
>

>  I think appid is the right solution to point 2.  More specifically, the
> problem appid is solving is better expressed as the correlation of packets
> from an RTP flow to an m-line; the correlation from the m-line to a MST is
> a separate step, and is msid's job.
>

Agree that msid should map m-line to MST, but it would seem recv-msid could
map RTP flow to m=line as well as indicate whether the receiver is in fact
interested in receiving the MST associated with said m= line. Note that
recv-msid doesn't specify an actual MSID; its purpose is to indicate that
the receiver wants to receive data from the remote side's MST, and provide
a mechanism for the sender to mark the data accordingly. (Which ends up
seeming a lot like appid.)

Generally, I am hesitant to depend on 3 things if we think we could get by
with only 2.

--001a1133ed3a51b0a304f1896696
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Mon, Feb 3, 2014 at 3:11 PM, Jonathan Lennox <span dir=3D"ltr">&=
lt;<a href=3D"mailto:jonathan@vidyo.com" target=3D"_blank">jonathan@vidyo.c=
om</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 style=3D"word-wrap:break-word">
<br>
<div>
<div>On Feb 3, 2014, at 6:00 PM, Justin Uberti &lt;<a href=3D"mailto:jubert=
i@google.com" target=3D"_blank">juberti@google.com</a>&gt;</div><div><div c=
lass=3D"h5">
<div>=C2=A0wrote:</div>
<br>
<blockquote type=3D"cite">
<div dir=3D"ltr">
<div class=3D"gmail_extra">On Mon, Feb 3, 2014 at 2:32 PM, Jonathan Lennox =
<span dir=3D"ltr">
&lt;<a href=3D"mailto:jonathan@vidyo.com" target=3D"_blank">jonathan@vidyo.=
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:1p=
x #ccc solid;padding-left:1ex">
<div style=3D"word-wrap:break-word">
<div>On Feb 3, 2014, at 2:27 PM, Justin Uberti &lt;<a href=3D"mailto:jubert=
i@google.com" target=3D"_blank">juberti@google.com</a>&gt;=C2=A0wrote:</div=
>
<div>
<div><br>
<blockquote type=3D"cite">
<div dir=3D"ltr">
<div style=3D"font-family:arial,sans-serif;font-size:13px">Also, this essen=
tially makes recv-appId the converse of MSID; for simplicity, I think we sh=
ould consider unifying these concepts into a single document.</div>
</div>
</blockquote>
</div>
<div><br>
</div>
</div>
AppID (as I envision it, as the author) and MSID (as I understand it) have =
rather different scopes, which are visible once sources are sent using more=
 than one packet stream.
<div><br>
</div>
<div>For example, an m-line which has both a primary encoded stream (of med=
ia) and a redundancy stream (of rtx packets) would have two appid values, b=
ut only one msid. =C2=A0On the receive side, if you wanted to receive both =
a primary encoded stream and a redundancy
 stream, you would have two recv-appid values.</div>
<div><br>
</div>
<div>As such, appid and msid are solving different (though related) problem=
s, and I don&#39;t think unifying them is a good idea.</div>
<div><br>
</div>
<div>Perhaps we need a recv-msid, explicitly?<br>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>That would be fine by me - it would allow a single invention (msid + r=
ecv-msid) to focus on solving the three identified problems with MSTs in un=
ified-plan:</div>
<div>- correlation of a MediaStreamTrack to m=3D line in signaling (msid)</=
div>
<div>- correlation of packets from a MST to m=3D line ahead of signaling (r=
ecv-msid)</div>
<div>- rejection of an individual MST (recv-msid)</div>
<div><br>
</div>
<div>Based on your description, appid seems to go one or two levels down fr=
om a MST to an individual RTP flow, but we can let that develop on its own,=
 separate from the timetable of webrtc 1.0 and unified plan.</div>
</div>
</div>
</div>
</blockquote>
<br>
</div></div></div>
<div>Points 1 and 3 make sense to me, and are I think what msid is for; but=
 I don&#39;t think point 2 will work.</div>
<div><br>
</div>
<div>I don&#39;t think a single, m-line-wide, identifier will work for the =
correlation-ahead-of-signaling=C2=A0problem, once you have multiple RTP flo=
ws in the m-line. =C2=A0That&#39;s why we&#39;ve designed appid the way we =
have.</div>

</div></blockquote><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 =
.8ex;border-left:1px #ccc solid;padding-left:1ex"><div style=3D"word-wrap:b=
reak-word">
<div><br>
</div>
<div>I think appid is the right solution to point 2. =C2=A0More specificall=
y, the problem appid is solving is better expressed as the correlation of p=
ackets from an RTP flow to an m-line; the correlation from the m-line to a =
MST is a separate step, and is msid&#39;s
 job.</div></div></blockquote><div><br></div><div>Agree that msid should ma=
p m-line to MST, but it would seem recv-msid could map RTP flow to m=3Dline=
 as well as indicate whether the receiver is in fact interested in receivin=
g the MST associated with said m=3D line. Note that recv-msid doesn&#39;t s=
pecify an actual MSID; its purpose is to indicate that the receiver wants t=
o receive data from the remote side&#39;s MST, and provide a mechanism for =
the sender to mark the data accordingly. (Which ends up seeming a lot like =
appid.)</div>

<div><br></div><div>Generally, I am hesitant to depend on 3 things if we th=
ink we could get by with only 2.</div></div><br></div></div>

--001a1133ed3a51b0a304f1896696--

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-----Original Message-----
From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of ext Paul Kyzivat
Sent: 31 January, 2014 20:53
To: mmusic@ietf.org
Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.txt

I just reviewed the new version. First I looked at just the diffs. But thos=
e raised some more general questions in my mind, so I re-reviewed more of t=
he draft. I'm left with some considerable confusion about how this is inten=
ded to work in practice, especially with extern correlation, but even with =
the other forms.

In particular, I am unclear of the sequencing of O/A processing, correlatio=
n processing, and signaling completion. (I realize this is an mmusic draft =
and theoretically independent of sip, but forgive me if I discuss this in t=
he context of sip.)

Normally, when first establishing a session with sip, the first O/A may occ=
ur before the signaling session is established. But early media may flow af=
ter the first O/A. Often it is expected that the user will be be aware and =
involved in this early media.

Once a session is up, it can be changed with a reinvite containing a new O/=
A. If the O/A and the reinvite are successful, then the results of the new =
O/A replace the old one. If the reinvite fails, then the session is suppose=
d to continue as it was before the reinvite. In the interim, from the sendi=
ng of the reinvite until its success or failure, things are a little confus=
ing - presumably both old and new media streams must be maintained, but whi=
ch are connected to the user is unspecified.

I see nothing in this draft that explains how the CS media, and correlation=
 verification, fit into the above process.

Consider a new call that offers CS media:

- The first O/A must complete before the CS call can be attempted.

- must the CS call complete and be correlated successfully before
   the INVITE gets a 200? If the invite is allowed to complete first,
   what happens if no CS call is established, or if it is but
   correlation fails?

[SV] I assume you mean the 200 carries the Answer. It may happen that the C=
S call arrives before the Answer, but in this case no correlation can be do=
ne. See Section 5.6.3. If the invite (I assume you mean O/A) completes firs=
t, but no CS call ever arrives, or correlation - including the "external" m=
echanism fails, the endpoint should treat the CS call as unrelated; this is=
 described in Section 5.3.3.

- when does CS media start to flow to the user as part of the
   SIP call? Is it before or after correlation? (E.g., does the
   user hear the DTMF exchange?)

[SV] CS media starts to flow after the call has been accepted, as in a norm=
al CS call. Correlation may be possible before accepting the call (for "cal=
lerid", "uuie", and "external" mechanisms), and for "dtmf" case only after =
the call has been accepted.

- if media flows to the user before correlation, then what if
   correlation fails? Will another call be accepted and correlation
   be attempted again?

[SV] I don' think we have explicitly defined this situation. If a call is a=
ccepted and *after* that correlation fails, it means that the DTMF mechanis=
m was used (for other correlation mechanisms, we know before accepting the =
call whether the correlation was successful). If the DTMF pattern is not re=
ceived correctly, as we cannot know whether it is related to the session or=
 not. Now, this raises a possible case of including both "dtmf" and "extern=
al" mechanisms in the O/A. If DTMF is successfully negotiated but it fails,=
 we can't really use the "external" mechanism since the call is already acc=
epted. Implementations could in this case release the CS call and try a new=
 O/A without the DTMF.


- what if multiple incoming calls arrive before correlation
   has been completed? (If correlation occurs before exposing the
   media to the user, then it isn't an issue.)

[SV] This would be an issue for the DTMF case, as there correlation can be =
only only after accepting the CS call. An unlikely case, but still possible=
. In this case, one of the CS calls could be related to the session, while =
other(s) would be normal CS calls. As only one CS call can be handled at a =
time, others would either wait or be terminated, depending on the callee's =
supplementary services settings.=20

- extern correlation complicates this. It isn't like the other
   correlation mechanisms - the user must be exposed to the media
   to decide if there is correlation. How would the user indicate
   "this is the wrong call" in a way that would allow correlation
   to be attempted on another call?

[SV] The external correlation mechanism does not assume the media flows bef=
ore "correlation" is made - if the user decides to accept the incoming call=
, it is considered as "correlated".=20

Now consider a call that is initially established with one RTP media stream=
. Later, a reinvite is sent with an offer proposing to to replace that with=
 a CS media stream:

- again, there must be an answer before the CS call can be attempted.


[SV] For being able to perform correlation, yes, the Answer needs to be rec=
eived.

- are you assuming that the reinvite will complete before the CS
   call does? If so, then you are in a break-before-make situation -
   the RTP session is gone before the CS session takes its place.

[SV] The O/A needs to be completed before correlation can be done, but CS c=
all may arrive before Answer. In this case the endpoint should wait until t=
he Answer is received. If the CS call does not succeed, or correlation is u=
nsuccessful, then yes there is no media as the RTP stream has been dropped.=
 But, this could also happen when re-negotiating RTP media types (endpoints=
 are not able to load correct codecs, new IP addr/port does not work etc.) =
so the situation not be specific to the CS media type.

- all the issues above also apply here.

ISTM that there is a bunch of missing explanation of how this is expected t=
o work.

[SV] Could you check the above whether my comments clarify the situation. W=
e can address them in more details in the draft as well once we know where =
more explanation is needed.

I also have a nit in section 5.3.2:

    The fourth correlation mechanism declares support for cases where
    correlation is done by external means.  Typically this means that the
    decision is left for the human user.  This is the way how some
    current conferencing systems operate: ...

"This is the way how some current conferencing systems operate"

should be:

"This is how some current conferencing systems operate"

[SV] Fill fix.

Thanks,
Simo

	Thanks,
	Paul

On 1/31/14 6:01 AM, Simo.Veikkolainen@nokia.com wrote:
> Hello WG,
>
> We have submitted a new version of=20
> http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-22
>
> To recap, there was discussion on the  MMUSIC WG list on which correlatio=
n mechanisms should be mandatory to implement (see Gonzalo's email http://w=
ww.ietf.org/mail-archive/web/mmusic/current/msg12274.html ). As stated by G=
onzalo, 3GPP is going to use this draft, more specifically the plan now is =
to use the caller id based correlation mechanism. This was the message I go=
t from Andrew after a recent 3GPP CT1 meeting.
>
> We have now made the following changes to the draft:
> -	the endpoint may choose which correlation mechanisms to implement (befo=
re all three were MTI).
> -	we have introduced a fourth correlation type ("external") which indicat=
es explicit support for cases where also some external means, typically the=
 human user, can do the correlation. This can be as a fallback in cases whe=
re no other common correlation mechanism is found, or the negotiated correl=
ation mechanisms fail due to being not supported in the PSTN network. This =
is much like current systems like Webex work.
>
> This was also the guidance from the IESG informal telechat in December wh=
ere Gonzalo presented the draft.
>
> We would like to hear from the WG whether there are any concerns with the=
se changes.
>
> On behalf of the authors,
> Simo
>
> PS. I'll add the proposed change in http://www.ietf.org/mail-archive/web/=
mmusic/current/msg12792.html to next version, slipped my mind this time.
>
>
> -----Original Message-----
> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of ext=20
> internet-drafts@ietf.org
> Sent: 31 January, 2014 12:25
> To: i-d-announce@ietf.org
> Cc: mmusic@ietf.org
> Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.txt
>
>
> A New Internet-Draft is available from the on-line Internet-Drafts direct=
ories.
>   This draft is a work item of the Multiparty Multimedia Session Control =
Working Group of the IETF.
>
>          Title           : Session Description Protocol (SDP) Extension F=
or Setting Audio and Video Media Streams Over Circuit-Switched Bearers In T=
he Public Switched Telephone Network (PSTN)
>          Authors         : Miguel A. Garcia-Martin
>                            Simo Veikkolainen
> 	Filename        : draft-ietf-mmusic-sdp-cs-22.txt
> 	Pages           : 37
> 	Date            : 2014-01-31
>
> 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-22
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sdp-cs-22
>
>
> Please note that it may take a couple of minutes from the time of submiss=
ion until the htmlized version and diff are available at tools.ietf.org.
>
> 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
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Justin Uberti <juberti@google.com>, Jonathan Lennox <jonathan@vidyo.com>
Thread-Topic: [rtcweb] Rejecting MediaStreamTracks in JSEP
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Cc: Cullen Jennings <fluffy@cisco.com>, Harald Alvestrand <hta@google.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] [rtcweb] Rejecting MediaStreamTracks in JSEP
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From: Jonathan Lennox <jonathan@vidyo.com>
To: Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] Rejecting MediaStreamTracks in JSEP
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On Feb 3, 2014, at 7:05 PM, Justin Uberti <juberti@google.com<mailto:jubert=
i@google.com>>
 wrote:

On Mon, Feb 3, 2014 at 3:11 PM, Jonathan Lennox <jonathan@vidyo.com<mailto:=
jonathan@vidyo.com>> wrote:

On Feb 3, 2014, at 6:00 PM, Justin Uberti <juberti@google.com<mailto:jubert=
i@google.com>>
 wrote:

On Mon, Feb 3, 2014 at 2:32 PM, Jonathan Lennox <jonathan@vidyo.com<mailto:=
jonathan@vidyo.com>> wrote:
On Feb 3, 2014, at 2:27 PM, Justin Uberti <juberti@google.com<mailto:jubert=
i@google.com>> wrote:

Also, this essentially makes recv-appId the converse of MSID; for simplicit=
y, I think we should consider unifying these concepts into a single documen=
t.

AppID (as I envision it, as the author) and MSID (as I understand it) have =
rather different scopes, which are visible once sources are sent using more=
 than one packet stream.

For example, an m-line which has both a primary encoded stream (of media) a=
nd a redundancy stream (of rtx packets) would have two appid values, but on=
ly one msid.  On the receive side, if you wanted to receive both a primary =
encoded stream and a redundancy stream, you would have two recv-appid value=
s.

As such, appid and msid are solving different (though related) problems, an=
d I don't think unifying them is a good idea.

Perhaps we need a recv-msid, explicitly?

That would be fine by me - it would allow a single invention (msid + recv-m=
sid) to focus on solving the three identified problems with MSTs in unified=
-plan:
- correlation of a MediaStreamTrack to m=3D line in signaling (msid)
- correlation of packets from a MST to m=3D line ahead of signaling (recv-m=
sid)
- rejection of an individual MST (recv-msid)

Based on your description, appid seems to go one or two levels down from a =
MST to an individual RTP flow, but we can let that develop on its own, sepa=
rate from the timetable of webrtc 1.0 and unified plan.

Points 1 and 3 make sense to me, and are I think what msid is for; but I do=
n't think point 2 will work.

I don't think a single, m-line-wide, identifier will work for the correlati=
on-ahead-of-signaling problem, once you have multiple RTP flows in the m-li=
ne.  That's why we've designed appid the way we have.

I think appid is the right solution to point 2.  More specifically, the pro=
blem appid is solving is better expressed as the correlation of packets fro=
m an RTP flow to an m-line; the correlation from the m-line to a MST is a s=
eparate step, and is msid's job.

Agree that msid should map m-line to MST, but it would seem recv-msid could=
 map RTP flow to m=3Dline as well as indicate whether the receiver is in fa=
ct interested in receiving the MST associated with said m=3D line. Note tha=
t recv-msid doesn't specify an actual MSID; its purpose is to indicate that=
 the receiver wants to receive data from the remote side's MST, and provide=
 a mechanism for the sender to mark the data accordingly. (Which ends up se=
eming a lot like appid.)

Generally, I am hesitant to depend on 3 things if we think we could get by =
with only 2.

I understand that, but I'm also hesitant to have one mechanism do two (it s=
eems to me) fairly unrelated things, especially where the scenarios in whic=
h the problems it's solving are applicable vary fairly widely.

As far as I can tell, the only use case anyone's identified for MSID is Web=
RTC, whereas the disambiguation problem occurs (and thus appid and recv-app=
id are applicable) for anything using Bundle with Unified-like semantics.


--_000_2DF6D66078B34F53AF39B3651E9BBB46vidyocom_
Content-Type: text/html; charset="us-ascii"
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; ">
<br>
<div>
<div>On Feb 3, 2014, at 7:05 PM, Justin Uberti &lt;<a href=3D"mailto:jubert=
i@google.com">juberti@google.com</a>&gt;</div>
<div>&nbsp;wrote:</div>
<div><br>
</div>
<blockquote type=3D"cite">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">On Mon, Feb 3, 2014 at 3:11 PM, Jonathan Lennox =
<span dir=3D"ltr">
&lt;<a href=3D"mailto:jonathan@vidyo.com" target=3D"_blank">jonathan@vidyo.=
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 style=3D"word-wrap:break-word"><br>
<div>
<div>On Feb 3, 2014, at 6:00 PM, Justin Uberti &lt;<a href=3D"mailto:jubert=
i@google.com" target=3D"_blank">juberti@google.com</a>&gt;</div>
<div>
<div class=3D"h5">
<div>&nbsp;wrote:</div>
<br>
<blockquote type=3D"cite">
<div dir=3D"ltr">
<div class=3D"gmail_extra">On Mon, Feb 3, 2014 at 2:32 PM, Jonathan Lennox =
<span dir=3D"ltr">
&lt;<a href=3D"mailto:jonathan@vidyo.com" target=3D"_blank">jonathan@vidyo.=
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:1p=
x #ccc solid;padding-left:1ex">
<div style=3D"word-wrap:break-word">
<div>On Feb 3, 2014, at 2:27 PM, Justin Uberti &lt;<a href=3D"mailto:jubert=
i@google.com" target=3D"_blank">juberti@google.com</a>&gt;&nbsp;wrote:</div=
>
<div>
<div><br>
<blockquote type=3D"cite">
<div dir=3D"ltr">
<div style=3D"font-family:arial,sans-serif;font-size:13px">Also, this essen=
tially makes recv-appId the converse of MSID; for simplicity, I think we sh=
ould consider unifying these concepts into a single document.</div>
</div>
</blockquote>
</div>
<div><br>
</div>
</div>
AppID (as I envision it, as the author) and MSID (as I understand it) have =
rather different scopes, which are visible once sources are sent using more=
 than one packet stream.
<div><br>
</div>
<div>For example, an m-line which has both a primary encoded stream (of med=
ia) and a redundancy stream (of rtx packets) would have two appid values, b=
ut only one msid. &nbsp;On the receive side, if you wanted to receive both =
a primary encoded stream and a redundancy
 stream, you would have two recv-appid values.</div>
<div><br>
</div>
<div>As such, appid and msid are solving different (though related) problem=
s, and I don't think unifying them is a good idea.</div>
<div><br>
</div>
<div>Perhaps we need a recv-msid, explicitly?<br>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>That would be fine by me - it would allow a single invention (msid &#4=
3; recv-msid) to focus on solving the three identified problems with MSTs i=
n unified-plan:</div>
<div>- correlation of a MediaStreamTrack to m=3D line in signaling (msid)</=
div>
<div>- correlation of packets from a MST to m=3D line ahead of signaling (r=
ecv-msid)</div>
<div>- rejection of an individual MST (recv-msid)</div>
<div><br>
</div>
<div>Based on your description, appid seems to go one or two levels down fr=
om a MST to an individual RTP flow, but we can let that develop on its own,=
 separate from the timetable of webrtc 1.0 and unified plan.</div>
</div>
</div>
</div>
</blockquote>
<br>
</div>
</div>
</div>
<div>Points 1 and 3 make sense to me, and are I think what msid is for; but=
 I don't think point 2 will work.</div>
<div><br>
</div>
<div>I don't think a single, m-line-wide, identifier will work for the corr=
elation-ahead-of-signaling&nbsp;problem, once you have multiple RTP flows i=
n the m-line. &nbsp;That's why we've designed appid the way we have.</div>
</div>
</blockquote>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div style=3D"word-wrap:break-word">
<div><br>
</div>
<div>I think appid is the right solution to point 2. &nbsp;More specificall=
y, the problem appid is solving is better expressed as the correlation of p=
ackets from an RTP flow to an m-line; the correlation from the m-line to a =
MST is a separate step, and is msid's
 job.</div>
</div>
</blockquote>
<div><br>
</div>
<div>Agree that msid should map m-line to MST, but it would seem recv-msid =
could map RTP flow to m=3Dline as well as indicate whether the receiver is =
in fact interested in receiving the MST associated with said m=3D line. Not=
e that recv-msid doesn't specify an
 actual MSID; its purpose is to indicate that the receiver wants to receive=
 data from the remote side's MST, and provide a mechanism for the sender to=
 mark the data accordingly. (Which ends up seeming a lot like appid.)</div>
<div><br>
</div>
<div>Generally, I am hesitant to depend on 3 things if we think we could ge=
t by with only 2.</div>
</div>
</div>
</div>
</blockquote>
<div><br>
</div>
I understand that, but I'm also hesitant to have one mechanism do two (it s=
eems to me) fairly unrelated things, especially where the scenarios in whic=
h the problems it's solving are applicable vary fairly widely.</div>
<div><br>
</div>
<div>As far as I can tell, the only use case anyone's identified for MSID i=
s WebRTC, whereas the disambiguation problem occurs (and thus appid and rec=
v-appid are applicable) for anything using Bundle with Unified-like semanti=
cs.</div>
<br>
</body>
</html>

--_000_2DF6D66078B34F53AF39B3651E9BBB46vidyocom_--

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From: Jonathan Lennox <jonathan@vidyo.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
Thread-Topic: [rtcweb] Rejecting MediaStreamTracks in JSEP
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Subject: Re: [MMUSIC] [rtcweb] Rejecting MediaStreamTracks in JSEP
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MSID doesn't affect the transport level.  Appid, however, defines an RTCP S=
DES item, and an RTP header extension.

On Feb 4, 2014, at 9:13 AM, Christer Holmberg <christer.holmberg@ericsson.c=
om<mailto:christer.holmberg@ericsson.com>>
 wrote:

Hi,

Just for my clarification: none of this would have any impact on transport =
level features related to RTCP, STUN, etc?

Regards,

Christer


From: rtcweb [mailto:rtcweb-bounces@ietf.org<mailto:bounces@ietf.org>] On B=
ehalf Of Justin Uberti
Sent: 4. helmikuuta 2014 2:05
To: Jonathan Lennox
Cc: Cullen Jennings; Harald Alvestrand; rtcweb@ietf.org<mailto:rtcweb@ietf.=
org>; mmusic
Subject: Re: [rtcweb] Rejecting MediaStreamTracks in JSEP



On Mon, Feb 3, 2014 at 3:11 PM, Jonathan Lennox <jonathan@vidyo.com<mailto:=
jonathan@vidyo.com>> wrote:

On Feb 3, 2014, at 6:00 PM, Justin Uberti <juberti@google.com<mailto:jubert=
i@google.com>>
 wrote:


On Mon, Feb 3, 2014 at 2:32 PM, Jonathan Lennox <jonathan@vidyo.com<mailto:=
jonathan@vidyo.com>> wrote:
On Feb 3, 2014, at 2:27 PM, Justin Uberti <juberti@google.com<mailto:jubert=
i@google.com>> wrote:


Also, this essentially makes recv-appId the converse of MSID; for simplicit=
y, I think we should consider unifying these concepts into a single documen=
t.

AppID (as I envision it, as the author) and MSID (as I understand it) have =
rather different scopes, which are visible once sources are sent using more=
 than one packet stream.

For example, an m-line which has both a primary encoded stream (of media) a=
nd a redundancy stream (of rtx packets) would have two appid values, but on=
ly one msid.  On the receive side, if you wanted to receive both a primary =
encoded stream and a redundancy stream, you would have two recv-appid value=
s.

As such, appid and msid are solving different (though related) problems, an=
d I don't think unifying them is a good idea.

Perhaps we need a recv-msid, explicitly?

That would be fine by me - it would allow a single invention (msid + recv-m=
sid) to focus on solving the three identified problems with MSTs in unified=
-plan:
- correlation of a MediaStreamTrack to m=3D line in signaling (msid)
- correlation of packets from a MST to m=3D line ahead of signaling (recv-m=
sid)
- rejection of an individual MST (recv-msid)

Based on your description, appid seems to go one or two levels down from a =
MST to an individual RTP flow, but we can let that develop on its own, sepa=
rate from the timetable of webrtc 1.0 and unified plan.

Points 1 and 3 make sense to me, and are I think what msid is for; but I do=
n't think point 2 will work.

I don't think a single, m-line-wide, identifier will work for the correlati=
on-ahead-of-signaling problem, once you have multiple RTP flows in the m-li=
ne.  That's why we've designed appid the way we have.

I think appid is the right solution to point 2.  More specifically, the pro=
blem appid is solving is better expressed as the correlation of packets fro=
m an RTP flow to an m-line; the correlation from the m-line to a MST is a s=
eparate step, and is msid's job.

Agree that msid should map m-line to MST, but it would seem recv-msid could=
 map RTP flow to m=3Dline as well as indicate whether the receiver is in fa=
ct interested in receiving the MST associated with said m=3D line. Note tha=
t recv-msid doesn't specify an actual MSID; its purpose is to indicate that=
 the receiver wants to receive data from the remote side's MST, and provide=
 a mechanism for the sender to mark the data accordingly. (Which ends up se=
eming a lot like appid.)

Generally, I am hesitant to depend on 3 things if we think we could get by =
with only 2.



--_000_B9399713DDEF4680B5E4E9C817372133vidyocom_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
<base href=3D"x-msg://35/">
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; ">
<div>MSID doesn't affect the transport level. &nbsp;Appid, however, defines=
 an RTCP SDES item, and an RTP header extension.</div>
<br>
<div>
<div>On Feb 4, 2014, at 9:13 AM, Christer Holmberg &lt;<a href=3D"mailto:ch=
rister.holmberg@ericsson.com">christer.holmberg@ericsson.com</a>&gt;</div>
<div>&nbsp;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: He=
lvetica; font-size: medium; font-style: normal; font-variant: normal; font-=
weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; te=
xt-align: -webkit-auto; text-indent: 0px; text-transform: none; white-space=
: normal; widows: 2; word-spacing: 0px; -webkit-text-size-adjust: auto; -we=
bkit-text-stroke-width: 0px; ">
<div class=3D"WordSection1" style=3D"page: WordSection1; ">
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<span style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: rg=
b(31, 73, 125); ">Hi,<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<span style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: rg=
b(31, 73, 125); ">&nbsp;</span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<span style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: rg=
b(31, 73, 125); ">Just for my clarification: none of this would have any im=
pact on transport level features related to RTCP, STUN, etc?<o:p></o:p></sp=
an></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<span style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: rg=
b(31, 73, 125); ">&nbsp;</span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<span style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: rg=
b(31, 73, 125); ">Regards,<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<span style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: rg=
b(31, 73, 125); ">&nbsp;</span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<span style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: rg=
b(31, 73, 125); ">Christer<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<span style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: rg=
b(31, 73, 125); ">&nbsp;</span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<span style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: rg=
b(31, 73, 125); ">&nbsp;</span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<b><span style=3D"font-size: 10pt; font-family: Tahoma, sans-serif; ">From:=
</span></b><span style=3D"font-size: 10pt; font-family: Tahoma, sans-serif;=
 "><span class=3D"Apple-converted-space">&nbsp;</span>rtcweb [mailto:rtcweb=
-<a href=3D"mailto:bounces@ietf.org" style=3D"color: purple; text-decoratio=
n: underline; ">bounces@ietf.org</a>]<span class=3D"Apple-converted-space">=
&nbsp;</span><b>On
 Behalf Of<span class=3D"Apple-converted-space">&nbsp;</span></b>Justin Ube=
rti<br>
<b>Sent:</b><span class=3D"Apple-converted-space">&nbsp;</span>4. helmikuut=
a 2014 2:05<br>
<b>To:</b><span class=3D"Apple-converted-space">&nbsp;</span>Jonathan Lenno=
x<br>
<b>Cc:</b><span class=3D"Apple-converted-space">&nbsp;</span>Cullen Jenning=
s; Harald Alvestrand;
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>; mmusic<br>
<b>Subject:</b><span class=3D"Apple-converted-space">&nbsp;</span>Re: [rtcw=
eb] Rejecting MediaStreamTracks in JSEP<o:p></o:p></span></div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<o:p>&nbsp;</o:p></div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<o:p>&nbsp;</o:p></div>
<div>
<p class=3D"MsoNormal" style=3D"margin: 0cm 0cm 12pt; font-size: 12pt; font=
-family: 'Times New Roman', serif; ">
<o:p>&nbsp;</o:p></p>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
On Mon, Feb 3, 2014 at 3:11 PM, Jonathan Lennox &lt;<a href=3D"mailto:jonat=
han@vidyo.com" target=3D"_blank" style=3D"color: purple; text-decoration: u=
nderline; ">jonathan@vidyo.com</a>&gt; wrote:<o:p></o:p></div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<o:p>&nbsp;</o:p></div>
<div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
On Feb 3, 2014, at 6:00 PM, Justin Uberti &lt;<a href=3D"mailto:juberti@goo=
gle.com" target=3D"_blank" style=3D"color: purple; text-decoration: underli=
ne; ">juberti@google.com</a>&gt;<o:p></o:p></div>
</div>
<div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
&nbsp;wrote:<o:p></o:p></div>
</div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<br>
<br>
<o:p></o:p></div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
On Mon, Feb 3, 2014 at 2:32 PM, Jonathan Lennox &lt;<a href=3D"mailto:jonat=
han@vidyo.com" target=3D"_blank" style=3D"color: purple; text-decoration: u=
nderline; ">jonathan@vidyo.com</a>&gt; wrote:<o:p></o:p></div>
<div>
<blockquote style=3D"border-style: none none none solid; border-left-width:=
 1pt; border-left-color: rgb(204, 204, 204); padding: 0cm 0cm 0cm 6pt; marg=
in-left: 4.8pt; margin-right: 0cm; ">
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
On Feb 3, 2014, at 2:27 PM, Justin Uberti &lt;<a href=3D"mailto:juberti@goo=
gle.com" target=3D"_blank" style=3D"color: purple; text-decoration: underli=
ne; ">juberti@google.com</a>&gt;&nbsp;wrote:<o:p></o:p></div>
</div>
<div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<br>
<br>
<o:p></o:p></div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<span style=3D"font-size: 10pt; font-family: Arial, sans-serif; ">Also, thi=
s essentially makes recv-appId the converse of MSID; for simplicity, I thin=
k we should consider unifying these concepts into a single document.<o:p></=
o:p></span></div>
</div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<o:p>&nbsp;</o:p></div>
</div>
</div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
AppID (as I envision it, as the author) and MSID (as I understand it) have =
rather different scopes, which are visible once sources are sent using more=
 than one packet stream.<o:p></o:p></div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<o:p>&nbsp;</o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
For example, an m-line which has both a primary encoded stream (of media) a=
nd a redundancy stream (of rtx packets) would have two appid values, but on=
ly one msid. &nbsp;On the receive side, if you wanted to receive both a pri=
mary encoded stream and a redundancy
 stream, you would have two recv-appid values.<o:p></o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<o:p>&nbsp;</o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
As such, appid and msid are solving different (though related) problems, an=
d I don't think unifying them is a good idea.<o:p></o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<o:p>&nbsp;</o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
Perhaps we need a recv-msid, explicitly?<o:p></o:p></div>
</div>
</blockquote>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<o:p>&nbsp;</o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
That would be fine by me - it would allow a single invention (msid &#43; re=
cv-msid) to focus on solving the three identified problems with MSTs in uni=
fied-plan:<o:p></o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
- correlation of a MediaStreamTrack to m=3D line in signaling (msid)<o:p></=
o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
- correlation of packets from a MST to m=3D line ahead of signaling (recv-m=
sid)<o:p></o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
- rejection of an individual MST (recv-msid)<o:p></o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<o:p>&nbsp;</o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
Based on your description, appid seems to go one or two levels down from a =
MST to an individual RTP flow, but we can let that develop on its own, sepa=
rate from the timetable of webrtc 1.0 and unified plan.<o:p></o:p></div>
</div>
</div>
</div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<o:p>&nbsp;</o:p></div>
</div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
Points 1 and 3 make sense to me, and are I think what msid is for; but I do=
n't think point 2 will work.<o:p></o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<o:p>&nbsp;</o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
I don't think a single, m-line-wide, identifier will work for the correlati=
on-ahead-of-signaling&nbsp;problem, once you have multiple RTP flows in the=
 m-line. &nbsp;That's why we've designed appid the way we have.<o:p></o:p><=
/div>
</div>
</div>
<blockquote style=3D"border-style: none none none solid; border-left-width:=
 1pt; border-left-color: rgb(204, 204, 204); padding: 0cm 0cm 0cm 6pt; marg=
in-left: 4.8pt; margin-right: 0cm; ">
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<o:p>&nbsp;</o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
I think appid is the right solution to point 2. &nbsp;More specifically, th=
e problem appid is solving is better expressed as the correlation of packet=
s from an RTP flow to an m-line; the correlation from the m-line to a MST i=
s a separate step, and is msid's job.<o:p></o:p></div>
</div>
</blockquote>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<o:p>&nbsp;</o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
Agree that msid should map m-line to MST, but it would seem recv-msid could=
 map RTP flow to m=3Dline as well as indicate whether the receiver is in fa=
ct interested in receiving the MST associated with said m=3D line. Note tha=
t recv-msid doesn't specify an actual
 MSID; its purpose is to indicate that the receiver wants to receive data f=
rom the remote side's MST, and provide a mechanism for the sender to mark t=
he data accordingly. (Which ends up seeming a lot like appid.)<o:p></o:p></=
div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<o:p>&nbsp;</o:p></div>
</div>
<div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
Generally, I am hesitant to depend on 3 things if we think we could get by =
with only 2.<o:p></o:p></div>
</div>
</div>
<div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Time=
s New Roman', serif; ">
<o:p>&nbsp;</o:p></div>
</div>
</div>
</div>
</div>
</blockquote>
</div>
<br>
</body>
</html>

--_000_B9399713DDEF4680B5E4E9C817372133vidyocom_--

From pkyzivat@alum.mit.edu  Tue Feb  4 08:59:32 2014
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Date: Tue, 04 Feb 2014 11:59:29 -0500
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: [MMUSIC] Comments on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00 (
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(Trying yet again, this time on the proper list)

Richard,

What is the status of this work?
Are you planning on a new version? (I guess a new one is needed just to 
bind it to mmusic rather than dispatch.)

Also, these naming issues extend to draft-ietf-mmusic-sctp-sdp-05.
Can I get your support in lobbying for naming changes there too?

	Thanks,
	Paul

On 10/28/13 12:08 PM, Paul Kyzivat wrote:
> On 10/28/13 11:07 AM, Ejzak, Richard P (Richard) wrote:
>> Thanks, Paul.
>>
>> I agree that the WebRTC specific aspects need to be clearly separated
>> from the core protocol aspects.  I'll do that in the next version.
>
> Thanks!
>
>> I also agree that "WebRTC" is not needed in either name.  Are you ok
>> with just "DataChannel" and "dcsa" instead of "WebRTC-DataChannel" and
>> "wdcsa"?  I'm not really picky about the names and am open to any
>> reasonable alternative...
>
> Its just an esthetic issue, so I'm not too picky.
>
> DataChannel is better than WebRTC-DataChannel.
> But IMO CamelCase identifiers aren't in stylistic agreement with SDP.
> So perhaps data-channel?
>
> "dcsa" seems reasonable.
>
>      Thanks,
>      Paul
>
>> BR, Richard
>>
>>> -----Original Message-----
>>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
>>> Behalf Of Paul Kyzivat
>>> Sent: Sunday, October 27, 2013 12:48 PM
>>> To: dispatch@ietf.org
>>> Subject: [dispatch] Comments on draft-ejzak-dispatch-webrtc-data-
>>> channel-sdpneg-00
>>>
>>> In general I think this document is looking pretty good. It is going in
>>> the right direction from my perspective. And I think its starting to
>>> become clear how to have this be an optional mechanism for webrtc. I'm
>>> starting to see how it could be used for CLUE, with and without webrtc.
>>>
>>> One thing bothers me: this is an ietf document, defining signaling. But
>>> it talks quite a lot about the webrtc API. It is certainly possible to
>>> use this specification in contexts where there are no browsers and no
>>> use of webrtc. ISTM it would be better to segregate all the webrtc
>>> stuff into an annex. Or maybe two documents are needed: one for SDP
>>> negotiation of sctp data channels, and one for webrtc external
>>> negotiation of data channels via the SDP mechanism.
>>>
>>> Esthetically I find the names "webrtc-DataChannel" and "wdcsa" quite
>>> unappealing. In particular, these data channels are usable (and will be
>>> used) in non-webrtc applications, so why do we need "webrtc" in the
>>> name?
>>>
>>>     Thanks,
>>>     Paul
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>


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From petithug@acm.org  Wed Feb  5 09:47:47 2014
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Subject: [MMUSIC] fmtp specific configuration vs parameters
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA256

RFC 3264 section 6.1:

  "The interpretation of fmtp parameters in an offer depends on the
   parameters.  In many cases, those parameters describe specific
   configurations of the media format, and should therefore be processed
   as the media format value itself would be.  This means that the same
   fmtp parameters with the same values MUST be present in the answer if
   the media format they describe is present in the answer.  Other fmtp
   parameters are more like parameters, for which it is perfectly
   acceptable for each agent to use different values.  In that case, the
   answer MAY contain fmtp parameters, and those MAY have the same
   values as those in the offer, or they MAY be different.  SDP
   extensions that define new parameters SHOULD specify the proper
   interpretation in offer/answer."

I am not sure I understand this paragraph.  It seems to say that there is fmtp
parameters that must be the same in the Offer and in the Answer, and some that
can be different.  But I went through the definition of 50 or so codecs, and
cannot find one fmtp parameter that is be required to be identical on both SDPs.

Can someone tell me if my understanding is correct, and if it is the case,
give me an example of parameter that must be identical on both sides?

Thanks.

- -- 
Marc Petit-Huguenin
Email: marc@petit-huguenin.org
Blog: http://blog.marc.petit-huguenin.org
Profile: http://www.linkedin.com/in/petithug
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From pkyzivat@alum.mit.edu  Wed Feb  5 11:11:34 2014
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Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.txt
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On 2/4/14 2:18 AM, Simo.Veikkolainen@nokia.com wrote:
>
>
> -----Original Message-----
> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of ext Paul Kyzivat
> Sent: 31 January, 2014 20:53
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.txt
>
> I just reviewed the new version. First I looked at just the diffs. But those raised some more general questions in my mind, so I re-reviewed more of the draft. I'm left with some considerable confusion about how this is intended to work in practice, especially with extern correlation, but even with the other forms.
>
> In particular, I am unclear of the sequencing of O/A processing, correlation processing, and signaling completion. (I realize this is an mmusic draft and theoretically independent of sip, but forgive me if I discuss this in the context of sip.)
>
> Normally, when first establishing a session with sip, the first O/A may occur before the signaling session is established. But early media may flow after the first O/A. Often it is expected that the user will be be aware and involved in this early media.
>
> Once a session is up, it can be changed with a reinvite containing a new O/A. If the O/A and the reinvite are successful, then the results of the new O/A replace the old one. If the reinvite fails, then the session is supposed to continue as it was before the reinvite. In the interim, from the sending of the reinvite until its success or failure, things are a little confusing - presumably both old and new media streams must be maintained, but which are connected to the user is unspecified.
>
> I see nothing in this draft that explains how the CS media, and correlation verification, fit into the above process.
>
> Consider a new call that offers CS media:
>
> - The first O/A must complete before the CS call can be attempted.
>
> - must the CS call complete and be correlated successfully before
>     the INVITE gets a 200? If the invite is allowed to complete first,
>     what happens if no CS call is established, or if it is but
>     correlation fails?
>
> [SV] I assume you mean the 200 carries the Answer.

Not necessarily. I was just trying to sort out the expectations for the 
timing of the events. The answer could be *before* the 200, but it must 
be before the call is established.

> It may happen that the CS call arrives before the Answer, but in this case no correlation can be done. See Section 5.6.3. If the invite (I assume you mean O/A) completes first, but no CS call ever arrives, or correlation - including the "external" mechanism fails, the endpoint should treat the CS call as unrelated; this is described in Section 5.3.3.

Sure, but my question is whether correlation is expected to occur before 
or after the INVITE completes.

> - when does CS media start to flow to the user as part of the
>     SIP call? Is it before or after correlation? (E.g., does the
>     user hear the DTMF exchange?)
>
> [SV] CS media starts to flow after the call has been accepted, as in a normal CS call. Correlation may be possible before accepting the call (for "callerid", "uuie", and "external" mechanisms), and for "dtmf" case only after the call has been accepted.

I should have been clearer. For DTMF and external correlation media 
clearly needs to flow on the CS call. My question pertains to whether 
this is considered part of the SIP session prior to correlation.

> - if media flows to the user before correlation, then what if
>     correlation fails? Will another call be accepted and correlation
>     be attempted again?
>
> [SV] I don' think we have explicitly defined this situation. If a call is accepted and *after* that correlation fails, it means that the DTMF mechanism was used (for other correlation mechanisms, we know before accepting the call whether the correlation was successful).

external correlation via also has this problem, and more.

> If the DTMF pattern is not received correctly, as we cannot know whether it is related to the session or not. Now, this raises a possible case of including both "dtmf" and "external" mechanisms in the O/A. If DTMF is successfully negotiated but it fails, we can't really use the "external" mechanism since the call is already accepted. Implementations could in this case release the CS call and try a new O/A without the DTMF.

I'm finding external correlation to be underspecified.

One one I could conceive of it working is that the incoming CS call is 
treated something like a call waiting situation. The existing (sip) call 
is "suspended" and the user is connected to the CS call, and asked to 
confirm if it is the intended one. If the user signals confirmation, 
then the sip call is resumed, with the CS audio stream in it.

If the external correlation fails, then the sip call is resumed without 
the CS media.

A problem with this is that the O/A completes prior to the CS call 
establishment and correlation. If this happens mid-call (because of a 
desire to switch from RTP to CS media) then the RTP media will be 
abandoned once the O/A succeeds, before correlation. So while waiting 
for a CS call to arrive and be correlated you have no RTP media. :-(

> - what if multiple incoming calls arrive before correlation
>     has been completed? (If correlation occurs before exposing the
>     media to the user, then it isn't an issue.)
>
> [SV] This would be an issue for the DTMF case, as there correlation can be only only after accepting the CS call. An unlikely case, but still possible. In this case, one of the CS calls could be related to the session, while other(s) would be normal CS calls. As only one CS call can be handled at a time, others would either wait or be terminated, depending on the callee's supplementary services settings.

For DTMF, the correlation could be done *before* relating the CS call to 
the session. Only bind it to the session after correlation is complete.

But that doesn't work as well for external correlation.

> - extern correlation complicates this. It isn't like the other
>     correlation mechanisms - the user must be exposed to the media
>     to decide if there is correlation. How would the user indicate
>     "this is the wrong call" in a way that would allow correlation
>     to be attempted on another call?
>
> [SV] The external correlation mechanism does not assume the media flows before "correlation" is made - if the user decides to accept the incoming call, it is considered as "correlated".

The user has to receive (hear) the media (and maybe send some too) 
before he can decide on correlation. Has the CS call be "related to the 
session" then, or not?

What happens if the user decides the CS call *isn't* correlated?

> Now consider a call that is initially established with one RTP media stream. Later, a reinvite is sent with an offer proposing to to replace that with a CS media stream:
>
> - again, there must be an answer before the CS call can be attempted.
>
>
> [SV] For being able to perform correlation, yes, the Answer needs to be received.
>
> - are you assuming that the reinvite will complete before the CS
>     call does? If so, then you are in a break-before-make situation -
>     the RTP session is gone before the CS session takes its place.
>
> [SV] The O/A needs to be completed before correlation can be done, but CS call may arrive before Answer. In this case the endpoint should wait until the Answer is received. If the CS call does not succeed, or correlation is unsuccessful, then yes there is no media as the RTP stream has been dropped. But, this could also happen when re-negotiating RTP media types (endpoints are not able to load correct codecs, new IP addr/port does not work etc.) so the situation not be specific to the CS media type.

Yes, it is not entirely unique, but usually updated O/As are done with 
an option in common to the prior one so that failure is unlikely. The 
failure modes are quite different when transitioning from RTP to CS. 
(And also when transitioning from CS to RTP.)

> - all the issues above also apply here.
>
> ISTM that there is a bunch of missing explanation of how this is expected to work.
>
> [SV] Could you check the above whether my comments clarify the situation. We can address them in more details in the draft as well once we know where more explanation is needed.

Upon rereading section 5.6.2 I realize that it does say that the answer 
must be received before the CS call is placed and correlated, so it 
implies a break-before-make approach on reINVITE.

What I still think is missing is:

* explanation of how things work when changing media mid call - from RTP 
to CS, CS to RTP, or even CS to different CS. Point out that this is 
break-before-make, and when the user will again perceive to have media.

* much about external correlation. Is this really a mechanism in the 
sense that the others are, that serves as a screen on CS calls until the 
one is found that correlates? Or does it really mean that any incoming 
CS call is taking as the right one, included in the sip session, with 
the user left to take independent action (such as hanging up) if he 
decides it isn't the right one?

* For DTMF correlation, does the user see media in the call before the 
correlation is successfully completed.

* potentially (but maybe not in this draft) a discussion of how to 
achieve make-before-break semantics. (E.g., Using multiple m-lines.)

	Thanks,
	Paul

> I also have a nit in section 5.3.2:
>
>      The fourth correlation mechanism declares support for cases where
>      correlation is done by external means.  Typically this means that the
>      decision is left for the human user.  This is the way how some
>      current conferencing systems operate: ...
>
> "This is the way how some current conferencing systems operate"
>
> should be:
>
> "This is how some current conferencing systems operate"
>
> [SV] Fill fix.
>
> Thanks,
> Simo
>
> 	Thanks,
> 	Paul
>
> On 1/31/14 6:01 AM, Simo.Veikkolainen@nokia.com wrote:
>> Hello WG,
>>
>> We have submitted a new version of
>> http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-22
>>
>> To recap, there was discussion on the  MMUSIC WG list on which correlation mechanisms should be mandatory to implement (see Gonzalo's email http://www.ietf.org/mail-archive/web/mmusic/current/msg12274.html ). As stated by Gonzalo, 3GPP is going to use this draft, more specifically the plan now is to use the caller id based correlation mechanism. This was the message I got from Andrew after a recent 3GPP CT1 meeting.
>>
>> We have now made the following changes to the draft:
>> -	the endpoint may choose which correlation mechanisms to implement (before all three were MTI).
>> -	we have introduced a fourth correlation type ("external") which indicates explicit support for cases where also some external means, typically the human user, can do the correlation. This can be as a fallback in cases where no other common correlation mechanism is found, or the negotiated correlation mechanisms fail due to being not supported in the PSTN network. This is much like current systems like Webex work.
>>
>> This was also the guidance from the IESG informal telechat in December where Gonzalo presented the draft.
>>
>> We would like to hear from the WG whether there are any concerns with these changes.
>>
>> On behalf of the authors,
>> Simo
>>
>> PS. I'll add the proposed change in http://www.ietf.org/mail-archive/web/mmusic/current/msg12792.html to next version, slipped my mind this time.
>>
>>
>> -----Original Message-----
>> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of ext
>> internet-drafts@ietf.org
>> Sent: 31 January, 2014 12:25
>> To: i-d-announce@ietf.org
>> Cc: mmusic@ietf.org
>> Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.txt
>>
>>
>> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>>    This draft is a work item of the Multiparty Multimedia Session Control Working Group of the IETF.
>>
>>           Title           : Session Description Protocol (SDP) Extension For Setting Audio and Video Media Streams Over Circuit-Switched Bearers In The Public Switched Telephone Network (PSTN)
>>           Authors         : Miguel A. Garcia-Martin
>>                             Simo Veikkolainen
>> 	Filename        : draft-ietf-mmusic-sdp-cs-22.txt
>> 	Pages           : 37
>> 	Date            : 2014-01-31
>>
>> 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-22
>>
>> A diff from the previous version is available at:
>> http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-sdp-cs-22
>>
>>
>> Please note that it may take a couple of minutes from the time of submission until the htmlized version and diff are available at tools.ietf.org.
>>
>> 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
>> _______________________________________________
>> 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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Date: Thu, 6 Feb 2014 09:13:20 -0800
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From: Oleg Moskalenko <mom040267@gmail.com>
To: "Tirumaleswar Reddy (tireddy)" <tireddy@cisco.com>
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Subject: Re: [MMUSIC] [tram] FW: New Version Notification for draft-wing-mmusic-ice-mobility-06.txt
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I have comments regarding the new Error Code "Mobility Forbidden".

1) Section 5.1.4:

Comment: remove two TBDs, if the error code has been already determined.

2) Section 7:

and to add a new STUN error code "Mobility Forbidden" with the value
501 to the STUN Error Codes registry [iana-stun].


Comment: I am not sure that 501 is the right code to be used. In SIP and
HTTP, 5xx errors are usually about a server code problem of some sort; 501
means "not implemented". That may be not exactly accurate. A mobility-aware
TURN server may reject the request simply because the administrator does
not allow it., and the server did not encounter an error. The reason why
the server reject the request is rather "not allowed" (which is the error
code 405).

May be be, the right approach would be to introduce two separate error
codes here: 405 "not allowed" (the server implements mobility but the
administrator forbade it) or 501 "not implemented" (the server knows about
mobility but it has no implementation for that). If the server truly is not
aware about mobility at all, the server will send no MOBILITY-TICKET in the
server response. Those are three distinct use cases: 405, 501 and an
"ignorant" server.

Thanks
Oleg



On Wed, Feb 5, 2014 at 6:17 AM, Tirumaleswar Reddy (tireddy) <
tireddy@cisco.com> wrote:

> Revised draft-wing-mmusic-ice-mobility-06 has been published. It addresses
> the comments received for Mobility using TURN and a new section with
> implementation details. Further comments and suggestions are welcome.
>
> Thanks and Regards,
> -Authors.
>
> -----Original Message-----
> From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]
> Sent: Wednesday, February 05, 2014 7:44 PM
> To: Prashanth Patil (praspati); Tirumaleswar Reddy (tireddy); Prashanth
> Patil (praspati); Dan Wing (dwing); Tirumaleswar Reddy (tireddy); Pal
> Martinsen (palmarti); Dan Wing (dwing); Pal Martinsen (palmarti)
> Subject: New Version Notification for draft-wing-mmusic-ice-mobility-06.txt
>
>
> A new version of I-D, draft-wing-mmusic-ice-mobility-06.txt
> has been successfully submitted by Tirumaleswar Reddy and posted to the
> IETF repository.
>
> Name:           draft-wing-mmusic-ice-mobility
> Revision:       06
> Title:          Mobility with ICE (MICE)
> Document date:  2014-02-05
> Group:          Individual Submission
> Pages:          21
> URL:
> http://www.ietf.org/internet-drafts/draft-wing-mmusic-ice-mobility-06.txt
> Status:
> https://datatracker.ietf.org/doc/draft-wing-mmusic-ice-mobility/
> Htmlized:
> http://tools.ietf.org/html/draft-wing-mmusic-ice-mobility-06
> Diff:
> http://www.ietf.org/rfcdiff?url2=draft-wing-mmusic-ice-mobility-06
>
> Abstract:
>    This specification describes how endpoint mobility can be achieved
>    using ICE.  Two mechanisms are shown, one where both endpoints
>    support MICE and another where only one endpoint supports MICE.
>
>
>
>
> Please note that it may take a couple of minutes from the time of
> submission until the htmlized version and diff are available at
> tools.ietf.org.
>
> The IETF Secretariat
>
> _______________________________________________
> tram mailing list
> tram@ietf.org
> https://www.ietf.org/mailman/listinfo/tram
>

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Content-Type: text/html; charset=ISO-8859-1
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<div dir=3D"ltr"><div><div><div><div>I have comments regarding the new Erro=
r Code &quot;Mobility Forbidden&quot;.<br><br></div>1) Section 5.1.4: <br><=
br>Comment: remove two TBDs, if the error code has been already determined.=
<br>
<br></div>2) Section 7:<br><br><pre>and to add a new STUN error code &quot;=
Mobility Forbidden&quot; with the value
501 to the STUN Error Codes registry [iana-stun].</pre><br></div>Comment: I=
 am not sure that 501 is the right code to be used. In SIP and HTTP, 5xx er=
rors are usually about a server code problem of some sort; 501 means &quot;=
not implemented&quot;. That may be not exactly accurate. A mobility-aware T=
URN server may reject the request simply because the administrator does not=
 allow it., and the server did not encounter an error. The reason why the s=
erver reject the request is rather &quot;not allowed&quot; (which is the er=
ror code 405).<br>
<br></div>May be be, the right approach would be to introduce two separate =
error codes here: 405 &quot;not allowed&quot; (the server implements mobili=
ty but the administrator forbade it) or 501 &quot;not implemented&quot; (th=
e server knows about mobility but it has no implementation for that). If th=
e server truly is not aware about mobility at all, the server will send no =
MOBILITY-TICKET in the server response. Those are three distinct use cases:=
 405, 501 and an &quot;ignorant&quot; server. <br>
<br><div><div>Thanks<br>Oleg<br><br></div></div></div><div class=3D"gmail_e=
xtra"><br><br><div class=3D"gmail_quote">On Wed, Feb 5, 2014 at 6:17 AM, Ti=
rumaleswar Reddy (tireddy) <span dir=3D"ltr">&lt;<a href=3D"mailto:tireddy@=
cisco.com" target=3D"_blank">tireddy@cisco.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">Revised draft-wing-mmusic-ice-mobility-06 ha=
s been published. It addresses the comments received for Mobility using TUR=
N and a new section with implementation details. Further comments and sugge=
stions are welcome.<br>

<br>
Thanks and Regards,<br>
-Authors.<br>
<br>
-----Original Message-----<br>
From: <a href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org<=
/a> [mailto:<a href=3D"mailto:internet-drafts@ietf.org">internet-drafts@iet=
f.org</a>]<br>
Sent: Wednesday, February 05, 2014 7:44 PM<br>
To: Prashanth Patil (praspati); Tirumaleswar Reddy (tireddy); Prashanth Pat=
il (praspati); Dan Wing (dwing); Tirumaleswar Reddy (tireddy); Pal Martinse=
n (palmarti); Dan Wing (dwing); Pal Martinsen (palmarti)<br>
Subject: New Version Notification for draft-wing-mmusic-ice-mobility-06.txt=
<br>
<br>
<br>
A new version of I-D, draft-wing-mmusic-ice-mobility-06.txt<br>
has been successfully submitted by Tirumaleswar Reddy and posted to the IET=
F repository.<br>
<br>
Name: =A0 =A0 =A0 =A0 =A0 draft-wing-mmusic-ice-mobility<br>
Revision: =A0 =A0 =A0 06<br>
Title: =A0 =A0 =A0 =A0 =A0Mobility with ICE (MICE)<br>
Document date: =A02014-02-05<br>
Group: =A0 =A0 =A0 =A0 =A0Individual Submission<br>
Pages: =A0 =A0 =A0 =A0 =A021<br>
URL: =A0 =A0 =A0 =A0 =A0 =A0<a href=3D"http://www.ietf.org/internet-drafts/=
draft-wing-mmusic-ice-mobility-06.txt" target=3D"_blank">http://www.ietf.or=
g/internet-drafts/draft-wing-mmusic-ice-mobility-06.txt</a><br>
Status: =A0 =A0 =A0 =A0 <a href=3D"https://datatracker.ietf.org/doc/draft-w=
ing-mmusic-ice-mobility/" target=3D"_blank">https://datatracker.ietf.org/do=
c/draft-wing-mmusic-ice-mobility/</a><br>
Htmlized: =A0 =A0 =A0 <a href=3D"http://tools.ietf.org/html/draft-wing-mmus=
ic-ice-mobility-06" target=3D"_blank">http://tools.ietf.org/html/draft-wing=
-mmusic-ice-mobility-06</a><br>
Diff: =A0 =A0 =A0 =A0 =A0 <a href=3D"http://www.ietf.org/rfcdiff?url2=3Ddra=
ft-wing-mmusic-ice-mobility-06" target=3D"_blank">http://www.ietf.org/rfcdi=
ff?url2=3Ddraft-wing-mmusic-ice-mobility-06</a><br>
<br>
Abstract:<br>
=A0 =A0This specification describes how endpoint mobility can be achieved<b=
r>
=A0 =A0using ICE. =A0Two mechanisms are shown, one where both endpoints<br>
=A0 =A0support MICE and another where only one endpoint supports MICE.<br>
<br>
<br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n until the htmlized version and diff are available at <a href=3D"http://to=
ols.ietf.org" target=3D"_blank">tools.ietf.org</a>.<br>
<br>
The IETF Secretariat<br>
<br>
_______________________________________________<br>
tram mailing list<br>
<a href=3D"mailto:tram@ietf.org">tram@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/tram" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/tram</a><br>
</blockquote></div><br></div>

--001a113453ec7d03ca04f1bffd76--

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... and another reason why we may want two different error codes for "not
allowed" or "not implemented" use cases is that we now have an "origin"
TURN draft, and the TURN server may reject or allow Mobility using the
"origin" field (according to a server policy). Then probably "not
implemented" error would be an incorrect error code if the "mobile
allocate" request was rejected because of the origin attribute.

But at the same time I am not sure that the "origin" field was really
designed as an input for "hard" security decisions (like allowing or
disallowing a piece of functionality). This is a grey area between
different drafts, and may be the authors or both "origin" draft and
"mobility" draft would like to clarify that, for future implementations.

Thanks
Oleg



On Thu, Feb 6, 2014 at 9:13 AM, Oleg Moskalenko <mom040267@gmail.com> wrote:

> I have comments regarding the new Error Code "Mobility Forbidden".
>
> 1) Section 5.1.4:
>
> Comment: remove two TBDs, if the error code has been already determined.
>
> 2) Section 7:
>
> and to add a new STUN error code "Mobility Forbidden" with the value
> 501 to the STUN Error Codes registry [iana-stun].
>
>
> Comment: I am not sure that 501 is the right code to be used. In SIP and
> HTTP, 5xx errors are usually about a server code problem of some sort; 501
> means "not implemented". That may be not exactly accurate. A mobility-aware
> TURN server may reject the request simply because the administrator does
> not allow it., and the server did not encounter an error. The reason why
> the server reject the request is rather "not allowed" (which is the error
> code 405).
>
> May be be, the right approach would be to introduce two separate error
> codes here: 405 "not allowed" (the server implements mobility but the
> administrator forbade it) or 501 "not implemented" (the server knows about
> mobility but it has no implementation for that). If the server truly is not
> aware about mobility at all, the server will send no MOBILITY-TICKET in the
> server response. Those are three distinct use cases: 405, 501 and an
> "ignorant" server.
>
> Thanks
> Oleg
>
>
>
> On Wed, Feb 5, 2014 at 6:17 AM, Tirumaleswar Reddy (tireddy) <
> tireddy@cisco.com> wrote:
>
>> Revised draft-wing-mmusic-ice-mobility-06 has been published. It
>> addresses the comments received for Mobility using TURN and a new section
>> with implementation details. Further comments and suggestions are welcome.
>>
>> Thanks and Regards,
>> -Authors.
>>
>> -----Original Message-----
>> From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]
>> Sent: Wednesday, February 05, 2014 7:44 PM
>> To: Prashanth Patil (praspati); Tirumaleswar Reddy (tireddy); Prashanth
>> Patil (praspati); Dan Wing (dwing); Tirumaleswar Reddy (tireddy); Pal
>> Martinsen (palmarti); Dan Wing (dwing); Pal Martinsen (palmarti)
>> Subject: New Version Notification for
>> draft-wing-mmusic-ice-mobility-06.txt
>>
>>
>> A new version of I-D, draft-wing-mmusic-ice-mobility-06.txt
>> has been successfully submitted by Tirumaleswar Reddy and posted to the
>> IETF repository.
>>
>> Name:           draft-wing-mmusic-ice-mobility
>> Revision:       06
>> Title:          Mobility with ICE (MICE)
>> Document date:  2014-02-05
>> Group:          Individual Submission
>> Pages:          21
>> URL:
>> http://www.ietf.org/internet-drafts/draft-wing-mmusic-ice-mobility-06.txt
>> Status:
>> https://datatracker.ietf.org/doc/draft-wing-mmusic-ice-mobility/
>> Htmlized:
>> http://tools.ietf.org/html/draft-wing-mmusic-ice-mobility-06
>> Diff:
>> http://www.ietf.org/rfcdiff?url2=draft-wing-mmusic-ice-mobility-06
>>
>> Abstract:
>>    This specification describes how endpoint mobility can be achieved
>>    using ICE.  Two mechanisms are shown, one where both endpoints
>>    support MICE and another where only one endpoint supports MICE.
>>
>>
>>
>>
>> Please note that it may take a couple of minutes from the time of
>> submission until the htmlized version and diff are available at
>> tools.ietf.org.
>>
>> The IETF Secretariat
>>
>> _______________________________________________
>> tram mailing list
>> tram@ietf.org
>> https://www.ietf.org/mailman/listinfo/tram
>>
>
>

--001a113453ec2f366704f1c413e5
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div>... and another reason why we may want two different =
error codes for &quot;not allowed&quot; or &quot;not implemented&quot; use =
cases is that we now have an &quot;origin&quot; TURN draft, and the TURN se=
rver may reject or allow Mobility using the &quot;origin&quot; field (accor=
ding to a server policy). Then probably &quot;not implemented&quot; error w=
ould be an incorrect error code if the &quot;mobile allocate&quot; request =
was rejected because of the origin attribute.<br>
<br></div><div>But at the same time I am not sure that the &quot;origin&quo=
t; field was really designed as an input for &quot;hard&quot; security deci=
sions (like allowing or disallowing a piece of functionality). This is a gr=
ey area between different drafts, and may be the authors or both &quot;orig=
in&quot; draft and &quot;mobility&quot; draft would like to clarify that, f=
or future implementations.<br>
</div><div><br></div><div>Thanks<br></div><div>Oleg<br><br></div></div><div=
 class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Thu, Feb 6, 20=
14 at 9:13 AM, Oleg Moskalenko <span dir=3D"ltr">&lt;<a href=3D"mailto:mom0=
40267@gmail.com" target=3D"_blank">mom040267@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 dir=3D"ltr"><div><div><div><div>I have =
comments regarding the new Error Code &quot;Mobility Forbidden&quot;.<br><b=
r>
</div>1) Section 5.1.4: <br><br>Comment: remove two TBDs, if the error code=
 has been already determined.<br>
<br></div>2) Section 7:<br><br><pre>and to add a new STUN error code &quot;=
Mobility Forbidden&quot; with the value
501 to the STUN Error Codes registry [iana-stun].</pre><br></div>Comment: I=
 am not sure that 501 is the right code to be used. In SIP and HTTP, 5xx er=
rors are usually about a server code problem of some sort; 501 means &quot;=
not implemented&quot;. That may be not exactly accurate. A mobility-aware T=
URN server may reject the request simply because the administrator does not=
 allow it., and the server did not encounter an error. The reason why the s=
erver reject the request is rather &quot;not allowed&quot; (which is the er=
ror code 405).<br>

<br></div>May be be, the right approach would be to introduce two separate =
error codes here: 405 &quot;not allowed&quot; (the server implements mobili=
ty but the administrator forbade it) or 501 &quot;not implemented&quot; (th=
e server knows about mobility but it has no implementation for that). If th=
e server truly is not aware about mobility at all, the server will send no =
MOBILITY-TICKET in the server response. Those are three distinct use cases:=
 405, 501 and an &quot;ignorant&quot; server. <br>

<br><div><div>Thanks<span class=3D"HOEnZb"><font color=3D"#888888"><br>Oleg=
<br><br></font></span></div></div></div><div class=3D"gmail_extra"><br><br>=
<div class=3D"gmail_quote"><div class=3D"im">On Wed, Feb 5, 2014 at 6:17 AM=
, Tirumaleswar Reddy (tireddy) <span dir=3D"ltr">&lt;<a href=3D"mailto:tire=
ddy@cisco.com" target=3D"_blank">tireddy@cisco.com</a>&gt;</span> wrote:<br=
>

</div><div><div class=3D"h5"><blockquote class=3D"gmail_quote" style=3D"mar=
gin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Revised draft-w=
ing-mmusic-ice-mobility-06 has been published. It addresses the comments re=
ceived for Mobility using TURN and a new section with implementation detail=
s. Further comments and suggestions are welcome.<br>


<br>
Thanks and Regards,<br>
-Authors.<br>
<br>
-----Original Message-----<br>
From: <a href=3D"mailto:internet-drafts@ietf.org" target=3D"_blank">interne=
t-drafts@ietf.org</a> [mailto:<a href=3D"mailto:internet-drafts@ietf.org" t=
arget=3D"_blank">internet-drafts@ietf.org</a>]<br>
Sent: Wednesday, February 05, 2014 7:44 PM<br>
To: Prashanth Patil (praspati); Tirumaleswar Reddy (tireddy); Prashanth Pat=
il (praspati); Dan Wing (dwing); Tirumaleswar Reddy (tireddy); Pal Martinse=
n (palmarti); Dan Wing (dwing); Pal Martinsen (palmarti)<br>
Subject: New Version Notification for draft-wing-mmusic-ice-mobility-06.txt=
<br>
<br>
<br>
A new version of I-D, draft-wing-mmusic-ice-mobility-06.txt<br>
has been successfully submitted by Tirumaleswar Reddy and posted to the IET=
F repository.<br>
<br>
Name: =A0 =A0 =A0 =A0 =A0 draft-wing-mmusic-ice-mobility<br>
Revision: =A0 =A0 =A0 06<br>
Title: =A0 =A0 =A0 =A0 =A0Mobility with ICE (MICE)<br>
Document date: =A02014-02-05<br>
Group: =A0 =A0 =A0 =A0 =A0Individual Submission<br>
Pages: =A0 =A0 =A0 =A0 =A021<br>
URL: =A0 =A0 =A0 =A0 =A0 =A0<a href=3D"http://www.ietf.org/internet-drafts/=
draft-wing-mmusic-ice-mobility-06.txt" target=3D"_blank">http://www.ietf.or=
g/internet-drafts/draft-wing-mmusic-ice-mobility-06.txt</a><br>
Status: =A0 =A0 =A0 =A0 <a href=3D"https://datatracker.ietf.org/doc/draft-w=
ing-mmusic-ice-mobility/" target=3D"_blank">https://datatracker.ietf.org/do=
c/draft-wing-mmusic-ice-mobility/</a><br>
Htmlized: =A0 =A0 =A0 <a href=3D"http://tools.ietf.org/html/draft-wing-mmus=
ic-ice-mobility-06" target=3D"_blank">http://tools.ietf.org/html/draft-wing=
-mmusic-ice-mobility-06</a><br>
Diff: =A0 =A0 =A0 =A0 =A0 <a href=3D"http://www.ietf.org/rfcdiff?url2=3Ddra=
ft-wing-mmusic-ice-mobility-06" target=3D"_blank">http://www.ietf.org/rfcdi=
ff?url2=3Ddraft-wing-mmusic-ice-mobility-06</a><br>
<br>
Abstract:<br>
=A0 =A0This specification describes how endpoint mobility can be achieved<b=
r>
=A0 =A0using ICE. =A0Two mechanisms are shown, one where both endpoints<br>
=A0 =A0support MICE and another where only one endpoint supports MICE.<br>
<br>
<br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n until the htmlized version and diff are available at <a href=3D"http://to=
ols.ietf.org" target=3D"_blank">tools.ietf.org</a>.<br>
<br>
The IETF Secretariat<br>
<br>
_______________________________________________<br>
tram mailing list<br>
<a href=3D"mailto:tram@ietf.org" target=3D"_blank">tram@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/tram" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/tram</a><br>
</blockquote></div></div></div><br></div>
</blockquote></div><br></div>

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See below [SV]

-----Original Message-----
From: ext Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]=20
Sent: 05 February, 2014 21:11
To: Veikkolainen Simo (Nokia-CTO/Espoo); mmusic@ietf.org
Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.txt

On 2/4/14 2:18 AM, Simo.Veikkolainen@nokia.com wrote:
>
>
> -----Original Message-----
> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of ext Paul=20
> Kyzivat
> Sent: 31 January, 2014 20:53
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.txt
>
> I just reviewed the new version. First I looked at just the diffs. But th=
ose raised some more general questions in my mind, so I re-reviewed more of=
 the draft. I'm left with some considerable confusion about how this is int=
ended to work in practice, especially with extern correlation, but even wit=
h the other forms.
>
> In particular, I am unclear of the sequencing of O/A processing,=20
> correlation processing, and signaling completion. (I realize this is=20
> an mmusic draft and theoretically independent of sip, but forgive me=20
> if I discuss this in the context of sip.)
>
> Normally, when first establishing a session with sip, the first O/A may o=
ccur before the signaling session is established. But early media may flow =
after the first O/A. Often it is expected that the user will be be aware an=
d involved in this early media.
>
> Once a session is up, it can be changed with a reinvite containing a new =
O/A. If the O/A and the reinvite are successful, then the results of the ne=
w O/A replace the old one. If the reinvite fails, then the session is suppo=
sed to continue as it was before the reinvite. In the interim, from the sen=
ding of the reinvite until its success or failure, things are a little conf=
using - presumably both old and new media streams must be maintained, but w=
hich are connected to the user is unspecified.
>
> I see nothing in this draft that explains how the CS media, and correlati=
on verification, fit into the above process.
>
> Consider a new call that offers CS media:
>
> - The first O/A must complete before the CS call can be attempted.
>
> - must the CS call complete and be correlated successfully before
>     the INVITE gets a 200? If the invite is allowed to complete first,
>     what happens if no CS call is established, or if it is but
>     correlation fails?
>
> [SV] I assume you mean the 200 carries the Answer.

Not necessarily. I was just trying to sort out the expectations for the tim=
ing of the events. The answer could be *before* the 200, but it must be bef=
ore the call is established.

> It may happen that the CS call arrives before the Answer, but in this cas=
e no correlation can be done. See Section 5.6.3. If the invite (I assume yo=
u mean O/A) completes first, but no CS call ever arrives, or correlation - =
including the "external" mechanism fails, the endpoint should treat the CS =
call as unrelated; this is described in Section 5.3.3.

Sure, but my question is whether correlation is expected to occur before or=
 after the INVITE completes.

[SV] Correlation can only occur after the correlation information has been =
exchanged, and the CS call is received. I'm not sure what you mean by INVIT=
E being completed, but I would say correlation can be done already before t=
he last request (ACK) has been sent, or after the whole INVITE related requ=
est-response exchange has been completed (in case the CS call is received l=
ate).

> - when does CS media start to flow to the user as part of the
>     SIP call? Is it before or after correlation? (E.g., does the
>     user hear the DTMF exchange?)
>
> [SV] CS media starts to flow after the call has been accepted, as in a no=
rmal CS call. Correlation may be possible before accepting the call (for "c=
allerid", "uuie", and "external" mechanisms), and for "dtmf" case only afte=
r the call has been accepted.

I should have been clearer. For DTMF and external correlation media clearly=
 needs to flow on the CS call. My question pertains to whether this is cons=
idered part of the SIP session prior to correlation.

[SV] Again, I'm not sure I'm able to follow your thought but correlation is=
 needed before the CS media is considered as part of the session.

> - if media flows to the user before correlation, then what if
>     correlation fails? Will another call be accepted and correlation
>     be attempted again?
>
> [SV] I don' think we have explicitly defined this situation. If a call is=
 accepted and *after* that correlation fails, it means that the DTMF mechan=
ism was used (for other correlation mechanisms, we know before accepting th=
e call whether the correlation was successful).

external correlation via also has this problem, and more.

> If the DTMF pattern is not received correctly, as we cannot know whether =
it is related to the session or not. Now, this raises a possible case of in=
cluding both "dtmf" and "external" mechanisms in the O/A. If DTMF is succes=
sfully negotiated but it fails, we can't really use the "external" mechanis=
m since the call is already accepted. Implementations could in this case re=
lease the CS call and try a new O/A without the DTMF.

I'm finding external correlation to be underspecified.

One one I could conceive of it working is that the incoming CS call is trea=
ted something like a call waiting situation. The existing (sip) call is "su=
spended" and the user is connected to the CS call, and asked to confirm if =
it is the intended one. If the user signals confirmation, then the sip call=
 is resumed, with the CS audio stream in it.

If the external correlation fails, then the sip call is resumed without the=
 CS media.

A problem with this is that the O/A completes prior to the CS call establis=
hment and correlation. If this happens mid-call (because of a desire to swi=
tch from RTP to CS media) then the RTP media will be abandoned once the O/A=
 succeeds, before correlation. So while waiting for a CS call to arrive and=
 be correlated you have no RTP media. :-(

[SV] Ok now I think I see your point. The O/A needs to be completed before =
correlation can take place. If this happens mid-call, it can indeed happen =
that the correlation fails and there is no media. On the other hand, there =
are a number of reasons why the same could happen to RTP based media as wel=
l. So, it can either be considered as an call failure case, or the other po=
ssibility would be to first add the CS media, and if it succeeds, do anothe=
r O/A to remove the RTP based media. The drawback in this case is that the =
endpoints would receive two media streams for a short while.

> - what if multiple incoming calls arrive before correlation
>     has been completed? (If correlation occurs before exposing the
>     media to the user, then it isn't an issue.)
>
> [SV] This would be an issue for the DTMF case, as there correlation can b=
e only only after accepting the CS call. An unlikely case, but still possib=
le. In this case, one of the CS calls could be related to the session, whil=
e other(s) would be normal CS calls. As only one CS call can be handled at =
a time, others would either wait or be terminated, depending on the callee'=
s supplementary services settings.

For DTMF, the correlation could be done *before* relating the CS call to th=
e session. Only bind it to the session after correlation is complete.

But that doesn't work as well for external correlation.

[SV] For both DTMF and external, the CS call needs to be accepted first, bu=
t when using DTMF it is possible (and advisable) to not play back the media=
 to the user until correlation has been done. For external case, the media =
would need to be played back .

> - extern correlation complicates this. It isn't like the other
>     correlation mechanisms - the user must be exposed to the media
>     to decide if there is correlation. How would the user indicate
>     "this is the wrong call" in a way that would allow correlation
>     to be attempted on another call?
>
> [SV] The external correlation mechanism does not assume the media flows b=
efore "correlation" is made - if the user decides to accept the incoming ca=
ll, it is considered as "correlated".

The user has to receive (hear) the media (and maybe send some too) before h=
e can decide on correlation. Has the CS call be "related to the session" th=
en, or not?

[SV] From a protocol point of view, I would say the call is considered as r=
elated to the session if the user accepts the call. But, you're right that =
the user cannot be really sure before some words are exchanged on the CS ca=
ll.

What happens if the user decides the CS call *isn't* correlated?

[SV] The user would likely hang up.

> Now consider a call that is initially established with one RTP media stre=
am. Later, a reinvite is sent with an offer proposing to to replace that wi=
th a CS media stream:
>
> - again, there must be an answer before the CS call can be attempted.
>
>
> [SV] For being able to perform correlation, yes, the Answer needs to be r=
eceived.
>
> - are you assuming that the reinvite will complete before the CS
>     call does? If so, then you are in a break-before-make situation -
>     the RTP session is gone before the CS session takes its place.
>
> [SV] The O/A needs to be completed before correlation can be done, but CS=
 call may arrive before Answer. In this case the endpoint should wait until=
 the Answer is received. If the CS call does not succeed, or correlation is=
 unsuccessful, then yes there is no media as the RTP stream has been droppe=
d. But, this could also happen when re-negotiating RTP media types (endpoin=
ts are not able to load correct codecs, new IP addr/port does not work etc.=
) so the situation not be specific to the CS media type.

Yes, it is not entirely unique, but usually updated O/As are done with an o=
ption in common to the prior one so that failure is unlikely. The failure m=
odes are quite different when transitioning from RTP to CS.=20
(And also when transitioning from CS to RTP.)

[SV] Agreed, adding a CS call introduces an additional level of uncertainty=
.

> - all the issues above also apply here.
>
> ISTM that there is a bunch of missing explanation of how this is expected=
 to work.
>
> [SV] Could you check the above whether my comments clarify the situation.=
 We can address them in more details in the draft as well once we know wher=
e more explanation is needed.

Upon rereading section 5.6.2 I realize that it does say that the answer mus=
t be received before the CS call is placed and correlated, so it implies a =
break-before-make approach on reINVITE.

What I still think is missing is:

* explanation of how things work when changing media mid call - from RTP to=
 CS, CS to RTP, or even CS to different CS. Point out that this is break-be=
fore-make, and when the user will again perceive to have media.

* much about external correlation. Is this really a mechanism in the sense =
that the others are, that serves as a screen on CS calls until the one is f=
ound that correlates? Or does it really mean that any incoming CS call is t=
aking as the right one, included in the sip session, with the user left to =
take independent action (such as hanging up) if he decides it isn't the rig=
ht one?

* For DTMF correlation, does the user see media in the call before the corr=
elation is successfully completed.

* potentially (but maybe not in this draft) a discussion of how to achieve =
make-before-break semantics. (E.g., Using multiple m-lines.)

[SV] I need to go through these with more thought with my co-author. We'll =
get back to this.

Thanks
Simo

	Thanks,
	Paul

> I also have a nit in section 5.3.2:
>
>      The fourth correlation mechanism declares support for cases where
>      correlation is done by external means.  Typically this means that th=
e
>      decision is left for the human user.  This is the way how some
>      current conferencing systems operate: ...
>
> "This is the way how some current conferencing systems operate"
>
> should be:
>
> "This is how some current conferencing systems operate"
>
> [SV] Fill fix.
>
> Thanks,
> Simo
>
> 	Thanks,
> 	Paul
>
> On 1/31/14 6:01 AM, Simo.Veikkolainen@nokia.com wrote:
>> Hello WG,
>>
>> We have submitted a new version of
>> http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-22
>>
>> To recap, there was discussion on the  MMUSIC WG list on which correlati=
on mechanisms should be mandatory to implement (see Gonzalo's email http://=
www.ietf.org/mail-archive/web/mmusic/current/msg12274.html ). As stated by =
Gonzalo, 3GPP is going to use this draft, more specifically the plan now is=
 to use the caller id based correlation mechanism. This was the message I g=
ot from Andrew after a recent 3GPP CT1 meeting.
>>
>> We have now made the following changes to the draft:
>> -	the endpoint may choose which correlation mechanisms to implement (bef=
ore all three were MTI).
>> -	we have introduced a fourth correlation type ("external") which indica=
tes explicit support for cases where also some external means, typically th=
e human user, can do the correlation. This can be as a fallback in cases wh=
ere no other common correlation mechanism is found, or the negotiated corre=
lation mechanisms fail due to being not supported in the PSTN network. This=
 is much like current systems like Webex work.
>>
>> This was also the guidance from the IESG informal telechat in December w=
here Gonzalo presented the draft.
>>
>> We would like to hear from the WG whether there are any concerns with th=
ese changes.
>>
>> On behalf of the authors,
>> Simo
>>
>> PS. I'll add the proposed change in http://www.ietf.org/mail-archive/web=
/mmusic/current/msg12792.html to next version, slipped my mind this time.
>>
>>
>> -----Original Message-----
>> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of ext=20
>> internet-drafts@ietf.org
>> Sent: 31 January, 2014 12:25
>> To: i-d-announce@ietf.org
>> Cc: mmusic@ietf.org
>> Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.txt
>>
>>
>> A New Internet-Draft is available from the on-line Internet-Drafts direc=
tories.
>>    This draft is a work item of the Multiparty Multimedia Session Contro=
l Working Group of the IETF.
>>
>>           Title           : Session Description Protocol (SDP) Extension=
 For Setting Audio and Video Media Streams Over Circuit-Switched Bearers In=
 The Public Switched Telephone Network (PSTN)
>>           Authors         : Miguel A. Garcia-Martin
>>                             Simo Veikkolainen
>> 	Filename        : draft-ietf-mmusic-sdp-cs-22.txt
>> 	Pages           : 37
>> 	Date            : 2014-01-31
>>
>> Abstract:
>>      This memo describes use cases, requirements, and protocol extension=
s
>>      for using the Session Description Protocol (SDP) Offer/Answer model
>>      for establishing audio and video media streams over circuit-switche=
d
>>      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-22
>>
>> A diff from the previous version is available at:
>> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sdp-cs-22
>>
>>
>> Please note that it may take a couple of minutes from the time of submis=
sion until the htmlized version and diff are available at tools.ietf.org.
>>
>> 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
>> _______________________________________________
>> 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 ari.keranen@ericsson.com  Fri Feb  7 02:43:58 2014
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Date: Fri, 7 Feb 2014 12:43:52 +0200
From: =?ISO-8859-1?Q?Ari_Ker=E4nen?= <ari.keranen@ericsson.com>
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Cc: Flemming Andreasen <fandreas@cisco.com>
Subject: [MMUSIC] MMUSIC Agenda Requests for IETF 89
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Hi all,

MMUSIC has two 2,5 hour sessions tentatively scheduled for the IETF 89 
meeting at London: http://tools.ietf.org/agenda/89/

If you want to request a time slot to discuss a draft, please send an 
e-mail to the chairs as soon as possible, but latest by the end of 
Wednesday 12th.

As usual, we want to ensure we take advantage of our meeting time in a 
productive manner that leads to forward progress. This means that we 
want presenters to come prepared to discuss open issues that have 
already been raised on the mailing list and ideally received some 
feedback already. Priority will be given to drafts that gather attention 
and discussion on the mailing list prior to the meeting. We also aim to 
give adequate time for drafts holding progress elsewhere. To facilitate 
this, please specify in your agenda request e-mail the following:

- name of the draft and the presenter

- how much time would you need

- outline of major open issues; what needs to be discussed at the meeting

- dependencies to your draft (if any)


Thanks,
Ari & Flemming

From internet-drafts@ietf.org  Fri Feb  7 13:23:03 2014
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-trickle-ice-01.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Multiparty Multimedia Session Control Working Group of the IETF.

        Title           : Trickle ICE: Incremental Provisioning of Candidates for the Interactive Connectivity Establishment (ICE) Protocol
        Authors         : Emil Ivov
                          Eric Rescorla
                          Justin Uberti
	Filename        : draft-ietf-mmusic-trickle-ice-01.txt
	Pages           : 25
	Date            : 2014-02-07

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

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


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-trickle-ice-01

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-trickle-ice-01


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

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


From emcho@sip-communicator.org  Fri Feb  7 13:27:56 2014
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From: Emil Ivov <emcho@jitsi.org>
Date: Fri, 7 Feb 2014 22:27:33 +0100
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Subject: [MMUSIC] Fwd: New Version Notification for draft-ietf-mmusic-trickle-ice-01.txt
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Hey all,

This version incorporates most of the comments we have received here
and closes the open issues (which are still in the text for
completeness).

A summary of the changes is available here:
http://tools.ietf.org/html/draft-ietf-mmusic-trickle-ice-01#appendix-B.1

The authors would hence like to ask the working group for feedback on
whether there is anything else left, that is not currently in the
draft.

Cheers,
Emil

P.S. An update of the SIP usage should follow in the next days.



---------- Forwarded message ----------
From:  <internet-drafts@ietf.org>
Date: Fri, Feb 7, 2014 at 10:23 PM
Subject: New Version Notification for draft-ietf-mmusic-trickle-ice-01.txt
To: Eric Rescorla <ekr@rtfm.com>, Emil Ivov <emcho@jitsi.org>, Justin
Uberti <justin@uberti.name>



A new version of I-D, draft-ietf-mmusic-trickle-ice-01.txt
has been successfully submitted by Emil Ivov and posted to the
IETF repository.

Name:           draft-ietf-mmusic-trickle-ice
Revision:       01
Title:          Trickle ICE: Incremental Provisioning of Candidates
for the Interactive Connectivity Establishment (ICE) Protocol
Document date:  2014-02-07
Group:          mmusic
Pages:          25
URL:
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-trickle-ice-01.txt
Status:         https://datatracker.ietf.org/doc/draft-ietf-mmusic-trickle-ice/
Htmlized:       http://tools.ietf.org/html/draft-ietf-mmusic-trickle-ice-01
Diff:
http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-trickle-ice-01

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

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




Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

The IETF Secretariat



-- 
Emil Ivov, Ph.D.                       67000 Strasbourg,
Project Lead                           France
Jitsi
emcho@jitsi.org                        PHONE: +33.1.77.62.43.30
https://jitsi.org                      FAX:   +33.1.77.62.47.31

From Simo.Veikkolainen@nokia.com  Mon Feb 10 02:45:46 2014
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From: <Simo.Veikkolainen@nokia.com>
To: <pkyzivat@alum.mit.edu>, <mmusic@ietf.org>
Thread-Topic: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.txt
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Date: Mon, 10 Feb 2014 10:44:21 +0000
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Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.txt
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Hello Paul, all,

I've copied the bullets from your email below, and propose the following te=
xt to address your concerns:

* explanation of how things work when changing media mid call - from RTP to=
 CS, CS to RTP, or even CS to different CS. Point out that this is break-be=
fore-make, and when the user will again perceive to have media.

-> Add in 5.6.4. Modifying the session

If either party would like to remove existing RTP based media from the sess=
ion and replace that with a circuit-switched bearer, it would create a new =
Offer to add the circuit-switched media as described in Section 5.6.1. abov=
e, and remove the RTP based media by setting the port number to zero. Once =
the Offer/Answer exchange is done, but the circuit-switched bearer is not y=
et established, there may be a period of time when no media is available. A=
lso, it may happen that correlating the circuit-switched call fails for rea=
sons discussed in Section 5.3.3. In this case, even if the Offer/Answer exc=
hange was successful, endpoints are not able to receive or send media. It i=
s up to the implementation to decide the behavior in this case; if nothing =
else is done, the user most likely hangs up after a while if there was no o=
ther media in the session. Note that this may also happen when switching fr=
om RTP media to another RTP media (for example when firewall blocks the new=
 media stream).

If either party would like to remove existing circuit-switched media from t=
he session and replace that with RTP based media, it would create a new Off=
er and set the circuit-switched port number to zero, and add a description =
for RTP based media. The endpoint MUST then terminate the circuit-switched =
bearer using whatever mechanism is appropriate for the technology in questi=
on.

* much about external correlation. Is this really a mechanism in the sense =
that the others are, that serves as a screen on CS calls until the one is f=
ound that correlates? Or does it really mean that any incoming CS call is t=
aking as the right one, included in the sip session, with the user left to =
take independent action (such as hanging up) if he decides it isn't the rig=
ht one?

-> add in 5.2.3.5. The External correlation mechanism

The external correlation mechanism relies on the human user to do the decis=
ion on whether the call is related to the ongoing session or not. After the=
 user accepts the call, that bearer is considered as related to the session=
. There is a small chance that the user receives at the same time another c=
ircuit-switched call which is not related to the ongoing session. The user =
may reject this call if he is able to determine (e.g. based on the calling =
line identification) that the call is not related to the session, and conti=
nue waiting for another call attempt. If the user accepts the incoming circ=
uit-switched call, but it turns out to be not related to the session, the e=
ndpoints need to rely on the human user to take appropriate action (typical=
ly, they would hang up).

* For DTMF correlation, does the user see media in the call before the corr=
elation is successfully completed.

I think this is an implementation specific matter. We could add a recommend=
ation of some sorts, but really it is up to the implementation to decide.

* potentially (but maybe not in this draft) a discussion of how to achieve =
make-before-break semantics. (E.g., Using multiple m-lines.)

This seems to be a larger issue which is not specific to circuit-switched m=
edia alone, to probably it would be best to address it in a standalone docu=
ment.

Regards,
Simo

-----Original Message-----
From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of Veikkolainen Sim=
o (Nokia-CTO/Espoo)
Sent: 07 February, 2014 10:38
To: pkyzivat@alum.mit.edu; mmusic@ietf.org
Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.txt

See below [SV]

-----Original Message-----
From: ext Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
Sent: 05 February, 2014 21:11
To: Veikkolainen Simo (Nokia-CTO/Espoo); mmusic@ietf.org
Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.txt

On 2/4/14 2:18 AM, Simo.Veikkolainen@nokia.com wrote:
>
>
> -----Original Message-----
> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of ext Paul=20
> Kyzivat
> Sent: 31 January, 2014 20:53
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.txt
>
> I just reviewed the new version. First I looked at just the diffs. But th=
ose raised some more general questions in my mind, so I re-reviewed more of=
 the draft. I'm left with some considerable confusion about how this is int=
ended to work in practice, especially with extern correlation, but even wit=
h the other forms.
>
> In particular, I am unclear of the sequencing of O/A processing,=20
> correlation processing, and signaling completion. (I realize this is=20
> an mmusic draft and theoretically independent of sip, but forgive me=20
> if I discuss this in the context of sip.)
>
> Normally, when first establishing a session with sip, the first O/A may o=
ccur before the signaling session is established. But early media may flow =
after the first O/A. Often it is expected that the user will be be aware an=
d involved in this early media.
>
> Once a session is up, it can be changed with a reinvite containing a new =
O/A. If the O/A and the reinvite are successful, then the results of the ne=
w O/A replace the old one. If the reinvite fails, then the session is suppo=
sed to continue as it was before the reinvite. In the interim, from the sen=
ding of the reinvite until its success or failure, things are a little conf=
using - presumably both old and new media streams must be maintained, but w=
hich are connected to the user is unspecified.
>
> I see nothing in this draft that explains how the CS media, and correlati=
on verification, fit into the above process.
>
> Consider a new call that offers CS media:
>
> - The first O/A must complete before the CS call can be attempted.
>
> - must the CS call complete and be correlated successfully before
>     the INVITE gets a 200? If the invite is allowed to complete first,
>     what happens if no CS call is established, or if it is but
>     correlation fails?
>
> [SV] I assume you mean the 200 carries the Answer.

Not necessarily. I was just trying to sort out the expectations for the tim=
ing of the events. The answer could be *before* the 200, but it must be bef=
ore the call is established.

> It may happen that the CS call arrives before the Answer, but in this cas=
e no correlation can be done. See Section 5.6.3. If the invite (I assume yo=
u mean O/A) completes first, but no CS call ever arrives, or correlation - =
including the "external" mechanism fails, the endpoint should treat the CS =
call as unrelated; this is described in Section 5.3.3.

Sure, but my question is whether correlation is expected to occur before or=
 after the INVITE completes.

[SV] Correlation can only occur after the correlation information has been =
exchanged, and the CS call is received. I'm not sure what you mean by INVIT=
E being completed, but I would say correlation can be done already before t=
he last request (ACK) has been sent, or after the whole INVITE related requ=
est-response exchange has been completed (in case the CS call is received l=
ate).

> - when does CS media start to flow to the user as part of the
>     SIP call? Is it before or after correlation? (E.g., does the
>     user hear the DTMF exchange?)
>
> [SV] CS media starts to flow after the call has been accepted, as in a no=
rmal CS call. Correlation may be possible before accepting the call (for "c=
allerid", "uuie", and "external" mechanisms), and for "dtmf" case only afte=
r the call has been accepted.

I should have been clearer. For DTMF and external correlation media clearly=
 needs to flow on the CS call. My question pertains to whether this is cons=
idered part of the SIP session prior to correlation.

[SV] Again, I'm not sure I'm able to follow your thought but correlation is=
 needed before the CS media is considered as part of the session.

> - if media flows to the user before correlation, then what if
>     correlation fails? Will another call be accepted and correlation
>     be attempted again?
>
> [SV] I don' think we have explicitly defined this situation. If a call is=
 accepted and *after* that correlation fails, it means that the DTMF mechan=
ism was used (for other correlation mechanisms, we know before accepting th=
e call whether the correlation was successful).

external correlation via also has this problem, and more.

> If the DTMF pattern is not received correctly, as we cannot know whether =
it is related to the session or not. Now, this raises a possible case of in=
cluding both "dtmf" and "external" mechanisms in the O/A. If DTMF is succes=
sfully negotiated but it fails, we can't really use the "external" mechanis=
m since the call is already accepted. Implementations could in this case re=
lease the CS call and try a new O/A without the DTMF.

I'm finding external correlation to be underspecified.

One one I could conceive of it working is that the incoming CS call is trea=
ted something like a call waiting situation. The existing (sip) call is "su=
spended" and the user is connected to the CS call, and asked to confirm if =
it is the intended one. If the user signals confirmation, then the sip call=
 is resumed, with the CS audio stream in it.

If the external correlation fails, then the sip call is resumed without the=
 CS media.

A problem with this is that the O/A completes prior to the CS call establis=
hment and correlation. If this happens mid-call (because of a desire to swi=
tch from RTP to CS media) then the RTP media will be abandoned once the O/A=
 succeeds, before correlation. So while waiting for a CS call to arrive and=
 be correlated you have no RTP media. :-(

[SV] Ok now I think I see your point. The O/A needs to be completed before =
correlation can take place. If this happens mid-call, it can indeed happen =
that the correlation fails and there is no media. On the other hand, there =
are a number of reasons why the same could happen to RTP based media as wel=
l. So, it can either be considered as an call failure case, or the other po=
ssibility would be to first add the CS media, and if it succeeds, do anothe=
r O/A to remove the RTP based media. The drawback in this case is that the =
endpoints would receive two media streams for a short while.

> - what if multiple incoming calls arrive before correlation
>     has been completed? (If correlation occurs before exposing the
>     media to the user, then it isn't an issue.)
>
> [SV] This would be an issue for the DTMF case, as there correlation can b=
e only only after accepting the CS call. An unlikely case, but still possib=
le. In this case, one of the CS calls could be related to the session, whil=
e other(s) would be normal CS calls. As only one CS call can be handled at =
a time, others would either wait or be terminated, depending on the callee'=
s supplementary services settings.

For DTMF, the correlation could be done *before* relating the CS call to th=
e session. Only bind it to the session after correlation is complete.

But that doesn't work as well for external correlation.

[SV] For both DTMF and external, the CS call needs to be accepted first, bu=
t when using DTMF it is possible (and advisable) to not play back the media=
 to the user until correlation has been done. For external case, the media =
would need to be played back .

> - extern correlation complicates this. It isn't like the other
>     correlation mechanisms - the user must be exposed to the media
>     to decide if there is correlation. How would the user indicate
>     "this is the wrong call" in a way that would allow correlation
>     to be attempted on another call?
>
> [SV] The external correlation mechanism does not assume the media flows b=
efore "correlation" is made - if the user decides to accept the incoming ca=
ll, it is considered as "correlated".

The user has to receive (hear) the media (and maybe send some too) before h=
e can decide on correlation. Has the CS call be "related to the session" th=
en, or not?

[SV] From a protocol point of view, I would say the call is considered as r=
elated to the session if the user accepts the call. But, you're right that =
the user cannot be really sure before some words are exchanged on the CS ca=
ll.

What happens if the user decides the CS call *isn't* correlated?

[SV] The user would likely hang up.

> Now consider a call that is initially established with one RTP media stre=
am. Later, a reinvite is sent with an offer proposing to to replace that wi=
th a CS media stream:
>
> - again, there must be an answer before the CS call can be attempted.
>
>
> [SV] For being able to perform correlation, yes, the Answer needs to be r=
eceived.
>
> - are you assuming that the reinvite will complete before the CS
>     call does? If so, then you are in a break-before-make situation -
>     the RTP session is gone before the CS session takes its place.
>
> [SV] The O/A needs to be completed before correlation can be done, but CS=
 call may arrive before Answer. In this case the endpoint should wait until=
 the Answer is received. If the CS call does not succeed, or correlation is=
 unsuccessful, then yes there is no media as the RTP stream has been droppe=
d. But, this could also happen when re-negotiating RTP media types (endpoin=
ts are not able to load correct codecs, new IP addr/port does not work etc.=
) so the situation not be specific to the CS media type.

Yes, it is not entirely unique, but usually updated O/As are done with an o=
ption in common to the prior one so that failure is unlikely. The failure m=
odes are quite different when transitioning from RTP to CS.=20
(And also when transitioning from CS to RTP.)

[SV] Agreed, adding a CS call introduces an additional level of uncertainty=
.

> - all the issues above also apply here.
>
> ISTM that there is a bunch of missing explanation of how this is expected=
 to work.
>
> [SV] Could you check the above whether my comments clarify the situation.=
 We can address them in more details in the draft as well once we know wher=
e more explanation is needed.

Upon rereading section 5.6.2 I realize that it does say that the answer mus=
t be received before the CS call is placed and correlated, so it implies a =
break-before-make approach on reINVITE.

What I still think is missing is:

* explanation of how things work when changing media mid call - from RTP to=
 CS, CS to RTP, or even CS to different CS. Point out that this is break-be=
fore-make, and when the user will again perceive to have media.

* much about external correlation. Is this really a mechanism in the sense =
that the others are, that serves as a screen on CS calls until the one is f=
ound that correlates? Or does it really mean that any incoming CS call is t=
aking as the right one, included in the sip session, with the user left to =
take independent action (such as hanging up) if he decides it isn't the rig=
ht one?

* For DTMF correlation, does the user see media in the call before the corr=
elation is successfully completed.

* potentially (but maybe not in this draft) a discussion of how to achieve =
make-before-break semantics. (E.g., Using multiple m-lines.)

[SV] I need to go through these with more thought with my co-author. We'll =
get back to this.

Thanks
Simo

	Thanks,
	Paul

> I also have a nit in section 5.3.2:
>
>      The fourth correlation mechanism declares support for cases where
>      correlation is done by external means.  Typically this means that th=
e
>      decision is left for the human user.  This is the way how some
>      current conferencing systems operate: ...
>
> "This is the way how some current conferencing systems operate"
>
> should be:
>
> "This is how some current conferencing systems operate"
>
> [SV] Fill fix.
>
> Thanks,
> Simo
>
> 	Thanks,
> 	Paul
>
> On 1/31/14 6:01 AM, Simo.Veikkolainen@nokia.com wrote:
>> Hello WG,
>>
>> We have submitted a new version of
>> http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-22
>>
>> To recap, there was discussion on the  MMUSIC WG list on which correlati=
on mechanisms should be mandatory to implement (see Gonzalo's email http://=
www.ietf.org/mail-archive/web/mmusic/current/msg12274.html ). As stated by =
Gonzalo, 3GPP is going to use this draft, more specifically the plan now is=
 to use the caller id based correlation mechanism. This was the message I g=
ot from Andrew after a recent 3GPP CT1 meeting.
>>
>> We have now made the following changes to the draft:
>> -	the endpoint may choose which correlation mechanisms to implement (bef=
ore all three were MTI).
>> -	we have introduced a fourth correlation type ("external") which indica=
tes explicit support for cases where also some external means, typically th=
e human user, can do the correlation. This can be as a fallback in cases wh=
ere no other common correlation mechanism is found, or the negotiated corre=
lation mechanisms fail due to being not supported in the PSTN network. This=
 is much like current systems like Webex work.
>>
>> This was also the guidance from the IESG informal telechat in December w=
here Gonzalo presented the draft.
>>
>> We would like to hear from the WG whether there are any concerns with th=
ese changes.
>>
>> On behalf of the authors,
>> Simo
>>
>> PS. I'll add the proposed change in http://www.ietf.org/mail-archive/web=
/mmusic/current/msg12792.html to next version, slipped my mind this time.
>>
>>
>> -----Original Message-----
>> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of ext=20
>> internet-drafts@ietf.org
>> Sent: 31 January, 2014 12:25
>> To: i-d-announce@ietf.org
>> Cc: mmusic@ietf.org
>> Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.txt
>>
>>
>> A New Internet-Draft is available from the on-line Internet-Drafts direc=
tories.
>>    This draft is a work item of the Multiparty Multimedia Session Contro=
l Working Group of the IETF.
>>
>>           Title           : Session Description Protocol (SDP) Extension=
 For Setting Audio and Video Media Streams Over Circuit-Switched Bearers In=
 The Public Switched Telephone Network (PSTN)
>>           Authors         : Miguel A. Garcia-Martin
>>                             Simo Veikkolainen
>> 	Filename        : draft-ietf-mmusic-sdp-cs-22.txt
>> 	Pages           : 37
>> 	Date            : 2014-01-31
>>
>> Abstract:
>>      This memo describes use cases, requirements, and protocol extension=
s
>>      for using the Session Description Protocol (SDP) Offer/Answer model
>>      for establishing audio and video media streams over circuit-switche=
d
>>      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-22
>>
>> A diff from the previous version is available at:
>> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sdp-cs-22
>>
>>
>> Please note that it may take a couple of minutes from the time of submis=
sion until the htmlized version and diff are available at tools.ietf.org.
>>
>> 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
>> _______________________________________________
>> 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: "Tirumaleswar Reddy (tireddy)" <tireddy@cisco.com>
To: Oleg Moskalenko <mom040267@gmail.com>
Thread-Topic: [tram] FW: New Version Notification for draft-wing-mmusic-ice-mobility-06.txt
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Subject: Re: [MMUSIC] [tram] FW: New Version Notification for draft-wing-mmusic-ice-mobility-06.txt
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Hi Oleg,

Inline [TR]

From: Oleg Moskalenko [mailto:mom040267@gmail.com]
Sent: Thursday, February 06, 2014 10:43 PM
To: Tirumaleswar Reddy (tireddy)
Cc: mmusic@ietf.org<mailto:mmusic@ietf.org>; tram@ietf.org<mailto:tram@ietf=
.org>
Subject: Re: [tram] FW: New Version Notification for draft-wing-mmusic-ice-=
mobility-06.txt

I have comments regarding the new Error Code "Mobility Forbidden".
1) Section 5.1.4:

Comment: remove two TBDs, if the error code has been already determined.
[TR] Removed
2) Section 7:

and to add a new STUN error code "Mobility Forbidden" with the value

501 to the STUN Error Codes registry [iana-stun].

Comment: I am not sure that 501 is the right code to be used. In SIP and HT=
TP, 5xx errors are usually about a server code problem of some sort; 501 me=
ans "not implemented". That may be not exactly accurate. A mobility-aware T=
URN server may reject the request simply because the administrator does not=
 allow it., and the server did not encounter an error. The reason why the s=
erver reject the request is rather "not allowed" (which is the error code 4=
05).
[TR] we can replace 501 error code with 403
May be be, the right approach would be to introduce two separate error code=
s here: 405 "not allowed" (the server implements mobility but the administr=
ator forbade it) or 501 "not implemented" (the server knows about mobility =
but it has no implementation for that).
[TR] Not sure if "not implemented" error will be of any help. TURN server i=
f it knows about mobility but has not implemented the functionality then it=
 can continue to ignore MOBILITY-TICKET in the request and not send MOBILIT=
Y-TICKET in the response.
-Tiru.
If the server truly is not aware about mobility at all, the server will sen=
d no MOBILITY-TICKET in the server response. Those are three distinct use c=
ases: 405, 501 and an "ignorant" server.
Thanks
Oleg

On Wed, Feb 5, 2014 at 6:17 AM, Tirumaleswar Reddy (tireddy) <tireddy@cisco=
.com<mailto:tireddy@cisco.com>> wrote:
Revised draft-wing-mmusic-ice-mobility-06 has been published. It addresses =
the comments received for Mobility using TURN and a new section with implem=
entation details. Further comments and suggestions are welcome.

Thanks and Regards,
-Authors.

-----Original Message-----
From: internet-drafts@ietf.org<mailto:internet-drafts@ietf.org> [mailto:int=
ernet-drafts@ietf.org<mailto:internet-drafts@ietf.org>]
Sent: Wednesday, February 05, 2014 7:44 PM
To: Prashanth Patil (praspati); Tirumaleswar Reddy (tireddy); Prashanth Pat=
il (praspati); Dan Wing (dwing); Tirumaleswar Reddy (tireddy); Pal Martinse=
n (palmarti); Dan Wing (dwing); Pal Martinsen (palmarti)
Subject: New Version Notification for draft-wing-mmusic-ice-mobility-06.txt


A new version of I-D, draft-wing-mmusic-ice-mobility-06.txt
has been successfully submitted by Tirumaleswar Reddy and posted to the IET=
F repository.

Name:           draft-wing-mmusic-ice-mobility
Revision:       06
Title:          Mobility with ICE (MICE)
Document date:  2014-02-05
Group:          Individual Submission
Pages:          21
URL:            http://www.ietf.org/internet-drafts/draft-wing-mmusic-ice-m=
obility-06.txt
Status:         https://datatracker.ietf.org/doc/draft-wing-mmusic-ice-mobi=
lity/
Htmlized:       http://tools.ietf.org/html/draft-wing-mmusic-ice-mobility-0=
6
Diff:           http://www.ietf.org/rfcdiff?url2=3Ddraft-wing-mmusic-ice-mo=
bility-06

Abstract:
   This specification describes how endpoint mobility can be achieved
   using ICE.  Two mechanisms are shown, one where both endpoints
   support MICE and another where only one endpoint supports MICE.




Please note that it may take a couple of minutes from the time of submissio=
n until the htmlized version and diff are available at tools.ietf.org<http:=
//tools.ietf.org>.

The IETF Secretariat

_______________________________________________
tram mailing list
tram@ietf.org<mailto:tram@ietf.org>
https://www.ietf.org/mailman/listinfo/tram


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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi Oleg,<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">Inline [TR]<o:p></o:p></s=
pan></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 style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Oleg Mos=
kalenko [</span><a href=3D"mailto:mom040267@gmail.com"><span style=3D"font-=
size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">mailto:m=
om040267@gmail.com</span></a><span style=3D"font-size:10.0pt;font-family:&q=
uot;Tahoma&quot;,&quot;sans-serif&quot;">]
<br>
<b>Sent:</b> Thursday, February 06, 2014 10:43 PM<br>
<b>To:</b> Tirumaleswar Reddy (tireddy)<br>
<b>Cc:</b> </span><a href=3D"mailto:mmusic@ietf.org"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">mmusic@iet=
f.org</span></a><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&qu=
ot;,&quot;sans-serif&quot;">;
</span><a href=3D"mailto:tram@ietf.org"><span style=3D"font-size:10.0pt;fon=
t-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">tram@ietf.org</span></a=
><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-=
serif&quot;"><br>
<b>Subject:</b> Re: [tram] FW: New Version Notification for draft-wing-mmus=
ic-ice-mobility-06.txt<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<div>
<div>
<div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">I have comments regar=
ding the new Error Code &quot;Mobility Forbidden&quot;.<o:p></o:p></p>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">1) Section 5.1.4: <br=
>
<br>
Comment: remove two TBDs, if the error code has been already determined.<o:=
p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"color:=
#1F497D">[TR] Removed<o:p></o:p></span></p>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">2) Section 7:<o:p></o=
:p></p>
<pre>and to add a new STUN error code &quot;Mobility Forbidden&quot; with t=
he value<o:p></o:p></pre>
<pre>501 to the STUN Error Codes registry [iana-stun].<o:p></o:p></pre>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">Comment: I am not sur=
e that 501 is the right code to be used. In SIP and HTTP, 5xx errors are us=
ually about a server code problem of some sort; 501 means &quot;not impleme=
nted&quot;. That may be not exactly accurate.
 A mobility-aware TURN server may reject the request simply because the adm=
inistrator does not allow it., and the server did not encounter an error. T=
he reason why the server reject the request is rather &quot;not allowed&quo=
t; (which is the error code 405).<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F=
497D">[TR] we can replace 501 error code with 403
<o:p></o:p></span></p>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">May be be, the right =
approach would be to introduce two separate error codes here: 405 &quot;not=
 allowed&quot; (the server implements mobility but the administrator forbad=
e it) or 501 &quot;not implemented&quot; (the server knows
 about mobility but it has no implementation for that). <span style=3D"colo=
r:#1F497D">
<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F=
497D">[TR] Not sure if &#8220;not implemented&#8221; error will be of any h=
elp. TURN server if it knows about mobility but has not implemented the
 functionality then it can continue to ignore MOBILITY-TICKET in the reques=
t and not send MOBILITY-TICKET in the response.<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F=
497D">-Tiru.<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">If the server truly i=
s not aware about mobility at all, the server will send no MOBILITY-TICKET =
in the server response. Those are three distinct use cases: 405, 501 and an=
 &quot;ignorant&quot; server.
<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">Thanks<br>
Oleg<o:p></o:p></p>
</div>
</div>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On Wed, Feb 5, 2014 at 6:17 AM, Tirumaleswar Reddy (=
tireddy) &lt;<a href=3D"mailto:tireddy@cisco.com" target=3D"_blank">tireddy=
@cisco.com</a>&gt; wrote:<o:p></o:p></p>
<p class=3D"MsoNormal">Revised draft-wing-mmusic-ice-mobility-06 has been p=
ublished. It addresses the comments received for Mobility using TURN and a =
new section with implementation details. Further comments and suggestions a=
re welcome.<br>
<br>
Thanks and Regards,<br>
-Authors.<br>
<br>
-----Original Message-----<br>
From: <a href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org<=
/a> [mailto:<a href=3D"mailto:internet-drafts@ietf.org">internet-drafts@iet=
f.org</a>]<br>
Sent: Wednesday, February 05, 2014 7:44 PM<br>
To: Prashanth Patil (praspati); Tirumaleswar Reddy (tireddy); Prashanth Pat=
il (praspati); Dan Wing (dwing); Tirumaleswar Reddy (tireddy); Pal Martinse=
n (palmarti); Dan Wing (dwing); Pal Martinsen (palmarti)<br>
Subject: New Version Notification for draft-wing-mmusic-ice-mobility-06.txt=
<br>
<br>
<br>
A new version of I-D, draft-wing-mmusic-ice-mobility-06.txt<br>
has been successfully submitted by Tirumaleswar Reddy and posted to the IET=
F repository.<br>
<br>
Name: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; draft-wing-mmusic-ice-mobility<br>
Revision: &nbsp; &nbsp; &nbsp; 06<br>
Title: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Mobility with ICE (MICE)<br>
Document date: &nbsp;2014-02-05<br>
Group: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Individual Submission<br>
Pages: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;21<br>
URL: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;<a href=3D"http://www.ietf.or=
g/internet-drafts/draft-wing-mmusic-ice-mobility-06.txt" target=3D"_blank">=
http://www.ietf.org/internet-drafts/draft-wing-mmusic-ice-mobility-06.txt</=
a><br>
Status: &nbsp; &nbsp; &nbsp; &nbsp; <a href=3D"https://datatracker.ietf.org=
/doc/draft-wing-mmusic-ice-mobility/" target=3D"_blank">
https://datatracker.ietf.org/doc/draft-wing-mmusic-ice-mobility/</a><br>
Htmlized: &nbsp; &nbsp; &nbsp; <a href=3D"http://tools.ietf.org/html/draft-=
wing-mmusic-ice-mobility-06" target=3D"_blank">
http://tools.ietf.org/html/draft-wing-mmusic-ice-mobility-06</a><br>
Diff: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; <a href=3D"http://www.ietf.org/rfc=
diff?url2=3Ddraft-wing-mmusic-ice-mobility-06" target=3D"_blank">
http://www.ietf.org/rfcdiff?url2=3Ddraft-wing-mmusic-ice-mobility-06</a><br=
>
<br>
Abstract:<br>
&nbsp; &nbsp;This specification describes how endpoint mobility can be achi=
eved<br>
&nbsp; &nbsp;using ICE. &nbsp;Two mechanisms are shown, one where both endp=
oints<br>
&nbsp; &nbsp;support MICE and another where only one endpoint supports MICE=
.<br>
<br>
<br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n until the htmlized version and diff are available at
<a href=3D"http://tools.ietf.org" target=3D"_blank">tools.ietf.org</a>.<br>
<br>
The IETF Secretariat<br>
<br>
_______________________________________________<br>
tram mailing list<br>
<a href=3D"mailto:tram@ietf.org">tram@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/tram" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/tram</a><o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</div>
</body>
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Date: Mon, 10 Feb 2014 05:52:23 -0800
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-rtsp-nat-20.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the 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)
        Authors         : Jeff Goldberg
                          Magnus Westerlund
                          Thomas Zeng
	Filename        : draft-ietf-mmusic-rtsp-nat-20.txt
	Pages           : 32
	Date            : 2014-02-10

Abstract:
   This document defines a solution for Network Address Translation
   (NAT) traversal for datagram based media streams set up 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 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-20

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-rtsp-nat-20


Please note that it may take a couple of minutes from the time of submission
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From internet-drafts@ietf.org  Mon Feb 10 05:53:06 2014
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Date: Mon, 10 Feb 2014 05:53:03 -0800
Cc: mmusic@ietf.org
Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-rtsp-nat-evaluation-13.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Multiparty Multimedia Session Control Working Group of the IETF.

        Title           : The Evaluation of Different Network Address Translator (NAT) Traversal Techniques for Media Controlled by Real-time Streaming Protocol (RTSP)
        Authors         : Magnus Westerlund
                          Thomas Zeng
	Filename        : draft-ietf-mmusic-rtsp-nat-evaluation-13.txt
	Pages           : 40
	Date            : 2014-02-10

Abstract:
   This document describes several Network Address Translator (NAT)
   traversal techniques that were considered to be used for establishing
   the RTP media flows controlled by the Real-time Streaming Protocol
   (RTSP).  Each technique includes a description of how it would be
   used, the security implications of using it and any other deployment
   considerations it has.  There are also discussions on how NAT
   traversal techniques relate to firewalls and how each technique can
   be applied in different use cases.  These findings were used when
   selecting the NAT traversal for RTSP 2.0, which is specified in a
   separate document.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-rtsp-nat-evaluation/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-rtsp-nat-evaluation-13

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http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-rtsp-nat-evaluation-13


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From magnus.westerlund@ericsson.com  Mon Feb 10 05:57:21 2014
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Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-rtsp-nat-20.txt
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WG,

I received some additional editorial feedback from Flemming. These have
now been addressed. I expect the publication request on this and the
draft-ietf-mmusic-rtsp-nat-evaluation-13 to be submitted shortly.

Cheers

Magnus

On 2014-02-10 14:52, internet-drafts@ietf.org wrote:
> 
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>  This draft is a work item of the Multiparty Multimedia Session Control Working Group of the IETF.
> 
>         Title           : A Network Address Translator (NAT) Traversal Mechanism for Media Controlled by Real-Time Streaming Protocol (RTSP)
>         Authors         : Jeff Goldberg
>                           Magnus Westerlund
>                           Thomas Zeng
> 	Filename        : draft-ietf-mmusic-rtsp-nat-20.txt
> 	Pages           : 32
> 	Date            : 2014-02-10
> 
> Abstract:
>    This document defines a solution for Network Address Translation
>    (NAT) traversal for datagram based media streams set up 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 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-20
> 
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-rtsp-nat-20
> 
> 
> Please note that it may take a couple of minutes from the time of submission
> until the htmlized version and diff are available at tools.ietf.org.
> 
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
> 
> _______________________________________________
> I-D-Announce mailing list
> I-D-Announce@ietf.org
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> Internet-Draft directories: http://www.ietf.org/shadow.html
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> 
> 


-- 

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From internet-drafts@ietf.org  Mon Feb 10 07:25:09 2014
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-rfc2326bis-40.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Multiparty Multimedia Session Control Working Group of the IETF.

        Title           : Real Time Streaming Protocol 2.0 (RTSP)
        Authors         : Henning Schulzrinne
                          Anup Rao
                          Rob Lanphier
                          Magnus Westerlund
                          Martin Stiemerling
	Filename        : draft-ietf-mmusic-rfc2326bis-40.txt
	Pages           : 309
	Date            : 2014-02-10

Abstract:
   This memorandum defines RTSP version 2.0 which obsoletes RTSP version
   1.0 defined in RFC 2326.

   The Real Time Streaming Protocol, or RTSP, is an application-layer
   protocol for setup and control of the delivery of data with real-time
   properties.  RTSP provides an extensible framework to enable
   controlled, on-demand delivery of real-time data, such as audio and
   video.  Sources of data can include both live data feeds and stored
   clips.  This protocol is intended to control multiple data delivery
   sessions, provide a means for choosing delivery channels such as UDP,
   multicast UDP and TCP, and provide a means for choosing delivery
   mechanisms based upon RTP (RFC 3550).


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-rfc2326bis/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-rfc2326bis-40

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From pkyzivat@alum.mit.edu  Mon Feb 10 07:49:17 2014
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Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.txt
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Simo,

What you describe below is enough to get me to shut up. :-)

On 2/10/14 5:44 AM, Simo.Veikkolainen@nokia.com wrote:
> Hello Paul, all,
>
> I've copied the bullets from your email below, and propose the following text to address your concerns:
>
> * explanation of how things work when changing media mid call - from RTP to CS, CS to RTP, or even CS to different CS. Point out that this is break-before-make, and when the user will again perceive to have media.
>
> -> Add in 5.6.4. Modifying the session
>
> If either party would like to remove existing RTP based media from the session and replace that with a circuit-switched bearer, it would create a new Offer to add the circuit-switched media as described in Section 5.6.1. above, and remove the RTP based media by setting the port number to zero. Once the Offer/Answer exchange is done, but the circuit-switched bearer is not yet established, there may be a period of time when no media is available. Also, it may happen that correlating the circuit-switched call fails for reasons discussed in Section 5.3.3. In this case, even if the Offer/Answer exchange was successful, endpoints are not able to receive or send media. It is up to the implementation to decide the behavior in this case; if nothing else is done, the user most likely hangs up after a while if there was no other media in the session. Note that this may also happen when switching from RTP media to another RTP media (for example when firewall blocks the new media stream).

I find it interesting that you describe adding a new m-line for the cs 
session and rejecting the old one. It should also be valid to simply 
replace the old (rtp) m-line with the new (cs) m-line.

If you are thinking of using two m-lines, then make-before-break is just 
a matter of keeping them both active for awhile.

	Thanks,
	Paul

> If either party would like to remove existing circuit-switched media from the session and replace that with RTP based media, it would create a new Offer and set the circuit-switched port number to zero, and add a description for RTP based media. The endpoint MUST then terminate the circuit-switched bearer using whatever mechanism is appropriate for the technology in question.
>
> * much about external correlation. Is this really a mechanism in the sense that the others are, that serves as a screen on CS calls until the one is found that correlates? Or does it really mean that any incoming CS call is taking as the right one, included in the sip session, with the user left to take independent action (such as hanging up) if he decides it isn't the right one?
>
> -> add in 5.2.3.5. The External correlation mechanism
>
> The external correlation mechanism relies on the human user to do the decision on whether the call is related to the ongoing session or not. After the user accepts the call, that bearer is considered as related to the session. There is a small chance that the user receives at the same time another circuit-switched call which is not related to the ongoing session. The user may reject this call if he is able to determine (e.g. based on the calling line identification) that the call is not related to the session, and continue waiting for another call attempt. If the user accepts the incoming circuit-switched call, but it turns out to be not related to the session, the endpoints need to rely on the human user to take appropriate action (typically, they would hang up).
>
> * For DTMF correlation, does the user see media in the call before the correlation is successfully completed.
>
> I think this is an implementation specific matter. We could add a recommendation of some sorts, but really it is up to the implementation to decide.
>
> * potentially (but maybe not in this draft) a discussion of how to achieve make-before-break semantics. (E.g., Using multiple m-lines.)
>
> This seems to be a larger issue which is not specific to circuit-switched media alone, to probably it would be best to address it in a standalone document.
>
> Regards,
> Simo
>
> -----Original Message-----
> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of Veikkolainen Simo (Nokia-CTO/Espoo)
> Sent: 07 February, 2014 10:38
> To: pkyzivat@alum.mit.edu; mmusic@ietf.org
> Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.txt
>
> See below [SV]
>
> -----Original Message-----
> From: ext Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
> Sent: 05 February, 2014 21:11
> To: Veikkolainen Simo (Nokia-CTO/Espoo); mmusic@ietf.org
> Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.txt
>
> On 2/4/14 2:18 AM, Simo.Veikkolainen@nokia.com wrote:
>>
>>
>> -----Original Message-----
>> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of ext Paul
>> Kyzivat
>> Sent: 31 January, 2014 20:53
>> To: mmusic@ietf.org
>> Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.txt
>>
>> I just reviewed the new version. First I looked at just the diffs. But those raised some more general questions in my mind, so I re-reviewed more of the draft. I'm left with some considerable confusion about how this is intended to work in practice, especially with extern correlation, but even with the other forms.
>>
>> In particular, I am unclear of the sequencing of O/A processing,
>> correlation processing, and signaling completion. (I realize this is
>> an mmusic draft and theoretically independent of sip, but forgive me
>> if I discuss this in the context of sip.)
>>
>> Normally, when first establishing a session with sip, the first O/A may occur before the signaling session is established. But early media may flow after the first O/A. Often it is expected that the user will be be aware and involved in this early media.
>>
>> Once a session is up, it can be changed with a reinvite containing a new O/A. If the O/A and the reinvite are successful, then the results of the new O/A replace the old one. If the reinvite fails, then the session is supposed to continue as it was before the reinvite. In the interim, from the sending of the reinvite until its success or failure, things are a little confusing - presumably both old and new media streams must be maintained, but which are connected to the user is unspecified.
>>
>> I see nothing in this draft that explains how the CS media, and correlation verification, fit into the above process.
>>
>> Consider a new call that offers CS media:
>>
>> - The first O/A must complete before the CS call can be attempted.
>>
>> - must the CS call complete and be correlated successfully before
>>      the INVITE gets a 200? If the invite is allowed to complete first,
>>      what happens if no CS call is established, or if it is but
>>      correlation fails?
>>
>> [SV] I assume you mean the 200 carries the Answer.
>
> Not necessarily. I was just trying to sort out the expectations for the timing of the events. The answer could be *before* the 200, but it must be before the call is established.
>
>> It may happen that the CS call arrives before the Answer, but in this case no correlation can be done. See Section 5.6.3. If the invite (I assume you mean O/A) completes first, but no CS call ever arrives, or correlation - including the "external" mechanism fails, the endpoint should treat the CS call as unrelated; this is described in Section 5.3.3.
>
> Sure, but my question is whether correlation is expected to occur before or after the INVITE completes.
>
> [SV] Correlation can only occur after the correlation information has been exchanged, and the CS call is received. I'm not sure what you mean by INVITE being completed, but I would say correlation can be done already before the last request (ACK) has been sent, or after the whole INVITE related request-response exchange has been completed (in case the CS call is received late).
>
>> - when does CS media start to flow to the user as part of the
>>      SIP call? Is it before or after correlation? (E.g., does the
>>      user hear the DTMF exchange?)
>>
>> [SV] CS media starts to flow after the call has been accepted, as in a normal CS call. Correlation may be possible before accepting the call (for "callerid", "uuie", and "external" mechanisms), and for "dtmf" case only after the call has been accepted.
>
> I should have been clearer. For DTMF and external correlation media clearly needs to flow on the CS call. My question pertains to whether this is considered part of the SIP session prior to correlation.
>
> [SV] Again, I'm not sure I'm able to follow your thought but correlation is needed before the CS media is considered as part of the session.
>
>> - if media flows to the user before correlation, then what if
>>      correlation fails? Will another call be accepted and correlation
>>      be attempted again?
>>
>> [SV] I don' think we have explicitly defined this situation. If a call is accepted and *after* that correlation fails, it means that the DTMF mechanism was used (for other correlation mechanisms, we know before accepting the call whether the correlation was successful).
>
> external correlation via also has this problem, and more.
>
>> If the DTMF pattern is not received correctly, as we cannot know whether it is related to the session or not. Now, this raises a possible case of including both "dtmf" and "external" mechanisms in the O/A. If DTMF is successfully negotiated but it fails, we can't really use the "external" mechanism since the call is already accepted. Implementations could in this case release the CS call and try a new O/A without the DTMF.
>
> I'm finding external correlation to be underspecified.
>
> One one I could conceive of it working is that the incoming CS call is treated something like a call waiting situation. The existing (sip) call is "suspended" and the user is connected to the CS call, and asked to confirm if it is the intended one. If the user signals confirmation, then the sip call is resumed, with the CS audio stream in it.
>
> If the external correlation fails, then the sip call is resumed without the CS media.
>
> A problem with this is that the O/A completes prior to the CS call establishment and correlation. If this happens mid-call (because of a desire to switch from RTP to CS media) then the RTP media will be abandoned once the O/A succeeds, before correlation. So while waiting for a CS call to arrive and be correlated you have no RTP media. :-(
>
> [SV] Ok now I think I see your point. The O/A needs to be completed before correlation can take place. If this happens mid-call, it can indeed happen that the correlation fails and there is no media. On the other hand, there are a number of reasons why the same could happen to RTP based media as well. So, it can either be considered as an call failure case, or the other possibility would be to first add the CS media, and if it succeeds, do another O/A to remove the RTP based media. The drawback in this case is that the endpoints would receive two media streams for a short while.
>
>> - what if multiple incoming calls arrive before correlation
>>      has been completed? (If correlation occurs before exposing the
>>      media to the user, then it isn't an issue.)
>>
>> [SV] This would be an issue for the DTMF case, as there correlation can be only only after accepting the CS call. An unlikely case, but still possible. In this case, one of the CS calls could be related to the session, while other(s) would be normal CS calls. As only one CS call can be handled at a time, others would either wait or be terminated, depending on the callee's supplementary services settings.
>
> For DTMF, the correlation could be done *before* relating the CS call to the session. Only bind it to the session after correlation is complete.
>
> But that doesn't work as well for external correlation.
>
> [SV] For both DTMF and external, the CS call needs to be accepted first, but when using DTMF it is possible (and advisable) to not play back the media to the user until correlation has been done. For external case, the media would need to be played back .
>
>> - extern correlation complicates this. It isn't like the other
>>      correlation mechanisms - the user must be exposed to the media
>>      to decide if there is correlation. How would the user indicate
>>      "this is the wrong call" in a way that would allow correlation
>>      to be attempted on another call?
>>
>> [SV] The external correlation mechanism does not assume the media flows before "correlation" is made - if the user decides to accept the incoming call, it is considered as "correlated".
>
> The user has to receive (hear) the media (and maybe send some too) before he can decide on correlation. Has the CS call be "related to the session" then, or not?
>
> [SV] From a protocol point of view, I would say the call is considered as related to the session if the user accepts the call. But, you're right that the user cannot be really sure before some words are exchanged on the CS call.
>
> What happens if the user decides the CS call *isn't* correlated?
>
> [SV] The user would likely hang up.
>
>> Now consider a call that is initially established with one RTP media stream. Later, a reinvite is sent with an offer proposing to to replace that with a CS media stream:
>>
>> - again, there must be an answer before the CS call can be attempted.
>>
>>
>> [SV] For being able to perform correlation, yes, the Answer needs to be received.
>>
>> - are you assuming that the reinvite will complete before the CS
>>      call does? If so, then you are in a break-before-make situation -
>>      the RTP session is gone before the CS session takes its place.
>>
>> [SV] The O/A needs to be completed before correlation can be done, but CS call may arrive before Answer. In this case the endpoint should wait until the Answer is received. If the CS call does not succeed, or correlation is unsuccessful, then yes there is no media as the RTP stream has been dropped. But, this could also happen when re-negotiating RTP media types (endpoints are not able to load correct codecs, new IP addr/port does not work etc.) so the situation not be specific to the CS media type.
>
> Yes, it is not entirely unique, but usually updated O/As are done with an option in common to the prior one so that failure is unlikely. The failure modes are quite different when transitioning from RTP to CS.
> (And also when transitioning from CS to RTP.)
>
> [SV] Agreed, adding a CS call introduces an additional level of uncertainty.
>
>> - all the issues above also apply here.
>>
>> ISTM that there is a bunch of missing explanation of how this is expected to work.
>>
>> [SV] Could you check the above whether my comments clarify the situation. We can address them in more details in the draft as well once we know where more explanation is needed.
>
> Upon rereading section 5.6.2 I realize that it does say that the answer must be received before the CS call is placed and correlated, so it implies a break-before-make approach on reINVITE.
>
> What I still think is missing is:
>
> * explanation of how things work when changing media mid call - from RTP to CS, CS to RTP, or even CS to different CS. Point out that this is break-before-make, and when the user will again perceive to have media.
>
> * much about external correlation. Is this really a mechanism in the sense that the others are, that serves as a screen on CS calls until the one is found that correlates? Or does it really mean that any incoming CS call is taking as the right one, included in the sip session, with the user left to take independent action (such as hanging up) if he decides it isn't the right one?
>
> * For DTMF correlation, does the user see media in the call before the correlation is successfully completed.
>
> * potentially (but maybe not in this draft) a discussion of how to achieve make-before-break semantics. (E.g., Using multiple m-lines.)
>
> [SV] I need to go through these with more thought with my co-author. We'll get back to this.
>
> Thanks
> Simo
>
> 	Thanks,
> 	Paul
>
>> I also have a nit in section 5.3.2:
>>
>>       The fourth correlation mechanism declares support for cases where
>>       correlation is done by external means.  Typically this means that the
>>       decision is left for the human user.  This is the way how some
>>       current conferencing systems operate: ...
>>
>> "This is the way how some current conferencing systems operate"
>>
>> should be:
>>
>> "This is how some current conferencing systems operate"
>>
>> [SV] Fill fix.
>>
>> Thanks,
>> Simo
>>
>> 	Thanks,
>> 	Paul
>>
>> On 1/31/14 6:01 AM, Simo.Veikkolainen@nokia.com wrote:
>>> Hello WG,
>>>
>>> We have submitted a new version of
>>> http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-22
>>>
>>> To recap, there was discussion on the  MMUSIC WG list on which correlation mechanisms should be mandatory to implement (see Gonzalo's email http://www.ietf.org/mail-archive/web/mmusic/current/msg12274.html ). As stated by Gonzalo, 3GPP is going to use this draft, more specifically the plan now is to use the caller id based correlation mechanism. This was the message I got from Andrew after a recent 3GPP CT1 meeting.
>>>
>>> We have now made the following changes to the draft:
>>> -	the endpoint may choose which correlation mechanisms to implement (before all three were MTI).
>>> -	we have introduced a fourth correlation type ("external") which indicates explicit support for cases where also some external means, typically the human user, can do the correlation. This can be as a fallback in cases where no other common correlation mechanism is found, or the negotiated correlation mechanisms fail due to being not supported in the PSTN network. This is much like current systems like Webex work.
>>>
>>> This was also the guidance from the IESG informal telechat in December where Gonzalo presented the draft.
>>>
>>> We would like to hear from the WG whether there are any concerns with these changes.
>>>
>>> On behalf of the authors,
>>> Simo
>>>
>>> PS. I'll add the proposed change in http://www.ietf.org/mail-archive/web/mmusic/current/msg12792.html to next version, slipped my mind this time.
>>>
>>>
>>> -----Original Message-----
>>> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of ext
>>> internet-drafts@ietf.org
>>> Sent: 31 January, 2014 12:25
>>> To: i-d-announce@ietf.org
>>> Cc: mmusic@ietf.org
>>> Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.txt
>>>
>>>
>>> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>>>     This draft is a work item of the Multiparty Multimedia Session Control Working Group of the IETF.
>>>
>>>            Title           : Session Description Protocol (SDP) Extension For Setting Audio and Video Media Streams Over Circuit-Switched Bearers In The Public Switched Telephone Network (PSTN)
>>>            Authors         : Miguel A. Garcia-Martin
>>>                              Simo Veikkolainen
>>> 	Filename        : draft-ietf-mmusic-sdp-cs-22.txt
>>> 	Pages           : 37
>>> 	Date            : 2014-01-31
>>>
>>> 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-22
>>>
>>> A diff from the previous version is available at:
>>> http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-sdp-cs-22
>>>
>>>
>>> Please note that it may take a couple of minutes from the time of submission until the htmlized version and diff are available at tools.ietf.org.
>>>
>>> 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
>>> _______________________________________________
>>> 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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> https://www.ietf.org/mailman/listinfo/mmusic
>


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Hi Tiru

I agree.

Thanks
Oleg

On Mon, Feb 10, 2014 at 4:09 AM, Tirumaleswar Reddy (tireddy) <
tireddy@cisco.com> wrote:

>   Comment: I am not sure that 501 is the right code to be used. In SIP
> and HTTP, 5xx errors are usually about a server code problem of some sort;
> 501 means "not implemented". That may be not exactly accurate. A
> mobility-aware TURN server may reject the request simply because the
> administrator does not allow it., and the server did not encounter an
> error. The reason why the server reject the request is rather "not allowed"
> (which is the error code 405).
>
> [TR] we can replace 501 error code with 403
>
> May be be, the right approach would be to introduce two separate error
> codes here: 405 "not allowed" (the server implements mobility but the
> administrator forbade it) or 501 "not implemented" (the server knows about
> mobility but it has no implementation for that).
>
> [TR] Not sure if "not implemented" error will be of any help. TURN server
> if it knows about mobility but has not implemented the functionality then
> it can continue to ignore MOBILITY-TICKET in the request and not send
> MOBILITY-TICKET in the response.
>
>

--001a11c361d21f7f0a04f21112d9
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra">Hi Tiru<br><br></div><div c=
lass=3D"gmail_extra">I agree.<br><br></div><div class=3D"gmail_extra">Thank=
s<br>Oleg<br></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote=
">On Mon, Feb 10, 2014 at 4:09 AM, Tirumaleswar Reddy (tireddy) <span dir=
=3D"ltr">&lt;<a href=3D"mailto:tireddy@cisco.com" target=3D"_blank">tireddy=
@cisco.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 link=3D"blue" vlink=3D"purple" lang=3D"EN-US">
<div><div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in=
 0in 4.0pt"><div><div><div><div class=3D"">
</div></div><div class=3D"">
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">Comment: I am not sur=
e that 501 is the right code to be used. In SIP and HTTP, 5xx errors are us=
ually about a server code problem of some sort; 501 means &quot;not impleme=
nted&quot;. That may be not exactly accurate.
 A mobility-aware TURN server may reject the request simply because the adm=
inistrator does not allow it., and the server did not encounter an error. T=
he reason why the server reject the request is rather &quot;not allowed&quo=
t; (which is the error code 405).<u></u><u></u></p>

</div><p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"=
font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;col=
or:#1f497d">[TR] we can replace 501 error code with 403
<u></u><u></u></span></p>
</div><div class=3D"">
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">May be be, the right =
approach would be to introduce two separate error codes here: 405 &quot;not=
 allowed&quot; (the server implements mobility but the administrator forbad=
e it) or 501 &quot;not implemented&quot; (the server knows
 about mobility but it has no implementation for that). <span style=3D"colo=
r:#1f497d">
<u></u><u></u></span></p>
</div><p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"=
font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;col=
or:#1f497d">[TR] Not sure if &ldquo;not implemented&rdquo; error will be of=
 any help. TURN server if it knows about mobility but has not implemented t=
he
 functionality then it can continue to ignore MOBILITY-TICKET in the reques=
t and not send MOBILITY-TICKET in the response.<u></u><u></u></span></p></d=
iv><br></div></div></div></blockquote></div><br></div></div>

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-----Original Message-----
From: ext Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]=20
Sent: 10 February, 2014 17:49
To: Veikkolainen Simo (Nokia-CTO/Espoo); mmusic@ietf.org
Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.txt

Simo,

What you describe below is enough to get me to shut up. :-)

On 2/10/14 5:44 AM, Simo.Veikkolainen@nokia.com wrote:
> Hello Paul, all,
>
> I've copied the bullets from your email below, and propose the following =
text to address your concerns:
>
> * explanation of how things work when changing media mid call - from RTP =
to CS, CS to RTP, or even CS to different CS. Point out that this is break-=
before-make, and when the user will again perceive to have media.
>
> -> Add in 5.6.4. Modifying the session
>
> If either party would like to remove existing RTP based media from the se=
ssion and replace that with a circuit-switched bearer, it would create a ne=
w Offer to add the circuit-switched media as described in Section 5.6.1. ab=
ove, and remove the RTP based media by setting the port number to zero. Onc=
e the Offer/Answer exchange is done, but the circuit-switched bearer is not=
 yet established, there may be a period of time when no media is available.=
 Also, it may happen that correlating the circuit-switched call fails for r=
easons discussed in Section 5.3.3. In this case, even if the Offer/Answer e=
xchange was successful, endpoints are not able to receive or send media. It=
 is up to the implementation to decide the behavior in this case; if nothin=
g else is done, the user most likely hangs up after a while if there was no=
 other media in the session. Note that this may also happen when switching =
from RTP media to another RTP media (for example when firewall blocks the n=
ew media strea!
 m).

I find it interesting that you describe adding a new m-line for the cs sess=
ion and rejecting the old one. It should also be valid to simply replace th=
e old (rtp) m-line with the new (cs) m-line.

[SV] This was actually something I was supposed to check before updating th=
e draft text, as I wasn't quite sure whether the old media description shou=
ld be explicitly rejected first and the new one added. But, as you say, it =
should be enough to simply replace the old m-line with the new m-line.

If you are thinking of using two m-lines, then make-before-break is just a =
matter of keeping them both active for awhile.

[SV] That is also possible, though in this case you would receive two media=
 streams for a while (typically, two audio streams from the same source). B=
ut of course, implementations could be clever enough to suppress the other =
until cut-over is done.=20

Simo

	Thanks,
	Paul

> If either party would like to remove existing circuit-switched media from=
 the session and replace that with RTP based media, it would create a new O=
ffer and set the circuit-switched port number to zero, and add a descriptio=
n for RTP based media. The endpoint MUST then terminate the circuit-switche=
d bearer using whatever mechanism is appropriate for the technology in ques=
tion.
>
> * much about external correlation. Is this really a mechanism in the sens=
e that the others are, that serves as a screen on CS calls until the one is=
 found that correlates? Or does it really mean that any incoming CS call is=
 taking as the right one, included in the sip session, with the user left t=
o take independent action (such as hanging up) if he decides it isn't the r=
ight one?
>
> -> add in 5.2.3.5. The External correlation mechanism
>
> The external correlation mechanism relies on the human user to do the dec=
ision on whether the call is related to the ongoing session or not. After t=
he user accepts the call, that bearer is considered as related to the sessi=
on. There is a small chance that the user receives at the same time another=
 circuit-switched call which is not related to the ongoing session. The use=
r may reject this call if he is able to determine (e.g. based on the callin=
g line identification) that the call is not related to the session, and con=
tinue waiting for another call attempt. If the user accepts the incoming ci=
rcuit-switched call, but it turns out to be not related to the session, the=
 endpoints need to rely on the human user to take appropriate action (typic=
ally, they would hang up).
>
> * For DTMF correlation, does the user see media in the call before the co=
rrelation is successfully completed.
>
> I think this is an implementation specific matter. We could add a recomme=
ndation of some sorts, but really it is up to the implementation to decide.
>
> * potentially (but maybe not in this draft) a discussion of how to=20
> achieve make-before-break semantics. (E.g., Using multiple m-lines.)
>
> This seems to be a larger issue which is not specific to circuit-switched=
 media alone, to probably it would be best to address it in a standalone do=
cument.
>
> Regards,
> Simo
>
> -----Original Message-----
> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of=20
> Veikkolainen Simo (Nokia-CTO/Espoo)
> Sent: 07 February, 2014 10:38
> To: pkyzivat@alum.mit.edu; mmusic@ietf.org
> Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.txt
>
> See below [SV]
>
> -----Original Message-----
> From: ext Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
> Sent: 05 February, 2014 21:11
> To: Veikkolainen Simo (Nokia-CTO/Espoo); mmusic@ietf.org
> Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.txt
>
> On 2/4/14 2:18 AM, Simo.Veikkolainen@nokia.com wrote:
>>
>>
>> -----Original Message-----
>> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of ext Paul=20
>> Kyzivat
>> Sent: 31 January, 2014 20:53
>> To: mmusic@ietf.org
>> Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.txt
>>
>> I just reviewed the new version. First I looked at just the diffs. But t=
hose raised some more general questions in my mind, so I re-reviewed more o=
f the draft. I'm left with some considerable confusion about how this is in=
tended to work in practice, especially with extern correlation, but even wi=
th the other forms.
>>
>> In particular, I am unclear of the sequencing of O/A processing,=20
>> correlation processing, and signaling completion. (I realize this is=20
>> an mmusic draft and theoretically independent of sip, but forgive me=20
>> if I discuss this in the context of sip.)
>>
>> Normally, when first establishing a session with sip, the first O/A may =
occur before the signaling session is established. But early media may flow=
 after the first O/A. Often it is expected that the user will be be aware a=
nd involved in this early media.
>>
>> Once a session is up, it can be changed with a reinvite containing a new=
 O/A. If the O/A and the reinvite are successful, then the results of the n=
ew O/A replace the old one. If the reinvite fails, then the session is supp=
osed to continue as it was before the reinvite. In the interim, from the se=
nding of the reinvite until its success or failure, things are a little con=
fusing - presumably both old and new media streams must be maintained, but =
which are connected to the user is unspecified.
>>
>> I see nothing in this draft that explains how the CS media, and correlat=
ion verification, fit into the above process.
>>
>> Consider a new call that offers CS media:
>>
>> - The first O/A must complete before the CS call can be attempted.
>>
>> - must the CS call complete and be correlated successfully before
>>      the INVITE gets a 200? If the invite is allowed to complete first,
>>      what happens if no CS call is established, or if it is but
>>      correlation fails?
>>
>> [SV] I assume you mean the 200 carries the Answer.
>
> Not necessarily. I was just trying to sort out the expectations for the t=
iming of the events. The answer could be *before* the 200, but it must be b=
efore the call is established.
>
>> It may happen that the CS call arrives before the Answer, but in this ca=
se no correlation can be done. See Section 5.6.3. If the invite (I assume y=
ou mean O/A) completes first, but no CS call ever arrives, or correlation -=
 including the "external" mechanism fails, the endpoint should treat the CS=
 call as unrelated; this is described in Section 5.3.3.
>
> Sure, but my question is whether correlation is expected to occur before =
or after the INVITE completes.
>
> [SV] Correlation can only occur after the correlation information has bee=
n exchanged, and the CS call is received. I'm not sure what you mean by INV=
ITE being completed, but I would say correlation can be done already before=
 the last request (ACK) has been sent, or after the whole INVITE related re=
quest-response exchange has been completed (in case the CS call is received=
 late).
>
>> - when does CS media start to flow to the user as part of the
>>      SIP call? Is it before or after correlation? (E.g., does the
>>      user hear the DTMF exchange?)
>>
>> [SV] CS media starts to flow after the call has been accepted, as in a n=
ormal CS call. Correlation may be possible before accepting the call (for "=
callerid", "uuie", and "external" mechanisms), and for "dtmf" case only aft=
er the call has been accepted.
>
> I should have been clearer. For DTMF and external correlation media clear=
ly needs to flow on the CS call. My question pertains to whether this is co=
nsidered part of the SIP session prior to correlation.
>
> [SV] Again, I'm not sure I'm able to follow your thought but correlation =
is needed before the CS media is considered as part of the session.
>
>> - if media flows to the user before correlation, then what if
>>      correlation fails? Will another call be accepted and correlation
>>      be attempted again?
>>
>> [SV] I don' think we have explicitly defined this situation. If a call i=
s accepted and *after* that correlation fails, it means that the DTMF mecha=
nism was used (for other correlation mechanisms, we know before accepting t=
he call whether the correlation was successful).
>
> external correlation via also has this problem, and more.
>
>> If the DTMF pattern is not received correctly, as we cannot know whether=
 it is related to the session or not. Now, this raises a possible case of i=
ncluding both "dtmf" and "external" mechanisms in the O/A. If DTMF is succe=
ssfully negotiated but it fails, we can't really use the "external" mechani=
sm since the call is already accepted. Implementations could in this case r=
elease the CS call and try a new O/A without the DTMF.
>
> I'm finding external correlation to be underspecified.
>
> One one I could conceive of it working is that the incoming CS call is tr=
eated something like a call waiting situation. The existing (sip) call is "=
suspended" and the user is connected to the CS call, and asked to confirm i=
f it is the intended one. If the user signals confirmation, then the sip ca=
ll is resumed, with the CS audio stream in it.
>
> If the external correlation fails, then the sip call is resumed without t=
he CS media.
>
> A problem with this is that the O/A completes prior to the CS call=20
> establishment and correlation. If this happens mid-call (because of a=20
> desire to switch from RTP to CS media) then the RTP media will be=20
> abandoned once the O/A succeeds, before correlation. So while waiting=20
> for a CS call to arrive and be correlated you have no RTP media. :-(
>
> [SV] Ok now I think I see your point. The O/A needs to be completed befor=
e correlation can take place. If this happens mid-call, it can indeed happe=
n that the correlation fails and there is no media. On the other hand, ther=
e are a number of reasons why the same could happen to RTP based media as w=
ell. So, it can either be considered as an call failure case, or the other =
possibility would be to first add the CS media, and if it succeeds, do anot=
her O/A to remove the RTP based media. The drawback in this case is that th=
e endpoints would receive two media streams for a short while.
>
>> - what if multiple incoming calls arrive before correlation
>>      has been completed? (If correlation occurs before exposing the
>>      media to the user, then it isn't an issue.)
>>
>> [SV] This would be an issue for the DTMF case, as there correlation can =
be only only after accepting the CS call. An unlikely case, but still possi=
ble. In this case, one of the CS calls could be related to the session, whi=
le other(s) would be normal CS calls. As only one CS call can be handled at=
 a time, others would either wait or be terminated, depending on the callee=
's supplementary services settings.
>
> For DTMF, the correlation could be done *before* relating the CS call to =
the session. Only bind it to the session after correlation is complete.
>
> But that doesn't work as well for external correlation.
>
> [SV] For both DTMF and external, the CS call needs to be accepted first, =
but when using DTMF it is possible (and advisable) to not play back the med=
ia to the user until correlation has been done. For external case, the medi=
a would need to be played back .
>
>> - extern correlation complicates this. It isn't like the other
>>      correlation mechanisms - the user must be exposed to the media
>>      to decide if there is correlation. How would the user indicate
>>      "this is the wrong call" in a way that would allow correlation
>>      to be attempted on another call?
>>
>> [SV] The external correlation mechanism does not assume the media flows =
before "correlation" is made - if the user decides to accept the incoming c=
all, it is considered as "correlated".
>
> The user has to receive (hear) the media (and maybe send some too) before=
 he can decide on correlation. Has the CS call be "related to the session" =
then, or not?
>
> [SV] From a protocol point of view, I would say the call is considered as=
 related to the session if the user accepts the call. But, you're right tha=
t the user cannot be really sure before some words are exchanged on the CS =
call.
>
> What happens if the user decides the CS call *isn't* correlated?
>
> [SV] The user would likely hang up.
>
>> Now consider a call that is initially established with one RTP media str=
eam. Later, a reinvite is sent with an offer proposing to to replace that w=
ith a CS media stream:
>>
>> - again, there must be an answer before the CS call can be attempted.
>>
>>
>> [SV] For being able to perform correlation, yes, the Answer needs to be =
received.
>>
>> - are you assuming that the reinvite will complete before the CS
>>      call does? If so, then you are in a break-before-make situation -
>>      the RTP session is gone before the CS session takes its place.
>>
>> [SV] The O/A needs to be completed before correlation can be done, but C=
S call may arrive before Answer. In this case the endpoint should wait unti=
l the Answer is received. If the CS call does not succeed, or correlation i=
s unsuccessful, then yes there is no media as the RTP stream has been dropp=
ed. But, this could also happen when re-negotiating RTP media types (endpoi=
nts are not able to load correct codecs, new IP addr/port does not work etc=
.) so the situation not be specific to the CS media type.
>
> Yes, it is not entirely unique, but usually updated O/As are done with an=
 option in common to the prior one so that failure is unlikely. The failure=
 modes are quite different when transitioning from RTP to CS.
> (And also when transitioning from CS to RTP.)
>
> [SV] Agreed, adding a CS call introduces an additional level of uncertain=
ty.
>
>> - all the issues above also apply here.
>>
>> ISTM that there is a bunch of missing explanation of how this is expecte=
d to work.
>>
>> [SV] Could you check the above whether my comments clarify the situation=
. We can address them in more details in the draft as well once we know whe=
re more explanation is needed.
>
> Upon rereading section 5.6.2 I realize that it does say that the answer m=
ust be received before the CS call is placed and correlated, so it implies =
a break-before-make approach on reINVITE.
>
> What I still think is missing is:
>
> * explanation of how things work when changing media mid call - from RTP =
to CS, CS to RTP, or even CS to different CS. Point out that this is break-=
before-make, and when the user will again perceive to have media.
>
> * much about external correlation. Is this really a mechanism in the sens=
e that the others are, that serves as a screen on CS calls until the one is=
 found that correlates? Or does it really mean that any incoming CS call is=
 taking as the right one, included in the sip session, with the user left t=
o take independent action (such as hanging up) if he decides it isn't the r=
ight one?
>
> * For DTMF correlation, does the user see media in the call before the co=
rrelation is successfully completed.
>
> * potentially (but maybe not in this draft) a discussion of how to=20
> achieve make-before-break semantics. (E.g., Using multiple m-lines.)
>
> [SV] I need to go through these with more thought with my co-author. We'l=
l get back to this.
>
> Thanks
> Simo
>
> 	Thanks,
> 	Paul
>
>> I also have a nit in section 5.3.2:
>>
>>       The fourth correlation mechanism declares support for cases where
>>       correlation is done by external means.  Typically this means that =
the
>>       decision is left for the human user.  This is the way how some
>>       current conferencing systems operate: ...
>>
>> "This is the way how some current conferencing systems operate"
>>
>> should be:
>>
>> "This is how some current conferencing systems operate"
>>
>> [SV] Fill fix.
>>
>> Thanks,
>> Simo
>>
>> 	Thanks,
>> 	Paul
>>
>> On 1/31/14 6:01 AM, Simo.Veikkolainen@nokia.com wrote:
>>> Hello WG,
>>>
>>> We have submitted a new version of
>>> http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-22
>>>
>>> To recap, there was discussion on the  MMUSIC WG list on which correlat=
ion mechanisms should be mandatory to implement (see Gonzalo's email http:/=
/www.ietf.org/mail-archive/web/mmusic/current/msg12274.html ). As stated by=
 Gonzalo, 3GPP is going to use this draft, more specifically the plan now i=
s to use the caller id based correlation mechanism. This was the message I =
got from Andrew after a recent 3GPP CT1 meeting.
>>>
>>> We have now made the following changes to the draft:
>>> -	the endpoint may choose which correlation mechanisms to implement (be=
fore all three were MTI).
>>> -	we have introduced a fourth correlation type ("external") which indic=
ates explicit support for cases where also some external means, typically t=
he human user, can do the correlation. This can be as a fallback in cases w=
here no other common correlation mechanism is found, or the negotiated corr=
elation mechanisms fail due to being not supported in the PSTN network. Thi=
s is much like current systems like Webex work.
>>>
>>> This was also the guidance from the IESG informal telechat in December =
where Gonzalo presented the draft.
>>>
>>> We would like to hear from the WG whether there are any concerns with t=
hese changes.
>>>
>>> On behalf of the authors,
>>> Simo
>>>
>>> PS. I'll add the proposed change in http://www.ietf.org/mail-archive/we=
b/mmusic/current/msg12792.html to next version, slipped my mind this time.
>>>
>>>
>>> -----Original Message-----
>>> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of ext=20
>>> internet-drafts@ietf.org
>>> Sent: 31 January, 2014 12:25
>>> To: i-d-announce@ietf.org
>>> Cc: mmusic@ietf.org
>>> Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.txt
>>>
>>>
>>> A New Internet-Draft is available from the on-line Internet-Drafts dire=
ctories.
>>>     This draft is a work item of the Multiparty Multimedia Session Cont=
rol Working Group of the IETF.
>>>
>>>            Title           : Session Description Protocol (SDP) Extensi=
on For Setting Audio and Video Media Streams Over Circuit-Switched Bearers =
In The Public Switched Telephone Network (PSTN)
>>>            Authors         : Miguel A. Garcia-Martin
>>>                              Simo Veikkolainen
>>> 	Filename        : draft-ietf-mmusic-sdp-cs-22.txt
>>> 	Pages           : 37
>>> 	Date            : 2014-01-31
>>>
>>> Abstract:
>>>       This memo describes use cases, requirements, and protocol extensi=
ons
>>>       for using the Session Description Protocol (SDP) Offer/Answer mod=
el
>>>       for establishing audio and video media streams over circuit-switc=
hed
>>>       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-22
>>>
>>> A diff from the previous version is available at:
>>> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sdp-cs-22
>>>
>>>
>>> Please note that it may take a couple of minutes from the time of submi=
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Subject: [MMUSIC] Protocol Action: 'Real Time Streaming Protocol 2.0 (RTSP)' to Proposed Standard (draft-ietf-mmusic-rfc2326bis-40.txt)
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The IESG has approved the following document:
- 'Real Time Streaming Protocol 2.0 (RTSP)'
  (draft-ietf-mmusic-rfc2326bis-40.txt) as Proposed Standard

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

The IESG contact persons are Gonzalo Camarillo and Richard Barnes.

A URL of this Internet Draft is:
http://datatracker.ietf.org/doc/draft-ietf-mmusic-rfc2326bis/




Technical Summary

The document defines RTSP version 2.0 which obsoletes RTSP version 1.0 defined in RFC 2326.

The Real Time Streaming Protocol, or RTSP, is an application-level protocol for setup and control of the delivery of data with real-time properties.  RTSP provides an extensible framework to enable controlled, on-demand delivery of real-time data, such as audio and video.  Sources of data can include both live data feeds and stored clips.  This protocol is intended to control multiple data delivery sessions, provide a means for choosing delivery channels such as UDP, multicast UDP and TCP, and provide a means for choosing delivery mechanisms based upon RTP (RFC 3550).

Working Group Summary

The document has been work in progress for an extended period of time dating back to 2002. Earlier versions saw decent WG participation however the later versions have primarily been driven by the document authors with limited overall discussion in the group, especially towards the end of the process. There are no known issues or major discussion points, and there has been no indication of lack of consensus in the WG.

Document Quality

The document has been reviewed in detail several times after WGLC and in preparation for the publication request and the authors have made several updates as a result of those. The document is considered to be of high quality at this point.

There is one known implementation of the specification, and many of the extensions compared to RTSP 1.0 have been implemented separately as well.

A Media type review was done for "text/parameters". The review thread can be found at: http://www.ietf.org/mail-archive/web/ietf-types/current/msg01656.html


Personnel
Document Shepherd: Flemming Andreasen
Responsible AD: Gonzalo Camarillo


RFC Editor's Note:

Section 2 of this document and its subsections provide "an informative overview
of the different mechanisms in the RTSP 2.0 protocol, to give the reader a high
level understanding".  As such, it's very important that it be clear and well
written.  Some reviewers have found the English to be difficult.  Please take
an especially critical pen to this section, making sure that the grammar and
usage are correct, and that the language is clear.


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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-23.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Multiparty Multimedia Session Control Working Group of the IETF.

        Title           : Session Description Protocol (SDP) Extension For Setting Audio and Video Media Streams Over Circuit-Switched Bearers In The Public Switched Telephone Network (PSTN)
        Authors         : Miguel A. Garcia-Martin
                          Simo Veikkolainen
	Filename        : draft-ietf-mmusic-sdp-cs-23.txt
	Pages           : 38
	Date            : 2014-02-11

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-23

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-sdp-cs-23


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Dear WG,=20

We just submitted version -23 addressing these comments.

The new version also includes the note on the E.164 format which slipped fr=
om version -22 (see http://www.ietf.org/mail-archive/web/mmusic/current/msg=
12792.html )

Regards,
Simo


-----Original Message-----
From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of Veikkolainen Sim=
o (Nokia-CTO/Espoo)
Sent: 11 February, 2014 12:03
To: pkyzivat@alum.mit.edu; mmusic@ietf.org
Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.txt



-----Original Message-----
From: ext Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]=20
Sent: 10 February, 2014 17:49
To: Veikkolainen Simo (Nokia-CTO/Espoo); mmusic@ietf.org
Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.txt

Simo,

What you describe below is enough to get me to shut up. :-)

On 2/10/14 5:44 AM, Simo.Veikkolainen@nokia.com wrote:
> Hello Paul, all,
>
> I've copied the bullets from your email below, and propose the following =
text to address your concerns:
>
> * explanation of how things work when changing media mid call - from RTP =
to CS, CS to RTP, or even CS to different CS. Point out that this is break-=
before-make, and when the user will again perceive to have media.
>
> -> Add in 5.6.4. Modifying the session
>
> If either party would like to remove existing RTP based media from the se=
ssion and replace that with a circuit-switched bearer, it would create a ne=
w Offer to add the circuit-switched media as described in Section 5.6.1. ab=
ove, and remove the RTP based media by setting the port number to zero. Onc=
e the Offer/Answer exchange is done, but the circuit-switched bearer is not=
 yet established, there may be a period of time when no media is available.=
 Also, it may happen that correlating the circuit-switched call fails for r=
easons discussed in Section 5.3.3. In this case, even if the Offer/Answer e=
xchange was successful, endpoints are not able to receive or send media. It=
 is up to the implementation to decide the behavior in this case; if nothin=
g else is done, the user most likely hangs up after a while if there was no=
 other media in the session. Note that this may also happen when switching =
from RTP media to another RTP media (for example when firewall blocks the n=
ew media strea!
 m).

I find it interesting that you describe adding a new m-line for the cs sess=
ion and rejecting the old one. It should also be valid to simply replace th=
e old (rtp) m-line with the new (cs) m-line.

[SV] This was actually something I was supposed to check before updating th=
e draft text, as I wasn't quite sure whether the old media description shou=
ld be explicitly rejected first and the new one added. But, as you say, it =
should be enough to simply replace the old m-line with the new m-line.

If you are thinking of using two m-lines, then make-before-break is just a =
matter of keeping them both active for awhile.

[SV] That is also possible, though in this case you would receive two media=
 streams for a while (typically, two audio streams from the same source). B=
ut of course, implementations could be clever enough to suppress the other =
until cut-over is done.=20

Simo

	Thanks,
	Paul

> If either party would like to remove existing circuit-switched media from=
 the session and replace that with RTP based media, it would create a new O=
ffer and set the circuit-switched port number to zero, and add a descriptio=
n for RTP based media. The endpoint MUST then terminate the circuit-switche=
d bearer using whatever mechanism is appropriate for the technology in ques=
tion.
>
> * much about external correlation. Is this really a mechanism in the sens=
e that the others are, that serves as a screen on CS calls until the one is=
 found that correlates? Or does it really mean that any incoming CS call is=
 taking as the right one, included in the sip session, with the user left t=
o take independent action (such as hanging up) if he decides it isn't the r=
ight one?
>
> -> add in 5.2.3.5. The External correlation mechanism
>
> The external correlation mechanism relies on the human user to do the dec=
ision on whether the call is related to the ongoing session or not. After t=
he user accepts the call, that bearer is considered as related to the sessi=
on. There is a small chance that the user receives at the same time another=
 circuit-switched call which is not related to the ongoing session. The use=
r may reject this call if he is able to determine (e.g. based on the callin=
g line identification) that the call is not related to the session, and con=
tinue waiting for another call attempt. If the user accepts the incoming ci=
rcuit-switched call, but it turns out to be not related to the session, the=
 endpoints need to rely on the human user to take appropriate action (typic=
ally, they would hang up).
>
> * For DTMF correlation, does the user see media in the call before the co=
rrelation is successfully completed.
>
> I think this is an implementation specific matter. We could add a recomme=
ndation of some sorts, but really it is up to the implementation to decide.
>
> * potentially (but maybe not in this draft) a discussion of how to=20
> achieve make-before-break semantics. (E.g., Using multiple m-lines.)
>
> This seems to be a larger issue which is not specific to circuit-switched=
 media alone, to probably it would be best to address it in a standalone do=
cument.
>
> Regards,
> Simo
>
> -----Original Message-----
> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of=20
> Veikkolainen Simo (Nokia-CTO/Espoo)
> Sent: 07 February, 2014 10:38
> To: pkyzivat@alum.mit.edu; mmusic@ietf.org
> Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.txt
>
> See below [SV]
>
> -----Original Message-----
> From: ext Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
> Sent: 05 February, 2014 21:11
> To: Veikkolainen Simo (Nokia-CTO/Espoo); mmusic@ietf.org
> Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.txt
>
> On 2/4/14 2:18 AM, Simo.Veikkolainen@nokia.com wrote:
>>
>>
>> -----Original Message-----
>> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of ext Paul=20
>> Kyzivat
>> Sent: 31 January, 2014 20:53
>> To: mmusic@ietf.org
>> Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.txt
>>
>> I just reviewed the new version. First I looked at just the diffs. But t=
hose raised some more general questions in my mind, so I re-reviewed more o=
f the draft. I'm left with some considerable confusion about how this is in=
tended to work in practice, especially with extern correlation, but even wi=
th the other forms.
>>
>> In particular, I am unclear of the sequencing of O/A processing,=20
>> correlation processing, and signaling completion. (I realize this is=20
>> an mmusic draft and theoretically independent of sip, but forgive me=20
>> if I discuss this in the context of sip.)
>>
>> Normally, when first establishing a session with sip, the first O/A may =
occur before the signaling session is established. But early media may flow=
 after the first O/A. Often it is expected that the user will be be aware a=
nd involved in this early media.
>>
>> Once a session is up, it can be changed with a reinvite containing a new=
 O/A. If the O/A and the reinvite are successful, then the results of the n=
ew O/A replace the old one. If the reinvite fails, then the session is supp=
osed to continue as it was before the reinvite. In the interim, from the se=
nding of the reinvite until its success or failure, things are a little con=
fusing - presumably both old and new media streams must be maintained, but =
which are connected to the user is unspecified.
>>
>> I see nothing in this draft that explains how the CS media, and correlat=
ion verification, fit into the above process.
>>
>> Consider a new call that offers CS media:
>>
>> - The first O/A must complete before the CS call can be attempted.
>>
>> - must the CS call complete and be correlated successfully before
>>      the INVITE gets a 200? If the invite is allowed to complete first,
>>      what happens if no CS call is established, or if it is but
>>      correlation fails?
>>
>> [SV] I assume you mean the 200 carries the Answer.
>
> Not necessarily. I was just trying to sort out the expectations for the t=
iming of the events. The answer could be *before* the 200, but it must be b=
efore the call is established.
>
>> It may happen that the CS call arrives before the Answer, but in this ca=
se no correlation can be done. See Section 5.6.3. If the invite (I assume y=
ou mean O/A) completes first, but no CS call ever arrives, or correlation -=
 including the "external" mechanism fails, the endpoint should treat the CS=
 call as unrelated; this is described in Section 5.3.3.
>
> Sure, but my question is whether correlation is expected to occur before =
or after the INVITE completes.
>
> [SV] Correlation can only occur after the correlation information has bee=
n exchanged, and the CS call is received. I'm not sure what you mean by INV=
ITE being completed, but I would say correlation can be done already before=
 the last request (ACK) has been sent, or after the whole INVITE related re=
quest-response exchange has been completed (in case the CS call is received=
 late).
>
>> - when does CS media start to flow to the user as part of the
>>      SIP call? Is it before or after correlation? (E.g., does the
>>      user hear the DTMF exchange?)
>>
>> [SV] CS media starts to flow after the call has been accepted, as in a n=
ormal CS call. Correlation may be possible before accepting the call (for "=
callerid", "uuie", and "external" mechanisms), and for "dtmf" case only aft=
er the call has been accepted.
>
> I should have been clearer. For DTMF and external correlation media clear=
ly needs to flow on the CS call. My question pertains to whether this is co=
nsidered part of the SIP session prior to correlation.
>
> [SV] Again, I'm not sure I'm able to follow your thought but correlation =
is needed before the CS media is considered as part of the session.
>
>> - if media flows to the user before correlation, then what if
>>      correlation fails? Will another call be accepted and correlation
>>      be attempted again?
>>
>> [SV] I don' think we have explicitly defined this situation. If a call i=
s accepted and *after* that correlation fails, it means that the DTMF mecha=
nism was used (for other correlation mechanisms, we know before accepting t=
he call whether the correlation was successful).
>
> external correlation via also has this problem, and more.
>
>> If the DTMF pattern is not received correctly, as we cannot know whether=
 it is related to the session or not. Now, this raises a possible case of i=
ncluding both "dtmf" and "external" mechanisms in the O/A. If DTMF is succe=
ssfully negotiated but it fails, we can't really use the "external" mechani=
sm since the call is already accepted. Implementations could in this case r=
elease the CS call and try a new O/A without the DTMF.
>
> I'm finding external correlation to be underspecified.
>
> One one I could conceive of it working is that the incoming CS call is tr=
eated something like a call waiting situation. The existing (sip) call is "=
suspended" and the user is connected to the CS call, and asked to confirm i=
f it is the intended one. If the user signals confirmation, then the sip ca=
ll is resumed, with the CS audio stream in it.
>
> If the external correlation fails, then the sip call is resumed without t=
he CS media.
>
> A problem with this is that the O/A completes prior to the CS call=20
> establishment and correlation. If this happens mid-call (because of a=20
> desire to switch from RTP to CS media) then the RTP media will be=20
> abandoned once the O/A succeeds, before correlation. So while waiting=20
> for a CS call to arrive and be correlated you have no RTP media. :-(
>
> [SV] Ok now I think I see your point. The O/A needs to be completed befor=
e correlation can take place. If this happens mid-call, it can indeed happe=
n that the correlation fails and there is no media. On the other hand, ther=
e are a number of reasons why the same could happen to RTP based media as w=
ell. So, it can either be considered as an call failure case, or the other =
possibility would be to first add the CS media, and if it succeeds, do anot=
her O/A to remove the RTP based media. The drawback in this case is that th=
e endpoints would receive two media streams for a short while.
>
>> - what if multiple incoming calls arrive before correlation
>>      has been completed? (If correlation occurs before exposing the
>>      media to the user, then it isn't an issue.)
>>
>> [SV] This would be an issue for the DTMF case, as there correlation can =
be only only after accepting the CS call. An unlikely case, but still possi=
ble. In this case, one of the CS calls could be related to the session, whi=
le other(s) would be normal CS calls. As only one CS call can be handled at=
 a time, others would either wait or be terminated, depending on the callee=
's supplementary services settings.
>
> For DTMF, the correlation could be done *before* relating the CS call to =
the session. Only bind it to the session after correlation is complete.
>
> But that doesn't work as well for external correlation.
>
> [SV] For both DTMF and external, the CS call needs to be accepted first, =
but when using DTMF it is possible (and advisable) to not play back the med=
ia to the user until correlation has been done. For external case, the medi=
a would need to be played back .
>
>> - extern correlation complicates this. It isn't like the other
>>      correlation mechanisms - the user must be exposed to the media
>>      to decide if there is correlation. How would the user indicate
>>      "this is the wrong call" in a way that would allow correlation
>>      to be attempted on another call?
>>
>> [SV] The external correlation mechanism does not assume the media flows =
before "correlation" is made - if the user decides to accept the incoming c=
all, it is considered as "correlated".
>
> The user has to receive (hear) the media (and maybe send some too) before=
 he can decide on correlation. Has the CS call be "related to the session" =
then, or not?
>
> [SV] From a protocol point of view, I would say the call is considered as=
 related to the session if the user accepts the call. But, you're right tha=
t the user cannot be really sure before some words are exchanged on the CS =
call.
>
> What happens if the user decides the CS call *isn't* correlated?
>
> [SV] The user would likely hang up.
>
>> Now consider a call that is initially established with one RTP media str=
eam. Later, a reinvite is sent with an offer proposing to to replace that w=
ith a CS media stream:
>>
>> - again, there must be an answer before the CS call can be attempted.
>>
>>
>> [SV] For being able to perform correlation, yes, the Answer needs to be =
received.
>>
>> - are you assuming that the reinvite will complete before the CS
>>      call does? If so, then you are in a break-before-make situation -
>>      the RTP session is gone before the CS session takes its place.
>>
>> [SV] The O/A needs to be completed before correlation can be done, but C=
S call may arrive before Answer. In this case the endpoint should wait unti=
l the Answer is received. If the CS call does not succeed, or correlation i=
s unsuccessful, then yes there is no media as the RTP stream has been dropp=
ed. But, this could also happen when re-negotiating RTP media types (endpoi=
nts are not able to load correct codecs, new IP addr/port does not work etc=
.) so the situation not be specific to the CS media type.
>
> Yes, it is not entirely unique, but usually updated O/As are done with an=
 option in common to the prior one so that failure is unlikely. The failure=
 modes are quite different when transitioning from RTP to CS.
> (And also when transitioning from CS to RTP.)
>
> [SV] Agreed, adding a CS call introduces an additional level of uncertain=
ty.
>
>> - all the issues above also apply here.
>>
>> ISTM that there is a bunch of missing explanation of how this is expecte=
d to work.
>>
>> [SV] Could you check the above whether my comments clarify the situation=
. We can address them in more details in the draft as well once we know whe=
re more explanation is needed.
>
> Upon rereading section 5.6.2 I realize that it does say that the answer m=
ust be received before the CS call is placed and correlated, so it implies =
a break-before-make approach on reINVITE.
>
> What I still think is missing is:
>
> * explanation of how things work when changing media mid call - from RTP =
to CS, CS to RTP, or even CS to different CS. Point out that this is break-=
before-make, and when the user will again perceive to have media.
>
> * much about external correlation. Is this really a mechanism in the sens=
e that the others are, that serves as a screen on CS calls until the one is=
 found that correlates? Or does it really mean that any incoming CS call is=
 taking as the right one, included in the sip session, with the user left t=
o take independent action (such as hanging up) if he decides it isn't the r=
ight one?
>
> * For DTMF correlation, does the user see media in the call before the co=
rrelation is successfully completed.
>
> * potentially (but maybe not in this draft) a discussion of how to=20
> achieve make-before-break semantics. (E.g., Using multiple m-lines.)
>
> [SV] I need to go through these with more thought with my co-author. We'l=
l get back to this.
>
> Thanks
> Simo
>
> 	Thanks,
> 	Paul
>
>> I also have a nit in section 5.3.2:
>>
>>       The fourth correlation mechanism declares support for cases where
>>       correlation is done by external means.  Typically this means that =
the
>>       decision is left for the human user.  This is the way how some
>>       current conferencing systems operate: ...
>>
>> "This is the way how some current conferencing systems operate"
>>
>> should be:
>>
>> "This is how some current conferencing systems operate"
>>
>> [SV] Fill fix.
>>
>> Thanks,
>> Simo
>>
>> 	Thanks,
>> 	Paul
>>
>> On 1/31/14 6:01 AM, Simo.Veikkolainen@nokia.com wrote:
>>> Hello WG,
>>>
>>> We have submitted a new version of
>>> http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-22
>>>
>>> To recap, there was discussion on the  MMUSIC WG list on which correlat=
ion mechanisms should be mandatory to implement (see Gonzalo's email http:/=
/www.ietf.org/mail-archive/web/mmusic/current/msg12274.html ). As stated by=
 Gonzalo, 3GPP is going to use this draft, more specifically the plan now i=
s to use the caller id based correlation mechanism. This was the message I =
got from Andrew after a recent 3GPP CT1 meeting.
>>>
>>> We have now made the following changes to the draft:
>>> -	the endpoint may choose which correlation mechanisms to implement (be=
fore all three were MTI).
>>> -	we have introduced a fourth correlation type ("external") which indic=
ates explicit support for cases where also some external means, typically t=
he human user, can do the correlation. This can be as a fallback in cases w=
here no other common correlation mechanism is found, or the negotiated corr=
elation mechanisms fail due to being not supported in the PSTN network. Thi=
s is much like current systems like Webex work.
>>>
>>> This was also the guidance from the IESG informal telechat in December =
where Gonzalo presented the draft.
>>>
>>> We would like to hear from the WG whether there are any concerns with t=
hese changes.
>>>
>>> On behalf of the authors,
>>> Simo
>>>
>>> PS. I'll add the proposed change in http://www.ietf.org/mail-archive/we=
b/mmusic/current/msg12792.html to next version, slipped my mind this time.
>>>
>>>
>>> -----Original Message-----
>>> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of ext=20
>>> internet-drafts@ietf.org
>>> Sent: 31 January, 2014 12:25
>>> To: i-d-announce@ietf.org
>>> Cc: mmusic@ietf.org
>>> Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.txt
>>>
>>>
>>> A New Internet-Draft is available from the on-line Internet-Drafts dire=
ctories.
>>>     This draft is a work item of the Multiparty Multimedia Session Cont=
rol Working Group of the IETF.
>>>
>>>            Title           : Session Description Protocol (SDP) Extensi=
on For Setting Audio and Video Media Streams Over Circuit-Switched Bearers =
In The Public Switched Telephone Network (PSTN)
>>>            Authors         : Miguel A. Garcia-Martin
>>>                              Simo Veikkolainen
>>> 	Filename        : draft-ietf-mmusic-sdp-cs-22.txt
>>> 	Pages           : 37
>>> 	Date            : 2014-01-31
>>>
>>> Abstract:
>>>       This memo describes use cases, requirements, and protocol extensi=
ons
>>>       for using the Session Description Protocol (SDP) Offer/Answer mod=
el
>>>       for establishing audio and video media streams over circuit-switc=
hed
>>>       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-22
>>>
>>> A diff from the previous version is available at:
>>> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sdp-cs-22
>>>
>>>
>>> Please note that it may take a couple of minutes from the time of submi=
ssion until the htmlized version and diff are available at tools.ietf.org.
>>>
>>> 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
>>> _______________________________________________
>>> 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 fandreas@cisco.com  Tue Feb 11 14:43:34 2014
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Date: Tue, 11 Feb 2014 17:43:30 -0500
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Subject: Re: [MMUSIC] Draft new version: draft-ietf-mmusic-udptl-dtls-04
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Hi Christer

I took a look at the latest draft and I have some comments:

- The Offer/Answer sections could use a bit more work. We try to have 
the following sub-sections with relevant procedures called out (in line 
with RFC 3264):
1) Generating the Initial Offer
2) Generating the Answer
3) Offerer Processing of the Answer
4) Modify the Session
and I'd much prefer the same here.

- Related to offer/answer, I find it underspecified in terms of things 
that are (or are not) at the session versus media-level. It should be 
more explicit here.

- NAT Traversal Without ICE
I'm not sure I follow this section. If the passive party is behind a NAT 
and then sends a STUN request to the active party in order to create a 
NAT binding, then the active party won't know the external NAT address 
and hence its ClientHello message won't get through (unless you are 
saying that it needs to look for such unauthenticated STUN messages and 
then start sending packet to their source address, at which point we are 
headed down a slippery slope on security implications, reinventing or 
ICE, or both).

- Backwards compatibility/interoperability considerations with vanilla 
UDPTL implementations; how do you negotiate successfully ?

I have a couple of nits that I'll send to you off-line as well.

Thanks

-- Flemming



On 2/1/14, 9:31 AM, Christer Holmberg wrote:
> Hi,
>
> Based on the review comments by Adam G and Ari K, I've submitted a new version (-04) of the udptl-dtls draft.
>
> Regards,
>
> Christer
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


From christer.holmberg@ericsson.com  Wed Feb 12 03:40:38 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Flemming Andreasen <fandreas@cisco.com>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Draft new version: draft-ietf-mmusic-udptl-dtls-04
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Hi,

I'll get back to your comment regarding SDP Offer/Answer procedures later.

> - NAT Traversal Without ICE
> I'm not sure I follow this section. If the passive party is behind a NAT =
and then sends a STUN request to the active party=20
> in order to create a NAT binding, then the active party won't know the ex=
ternal NAT address and hence its ClientHello=20
> message won't get through (unless you are saying that it needs to look fo=
r such unauthenticated STUN messages and=20
> then start sending packet to their source address, at which point we are =
headed down a slippery slope on security implications, reinventing or ICE, =
or both).

This part is re-used from RFC 5763, as part of the Middlebox Interaction se=
ction, so I assume the idea is that the STUN messages are terminated by an =
SBC (5763 talks about latching).

However, my suggestion would be to remove section 4.2.2 completely. We need=
 4.2.1 and 4.2.3, which are related to ICE. Then, whatever other mechanisms=
 people may use to do NAT traversal, are out of the scope.

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

> - Backwards compatibility/interoperability considerations with vanilla UD=
PTL implementations; how do you negotiate successfully ?

The draft defines a mechanism to negotiate a secure fax channel. If one end=
point doesn't support it, there will be no successful negotiation, and no s=
ecure fax channel.=20

However, I guess I can add some informative text, saying that if the user i=
s ok to have a non-secure fax channel in case of failure, it can send a new=
 offer with non-secure fax. Or, it can use CapNeg.

Regards,

Christer


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Flemming Andreasen <fandreas@cisco.com>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Draft new version: draft-ietf-mmusic-udptl-dtls-04
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Subject: Re: [MMUSIC] Draft new version: draft-ietf-mmusic-udptl-dtls-04
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Hi Flemming,

>- The Offer/Answer sections could use a bit more work. We try to have the =
following sub-sections with relevant procedures called out (in line with RF=
C 3264):
>1) Generating the Initial Offer
>2) Generating the Answer
>3) Offerer Processing of the Answer
>4) Modify the Session
>and I'd much prefer the same here.
>
>- Related to offer/answer, I find it underspecified in terms of things tha=
t are (or are not) at the session versus media-level. It should be more exp=
licit here.

It did require some restructuring, but I was able to keep most of the text =
(with some clarifications).

Something like (Note that the compatibility with UDPTL over UDP will be cov=
ered in Section 5):

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

3.  Secure Channel

   The UDPTL over DTLS media stream is negotiated using the SDP offer/
   answer mechanism [RFC3264].

   DTLS is used as specified in [RFC6347].  Once the DTLS handshake is
   successfully completed (in order to prevent facsimile data from being
   transmitted insecurely), the UDPTL packets SHALL be transported in
   DTLS record layer "application_data" packets.

4.  SDP Offerer/Answerer Procedures

4.1.  General

   The offerer and answerer MUST create an SDP media description ("m=3D"
   line) for each UDPTL over DTLS media stream, and MUST assign a UDP/
   TLS/UDPTL value to the "proto" field of the "m=3D" line.  The
   procedures in this section apply to such "m=3D" line.

   In negotiation of a UDPTL over DTLS media stream, the following SDP
   attbibutes are used:

   o  The SDP attributes defined for UDPTL over UDP, as described in
      [ITU.T38.2010]; and
   o  The SDP attributes, defined in [RFC4145] and [RFC4572], as
      described in this section.

   The offerer and answerer MUST NOT use the SDP connection attribute
   [RFC4145].

   In order to negotiate the TLS roles for the UDPTL over DTLS transport
   connection, the offerer and answerer MUST use the SDP setup attribute
   [RFC4145].

   If an endpoint supports, and is willing to use, a cipher suite with
   an associated certificate, it MUST include an SDP fingerprint
   attribute [RFC4572].

   If a cipher suite with an associated certificate is selected during
   the DTLS handshake, the certificate received during the DTLS
   handshake MUST match the fingerprint received in the SDP fingerprint
   attribute.  If the fingerprint does not match the hashed certificate,
   then the endpoint MUST tear down the media session immediately.  Note
   that it is permissible to wait until the other side's fingerprint has
   been received before establishing the connection; however, this may
   have undesirable latency effects.

4.2.  SDP Offerer Procedures

   The offerer SHOULD assign the SDP setup attribute with a
   setup:actpass value.  Alternatively, the offerer MAY assign the SDP
   setup attribute with a setup:active value or setup:passive value.
   The offerer MUST NOT assign an SDP setup attribute with a
   setup:holdconn value.

   If the offerer assigns the SDP setup attribute with a setup:actpass
   value or setup:passive value, the offerer MUST be prepared to receive
   a DTLS client_hello message before it receives the SDP answer.

   When the offerer receives the associated answer, if the offerer ends
   up being active it MUST initiate a DTLS handshake by sending a
   ClientHello message on the negotiated media stream, towards the IP
   address and port of the answerer.

4.3.  SDP Answerer Procedures

   If the answerer accepts the offered UDPTL over DTLS transport
   connection, in the associated answer the answerer MUST assign an SDP
   setup attribute with either a setup:active value or setup:passive
   value, according to the procedures in [RFC4145].  The answerer MUST
   NOT assign an SDP setup attribute with a setup:holdconn value.

   If the answerer assigns a setup:active value, the answerer MUST
   initiate a DTLS handshake by sending a ClientHello message on the
   negotiated media stream, towards the IP address and port of the
   offerer.

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


From fandreas@cisco.com  Wed Feb 12 07:10:59 2014
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Subject: Re: [MMUSIC] Draft new version: draft-ietf-mmusic-udptl-dtls-04
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On 2/12/14, 6:40 AM, Christer Holmberg wrote:
> Hi,
>
> I'll get back to your comment regarding SDP Offer/Answer procedures later.
>
>> - NAT Traversal Without ICE
>> I'm not sure I follow this section. If the passive party is behind a NAT and then sends a STUN request to the active party
>> in order to create a NAT binding, then the active party won't know the external NAT address and hence its ClientHello
>> message won't get through (unless you are saying that it needs to look for such unauthenticated STUN messages and
>> then start sending packet to their source address, at which point we are headed down a slippery slope on security implications, reinventing or ICE, or both).
> This part is re-used from RFC 5763, as part of the Middlebox Interaction section, so I assume the idea is that the STUN messages are terminated by an SBC (5763 talks about latching).
Ok - the draft doesn't say that and in general we don't recommend 
latching since it introduces various security issues (see the latching 
draft for more on all of this).


> However, my suggestion would be to remove section 4.2.2 completely. We need 4.2.1 and 4.2.3, which are related to ICE. Then, whatever other mechanisms people may use to do NAT traversal, are out of the scope.
I would agree with that.

>
> ------------------
>
>> - Backwards compatibility/interoperability considerations with vanilla UDPTL implementations; how do you negotiate successfully ?
> The draft defines a mechanism to negotiate a secure fax channel. If one endpoint doesn't support it, there will be no successful negotiation, and no secure fax channel.
>
> However, I guess I can add some informative text, saying that if the user is ok to have a non-secure fax channel in case of failure, it can send a new offer with non-secure fax. Or, it can use CapNeg.
Sounds good.

Thanks

-- Flemming


> Regards,
>
> Christer
>
> .
>


From suhasietf@gmail.com  Wed Feb 12 11:10:03 2014
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To: mmusic WG <mmusic@ietf.org>
Content-Type: multipart/alternative; boundary=001a11c25d36b8433304f23a5122
Subject: [MMUSIC] BUNDLE and RFC 4091 ANAT Semantics
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--001a11c25d36b8433304f23a5122
Content-Type: text/plain; charset=ISO-8859-1

Hello All,

I am trying to analyze the multiplexing categorization for media streams
that has ANAT semantics.

Say we have the below SDP example

v=0
o=bob 280744730 28977631 IN IP4 host.example.com
s=
t=0 0
a=group:ANAT 1 2
a=group:ANAT 3 4
m=audio 25000 RTP/AVP 0
c=IN IP6 2001:DB8::1
a= <ICE-encoded additional IPv6 addresses (and ports)>
a=mid:1
m=audio 22334 RTP/AVP 0
c=IN IP4 192.0.2.1
a= <ICE-encoded additional IPv4 addresses (and ports)>
a=mid:2
m=video 25112 RTP/AVP 0
c=IN IP6 2001:DB8::1
a= <ICE-encoded additional IPv6 addresses (and ports)>
a=mid:3
m=video 24544 RTP/AVP 0
c=IN IP4 192.0.2.1
a= <ICE-encoded additional IPv4 addresses (and ports)>
a=mid:4


Here we have audio stream with IPV6 and IPV6 network address, video
streams doing the same as well.

Now if we want to BUNDLE these m=lines, one possible way is to split
BUNDLE into groups that belong to the same network address type (IPV4
or IPV6) as shown below

v=0
o=bob 280744730 28977631 IN IP4 host.example.com
s=
t=0 0
a=group:ANAT 1 2
a=group:ANAT 3 4
a=group:BUNDLE 1 3     // BUNDLE IPv4 audio and video streams
a=group:BUNDLE 2 4     // BUNDLE IPv6 audio and video streams.
m=audio 25000 RTP/AVP 0
c=IN IP6 2001:DB8::1
a= <ICE-encoded additional IPv6 addresses (and ports)>
a=mid:1
m=audio 22334 RTP/AVP 0
c=IN IP4 192.0.2.1
a= <ICE-encoded additional IPv4 addresses (and ports)>
a=mid:2
m=video 25112 RTP/AVP 0
c=IN IP6 2001:DB8::1
a= <ICE-encoded additional IPv6 addresses (and ports)>
a=mid:3
m=video 24544 RTP/AVP 0
c=IN IP4 192.0.2.1
a= <ICE-encoded additional IPv4 addresses (and ports)>
a=mid:4


Any thoughts if this sounds like the right approach ?



Cheers

Suhas

--001a11c25d36b8433304f23a5122
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div>Hello All,</div><div><br></div>I am trying to analyze=
 the multiplexing categorization for media streams that has ANAT semantics.=
<div><br></div><div>Say we have the below SDP example</div><div><pre>v=3D0
o=3Dbob 280744730 28977631 IN IP4 <a href=3D"http://host.example.com">host.=
example.com</a>
s=3D
t=3D0 0
a=3Dgroup:ANAT 1 2
a=3Dgroup:ANAT 3 4
m=3Daudio 25000 RTP/AVP 0
c=3DIN IP6 2001:DB8::1
a=3D &lt;ICE-encoded additional IPv6 addresses (and ports)&gt;
a=3Dmid:1
m=3Daudio 22334 RTP/AVP 0
c=3DIN IP4 192.0.2.1
a=3D &lt;ICE-encoded additional IPv4 addresses (and ports)&gt;
a=3Dmid:2
m=3Dvideo 25112 RTP/AVP 0
c=3DIN IP6 2001:DB8::1
a=3D &lt;ICE-encoded additional IPv6 addresses (and ports)&gt;
a=3Dmid:3
m=3Dvideo 24544 RTP/AVP 0
c=3DIN IP4 192.0.2.1
a=3D &lt;ICE-encoded additional IPv4 addresses (and ports)&gt;
a=3Dmid:4</pre><pre><br></pre><pre><font face=3D"arial"><span style=3D"whit=
e-space:normal">Here we have audio stream with IPV6 and IPV6 network addres=
s, video streams doing the same as well.</span></font></pre><pre><font face=
=3D"arial"><span style=3D"white-space:normal">Now if we want to BUNDLE thes=
e m=3Dlines, one possible way is to split BUNDLE into groups that belong to=
 the same network address type (IPV4 or IPV6) as shown below</span></font><=
/pre>
<pre>v=3D0
o=3Dbob 280744730 28977631 IN IP4 <a href=3D"http://host.example.com">host.=
example.com</a>
s=3D
t=3D0 0
a=3Dgroup:ANAT 1 2
a=3Dgroup:ANAT 3 4
a=3Dgroup:BUNDLE 1 3     // BUNDLE IPv4 audio and video streams
a=3Dgroup:BUNDLE 2 4     // BUNDLE IPv6 audio and video streams.
m=3Daudio 25000 RTP/AVP 0
c=3DIN IP6 2001:DB8::1
a=3D &lt;ICE-encoded additional IPv6 addresses (and ports)&gt;
a=3Dmid:1
m=3Daudio 22334 RTP/AVP 0
c=3DIN IP4 192.0.2.1
a=3D &lt;ICE-encoded additional IPv4 addresses (and ports)&gt;
a=3Dmid:2
m=3Dvideo 25112 RTP/AVP 0
c=3DIN IP6 2001:DB8::1
a=3D &lt;ICE-encoded additional IPv6 addresses (and ports)&gt;
a=3Dmid:3
m=3Dvideo 24544 RTP/AVP 0
c=3DIN IP4 192.0.2.1
a=3D &lt;ICE-encoded additional IPv4 addresses (and ports)&gt;
a=3Dmid:4</pre><pre><br></pre><pre><font face=3D"arial"><span style=3D"whit=
e-space:normal">Any thoughts if this sounds like the right approach ?</span=
></font></pre><pre><font face=3D"arial"><span style=3D"white-space:normal">=
<br></span></font></pre>
<pre><font face=3D"arial"><span style=3D"white-space:normal"><br></span></f=
ont></pre><pre><font face=3D"arial"><span style=3D"white-space:normal">Chee=
rs</span></font></pre><pre><font face=3D"arial"><span style=3D"white-space:=
normal">Suhas</span></font></pre>
<pre><br></pre><pre><font face=3D"arial"><span style=3D"white-space:normal"=
><br></span></font></pre><pre><font face=3D"arial"><span style=3D"white-spa=
ce:normal"><br></span></font></pre><pre><font face=3D"arial"><span style=3D=
"white-space:normal"><br>
</span></font></pre></div></div>

--001a11c25d36b8433304f23a5122--


From martin.thomson@gmail.com  Wed Feb 12 11:26:48 2014
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Subject: Re: [MMUSIC] BUNDLE and RFC 4091 ANAT Semantics
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On 12 February 2014 11:09, Suhas Nandakumar <suhasietf@gmail.com> wrote:
> ANAT semantics

Since RFC 4091 is obsolete, is this really that important?


From suhasietf@gmail.com  Wed Feb 12 11:31:48 2014
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On Wed, Feb 12, 2014 at 11:26 AM, Martin Thomson
<martin.thomson@gmail.com>wrote:

> On 12 February 2014 11:09, Suhas Nandakumar <suhasietf@gmail.com> wrote:
> > ANAT semantics
>
> Since RFC 4091 is obsolete, is this really that important?
>

Good point. I feel we should just assign NOT RECOMMENDED in the Mux draft
and leave it there.

Cheers
Suhas

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Wed, Feb 12, 2014 at 11:26 AM, Martin Thomson <span dir=3D"ltr">=
&lt;<a href=3D"mailto:martin.thomson@gmail.com" target=3D"_blank">martin.th=
omson@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">On 12 February 2014 11:09, Suhas Nandakumar =
&lt;<a href=3D"mailto:suhasietf@gmail.com">suhasietf@gmail.com</a>&gt; wrot=
e:<br>

&gt; ANAT semantics<br>
<br>
Since RFC 4091 is obsolete, is this really that important?<br></blockquote>=
<div><br></div><div>Good point. I feel we should just assign NOT RECOMMENDE=
D in the Mux draft and leave it there.</div><div><br></div><div>Cheers</div=
>
<div>Suhas=A0<br></div></div><br></div></div>

--f46d04428070836fc004f23a9fc8--


From magnus.westerlund@ericsson.com  Thu Feb 13 01:05:54 2014
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Subject: Re: [MMUSIC] [rtcweb] Rejecting MediaStreamTracks in JSEP
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On 2014-02-04 01:05, Justin Uberti wrote:

> 
> Agree that msid should map m-line to MST, but it would seem recv-msid
> could map RTP flow to m=line as well as indicate whether the receiver is
> in fact interested in receiving the MST associated with said m= line.
> Note that recv-msid doesn't specify an actual MSID; its purpose is to
> indicate that the receiver wants to receive data from the remote side's
> MST, and provide a mechanism for the sender to mark the data
> accordingly. (Which ends up seeming a lot like appid.)
> 

A question here, what are you really trying to reject?

Is it the MST, a particular RTP Stream (Id by SSRC) or the usage of this
media description (m= block)  in a particular direction (i.e. SDP internal)?

I think the solution differs quite a lot depending on what you really
try to accomplish.

cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
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On 14 January 2014 03:05, Charles Eckel (eckelcu) <eckelcu@cisco.com> wrote:

> I am afraid Ali is right. Let's take the SHOULDs 1 by 1 and see how to
> deal with them.
>

I assume Ali and Charles are right, so we better brush up the mandatory
language before issuing the next (and hopefully last) version of the draft.


> 1) From section 3:
> "The fmt (format) list is ignored for BFCP.  The fmt list of BFCP 'm'
> The fmt (format) list is ignored for BFCP.  The fmt list of BFCP 'm'
>          lines SHOULD contain a single "*" character."
>
>
> This is inherited from RFC 4583. I see no reason to not use '*'. Unless
> someone can identify one, I suggest changing to "MUST".
>

Yes, but we have seen an implementation (old, but might be out in the wild)
using a digit and not *. So, I propose we use the SHOULD here still, but
introduce the text proposed by Mary Barnes in the accompanying BFCPbis WG
PROTO review thread:

  "The fmt (format) list is not applicable to BFCP. The fmt list of 'm'
   lines in the case of any proto field value related to BFCP SHOULD
   contain a single "*" character. If the the fmt list contains any other
   value it is ignored."


> 2) From section 4:
> "Endpoints that use the offer/answer model to establish BFCP
>    connections MUST support the 'floorctrl' attribute.  A floor control
>    server acting as an offerer or as an answerer SHOULD include this
>    attribute in its session descriptions.
>    If the 'floorctrl' attribute is not used in an offer/answer exchange,
>    by default the offerer and the answerer will act as a floor control
>    client and as a floor control server, respectively."
>
> I think this is okay as is.
>

I agree, since the default behaviour is specified.


> 3) From section 5:
> "Endpoints that use the offer/answer model to establish BFCP
>    connections MUST support the 'confid' and the 'userid' attributes.  A
>    floor control server acting as an offerer or as an answerer SHOULD
>    include these attributes in its session descriptions."
>
> This is inherited form RFC 4583. I have seen clients not including these
> in an offer, but I cannot think of a case in which a server would not
> include them these. I suggest changing to "MUST".
>

Yes, I agree that the floor control server must include these
identificators.


> 4) From section 6:
> "Endpoints that use the offer/answer model to establish BFCP
>    connections MUST support the 'floorid' and the 'label' attributes.  A
>    floor control server acting as an offerer or as an answerer SHOULD
>    include these attributes in its session descriptions."
>
>
> This is inherited form RFC 4583. I cannot think of a case in which a
> server would not include these. I suggest changing to "MUST".
>

Yes, I agree here as well.


> 5) There are several SHOULDs in section 7, which defines the new bfcpver
> attribute. As this attribute did not exist in RFC 4583, the SHOULDs are
> warranted for backward compatibility. However, they can be made mandatory
> for implementations of BFCP/UDP as well as those compliant with this
> update to RFC 4583.
>

True, and as proposed in the accompanying BFCPbis WG PROTO review thread
merging in this requirement for implementations of the upcoming RFCXXX of
this draft. The paragraph will then read:

  "Endpoints that use the offer/answer model to establish BFCP
   connections SHOULD support the 'bfcpver' attribute. A floor control
   server acting as an offerer or as an answerer SHOULD include
   this attribute in its session descriptions.   However, endpoints that
   support RFCXXXX, and not only the RFC 4583 subset, are
   REQUIRED to support and, when acting as a floor control server
   to use the 'bfcpver' attribute.  [...]"


> The only should for which this is not the case is the following:
>
> "Tokens representing versions SHOULD be
>    integers matching the "Version" field that would be presented in the
>    BFCP COMMON-HEADER [8
> <http://tools.ietf.org/html/draft-ietf-bfcpbis-rfc4583bis-08#ref-8>]."
>
> I think this can be changed to "MUST" for all cases.
>

Yes, will be done.


> 6) From section 8.1:
> "When the existing TCP connection is reset following the rules in [8
> <http://tools.ietf.org/html/draft-ietf-bfcpbis-rfc4583bis-08#ref-8>],
>    the client SHOULD generate an offer towards the floor control server
>    in order to reestablish the connection.  If a TCP connection cannot
>    deliver a BFCP message and times out, the entity that attempted to
>    send the message (i.e., the one that detected the TCP timeout) SHOULD
>    generate an offer in order to reestablish the TCP connection."
>
> I am not sure about these. I think they could be changed to "MUST".
> Hopefully others can chime in.
>

This might be considered being part of application logic/policy, but I
can't see a reason for not trying to reestablish the connection here and
now. Anyway, I'll leave these two as "SHOULD". Shout out if you disagree.


> 7) Section 9 reads as follows:
> "Endpoints that use the offer/
>    answer model to establish BFCP connections MUST support the
>    'fingerprint' attribute and SHOULD include it in their session
>    descriptions."
>
> I think this can be changed to "MUST"; however, I would appreciate
> additional opinions on this one as well.


To actually get the security/authentication from TLS and establish a secure
channel, the fingerprint attribute must be used of course. So, I'm fine
with mandating the use of it. Will change to "MUST".

 Thanks to Charles for going through all the "SHOULDs" in the draft an
discuss them, the fixes will be in the upcoming version of the draft to be
issued very very soon.

In addition Ali's original review issue #1 is fixed, but #2 is considered
taken care of in the existing text as described earlier.

-- Tom

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<div dir=3D"ltr">On 14 January 2014 03:05, Charles Eckel (eckelcu) <span di=
r=3D"ltr">&lt;<a href=3D"mailto:eckelcu@cisco.com" target=3D"_blank">eckelc=
u@cisco.com</a>&gt;</span> wrote:<br><div class=3D"gmail_extra"><div class=
=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">I am afraid Ali is right. Let&#39;s take the SHOULDs 1 by =
1 and see how to<br>

deal with them.<br></blockquote><div><br></div><div style>I assume Ali and =
Charles are right, so we better brush up the mandatory language before issu=
ing the next (and hopefully last) version of the draft.</div><div>=A0</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
1) From section 3:<br>
&quot;The fmt (format) list is ignored for BFCP. =A0The fmt list of BFCP &#=
39;m&#39;<br>
The fmt (format) list is ignored for BFCP. =A0The fmt list of BFCP &#39;m&#=
39;<br>
=A0 =A0 =A0 =A0 =A0lines SHOULD contain a single &quot;*&quot; character.&q=
uot;<br>
<br>
<br>
This is inherited from RFC 4583. I see no reason to not use &#39;*&#39;. Un=
less<br>
someone can identify one, I suggest changing to &quot;MUST&quot;.<br></bloc=
kquote><div><br></div><div style>Yes, but we have seen an implementation (o=
ld, but might be out in the wild) using a digit and not *. So, I propose we=
 use the SHOULD here still, but introduce the text proposed by Mary Barnes =
in the accompanying BFCPbis WG PROTO review thread:</div>
<div style><br></div><div style><div>=A0 &quot;The fmt (format) list is not=
 applicable to BFCP. The fmt list of &#39;m&#39;</div><div>=A0 =A0lines in =
the case of any proto field value related to BFCP SHOULD</div><div>=A0 =A0c=
ontain a single &quot;*&quot; character. If the the fmt list contains any o=
ther</div>
<div>=A0 =A0value it is ignored.&quot;</div></div><div>=A0</div><blockquote=
 class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:=
1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left=
:1ex">

2) From section 4:<br>
&quot;Endpoints that use the offer/answer model to establish BFCP<br>
=A0 =A0connections MUST support the &#39;floorctrl&#39; attribute. =A0A flo=
or control<br>
=A0 =A0server acting as an offerer or as an answerer SHOULD include this<br=
>
=A0 =A0attribute in its session descriptions.<br>
=A0 =A0If the &#39;floorctrl&#39; attribute is not used in an offer/answer =
exchange,<br>
=A0 =A0by default the offerer and the answerer will act as a floor control<=
br>
=A0 =A0client and as a floor control server, respectively.&quot;<br><br>
I think this is okay as is.<br></blockquote><div><br></div><div style>I agr=
ee, since the default behaviour is specified.</div><div>=A0</div><blockquot=
e class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width=
:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-lef=
t:1ex">

3) From section 5:<br>
&quot;Endpoints that use the offer/answer model to establish BFCP<br>
=A0 =A0connections MUST support the &#39;confid&#39; and the &#39;userid&#3=
9; attributes. =A0A<br>
=A0 =A0floor control server acting as an offerer or as an answerer SHOULD<b=
r>
=A0 =A0include these attributes in its session descriptions.&quot;<br>
<br>
This is inherited form RFC 4583. I have seen clients not including these<br=
>
in an offer, but I cannot think of a case in which a server would not<br>
include them these. I suggest changing to &quot;MUST&quot;.<br></blockquote=
><div><br></div><div style>Yes, I agree that the floor control server must =
include these identificators.</div><div>=A0</div><blockquote class=3D"gmail=
_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left=
-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">

4) From section 6:<br>
&quot;Endpoints that use the offer/answer model to establish BFCP<br>
=A0 =A0connections MUST support the &#39;floorid&#39; and the &#39;label&#3=
9; attributes. =A0A<br>
=A0 =A0floor control server acting as an offerer or as an answerer SHOULD<b=
r>
=A0 =A0include these attributes in its session descriptions.&quot;<br>
<br>
<br>
This is inherited form RFC 4583. I cannot think of a case in which a<br>
server would not include these. I suggest changing to &quot;MUST&quot;.<br>=
</blockquote><div><br></div><div style>Yes, I agree here as well.</div><div=
>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8=
ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-sty=
le:solid;padding-left:1ex">

5) There are several SHOULDs in section 7, which defines the new bfcpver<br=
>
attribute. As this attribute did not exist in RFC 4583, the SHOULDs are<br>
warranted for backward compatibility. However, they can be made mandatory<b=
r>
for implementations of BFCP/UDP as well as those compliant with this<br>
update to RFC 4583.<br></blockquote><div><br></div><div style>True, and as =
proposed in the accompanying BFCPbis WG PROTO review thread merging in this=
 requirement for implementations of the upcoming RFCXXX of this draft. The =
paragraph will then read:</div>
<div style><br></div><div style>=A0 &quot;Endpoints that use the offer/answ=
er model to establish BFCP</div><div style>=A0 =A0connections SHOULD suppor=
t the &#39;bfcpver&#39; attribute. A floor control</div><div style>=A0 =A0s=
erver acting as an offerer or as an answerer SHOULD include</div>
<div style>=A0 =A0this attribute in its session descriptions. =A0 However, =
endpoints that</div><div style>=A0 =A0support RFCXXXX, and not only the RFC=
 4583 subset, are</div><div style>=A0 =A0REQUIRED to support and, when acti=
ng as a floor control server</div>
<div style>=A0 =A0to use the &#39;bfcpver&#39; attribute. =A0[...]&quot;</d=
iv><div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px =
0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-l=
eft-style:solid;padding-left:1ex">

The only should for which this is not the case is the following:<br>
<br>
&quot;Tokens representing versions SHOULD be<br>
=A0 =A0integers matching the &quot;Version&quot; field that would be presen=
ted in the<br>
=A0 =A0BFCP COMMON-HEADER [8<br>
&lt;<a href=3D"http://tools.ietf.org/html/draft-ietf-bfcpbis-rfc4583bis-08#=
ref-8" target=3D"_blank">http://tools.ietf.org/html/draft-ietf-bfcpbis-rfc4=
583bis-08#ref-8</a>&gt;].&quot;<br><br>
I think this can be changed to &quot;MUST&quot; for all cases.<br></blockqu=
ote><div><br></div><div style>Yes, will be done.</div><div>=A0</div><blockq=
uote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-wi=
dth:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-=
left:1ex">

6) From section 8.1:<br>
&quot;When the existing TCP connection is reset following the rules in [8<b=
r>
&lt;<a href=3D"http://tools.ietf.org/html/draft-ietf-bfcpbis-rfc4583bis-08#=
ref-8" target=3D"_blank">http://tools.ietf.org/html/draft-ietf-bfcpbis-rfc4=
583bis-08#ref-8</a>&gt;],<br>
=A0 =A0the client SHOULD generate an offer towards the floor control server=
<br>
=A0 =A0in order to reestablish the connection. =A0If a TCP connection canno=
t<br>
=A0 =A0deliver a BFCP message and times out, the entity that attempted to<b=
r>
=A0 =A0send the message (i.e., the one that detected the TCP timeout) SHOUL=
D<br>
=A0 =A0generate an offer in order to reestablish the TCP connection.&quot;<=
br>
<br>
I am not sure about these. I think they could be changed to &quot;MUST&quot=
;.<br>
Hopefully others can chime in.<br></blockquote><div><br></div><div style>Th=
is might be considered being part of application logic/policy, but I can&#3=
9;t see a reason for not trying to reestablish the connection here and now.=
 Anyway, I&#39;ll leave these two as &quot;SHOULD&quot;. Shout out if you d=
isagree.</div>
<div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px=
 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left=
-style:solid;padding-left:1ex">
7) Section 9 reads as follows:<br>
&quot;Endpoints that use the offer/<br>
=A0 =A0answer model to establish BFCP connections MUST support the<br>
=A0 =A0&#39;fingerprint&#39; attribute and SHOULD include it in their sessi=
on<br>
=A0 =A0descriptions.&quot;<br>
<br>
I think this can be changed to &quot;MUST&quot;; however, I would appreciat=
e<br>
additional opinions on this one as well.=A0</blockquote><div><br></div><div=
 style>To actually get the security/authentication from TLS and establish a=
 secure channel, the fingerprint attribute must be used of course. So, I&#3=
9;m fine with mandating the use of it. Will change to &quot;MUST&quot;.</di=
v>
<div style><br></div><div style>=A0Thanks to Charles for going through all =
the &quot;SHOULDs&quot; in the draft an discuss them, the fixes will be in =
the upcoming version of the draft to be issued very very soon.</div><div>
<br></div><div style>In addition Ali&#39;s original review issue #1 is fixe=
d, but #2 is considered taken care of in the existing text as described ear=
lier.</div><div style><br></div><div style>-- Tom=A0</div></div></div></div=
>

--001a113aadc268837b04f249b60d--


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On 02/13/2014 02:31 PM, Tom Kristensen wrote:
> On 14 January 2014 03:05, Charles Eckel (eckelcu) <eckelcu@cisco.com 
> <mailto:eckelcu@cisco.com>> wrote:
>
>     I am afraid Ali is right. Let's take the SHOULDs 1 by 1 and see how to
>     deal with them.
>

[...]

>     6) From section 8.1:
>     "When the existing TCP connection is reset following the rules in [8
>     <http://tools.ietf.org/html/draft-ietf-bfcpbis-rfc4583bis-08#ref-8>],
>        the client SHOULD generate an offer towards the floor control
>     server
>        in order to reestablish the connection.  If a TCP connection cannot
>        deliver a BFCP message and times out, the entity that attempted to
>        send the message (i.e., the one that detected the TCP timeout)
>     SHOULD
>        generate an offer in order to reestablish the TCP connection."
>
>     I am not sure about these. I think they could be changed to "MUST".
>     Hopefully others can chime in.
>
>
> This might be considered being part of application logic/policy, but I 
> can't see a reason for not trying to reestablish the connection here 
> and now. Anyway, I'll leave these two as "SHOULD". Shout out if you 
> disagree.

I'm disagreeing with the previous version of me. This will be changed to 
"MUSTs" for the two cases, I cannot really see a reason for keeping the 
"SHOULDs".

-- Tom

<snip>

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On 02/13/2014 02:31 PM, Tom Kristensen wrote:
<blockquote
 cite="mid:CAFHv=r_MvOubfhCp8m0w09dJpszVabOekn=sE4KG+UvxygUutQ@mail.gmail.com"
 type="cite">
  <meta http-equiv="Content-Type"
 content="text/html; charset=ISO-8859-1">
  <div dir="ltr">On 14 January 2014 03:05, Charles Eckel (eckelcu) <span
 dir="ltr">&lt;<a moz-do-not-send="true" href="mailto:eckelcu@cisco.com"
 target="_blank">eckelcu@cisco.com</a>&gt;</span> wrote:<br>
  <div class="gmail_extra">
  <div class="gmail_quote">
  <blockquote class="gmail_quote"
 style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;">I
am afraid Ali is right. Let's take the SHOULDs 1 by 1 and see how to<br>
deal with them.<br>
  </blockquote>
  </div>
  </div>
  </div>
</blockquote>
<br>
[...]<br>
&nbsp;<br>
<blockquote
 cite="mid:CAFHv=r_MvOubfhCp8m0w09dJpszVabOekn=sE4KG+UvxygUutQ@mail.gmail.com"
 type="cite">
  <div dir="ltr">
  <div class="gmail_extra">
  <div class="gmail_quote">
  <blockquote class="gmail_quote"
 style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;">6)
>From section 8.1:<br>
"When the existing TCP connection is reset following the rules in [8<br>
&lt;<a moz-do-not-send="true"
 href="http://tools.ietf.org/html/draft-ietf-bfcpbis-rfc4583bis-08#ref-8"
 target="_blank">http://tools.ietf.org/html/draft-ietf-bfcpbis-rfc4583bis-08#ref-8</a>&gt;],<br>
&nbsp; &nbsp;the client SHOULD generate an offer towards the floor control server<br>
&nbsp; &nbsp;in order to reestablish the connection. &nbsp;If a TCP connection cannot<br>
&nbsp; &nbsp;deliver a BFCP message and times out, the entity that attempted to<br>
&nbsp; &nbsp;send the message (i.e., the one that detected the TCP timeout) SHOULD<br>
&nbsp; &nbsp;generate an offer in order to reestablish the TCP connection."<br>
    <br>
I am not sure about these. I think they could be changed to "MUST".<br>
Hopefully others can chime in.<br>
  </blockquote>
  <div><br>
  </div>
  <div style="">This might be considered being part of application
logic/policy, but I can't see a reason for not trying to reestablish
the connection here and now. Anyway, I'll leave these two as "SHOULD".
Shout out if you disagree.</div>
  </div>
  </div>
  </div>
</blockquote>
<br>
I'm disagreeing with the previous version of me. This will be changed
to "MUSTs" for the two cases, I cannot really see a reason for keeping
the "SHOULDs".<br>
<br>
-- Tom<br>
<br>
&lt;snip&gt;<br>
</body>
</html>

--------------010708080104010105000003--


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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-06.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Multiparty Multimedia Session Control Working Group of the IETF.

        Title           : Stream Control Transmission Protocol (SCTP)-Based Media Transport in the Session Description Protocol (SDP)
        Authors         : Salvatore Loreto
                          Gonzalo Camarillo
	Filename        : draft-ietf-mmusic-sctp-sdp-06.txt
	Pages           : 14
	Date            : 2014-02-13

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


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

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

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-sctp-sdp-06


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

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


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From: Jonathan Lennox <jonathan@vidyo.com>
To: mmusic <mmusic@ietf.org>
Thread-Topic: MSID and AppID: solving different problems
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The chairs asked that the authors of the MSID and AppID drafts discuss the =
relationship between the two documents.

My personal opinion (which I mentioned in the RTCWeb Rejecting MediaStreamT=
racks thread, but folks might not have seen) is that they're solving differ=
ent problems, and so are non-conflicting.

MSID provides an identifier for SDP m=3D lines.  Its primary use case is pr=
oviding a Javascript-level handle in WebRTC.

AppID -- which needs a better name, suggestions welcome -- provides an indi=
rection label for identifying individual RTP packet streams in SDP, in scen=
arios where the SSRC is unavailable to the signaling.

The scenarios we've identified so far for AppID where an SSRC might be unav=
ailable are: "early media" race conditions where media outraces an SDP answ=
er; SSRC collisions; applications which wish to dynamically switch which SS=
RC is fulfilling a higher-level application role; and architectures where t=
he signaling stack can't control/doesn't have visibility onto the SSRC sele=
ction.

AppID should be applicable to anything using BUNDLE, I think, and certainly=
 anything using it with a vaguely Unified-style semantic.  (I think it woul=
d apply to a "Plan B"-style semantic, as well, which I know some people are=
 still advocating for some non-WebRTC contexts; however, I'm not positive a=
ll the syntax necessary for that is currently complete.)

The difference between AppID and MSID should be most clear in scenarios whe=
re an m=3D line group has more than one packet stream (my canonical example=
 is a primary stream and a retransmission stream).  In this case, the m=3D =
line group has one msid attribute, but multiple appid attributes.=


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On 31/01/14 15:44, Pal Martinsen (palmarti) wrote:
> On 30 Jan 2014, at 12:34 pm, Ari Keränen <ari.keranen@ericsson.com> wrote:
>> On 28/01/14 15:49, Pal Martinsen (palmarti) wrote:
[...]
>> What I would find helpful for discussion is that if you could provide a step-by-step example of how the candidate lists are formed and connectivity checks done in cases where a) both support this with same promo algo b) both support this but with different promo algo c) only one endpoint supports this.
>>
>
> As I do believe in running code, I wrote a simple implementation of the algorithm described below.

That's very nice, but unfortunately that didn't compile on my Mac so I 
couldn't give it a try.

>> Also maybe a recommended promo algo would make sense so that there is at least a fair chance that both will use the same algo/values and get most out of this.
>>
> First shot an an algorithm:
>
> 5.  Example Algorithm for Choosing the Local Preference
>
>     The value space for the local preference is from 0 to 65535
>     inclusive.  This value space can be divided up in chunks for each IP
>     address family.
>
>     An IPv6 and IPv4 start priority must be given.  In this example IPv6
>     starts at 60000 and IPv4 at 59000.  This leaves enough address space
>     to further play with the values if pr interface priorities needs to
>     be added.  The highest value should be given to the address family
>     that should be prioritized.
>
>          IPv6    IPv4
>           Start   Start
>     65535  60k     59k    58k    57k    56k    55k                    0
>     +--------+------+------+------+------+------+---------------------+
>     |        | IPv6 | IPv4 | IPv6 | IPv4 | IPv6 |                     |
>     |        | (1)  |  (1) |  (2) |  (2) |  (3) |                     |
>     +--------+------+------+------+------+------+---------------------+
>               <- N->
>
>
>     The local preference can be calculated by the given formula:
>
>
>           local_preference = N*2*(Cn/Cmax)
>
>     Where N is the absolute value of IPv6_start-IPv4_start.  This ensures
>     a positive number even if IPv4 is the highest priority.  Cn is the
>     number of current candidates of a specific IP address type and
>     candidate type (HOST, SRFLX, RELAY).  Cmax is the number of allowed
>     consecutive candidates of the same IP address type.
>

Could you give an example how this algorithm and some priority values, 
e.g., from your program, result in the candidate pairs and order of 
checks? Especially the pairing part with respect to different types of 
candidates and IP families is where I'm afraid things may go wrong.


Cheers,
Ari


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Subject: Re: [MMUSIC] [rtcweb] Rejecting MediaStreamTracks in JSEP
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What's being asked for is a way to control MediaStreamTracks.

The RTP streams are only incidentally impacted.

I'll put a strawman proposal into -msid- today.


On Thu, Feb 13, 2014 at 10:05 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> On 2014-02-04 01:05, Justin Uberti wrote:
>
> >
> > Agree that msid should map m-line to MST, but it would seem recv-msid
> > could map RTP flow to m=3Dline as well as indicate whether the receiver=
 is
> > in fact interested in receiving the MST associated with said m=3D line.
> > Note that recv-msid doesn't specify an actual MSID; its purpose is to
> > indicate that the receiver wants to receive data from the remote side's
> > MST, and provide a mechanism for the sender to mark the data
> > accordingly. (Which ends up seeming a lot like appid.)
> >
>
> A question here, what are you really trying to reject?
>
> Is it the MST, a particular RTP Stream (Id by SSRC) or the usage of this
> media description (m=3D block)  in a particular direction (i.e. SDP
> internal)?
>
> I think the solution differs quite a lot depending on what you really
> try to accomplish.
>
> cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=C3=A4r=C3=B6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
>

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<div dir=3D"ltr">What&#39;s being asked for is a way to control MediaStream=
Tracks.<div><br></div><div>The RTP streams are only incidentally impacted.<=
/div><div><br></div><div>I&#39;ll put a strawman proposal into -msid- today=
.</div>

</div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Thu,=
 Feb 13, 2014 at 10:05 AM, Magnus Westerlund <span dir=3D"ltr">&lt;<a href=
=3D"mailto:magnus.westerlund@ericsson.com" target=3D"_blank">magnus.westerl=
und@ericsson.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 class=3D"">On 2014-02-04 01:05, Justin =
Uberti wrote:<br>
<br>
&gt;<br>
&gt; Agree that msid should map m-line to MST, but it would seem recv-msid<=
br>
&gt; could map RTP flow to m=3Dline as well as indicate whether the receive=
r is<br>
&gt; in fact interested in receiving the MST associated with said m=3D line=
.<br>
&gt; Note that recv-msid doesn&#39;t specify an actual MSID; its purpose is=
 to<br>
&gt; indicate that the receiver wants to receive data from the remote side&=
#39;s<br>
&gt; MST, and provide a mechanism for the sender to mark the data<br>
&gt; accordingly. (Which ends up seeming a lot like appid.)<br>
&gt;<br>
<br>
</div>A question here, what are you really trying to reject?<br>
<br>
Is it the MST, a particular RTP Stream (Id by SSRC) or the usage of this<br=
>
media description (m=3D block) =C2=A0in a particular direction (i.e. SDP in=
ternal)?<br>
<br>
I think the solution differs quite a lot depending on what you really<br>
try to accomplish.<br>
<br>
cheers<br>
<br>
Magnus Westerlund<br>
<br>
----------------------------------------------------------------------<br>
Services, Media and Network features, Ericsson Research EAB/TXM<br>
----------------------------------------------------------------------<br>
Ericsson AB =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | Phone=
 =C2=A0<a href=3D"tel:%2B46%2010%207148287" value=3D"+46107148287">+46 10 7=
148287</a><br>
F=C3=A4r=C3=B6gatan 6 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 | Mobile <a href=3D"tel:%2B46%2073%200949079" value=3D"+46730949079">+4=
6 73 0949079</a><br>
SE-164 80 Stockholm, Sweden | mailto: <a href=3D"mailto:magnus.westerlund@e=
ricsson.com">magnus.westerlund@ericsson.com</a><br>
----------------------------------------------------------------------<br>
<br>
</blockquote></div><br></div>

--089e0117696f994d5b04f24da06a--


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Thread-Topic: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-06.txt - Christer's initial comments
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Hi,

I gave some comments off-line to Salvatore, which have been addressed in th=
e new version.

Some additional comments. I do realize some are pure editorial, but rather =
to write then down in some notes I put them directly into the e-mail :)


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

Q1 (EDITORIAL):

The text says:

   "Since
   no-one so far has implemented SCTP over TLS, due to some serious
   limitations described in [RFC6083], this document does not make use
   of TLS over SCTP as described in RFC3436 [RFC3436]."

I suggest to simply say something like:

   "As described in [RFC6083], there are limitations associated with implem=
enting
  SCTP over TLS. For that reason, SCTP over TLS is outside the scope of thi=
s document."


Q2 (EDITORIAL):

The text says:

  "Additionally, this document specifies the use of the 'setup' and
   'connection' SDP attributes to establish SCTP associations.  These
   attributes were defined in RFC4145 [RFC4145] for TCP.  This document
   discusses their use with SCTP."

I suggest to remove the last sentence, as it only repeats what is said in t=
he first sentence.


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

Q3 (EDITORIAL):

The text says:

   "This document defines three new values for the 'proto' field: 'SCTP', '=
SCTP/DTLS' and 'DTLS/SCTP'."

I would suggest to say:

   "This document defines three new protocol identifier values for the 'pro=
to' field: 'SCTP', 'SCTP/DTLS' and 'DTLS/SCTP'."

...because, you use "protocol identifier" terminology later.


Q4 (CLARIFICATION):

The text says:

   "An 'm' line that specifies 'SCTP' or 'SCTP/DTLS' or 'DTLS/SCTP' MUST
   further qualify the application-layer protocol using an fmt
   identifier."

I do understand what you are trying to say, but I think the "qualify" word =
is a little confusing.

Maybe something like:

   "An 'm' line that specifies 'SCTP' or 'SCTP/DTLS' or 'DTLS/SCTP' MUST
   further include an fmt value that identifies the application-layer proto=
col
   carried on top of the SCTP association."


Section 4.1:
--------------

Q5 (TECHNICAL):

The text says:

   "For the payload type assignments the "a=3Dsctpmap:"
   attribute (see Section 5.1) 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."

It is very difficult for me to parse this, and figure out the semantics of =
the sctpmap attribute.

First, what do you mean by "media encoding name"?

Second, when you talk about "application protocol", I assume you refer to t=
he SCTP protocol?

Third, the text say SHOULD. When is it ok to not use the sctpmap attribute?


Section 5:
------------

Q6 (EDITORIAL):

I suggest to rename section 5 to "SDP sctpmap attribute", rename section 5.=
1 to "General", and add a new section 5.2 named "ABNF".


Q7 (TECHNICAL):

The text says:

   "The sctpmap MUST include the app parameter indicating the application
   running on top of the association."

This is fine if the sctpmap attribute maps to an application. But, what if =
the sctpmap attribute maps to a "media encoding" or "payload"?


Q8 (EDITORIAL):

The text says:

   "The sctpmap line should also contain the max-message-size parameter
   indicating the maximum message size, in bytes, the endpoint is
   willing to accept."

I suggest to replace "accept" with "receive on the associated SCTP associat=
ion".


Q9 (EDITORIAL):

The text says:

      "The peer should assume that larger message will be rejected by the
      endpoint, though it is up to the endpoint decide the appropriate
      behaviour."

I suggest to either remove this paragraph, OR replace it with a sentence sa=
ying that a peer must/should not send messages bigger than the indicated ma=
x value.


Q10 (CLARIFICATION):

The text says:

      "If the parameter is not present, the implementation should provide
      a default, with a suggested value of 64K."

I don't understand "suggested". If the default value is 64K the text should=
 specify (rather than suggest) that.

The same comment applies to the stream parameter, described later in the se=
ction.


Q11 (CLARIFICATION):

The text says:

   "It may also provide the stream parameter to specify the initial
   number of incoming streams to be supported by each side of the
   association."

First, to be consistent with previous text, I suggest to replace "It may al=
so provide" with "The sctpmap line may also contain"

Second, I don't understand "incoming streams to be supported by each side o=
f the association". Doesn't each peer indicate the number of streams it is =
willing to receive? Then, we can always recommend that the peer indicates t=
he same number of streams, if possible.


Q12 (EDITORIAL):

The text that follows the ABNF should be put into a separate "SDP Offer/Ans=
wer Procedures" section.


Section 7:
------------

Q13 (EDITORIAL):

Please don't talk about re-INVITEs, or other SIP messages. Simply talk abou=
t offers and answers.


Q14 (EDITORIAL):

The text says:

   "It is worth to underline that when using SCTP on top of DTLS,.."

To have consistent terminology, please say "SCTP over DTLS".



That's all for now! :)

Regards,

Christer





-----Alkuper=E4inen viesti-----
L=E4hett=E4j=E4: mmusic [mailto:mmusic-bounces@ietf.org] Puolesta internet-=
drafts@ietf.org
L=E4hetetty: 13. helmikuuta 2014 16:21
Vastaanottaja: i-d-announce@ietf.org
Kopio: mmusic@ietf.org
Aihe: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-06.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           : Stream Control Transmission Protocol (SCTP)-Based=
 Media Transport in the Session Description Protocol (SDP)
        Authors         : Salvatore Loreto
                          Gonzalo Camarillo
        Filename        : draft-ietf-mmusic-sctp-sdp-06.txt
        Pages           : 14
        Date            : 2014-02-13

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


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

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

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


Please note that it may take a couple of minutes from the time of submissio=
n until the htmlized version and diff are available at tools.ietf.org.

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

_______________________________________________
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<font face=3D"Calibri" size=3D"2"><span style=3D"font-size:11pt;">
<div>Hi,</div>
<div>&nbsp;</div>
<div>I gave some comments off-line to Salvatore, which have been addressed =
in the new version.</div>
<div>&nbsp;</div>
<div>Some additional comments. I do realize some are pure editorial, but ra=
ther to write then down in some notes I put them directly into the e-mail <=
font face=3D"Wingdings">J</font></div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div><b>Section 1:</b></div>
<div><b>------------</b></div>
<div>&nbsp;</div>
<div><b>Q1</b><b> (EDITORIAL)</b><b>:</b></div>
<div>&nbsp;</div>
<div>The text says:</div>
<div>&nbsp;</div>
<div>&nbsp;&nbsp; &quot;Since</div>
<div>&nbsp;&nbsp; no-one so far has implemented SCTP over TLS, due to some =
serious</div>
<div>&nbsp;&nbsp; limitations described in [RFC6083], this document does no=
t make use</div>
<div>&nbsp;&nbsp; of TLS over SCTP as described in RFC3436 [RFC3436].&quot;=
</div>
<div>&nbsp;</div>
<div>I suggest to simply say something like:</div>
<div>&nbsp;</div>
<div>&nbsp;&nbsp; &quot;As described in [RFC6083], there are limitations as=
sociated with implementing</div>
<div>&nbsp; SCTP over TLS. For that reason, SCTP over TLS is outside the sc=
ope of this document.&quot;</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div><b>Q2</b><b> (EDITORIAL)</b><b>:</b></div>
<div>&nbsp;</div>
<div>The text says:</div>
<div>&nbsp;</div>
<div>&nbsp; &quot;Additionally, this document specifies the use of the 'set=
up' and</div>
<div>&nbsp;&nbsp; 'connection' SDP attributes to establish SCTP association=
s.&nbsp; These</div>
<div>&nbsp;&nbsp; attributes were defined in RFC4145 [RFC4145] for TCP.&nbs=
p; This document</div>
<div>&nbsp;&nbsp; discusses their use with SCTP.&quot;</div>
<div>&nbsp;</div>
<div>I suggest to remove the last sentence, as it only repeats what is said=
 in the first sentence.</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div><b>Section 3:</b></div>
<div><b>------------</b></div>
<div>&nbsp;</div>
<div><b>Q3 (EDITORIAL):</b></div>
<div>&nbsp;</div>
<div>The text says:</div>
<div>&nbsp;</div>
<div>&nbsp;&nbsp; &#8220;This document defines three new values for the 'pr=
oto' field: 'SCTP', 'SCTP/DTLS' and 'DTLS/SCTP'.&#8221;</div>
<div>&nbsp;</div>
<div>I would suggest to say:</div>
<div>&nbsp;</div>
<div>&nbsp;&nbsp; &#8220;This document defines three new <b>protocol identi=
fier</b> values for the 'proto' field: 'SCTP', 'SCTP/DTLS' and 'DTLS/SCTP'.=
&#8221;</div>
<div>&nbsp;</div>
<div>&#8230;because, you use &#8220;protocol identifier&#8221; terminology =
later.</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div><b>Q4 (CLARIFICATION):</b></div>
<div>&nbsp;</div>
<div>The text says:</div>
<div>&nbsp;</div>
<div>&nbsp;&nbsp; &#8220;An 'm' line that specifies 'SCTP' or 'SCTP/DTLS' o=
r 'DTLS/SCTP' MUST</div>
<div>&nbsp;&nbsp; further qualify the application-layer protocol using an f=
mt</div>
<div>&nbsp;&nbsp; identifier.&#8221;</div>
<div>&nbsp;</div>
<div>I do understand what you are trying to say, but I think the &#8220;qua=
lify&#8221; word is a little confusing.</div>
<div>&nbsp;</div>
<div>Maybe something like:</div>
<div>&nbsp;</div>
<div>&nbsp;&nbsp; &#8220;An 'm' line that specifies 'SCTP' or 'SCTP/DTLS' o=
r 'DTLS/SCTP' MUST</div>
<div>&nbsp;&nbsp; further include an fmt value that identifies the applicat=
ion-layer protocol</div>
<div>&nbsp;&nbsp; carried on top of the SCTP association.&#8221;</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div><b>Section 4.1:</b></div>
<div><b>--------------</b></div>
<div>&nbsp;</div>
<div><b>Q5 (TECHNICAL):</b></div>
<div>&nbsp;</div>
<div>The text says:</div>
<div>&nbsp;</div>
<div>&nbsp;&nbsp; &#8220;For the payload type assignments the &quot;a=3Dsct=
pmap:&quot;</div>
<div>&nbsp;&nbsp; attribute (see Section 5.1) SHOULD be used to map from a =
port number</div>
<div>&nbsp;&nbsp; to a media encoding name that identifies the payload form=
at</div>
<div>&nbsp;&nbsp; transported by the association or the actual application =
protocol</div>
<div>&nbsp;&nbsp; running on top of it.&#8221;</div>
<div>&nbsp;</div>
<div>It is very difficult for me to parse this, and figure out the semantic=
s of the sctpmap attribute.</div>
<div>&nbsp;</div>
<div>First, what do you mean by &#8220;media encoding name&#8221;?</div>
<div>&nbsp;</div>
<div>Second, when you talk about &#8220;application protocol&#8221;, I assu=
me you refer to the SCTP protocol?</div>
<div>&nbsp;</div>
<div>Third, the text say SHOULD. When is it ok to not use the sctpmap attri=
bute?</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div><b>Section </b><b>5</b><b>:</b></div>
<div><b>------------</b></div>
<div>&nbsp;</div>
<div><b>Q6 (EDITORIAL):</b></div>
<div>&nbsp;</div>
<div>I suggest to rename section 5 to &#8220;SDP sctpmap attribute&#8221;, =
rename section 5.1 to &#8220;General&#8221;, and add a new section 5.2 name=
d &#8220;ABNF&#8221;.</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div><b>Q7 (TECHNICAL):</b></div>
<div>&nbsp;</div>
<div>The text says:</div>
<div>&nbsp;</div>
<div>&nbsp;&nbsp; &#8220;The sctpmap MUST include the app parameter indicat=
ing the application</div>
<div>&nbsp;&nbsp; running on top of the association.&#8221;</div>
<div>&nbsp;</div>
<div>This is fine if the sctpmap attribute maps to an application. But, wha=
t if the sctpmap attribute maps to a &#8220;media encoding&#8221; or &#8220=
;payload&#8221;?</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div><b>Q8 </b><b>(EDITORIAL):</b></div>
<div>&nbsp;</div>
<div>The text says:</div>
<div>&nbsp;</div>
<div>&nbsp;&nbsp; &#8220;The sctpmap line should also contain the max-messa=
ge-size parameter</div>
<div>&nbsp;&nbsp; indicating the maximum message size, in bytes, the endpoi=
nt is</div>
<div>&nbsp;&nbsp; willing to accept.&#8221;</div>
<div>&nbsp;</div>
<div>I suggest to replace &#8220;accept&#8221; with &#8220;receive on the a=
ssociated SCTP association&#8221;.</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div><b>Q9 (EDITORIAL):</b></div>
<div>&nbsp;</div>
<div>The text says:</div>
<div>&nbsp;</div>
<div>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &#8220;The peer should assume that larg=
er message will be rejected by the</div>
<div>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; endpoint, though it is up to the endpoi=
nt decide the appropriate</div>
<div>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; behaviour.&#8221;</div>
<div>&nbsp;</div>
<div>I suggest to either remove this paragraph, OR replace it with a senten=
ce saying that a peer must/should not send messages bigger than the indicat=
ed max value.</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div><b>Q10 (CLARIFICATION):</b></div>
<div>&nbsp;</div>
<div>The text says:</div>
<div>&nbsp;</div>
<div>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &#8220;If the parameter is not present,=
 the implementation should provide</div>
<div>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a default, with a suggested value of 64=
K.&#8221;</div>
<div>&nbsp;</div>
<div>I don&#8217;t understand &#8220;suggested&#8221;. If the default value=
 is 64K the text should specify (rather than suggest) that.</div>
<div>&nbsp;</div>
<div>The same comment applies to the stream parameter, described later in t=
he section.</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div><b>Q11 (CLARIFICATION):</b></div>
<div>&nbsp;</div>
<div>The text says:</div>
<div>&nbsp;</div>
<div>&nbsp;&nbsp; &#8220;It may also provide the stream parameter to specif=
y the initial</div>
<div>&nbsp;&nbsp; number of incoming streams to be supported by each side o=
f the</div>
<div>&nbsp;&nbsp; association.&#8221;</div>
<div>&nbsp;</div>
<div>First, to be consistent with previous text, I suggest to replace &#822=
0;It may also provide&#8221; with &#8220;The sctpmap line may also contain&=
#8221;</div>
<div>&nbsp;</div>
<div>Second, I don&#8217;t understand &#8220;incoming streams to be support=
ed by each side of the association&#8221;. Doesn&#8217;t each peer indicate=
 the number of streams it is willing to receive? Then, we can always recomm=
end that the peer indicates the same number of streams,
if possible.</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div><b>Q12 (EDITORIAL):</b></div>
<div>&nbsp;</div>
<div>The text that follows the ABNF should be put into a separate &#8220;SD=
P Offer/Answer Procedures&#8221; section.</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div><b>Section 7:</b></div>
<div><b>------------</b></div>
<div>&nbsp;</div>
<div><b>Q13 (EDITORIAL):</b></div>
<div>&nbsp;</div>
<div>Please don&#8217;t talk about re-INVITEs, or other SIP messages. Simpl=
y talk about offers and answers.</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div><b>Q14 (EDITORIAL):</b></div>
<div>&nbsp;</div>
<div>The text says:</div>
<div>&nbsp;</div>
<div>&nbsp;&nbsp; &#8220;It is worth to underline that when using SCTP on t=
op of DTLS,..&#8221;</div>
<div>&nbsp;</div>
<div>To have consistent terminology, please say &#8220;SCTP over DTLS&#8221=
;.</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div>That&#8217;s all for now! :)</div>
<div>&nbsp;</div>
<div>Regards,</div>
<div>&nbsp;</div>
<div>Christer</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div>-----Alkuper=E4inen viesti-----<br>

L=E4hett=E4j=E4: mmusic [<a href=3D"mailto:mmusic-bounces@ietf.org">mailto:=
mmusic-bounces@ietf.org</a>] Puolesta internet-drafts@ietf.org<br>

L=E4hetetty: 13. helmikuuta 2014 16:21<br>

Vastaanottaja: i-d-announce@ietf.org<br>

Kopio: mmusic@ietf.org<br>

Aihe: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-06.txt</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div>A New Internet-Draft is available from the on-line Internet-Drafts dir=
ectories.</div>
<div> This draft is a work item of the Multiparty Multimedia Session Contro=
l Working Group of the IETF.</div>
<div>&nbsp;</div>
<div>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Title&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : Stream Control Transmission Protoc=
ol (SCTP)-Based Media Transport in the Session Description Protocol (SDP)</=
div>
<div>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Authors&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp; : Salvatore Loreto</div>
<div>&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; Gonzalo Camarillo</div>
<div>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Filename&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp; : draft-ietf-mmusic-sctp-sdp-06.txt</div>
<div>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Pages&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 14</div>
<div>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Date&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 2014-02-13</div>
<div>&nbsp;</div>
<div>Abstract:</div>
<div>&nbsp;&nbsp; SCTP (Stream Control Transmission Protocol) is a transpor=
t protocol</div>
<div>&nbsp;&nbsp; used to establish associations between two endpoints.&nbs=
p; This document</div>
<div>&nbsp;&nbsp; describes how to express media transport over SCTP in SDP=
 (Session</div>
<div>&nbsp;&nbsp; Description Protocol).&nbsp; This document defines the 'S=
CTP', 'SCTP/DTLS'</div>
<div>&nbsp;&nbsp; and 'DTLS/SCTP' protocol identifiers for SDP.</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div>The IETF datatracker status page for this draft is:</div>
<div><a href=3D"https://datatracker.ietf.org/doc/draft-ietf-mmusic-sctp-sdp=
/">https://datatracker.ietf.org/doc/draft-ietf-mmusic-sctp-sdp/</a></div>
<div>&nbsp;</div>
<div>There's also a htmlized version available at:</div>
<div><a href=3D"http://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-06">h=
ttp://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-06</a></div>
<div>&nbsp;</div>
<div>A diff from the previous version is available at:</div>
<div><a href=3D"http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sctp-s=
dp-06">http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sctp-sdp-06</a>=
</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div>Please note that it may take a couple of minutes from the time of subm=
ission until the htmlized version and diff are available at tools.ietf.org.=
</div>
<div>&nbsp;</div>
<div>Internet-Drafts are also available by anonymous FTP at:</div>
<div><a href=3D"ftp://ftp.ietf.org/internet-drafts/">ftp://ftp.ietf.org/int=
ernet-drafts/</a></div>
<div>&nbsp;</div>
<div>_______________________________________________</div>
<div>mmusic mailing list</div>
<div><a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a></div>
<div><a href=3D"https://www.ietf.org/mailman/listinfo/mmusic">https://www.i=
etf.org/mailman/listinfo/mmusic</a></div>
<div>&nbsp;</div>
</span></font>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1D17183EESESSMB209erics_--


From nobody Thu Feb 13 13:18:09 2014
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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-06.txt - Christer's initial comments
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On 2/13/14 3:57 PM, Christer Holmberg wrote:

> *Q4 (CLARIFICATION):*
> The text says:
>     “An 'm' line that specifies 'SCTP' or 'SCTP/DTLS' or 'DTLS/SCTP' MUST
>     further qualify the application-layer protocol using an fmt
>     identifier.”
> I do understand what you are trying to say, but I think the “qualify”
> word is a little confusing.
> Maybe something like:
>     “An 'm' line that specifies 'SCTP' or 'SCTP/DTLS' or 'DTLS/SCTP' MUST
>     further include an fmt value that identifies the application-layer
> protocol
>     carried on top of the SCTP association.”

The clarification would be fine, but both the original and the 
clarification are misleading. The fmt value does not identify an 
application-layer protocol - it identifies a port number. This is 
explained later in section 4.1.

I think this paragraph could just be omitted, or else be replaced by a 
forward reference to section 4.1.

	Thanks,
	Paul


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#4: #4: Multi-stream demultiplexing

Changes (by fandreas@cisco.com):

 * status:  new => closed
 * resolution:   => worksforme


-- 
-------------------------+-------------------------------------------------
  Reporter:              |      Owner:  draft-ietf-mmusic-media-
  hkaplan@acmepacket.com |  loopback@tools.ietf.org
      Type:  defect      |     Status:  closed
  Priority:  minor       |  Component:  media-loopback
   Version:  1.0         |   Severity:  In WG Last Call
Resolution:  worksforme  |   Keywords:
-------------------------+-------------------------------------------------

Ticket URL: <http://tools.ietf.org/wg/mmusic/trac/ticket/4#comment:2>
mmusic <http://tools.ietf.org/mmusic/>


From nobody Thu Feb 13 13:52:50 2014
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#3: Packet size and fragmentation avoidance

Changes (by fandreas@cisco.com):

 * status:  new => closed
 * resolution:   => worksforme


-- 
-------------------------------------+------------------------------------
  Reporter:  hkaplan@acmepacket.com  |      Owner:  hkaplan@acmepacket.com
      Type:  defect                  |     Status:  closed
  Priority:  major                   |  Component:  media-loopback
   Version:  1.0                     |   Severity:  In WG Last Call
Resolution:  worksforme              |   Keywords:
-------------------------------------+------------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/mmusic/trac/ticket/3#comment:2>
mmusic <http://tools.ietf.org/mmusic/>


From nobody Thu Feb 13 13:53:45 2014
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#5: #5: Marker bit handling

Changes (by fandreas@cisco.com):

 * status:  new => closed
 * resolution:   => worksforme


-- 
-------------------------+-------------------------------------------------
  Reporter:              |      Owner:  draft-ietf-mmusic-media-
  hkaplan@acmepacket.com |  loopback@tools.ietf.org
      Type:  defect      |     Status:  closed
  Priority:  minor       |  Component:  media-loopback
   Version:  1.0         |   Severity:  In WG Last Call
Resolution:  worksforme  |   Keywords:
-------------------------+-------------------------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/mmusic/trac/ticket/5#comment:2>
mmusic <http://tools.ietf.org/mmusic/>


From nobody Thu Feb 13 13:54:36 2014
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#6: Clock-rate per payload type

Changes (by fandreas@cisco.com):

 * status:  new => closed
 * resolution:   => worksforme


-- 
-------------------------+-------------------------------------------------
  Reporter:              |      Owner:  draft-ietf-mmusic-media-
  hkaplan@acmepacket.com |  loopback@tools.ietf.org
      Type:  defect      |     Status:  closed
  Priority:  minor       |  Component:  media-loopback
   Version:  1.0         |   Severity:  In WG Last Call
Resolution:  worksforme  |   Keywords:
-------------------------+-------------------------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/mmusic/trac/ticket/6#comment:2>
mmusic <http://tools.ietf.org/mmusic/>


From nobody Thu Feb 13 13:55:18 2014
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#7: Encapsulated packet format

Changes (by fandreas@cisco.com):

 * status:  new => closed
 * resolution:   => worksforme


-- 
-------------------------+-------------------------------------------------
  Reporter:              |      Owner:  draft-ietf-mmusic-media-
  hkaplan@acmepacket.com |  loopback@tools.ietf.org
      Type:  defect      |     Status:  closed
  Priority:  minor       |  Component:  media-loopback
   Version:  1.0         |   Severity:  In WG Last Call
Resolution:  worksforme  |   Keywords:
-------------------------+-------------------------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/mmusic/trac/ticket/7#comment:2>
mmusic <http://tools.ietf.org/mmusic/>


From nobody Thu Feb 13 13:55:40 2014
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#8: Mirrored packet field overwriting

Changes (by fandreas@cisco.com):

 * status:  new => closed
 * resolution:   => worksforme


-- 
-------------------------+-------------------------------------------------
  Reporter:              |      Owner:  draft-ietf-mmusic-media-
  hkaplan@acmepacket.com |  loopback@tools.ietf.org
      Type:  defect      |     Status:  closed
  Priority:  minor       |  Component:  media-loopback
   Version:  1.0         |   Severity:  In WG Last Call
Resolution:  worksforme  |   Keywords:
-------------------------+-------------------------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/mmusic/trac/ticket/8#comment:2>
mmusic <http://tools.ietf.org/mmusic/>


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Subject: Re: [MMUSIC] [mmusic] #9: Header extensions in fragmented mirror packet
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#9: Header extensions in fragmented mirror packet

Changes (by fandreas@cisco.com):

 * status:  new => closed
 * resolution:   => worksforme


-- 
-------------------------+-------------------------------------------------
  Reporter:              |      Owner:  draft-ietf-mmusic-media-
  hkaplan@acmepacket.com |  loopback@tools.ietf.org
      Type:  defect      |     Status:  closed
  Priority:  minor       |  Component:  media-loopback
   Version:  1.0         |   Severity:  In WG Last Call
Resolution:  worksforme  |   Keywords:
-------------------------+-------------------------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/mmusic/trac/ticket/9#comment:2>
mmusic <http://tools.ietf.org/mmusic/>


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#10: Source and mirror clock synchronization

Changes (by fandreas@cisco.com):

 * status:  new => closed
 * resolution:   => worksforme


-- 
-------------------------+-------------------------------------------------
  Reporter:              |      Owner:  draft-ietf-mmusic-media-
  hkaplan@acmepacket.com |  loopback@tools.ietf.org
      Type:  defect      |     Status:  closed
  Priority:  minor       |  Component:  media-loopback
   Version:  1.0         |   Severity:  In WG Last Call
Resolution:  worksforme  |   Keywords:
-------------------------+-------------------------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/mmusic/trac/ticket/10#comment:2>
mmusic <http://tools.ietf.org/mmusic/>


From nobody Thu Feb 13 13:56:36 2014
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#11: Padding in mirrored packets

Changes (by fandreas@cisco.com):

 * status:  new => closed
 * resolution:   => worksforme


-- 
-------------------------+-------------------------------------------------
  Reporter:              |      Owner:  draft-ietf-mmusic-media-
  hkaplan@acmepacket.com |  loopback@tools.ietf.org
      Type:  defect      |     Status:  closed
  Priority:  minor       |  Component:  media-loopback
   Version:  1.0         |   Severity:  In WG Last Call
Resolution:  worksforme  |   Keywords:
-------------------------+-------------------------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/mmusic/trac/ticket/11#comment:2>
mmusic <http://tools.ietf.org/mmusic/>


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Subject: Re: [MMUSIC] [mmusic] #12: MUST vs. SHOULD for mirror RTP seq number
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#12: MUST vs. SHOULD for mirror RTP seq number

Changes (by fandreas@cisco.com):

 * status:  new => closed
 * resolution:   => worksforme


-- 
-------------------------+-------------------------------------------------
  Reporter:              |      Owner:  draft-ietf-mmusic-media-
  hkaplan@acmepacket.com |  loopback@tools.ietf.org
      Type:  defect      |     Status:  closed
  Priority:  minor       |  Component:  media-loopback
   Version:  1.0         |   Severity:  In WG Last Call
Resolution:  worksforme  |   Keywords:
-------------------------+-------------------------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/mmusic/trac/ticket/12#comment:2>
mmusic <http://tools.ietf.org/mmusic/>


From nobody Thu Feb 13 14:44:05 2014
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Date: Thu, 13 Feb 2014 14:43:57 -0800
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Cc: mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] MSID and AppID: solving different problems
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On Thu, Feb 13, 2014 at 10:09 AM, Jonathan Lennox <jonathan@vidyo.com>wrote:

> The chairs asked that the authors of the MSID and AppID drafts discuss the
> relationship between the two documents.
>
> My personal opinion (which I mentioned in the RTCWeb Rejecting
> MediaStreamTracks thread, but folks might not have seen) is that they're
> solving different problems, and so are non-conflicting.
>
MSID provides an identifier for SDP m= lines.  Its primary use case is
> providing a Javascript-level handle in WebRTC.
>
> AppID -- which needs a better name, suggestions welcome -- provides an
> indirection label for identifying individual RTP packet streams in SDP, in
> scenarios where the SSRC is unavailable to the signaling.
>

[Suhas] How about the name SSID (Source Stream ID) to identify the
individual RTP packet streams


>
> The scenarios we've identified so far for AppID where an SSRC might be
> unavailable are: "early media" race conditions where media outraces an SDP
> answer; SSRC collisions; applications which wish to dynamically switch
> which SSRC is fulfilling a higher-level application role; and architectures
> where the signaling stack can't control/doesn't have visibility onto the
> SSRC selection.
>
> AppID should be applicable to anything using BUNDLE, I think, and
> certainly anything using it with a vaguely Unified-style semantic.  (I
> think it would apply to a "Plan B"-style semantic, as well, which I know
> some people are still advocating for some non-WebRTC contexts; however, I'm
> not positive all the syntax necessary for that is currently complete.)
>
> The difference between AppID and MSID should be most clear in scenarios
> where an m= line group has more than one packet stream (my canonical
> example is a primary stream and a retransmission stream).  In this case,
> the m= line group has one msid attribute, but multiple appid attributes.
>

[Suhas] +1 on the difference be made clear.


> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Thu, Feb 13, 2014 at 10:09 AM, Jonathan Lennox <span dir=3D"ltr"=
>&lt;<a href=3D"mailto:jonathan@vidyo.com" target=3D"_blank">jonathan@vidyo=
.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">The chairs asked that the authors of the MSI=
D and AppID drafts discuss the relationship between the two documents.<br>
<br>
My personal opinion (which I mentioned in the RTCWeb Rejecting MediaStreamT=
racks thread, but folks might not have seen) is that they&#39;re solving di=
fferent problems, and so are non-conflicting.<br></blockquote><blockquote c=
lass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;=
padding-left:1ex">

MSID provides an identifier for SDP m=3D lines. =A0Its primary use case is =
providing a Javascript-level handle in WebRTC.<br>
<br>
AppID -- which needs a better name, suggestions welcome -- provides an indi=
rection label for identifying individual RTP packet streams in SDP, in scen=
arios where the SSRC is unavailable to the signaling.<br></blockquote>
<div><br></div><div>[Suhas] How about the name SSID (Source Stream ID) to i=
dentify the individual RTP packet streams</div><div>=A0</div><blockquote cl=
ass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;p=
adding-left:1ex">

<br>
The scenarios we&#39;ve identified so far for AppID where an SSRC might be =
unavailable are: &quot;early media&quot; race conditions where media outrac=
es an SDP answer; SSRC collisions; applications which wish to dynamically s=
witch which SSRC is fulfilling a higher-level application role; and archite=
ctures where the signaling stack can&#39;t control/doesn&#39;t have visibil=
ity onto the SSRC selection.<br>

<br>
AppID should be applicable to anything using BUNDLE, I think, and certainly=
 anything using it with a vaguely Unified-style semantic. =A0(I think it wo=
uld apply to a &quot;Plan B&quot;-style semantic, as well, which I know som=
e people are still advocating for some non-WebRTC contexts; however, I&#39;=
m not positive all the syntax necessary for that is currently complete.)<br=
>

<br>
The difference between AppID and MSID should be most clear in scenarios whe=
re an m=3D line group has more than one packet stream (my canonical example=
 is a primary stream and a retransmission stream). =A0In this case, the m=
=3D line group has one msid attribute, but multiple appid attributes.<br>
</blockquote><div><br></div><div>[Suhas] +1 on the difference be made clear=
.=A0</div><div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0=
 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
_______________________________________________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/mmusic</a><br>
</blockquote></div><br></div></div>

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From nobody Thu Feb 13 15:49:50 2014
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Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-06.txt - Christer's initial comments
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Hello,

Some further comments:

Section 11.1 (Editorial) - 11.1 The first paragraph regarding 'sctpmap' 
attribute appears to be repeated.

Section 4.1 (Technical): I see the draft has removed the using using 
"bfcp" and "t38" as values for the "App" field under one m-line. However 
that also removes any hint about multiple a=sctpmaps being used under 
one m-line. For example should there be one a=sctpmap per 'fmt' or could 
there be zero or more a=sctpmaps per 'fmt' in the m-line? If think this 
should be clarified.

General: There was a discussion on the CLUE mailing list whether there 
could be multiple CLUE protocol instances per SCTP association. We 
couldn't identify a use case but couldn't rule it out either. I assume 
this would apply to other "apps" also.
For example something like:
m=application 54111 DTLS/SCTP 5000
c=IN IP4 79.97.215.79
a=sctpmap:5000 clue max-message-size=100000 streams=1
a=sctpmap:5000 clue max-message-size=100000 streams=1
a=sctpmap:5000 clue max-message-size=100000 streams=1
Do we need a "label" to uniquely identify a particular 'app' instance? 
i.e. Similar to the optional label that draft-ietf-rtcweb-data-protocol 
uses.
e.g. a=sctpmap:5000 clue max-message-size=100000 streams=1 label1
a=sctpmap:5000 clue max-message-size=100000 streams=1 label2
a=sctpmap:5000 clue max-message-size=100000 streams=1 label3

Regards, Christian


On 14/02/2014 7:57 AM, Christer Holmberg wrote:
> Hi,
> I gave some comments off-line to Salvatore, which have been addressed 
> in the new version.
> Some additional comments. I do realize some are pure editorial, but 
> rather to write then down in some notes I put them directly into the 
> e-mail J
> *Section 1:*
> *------------*
> *Q1**(EDITORIAL)**:*
> The text says:
> "Since
> no-one so far has implemented SCTP over TLS, due to some serious
> limitations described in [RFC6083], this document does not make use
> of TLS over SCTP as described in RFC3436 [RFC3436]."
> I suggest to simply say something like:
> "As described in [RFC6083], there are limitations associated with 
> implementing
> SCTP over TLS. For that reason, SCTP over TLS is outside the scope of 
> this document."
> *Q2**(EDITORIAL)**:*
> The text says:
> "Additionally, this document specifies the use of the 'setup' and
> 'connection' SDP attributes to establish SCTP associations. These
> attributes were defined in RFC4145 [RFC4145] for TCP. This document
> discusses their use with SCTP."
> I suggest to remove the last sentence, as it only repeats what is said 
> in the first sentence.
> *Section 3:*
> *------------*
> *Q3 (EDITORIAL):*
> The text says:
> “This document defines three new values for the 'proto' field: 'SCTP', 
> 'SCTP/DTLS' and 'DTLS/SCTP'.”
> I would suggest to say:
> “This document defines three new *protocol identifier* values for the 
> 'proto' field: 'SCTP', 'SCTP/DTLS' and 'DTLS/SCTP'.”
> …because, you use “protocol identifier” terminology later.
> *Q4 (CLARIFICATION):*
> The text says:
> “An 'm' line that specifies 'SCTP' or 'SCTP/DTLS' or 'DTLS/SCTP' MUST
> further qualify the application-layer protocol using an fmt
> identifier.”
> I do understand what you are trying to say, but I think the “qualify” 
> word is a little confusing.
> Maybe something like:
> “An 'm' line that specifies 'SCTP' or 'SCTP/DTLS' or 'DTLS/SCTP' MUST
> further include an fmt value that identifies the application-layer 
> protocol
> carried on top of the SCTP association.”
> *Section 4.1:*
> *--------------*
> *Q5 (TECHNICAL):*
> The text says:
> “For the payload type assignments the "a=sctpmap:"
> attribute (see Section 5.1) 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.”
> It is very difficult for me to parse this, and figure out the 
> semantics of the sctpmap attribute.
> First, what do you mean by “media encoding name”?
> Second, when you talk about “application protocol”, I assume you refer 
> to the SCTP protocol?
> Third, the text say SHOULD. When is it ok to not use the sctpmap 
> attribute?
> *Section **5**:*
> *------------*
> *Q6 (EDITORIAL):*
> I suggest to rename section 5 to “SDP sctpmap attribute”, rename 
> section 5.1 to “General”, and add a new section 5.2 named “ABNF”.
> *Q7 (TECHNICAL):*
> The text says:
> “The sctpmap MUST include the app parameter indicating the application
> running on top of the association.”
> This is fine if the sctpmap attribute maps to an application. But, 
> what if the sctpmap attribute maps to a “media encoding” or “payload”?
> *Q8 **(EDITORIAL):*
> The text says:
> “The sctpmap line should also contain the max-message-size parameter
> indicating the maximum message size, in bytes, the endpoint is
> willing to accept.”
> I suggest to replace “accept” with “receive on the associated SCTP 
> association”.
> *Q9 (EDITORIAL):*
> The text says:
> “The peer should assume that larger message will be rejected by the
> endpoint, though it is up to the endpoint decide the appropriate
> behaviour.”
> I suggest to either remove this paragraph, OR replace it with a 
> sentence saying that a peer must/should not send messages bigger than 
> the indicated max value.
> *Q10 (CLARIFICATION):*
> The text says:
> “If the parameter is not present, the implementation should provide
> a default, with a suggested value of 64K.”
> I don’t understand “suggested”. If the default value is 64K the text 
> should specify (rather than suggest) that.
> The same comment applies to the stream parameter, described later in 
> the section.
> *Q11 (CLARIFICATION):*
> The text says:
> “It may also provide the stream parameter to specify the initial
> number of incoming streams to be supported by each side of the
> association.”
> First, to be consistent with previous text, I suggest to replace “It 
> may also provide” with “The sctpmap line may also contain”
> Second, I don’t understand “incoming streams to be supported by each 
> side of the association”. Doesn’t each peer indicate the number of 
> streams it is willing to receive? Then, we can always recommend that 
> the peer indicates the same number of streams, if possible.
> *Q12 (EDITORIAL):*
> The text that follows the ABNF should be put into a separate “SDP 
> Offer/Answer Procedures” section.
> *Section 7:*
> *------------*
> *Q13 (EDITORIAL):*
> Please don’t talk about re-INVITEs, or other SIP messages. Simply talk 
> about offers and answers.
> *Q14 (EDITORIAL):*
> The text says:
> “It is worth to underline that when using SCTP on top of DTLS,..”
> To have consistent terminology, please say “SCTP over DTLS”.
> That’s all for now! :)
> Regards,
> Christer
> -----Alkuperäinen viesti-----
> Lähettäjä: mmusic [mailto:mmusic-bounces@ietf.org] Puolesta 
> internet-drafts@ietf.org
> Lähetetty: 13. helmikuuta 2014 16:21
> Vastaanottaja: i-d-announce@ietf.org
> Kopio: mmusic@ietf.org
> Aihe: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-06.txt
> A New Internet-Draft is available from the on-line Internet-Drafts 
> directories.
> This draft is a work item of the Multiparty Multimedia Session Control 
> Working Group of the IETF.
> Title : Stream Control Transmission Protocol (SCTP)-Based Media 
> Transport in the Session Description Protocol (SDP)
> Authors : Salvatore Loreto
> Gonzalo Camarillo
> Filename : draft-ietf-mmusic-sctp-sdp-06.txt
> Pages : 14
> Date : 2014-02-13
> Abstract:
> SCTP (Stream Control Transmission Protocol) is a transport protocol
> used to establish associations between two endpoints. This document
> describes how to express media transport over SCTP in SDP (Session
> Description Protocol). This document defines the 'SCTP', 'SCTP/DTLS'
> and 'DTLS/SCTP' protocol identifiers for SDP.
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-mmusic-sctp-sdp/
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-06
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-sctp-sdp-06
> Please note that it may take a couple of minutes from the time of 
> submission until the htmlized version and diff are available at 
> tools.ietf.org.
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
> _______________________________________________
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>
>
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Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-06.txt - Christer's initial comments
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Hi Christian,

A comment, based on your example. I don't think we would indicate "clue" in=
 the sctpmap attribute. We would indicate "webrtc-datachannel", in order to=
 be able to interwork with browsers. Then, on that webrtc-datachannel, we w=
ould run the CLUE protocol (which we would negotiate e.g. using the DCP, Ri=
chard's draft, or some other mechanism).

Regards,

Christer

-----Original Message-----
From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of Christian Groves
Sent: 14. helmikuuta 2014 1:49
To: mmusic@ietf.org
Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-06.txt - Chris=
ter's initial comments

Hello,

Some further comments:

Section 11.1 (Editorial) - 11.1 The first paragraph regarding 'sctpmap'=20
attribute appears to be repeated.

Section 4.1 (Technical): I see the draft has removed the using using "bfcp"=
 and "t38" as values for the "App" field under one m-line. However that als=
o removes any hint about multiple a=3Dsctpmaps being used under one m-line.=
 For example should there be one a=3Dsctpmap per 'fmt' or could there be ze=
ro or more a=3Dsctpmaps per 'fmt' in the m-line? If think this should be cl=
arified.

General: There was a discussion on the CLUE mailing list whether there coul=
d be multiple CLUE protocol instances per SCTP association. We couldn't ide=
ntify a use case but couldn't rule it out either. I assume this would apply=
 to other "apps" also.
For example something like:
m=3Dapplication 54111 DTLS/SCTP 5000
c=3DIN IP4 79.97.215.79
a=3Dsctpmap:5000 clue max-message-size=3D100000 streams=3D1
a=3Dsctpmap:5000 clue max-message-size=3D100000 streams=3D1
a=3Dsctpmap:5000 clue max-message-size=3D100000 streams=3D1 Do we need a "l=
abel" to uniquely identify a particular 'app' instance?=20
i.e. Similar to the optional label that draft-ietf-rtcweb-data-protocol use=
s.
e.g. a=3Dsctpmap:5000 clue max-message-size=3D100000 streams=3D1 label1
a=3Dsctpmap:5000 clue max-message-size=3D100000 streams=3D1 label2
a=3Dsctpmap:5000 clue max-message-size=3D100000 streams=3D1 label3

Regards, Christian


On 14/02/2014 7:57 AM, Christer Holmberg wrote:
> Hi,
> I gave some comments off-line to Salvatore, which have been addressed=20
> in the new version.
> Some additional comments. I do realize some are pure editorial, but=20
> rather to write then down in some notes I put them directly into the=20
> e-mail J *Section 1:*
> *------------*
> *Q1**(EDITORIAL)**:*
> The text says:
> "Since
> no-one so far has implemented SCTP over TLS, due to some serious=20
> limitations described in [RFC6083], this document does not make use of=20
> TLS over SCTP as described in RFC3436 [RFC3436]."
> I suggest to simply say something like:
> "As described in [RFC6083], there are limitations associated with=20
> implementing SCTP over TLS. For that reason, SCTP over TLS is outside=20
> the scope of this document."
> *Q2**(EDITORIAL)**:*
> The text says:
> "Additionally, this document specifies the use of the 'setup' and=20
> 'connection' SDP attributes to establish SCTP associations. These=20
> attributes were defined in RFC4145 [RFC4145] for TCP. This document=20
> discusses their use with SCTP."
> I suggest to remove the last sentence, as it only repeats what is said=20
> in the first sentence.
> *Section 3:*
> *------------*
> *Q3 (EDITORIAL):*
> The text says:
> "This document defines three new values for the 'proto' field: 'SCTP',=20
> 'SCTP/DTLS' and 'DTLS/SCTP'."
> I would suggest to say:
> "This document defines three new *protocol identifier* values for the=20
> 'proto' field: 'SCTP', 'SCTP/DTLS' and 'DTLS/SCTP'."
> .because, you use "protocol identifier" terminology later.
> *Q4 (CLARIFICATION):*
> The text says:
> "An 'm' line that specifies 'SCTP' or 'SCTP/DTLS' or 'DTLS/SCTP' MUST=20
> further qualify the application-layer protocol using an fmt=20
> identifier."
> I do understand what you are trying to say, but I think the "qualify"=20
> word is a little confusing.
> Maybe something like:
> "An 'm' line that specifies 'SCTP' or 'SCTP/DTLS' or 'DTLS/SCTP' MUST=20
> further include an fmt value that identifies the application-layer=20
> protocol carried on top of the SCTP association."
> *Section 4.1:*
> *--------------*
> *Q5 (TECHNICAL):*
> The text says:
> "For the payload type assignments the "a=3Dsctpmap:"
> attribute (see Section 5.1) SHOULD be used to map from a port number=20
> to a media encoding name that identifies the payload format=20
> transported by the association or the actual application protocol=20
> running on top of it."
> It is very difficult for me to parse this, and figure out the=20
> semantics of the sctpmap attribute.
> First, what do you mean by "media encoding name"?
> Second, when you talk about "application protocol", I assume you refer=20
> to the SCTP protocol?
> Third, the text say SHOULD. When is it ok to not use the sctpmap=20
> attribute?
> *Section **5**:*
> *------------*
> *Q6 (EDITORIAL):*
> I suggest to rename section 5 to "SDP sctpmap attribute", rename=20
> section 5.1 to "General", and add a new section 5.2 named "ABNF".
> *Q7 (TECHNICAL):*
> The text says:
> "The sctpmap MUST include the app parameter indicating the application=20
> running on top of the association."
> This is fine if the sctpmap attribute maps to an application. But,=20
> what if the sctpmap attribute maps to a "media encoding" or "payload"?
> *Q8 **(EDITORIAL):*
> The text says:
> "The sctpmap line should also contain the max-message-size parameter=20
> indicating the maximum message size, in bytes, the endpoint is willing=20
> to accept."
> I suggest to replace "accept" with "receive on the associated SCTP=20
> association".
> *Q9 (EDITORIAL):*
> The text says:
> "The peer should assume that larger message will be rejected by the=20
> endpoint, though it is up to the endpoint decide the appropriate=20
> behaviour."
> I suggest to either remove this paragraph, OR replace it with a=20
> sentence saying that a peer must/should not send messages bigger than=20
> the indicated max value.
> *Q10 (CLARIFICATION):*
> The text says:
> "If the parameter is not present, the implementation should provide a=20
> default, with a suggested value of 64K."
> I don't understand "suggested". If the default value is 64K the text=20
> should specify (rather than suggest) that.
> The same comment applies to the stream parameter, described later in=20
> the section.
> *Q11 (CLARIFICATION):*
> The text says:
> "It may also provide the stream parameter to specify the initial=20
> number of incoming streams to be supported by each side of the=20
> association."
> First, to be consistent with previous text, I suggest to replace "It=20
> may also provide" with "The sctpmap line may also contain"
> Second, I don't understand "incoming streams to be supported by each=20
> side of the association". Doesn't each peer indicate the number of=20
> streams it is willing to receive? Then, we can always recommend that=20
> the peer indicates the same number of streams, if possible.
> *Q12 (EDITORIAL):*
> The text that follows the ABNF should be put into a separate "SDP=20
> Offer/Answer Procedures" section.
> *Section 7:*
> *------------*
> *Q13 (EDITORIAL):*
> Please don't talk about re-INVITEs, or other SIP messages. Simply talk=20
> about offers and answers.
> *Q14 (EDITORIAL):*
> The text says:
> "It is worth to underline that when using SCTP on top of DTLS,.."
> To have consistent terminology, please say "SCTP over DTLS".
> That's all for now! :)
> Regards,
> Christer
> -----Alkuper=E4inen viesti-----
> L=E4hett=E4j=E4: mmusic [mailto:mmusic-bounces@ietf.org] Puolesta=20
> internet-drafts@ietf.org
> L=E4hetetty: 13. helmikuuta 2014 16:21
> Vastaanottaja: i-d-announce@ietf.org
> Kopio: mmusic@ietf.org
> Aihe: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-06.txt A New=20
> Internet-Draft is available from the on-line Internet-Drafts=20
> directories.
> This draft is a work item of the Multiparty Multimedia Session Control=20
> Working Group of the IETF.
> Title : Stream Control Transmission Protocol (SCTP)-Based Media=20
> Transport in the Session Description Protocol (SDP) Authors :=20
> Salvatore Loreto Gonzalo Camarillo Filename :=20
> draft-ietf-mmusic-sctp-sdp-06.txt Pages : 14 Date : 2014-02-13
> Abstract:
> SCTP (Stream Control Transmission Protocol) is a transport protocol=20
> used to establish associations between two endpoints. This document=20
> describes how to express media transport over SCTP in SDP (Session=20
> Description Protocol). This document defines the 'SCTP', 'SCTP/DTLS'
> and 'DTLS/SCTP' protocol identifiers for SDP.
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-mmusic-sctp-sdp/
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-06
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sctp-sdp-06
> Please note that it may take a couple of minutes from the time of=20
> submission until the htmlized version and diff are available at=20
> tools.ietf.org.
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org <mailto:mmusic@ietf.org>=20
> https://www.ietf.org/mailman/listinfo/mmusic
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

_______________________________________________
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Multiparty Multimedia Session Control Working Group of the IETF.

        Title           : WebRTC MediaStream Identification in the Session Description Protocol
        Author          : Harald Alvestrand
	Filename        : draft-ietf-mmusic-msid-04.txt
	Pages           : 17
	Date            : 2014-02-13

Abstract:
   This document specifies a grouping mechanism for RTP media streams
   that can be used to specify relations between media streams.

   This mechanism is used to signal the association between the SDP
   concept of "m-line" and the WebRTC concept of "MediaStream" /
   "MediaStreamTrack" using SDP signaling.

   This document is a work item of the MMUSIC WG, whose discussion list
   is mmusic@ietf.org.



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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-msid-04

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-msid-04


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

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


From nobody Thu Feb 13 23:11:05 2014
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I have submitted a new version of the MSID draft, -04.

It tries to make it crystal clear that the msid-semantic:wms is only
intended for signalling MediaStreamTracks and MediaStreams.

It also adds a new functionality: msid-control, which can be used by a
receiver of MediaStreamTracks to signal the sender of a MediaStreamTrack
what the receiver wishes with regard to that MediaStreamTrack; this need
has been identified within the WebRTC and RTCWEB WGs.

It seemed logical to extend this draft with that functionality, since it
is closely allied functionality.

Comments are welcome.

          Harald

-- 
Surveillance is pervasive. Go Dark.


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Subject: Re: [MMUSIC] MSID and AppID: solving different problems
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On 02/13/2014 11:43 PM, Suhas Nandakumar wrote:
>
>
>
> On Thu, Feb 13, 2014 at 10:09 AM, Jonathan Lennox <jonathan@vidyo.com
> <mailto:jonathan@vidyo.com>> wrote:
>
>     The chairs asked that the authors of the MSID and AppID drafts
>     discuss the relationship between the two documents.
>
>     My personal opinion (which I mentioned in the RTCWeb Rejecting
>     MediaStreamTracks thread, but folks might not have seen) is that
>     they're solving different problems, and so are non-conflicting.
>
>     MSID provides an identifier for SDP m= lines.  Its primary use
>     case is providing a Javascript-level handle in WebRTC.
>
>     AppID -- which needs a better name, suggestions welcome --
>     provides an indirection label for identifying individual RTP
>     packet streams in SDP, in scenarios where the SSRC is unavailable
>     to the signaling.
>

I am certainly happy with drawing a distinction along these lines, and
let each find its own user base.

>
> [Suhas] How about the name SSID (Source Stream ID) to identify the
> individual RTP packet streams

I would advise against using such a frequently used acronym.
If inventing a new one, what about RTP Packet Stream Correlation Tag -
RPSCT?
I don't think I've encountered that in the IETF yet....



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<html>
  <head>
    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    <div class="moz-cite-prefix">On 02/13/2014 11:43 PM, Suhas
      Nandakumar wrote:<br>
    </div>
    <blockquote
cite="mid:CAMRcRGT-KVsMpSGcOYkx9Fop90Nrm3FQmTVms6RZCAm3AS0hNg@mail.gmail.com"
      type="cite">
      <div dir="ltr"><br>
        <div class="gmail_extra"><br>
          <br>
          <div class="gmail_quote">On Thu, Feb 13, 2014 at 10:09 AM,
            Jonathan Lennox <span dir="ltr">&lt;<a
                moz-do-not-send="true" href="mailto:jonathan@vidyo.com"
                target="_blank">jonathan@vidyo.com</a>&gt;</span> wrote:<br>
            <blockquote class="gmail_quote" style="margin:0 0 0
              .8ex;border-left:1px #ccc solid;padding-left:1ex">The
              chairs asked that the authors of the MSID and AppID drafts
              discuss the relationship between the two documents.<br>
              <br>
              My personal opinion (which I mentioned in the RTCWeb
              Rejecting MediaStreamTracks thread, but folks might not
              have seen) is that they're solving different problems, and
              so are non-conflicting.<br>
            </blockquote>
            <blockquote class="gmail_quote" style="margin:0 0 0
              .8ex;border-left:1px #ccc solid;padding-left:1ex">
              MSID provides an identifier for SDP m= lines. &nbsp;Its primary
              use case is providing a Javascript-level handle in WebRTC.<br>
              <br>
              AppID -- which needs a better name, suggestions welcome --
              provides an indirection label for identifying individual
              RTP packet streams in SDP, in scenarios where the SSRC is
              unavailable to the signaling.<br>
            </blockquote>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
    I am certainly happy with drawing a distinction along these lines,
    and let each find its own user base.<br>
    <br>
    <blockquote
cite="mid:CAMRcRGT-KVsMpSGcOYkx9Fop90Nrm3FQmTVms6RZCAm3AS0hNg@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div class="gmail_extra">
          <div class="gmail_quote">
            <div><br>
            </div>
            <div>[Suhas] How about the name SSID (Source Stream ID) to
              identify the individual RTP packet streams</div>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
    I would advise against using such a frequently used acronym.<br>
    If inventing a new one, what about RTP Packet Stream Correlation Tag
    - RPSCT?<br>
    I don't think I've encountered that in the IETF yet....<br>
    <br>
    <br>
  </body>
</html>

--------------060006040203020604020403--


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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Multiparty Multimedia Session Control Working Group of the IETF.

        Title           : UDP Transport Layer (UDPTL) over Datagram Transport Layer Security (DTLS)
        Authors         : Christer Holmberg
                          Ivo Sedlacek
                          Gonzalo Salgueiro
	Filename        : draft-ietf-mmusic-udptl-dtls-05.txt
	Pages           : 23
	Date            : 2014-02-13

Abstract:
   This document specifies how the UDP Transport Layer (UDPTL) protocol,
   the predominant transport protocol for T.38 fax, can be transported
   over the Datagram Transport Layer Security (DTLS) protocol, how the
   usage of UDPTL over DTLS is indicated in the Session Description
   Protocol (SDP), and how UDPTL over DTLS is negotiated in a session
   established using the Session Initiation Protocol (SIP).


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-udptl-dtls-05

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-udptl-dtls-05


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

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


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: Draft new version: draft-ietf-mmusic-udptl-dtls-05
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Hi,

Based on the review comments by Flemming, I've submitted a new version of t=
he udptl-dtls draft.

Regards,

Christer

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<p class=3D"MsoNormal">Hi,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Based on the review comments by Flemming, I&#8217;ve=
 submitted a new version of the udptl-dtls draft.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Christer<o:p></o:p></p>
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#13: This is a test ticket for 4566bis

 This is a test, please ignore.

-- 
-----------------------------+--------------------------------
Reporter:  abegen@cisco.com  |      Owner:  abegen@cisco.com
    Type:  defect            |     Status:  new
Priority:  trivial           |  Component:  rfc4566bis
 Version:  2.0               |   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: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-06.txt - Christer's initial comments
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Hi,

>> *Q4 (CLARIFICATION):*
>> The text says:
>>     "An 'm' line that specifies 'SCTP' or 'SCTP/DTLS' or 'DTLS/SCTP' MUS=
T
>>     further qualify the application-layer protocol using an fmt
>>     identifier."
>> I do understand what you are trying to say, but I think the "qualify"
>> word is a little confusing.
>> Maybe something like:
>>     "An 'm' line that specifies 'SCTP' or 'SCTP/DTLS' or 'DTLS/SCTP' MUS=
T
>>     further include an fmt value that identifies the application-layer=20
>> protocol
>>     carried on top of the SCTP association."
>
> The clarification would be fine, but both the original and the clarificat=
ion are misleading. The fmt value does not identify an application-layer pr=
otocol - it identifies a port number. This is explained later in section 4.=
1.

Correct. The SDP sctpmap attribute then associates that port number with an=
 SCTP application.

Regards,

Christer


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Archived-At: http://mailarchive.ietf.org/arch/msg/mmusic/63HetNemVagzDtTkzBicGfSi4Ts
Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-06.txt - Christer's initial comments
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Hello Christer,

I'm not assuming the use of webrtc-data channel. The issue would apply 
to any "app" being used multiple times in an SCTP association.

Are you suggesting any time that you want to use multiple instances of 
an "app" that you must use "webrtc-datachannel" even if there are no web 
browsers involved?

Regards, Christian

On 14/02/2014 5:56 PM, Christer Holmberg wrote:
> Hi Christian,
>
> A comment, based on your example. I don't think we would indicate "clue" in the sctpmap attribute. We would indicate "webrtc-datachannel", in order to be able to interwork with browsers. Then, on that webrtc-datachannel, we would run the CLUE protocol (which we would negotiate e.g. using the DCP, Richard's draft, or some other mechanism).
>
> Regards,
>
> Christer
>
> -----Original Message-----
> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of Christian Groves
> Sent: 14. helmikuuta 2014 1:49
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-06.txt - Christer's initial comments
>
> Hello,
>
> Some further comments:
>
> Section 11.1 (Editorial) - 11.1 The first paragraph regarding 'sctpmap'
> attribute appears to be repeated.
>
> Section 4.1 (Technical): I see the draft has removed the using using "bfcp" and "t38" as values for the "App" field under one m-line. However that also removes any hint about multiple a=sctpmaps being used under one m-line. For example should there be one a=sctpmap per 'fmt' or could there be zero or more a=sctpmaps per 'fmt' in the m-line? If think this should be clarified.
>
> General: There was a discussion on the CLUE mailing list whether there could be multiple CLUE protocol instances per SCTP association. We couldn't identify a use case but couldn't rule it out either. I assume this would apply to other "apps" also.
> For example something like:
> m=application 54111 DTLS/SCTP 5000
> c=IN IP4 79.97.215.79
> a=sctpmap:5000 clue max-message-size=100000 streams=1
> a=sctpmap:5000 clue max-message-size=100000 streams=1
> a=sctpmap:5000 clue max-message-size=100000 streams=1 Do we need a "label" to uniquely identify a particular 'app' instance?
> i.e. Similar to the optional label that draft-ietf-rtcweb-data-protocol uses.
> e.g. a=sctpmap:5000 clue max-message-size=100000 streams=1 label1
> a=sctpmap:5000 clue max-message-size=100000 streams=1 label2
> a=sctpmap:5000 clue max-message-size=100000 streams=1 label3
>
> Regards, Christian
>
>
> On 14/02/2014 7:57 AM, Christer Holmberg wrote:
>> Hi,
>> I gave some comments off-line to Salvatore, which have been addressed
>> in the new version.
>> Some additional comments. I do realize some are pure editorial, but
>> rather to write then down in some notes I put them directly into the
>> e-mail J *Section 1:*
>> *------------*
>> *Q1**(EDITORIAL)**:*
>> The text says:
>> "Since
>> no-one so far has implemented SCTP over TLS, due to some serious
>> limitations described in [RFC6083], this document does not make use of
>> TLS over SCTP as described in RFC3436 [RFC3436]."
>> I suggest to simply say something like:
>> "As described in [RFC6083], there are limitations associated with
>> implementing SCTP over TLS. For that reason, SCTP over TLS is outside
>> the scope of this document."
>> *Q2**(EDITORIAL)**:*
>> The text says:
>> "Additionally, this document specifies the use of the 'setup' and
>> 'connection' SDP attributes to establish SCTP associations. These
>> attributes were defined in RFC4145 [RFC4145] for TCP. This document
>> discusses their use with SCTP."
>> I suggest to remove the last sentence, as it only repeats what is said
>> in the first sentence.
>> *Section 3:*
>> *------------*
>> *Q3 (EDITORIAL):*
>> The text says:
>> "This document defines three new values for the 'proto' field: 'SCTP',
>> 'SCTP/DTLS' and 'DTLS/SCTP'."
>> I would suggest to say:
>> "This document defines three new *protocol identifier* values for the
>> 'proto' field: 'SCTP', 'SCTP/DTLS' and 'DTLS/SCTP'."
>> .because, you use "protocol identifier" terminology later.
>> *Q4 (CLARIFICATION):*
>> The text says:
>> "An 'm' line that specifies 'SCTP' or 'SCTP/DTLS' or 'DTLS/SCTP' MUST
>> further qualify the application-layer protocol using an fmt
>> identifier."
>> I do understand what you are trying to say, but I think the "qualify"
>> word is a little confusing.
>> Maybe something like:
>> "An 'm' line that specifies 'SCTP' or 'SCTP/DTLS' or 'DTLS/SCTP' MUST
>> further include an fmt value that identifies the application-layer
>> protocol carried on top of the SCTP association."
>> *Section 4.1:*
>> *--------------*
>> *Q5 (TECHNICAL):*
>> The text says:
>> "For the payload type assignments the "a=sctpmap:"
>> attribute (see Section 5.1) 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."
>> It is very difficult for me to parse this, and figure out the
>> semantics of the sctpmap attribute.
>> First, what do you mean by "media encoding name"?
>> Second, when you talk about "application protocol", I assume you refer
>> to the SCTP protocol?
>> Third, the text say SHOULD. When is it ok to not use the sctpmap
>> attribute?
>> *Section **5**:*
>> *------------*
>> *Q6 (EDITORIAL):*
>> I suggest to rename section 5 to "SDP sctpmap attribute", rename
>> section 5.1 to "General", and add a new section 5.2 named "ABNF".
>> *Q7 (TECHNICAL):*
>> The text says:
>> "The sctpmap MUST include the app parameter indicating the application
>> running on top of the association."
>> This is fine if the sctpmap attribute maps to an application. But,
>> what if the sctpmap attribute maps to a "media encoding" or "payload"?
>> *Q8 **(EDITORIAL):*
>> The text says:
>> "The sctpmap line should also contain the max-message-size parameter
>> indicating the maximum message size, in bytes, the endpoint is willing
>> to accept."
>> I suggest to replace "accept" with "receive on the associated SCTP
>> association".
>> *Q9 (EDITORIAL):*
>> The text says:
>> "The peer should assume that larger message will be rejected by the
>> endpoint, though it is up to the endpoint decide the appropriate
>> behaviour."
>> I suggest to either remove this paragraph, OR replace it with a
>> sentence saying that a peer must/should not send messages bigger than
>> the indicated max value.
>> *Q10 (CLARIFICATION):*
>> The text says:
>> "If the parameter is not present, the implementation should provide a
>> default, with a suggested value of 64K."
>> I don't understand "suggested". If the default value is 64K the text
>> should specify (rather than suggest) that.
>> The same comment applies to the stream parameter, described later in
>> the section.
>> *Q11 (CLARIFICATION):*
>> The text says:
>> "It may also provide the stream parameter to specify the initial
>> number of incoming streams to be supported by each side of the
>> association."
>> First, to be consistent with previous text, I suggest to replace "It
>> may also provide" with "The sctpmap line may also contain"
>> Second, I don't understand "incoming streams to be supported by each
>> side of the association". Doesn't each peer indicate the number of
>> streams it is willing to receive? Then, we can always recommend that
>> the peer indicates the same number of streams, if possible.
>> *Q12 (EDITORIAL):*
>> The text that follows the ABNF should be put into a separate "SDP
>> Offer/Answer Procedures" section.
>> *Section 7:*
>> *------------*
>> *Q13 (EDITORIAL):*
>> Please don't talk about re-INVITEs, or other SIP messages. Simply talk
>> about offers and answers.
>> *Q14 (EDITORIAL):*
>> The text says:
>> "It is worth to underline that when using SCTP on top of DTLS,.."
>> To have consistent terminology, please say "SCTP over DTLS".
>> That's all for now! :)
>> Regards,
>> Christer
>> -----Alkuperäinen viesti-----
>> Lähettäjä: mmusic [mailto:mmusic-bounces@ietf.org] Puolesta
>> internet-drafts@ietf.org
>> Lähetetty: 13. helmikuuta 2014 16:21
>> Vastaanottaja: i-d-announce@ietf.org
>> Kopio: mmusic@ietf.org
>> Aihe: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-06.txt A New
>> Internet-Draft is available from the on-line Internet-Drafts
>> directories.
>> This draft is a work item of the Multiparty Multimedia Session Control
>> Working Group of the IETF.
>> Title : Stream Control Transmission Protocol (SCTP)-Based Media
>> Transport in the Session Description Protocol (SDP) Authors :
>> Salvatore Loreto Gonzalo Camarillo Filename :
>> draft-ietf-mmusic-sctp-sdp-06.txt Pages : 14 Date : 2014-02-13
>> Abstract:
>> SCTP (Stream Control Transmission Protocol) is a transport protocol
>> used to establish associations between two endpoints. This document
>> describes how to express media transport over SCTP in SDP (Session
>> Description Protocol). This document defines the 'SCTP', 'SCTP/DTLS'
>> and 'DTLS/SCTP' protocol identifiers for SDP.
>> The IETF datatracker status page for this draft is:
>> https://datatracker.ietf.org/doc/draft-ietf-mmusic-sctp-sdp/
>> There's also a htmlized version available at:
>> http://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-06
>> A diff from the previous version is available at:
>> http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-sctp-sdp-06
>> Please note that it may take a couple of minutes from the time of
>> submission until the htmlized version and diff are available at
>> tools.ietf.org.
>> Internet-Drafts are also available by anonymous FTP at:
>> ftp://ftp.ietf.org/internet-drafts/
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org <mailto:mmusic@ietf.org>
>> https://www.ietf.org/mailman/listinfo/mmusic
>>
>>
>> _______________________________________________
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>> https://www.ietf.org/mailman/listinfo/mmusic
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Thread-Topic: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-06.txt - Christer's initial comments
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Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-06.txt - Christer's initial comments
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Hi Christian,

> I'm not assuming the use of webrtc-data channel. The issue would apply to=
 any "app" being used multiple times in an SCTP association.

Correct.=20

My comment was not on multiple apps per SCTP association (which I agree was=
 the main focus of your e-mail), but simply on the example :)

> Are you suggesting any time that you want to use multiple instances of an=
 "app" that you must use "webrtc-datachannel" even if there are no web brow=
sers involved?

No. Only if you want interoperability with a browser.

BUT, even if there are no browser involved, you can still use the webrtc-da=
tachannel if you want. The name should probably be changed to something mor=
e generic, though.

Regards,

Christer


On 14/02/2014 5:56 PM, Christer Holmberg wrote:
> Hi Christian,
>
> A comment, based on your example. I don't think we would indicate "clue" =
in the sctpmap attribute. We would indicate "webrtc-datachannel", in order =
to be able to interwork with browsers. Then, on that webrtc-datachannel, we=
 would run the CLUE protocol (which we would negotiate e.g. using the DCP, =
Richard's draft, or some other mechanism).
>
> Regards,
>
> Christer
>
> -----Original Message-----
> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of Christian=20
> Groves
> Sent: 14. helmikuuta 2014 1:49
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-06.txt -=20
> Christer's initial comments
>
> Hello,
>
> Some further comments:
>
> Section 11.1 (Editorial) - 11.1 The first paragraph regarding 'sctpmap'
> attribute appears to be repeated.
>
> Section 4.1 (Technical): I see the draft has removed the using using "bfc=
p" and "t38" as values for the "App" field under one m-line. However that a=
lso removes any hint about multiple a=3Dsctpmaps being used under one m-lin=
e. For example should there be one a=3Dsctpmap per 'fmt' or could there be =
zero or more a=3Dsctpmaps per 'fmt' in the m-line? If think this should be =
clarified.
>
> General: There was a discussion on the CLUE mailing list whether there co=
uld be multiple CLUE protocol instances per SCTP association. We couldn't i=
dentify a use case but couldn't rule it out either. I assume this would app=
ly to other "apps" also.
> For example something like:
> m=3Dapplication 54111 DTLS/SCTP 5000
> c=3DIN IP4 79.97.215.79
> a=3Dsctpmap:5000 clue max-message-size=3D100000 streams=3D1
> a=3Dsctpmap:5000 clue max-message-size=3D100000 streams=3D1
> a=3Dsctpmap:5000 clue max-message-size=3D100000 streams=3D1 Do we need a =
"label" to uniquely identify a particular 'app' instance?
> i.e. Similar to the optional label that draft-ietf-rtcweb-data-protocol u=
ses.
> e.g. a=3Dsctpmap:5000 clue max-message-size=3D100000 streams=3D1 label1
> a=3Dsctpmap:5000 clue max-message-size=3D100000 streams=3D1 label2
> a=3Dsctpmap:5000 clue max-message-size=3D100000 streams=3D1 label3
>
> Regards, Christian
>
>
> On 14/02/2014 7:57 AM, Christer Holmberg wrote:
>> Hi,
>> I gave some comments off-line to Salvatore, which have been addressed=20
>> in the new version.
>> Some additional comments. I do realize some are pure editorial, but=20
>> rather to write then down in some notes I put them directly into the=20
>> e-mail J *Section 1:*
>> *------------*
>> *Q1**(EDITORIAL)**:*
>> The text says:
>> "Since
>> no-one so far has implemented SCTP over TLS, due to some serious=20
>> limitations described in [RFC6083], this document does not make use=20
>> of TLS over SCTP as described in RFC3436 [RFC3436]."
>> I suggest to simply say something like:
>> "As described in [RFC6083], there are limitations associated with=20
>> implementing SCTP over TLS. For that reason, SCTP over TLS is outside=20
>> the scope of this document."
>> *Q2**(EDITORIAL)**:*
>> The text says:
>> "Additionally, this document specifies the use of the 'setup' and=20
>> 'connection' SDP attributes to establish SCTP associations. These=20
>> attributes were defined in RFC4145 [RFC4145] for TCP. This document=20
>> discusses their use with SCTP."
>> I suggest to remove the last sentence, as it only repeats what is=20
>> said in the first sentence.
>> *Section 3:*
>> *------------*
>> *Q3 (EDITORIAL):*
>> The text says:
>> "This document defines three new values for the 'proto' field:=20
>> 'SCTP', 'SCTP/DTLS' and 'DTLS/SCTP'."
>> I would suggest to say:
>> "This document defines three new *protocol identifier* values for the=20
>> 'proto' field: 'SCTP', 'SCTP/DTLS' and 'DTLS/SCTP'."
>> .because, you use "protocol identifier" terminology later.
>> *Q4 (CLARIFICATION):*
>> The text says:
>> "An 'm' line that specifies 'SCTP' or 'SCTP/DTLS' or 'DTLS/SCTP' MUST=20
>> further qualify the application-layer protocol using an fmt=20
>> identifier."
>> I do understand what you are trying to say, but I think the "qualify"
>> word is a little confusing.
>> Maybe something like:
>> "An 'm' line that specifies 'SCTP' or 'SCTP/DTLS' or 'DTLS/SCTP' MUST=20
>> further include an fmt value that identifies the application-layer=20
>> protocol carried on top of the SCTP association."
>> *Section 4.1:*
>> *--------------*
>> *Q5 (TECHNICAL):*
>> The text says:
>> "For the payload type assignments the "a=3Dsctpmap:"
>> attribute (see Section 5.1) SHOULD be used to map from a port number=20
>> to a media encoding name that identifies the payload format=20
>> transported by the association or the actual application protocol=20
>> running on top of it."
>> It is very difficult for me to parse this, and figure out the=20
>> semantics of the sctpmap attribute.
>> First, what do you mean by "media encoding name"?
>> Second, when you talk about "application protocol", I assume you=20
>> refer to the SCTP protocol?
>> Third, the text say SHOULD. When is it ok to not use the sctpmap=20
>> attribute?
>> *Section **5**:*
>> *------------*
>> *Q6 (EDITORIAL):*
>> I suggest to rename section 5 to "SDP sctpmap attribute", rename=20
>> section 5.1 to "General", and add a new section 5.2 named "ABNF".
>> *Q7 (TECHNICAL):*
>> The text says:
>> "The sctpmap MUST include the app parameter indicating the=20
>> application running on top of the association."
>> This is fine if the sctpmap attribute maps to an application. But,=20
>> what if the sctpmap attribute maps to a "media encoding" or "payload"?
>> *Q8 **(EDITORIAL):*
>> The text says:
>> "The sctpmap line should also contain the max-message-size parameter=20
>> indicating the maximum message size, in bytes, the endpoint is=20
>> willing to accept."
>> I suggest to replace "accept" with "receive on the associated SCTP=20
>> association".
>> *Q9 (EDITORIAL):*
>> The text says:
>> "The peer should assume that larger message will be rejected by the=20
>> endpoint, though it is up to the endpoint decide the appropriate=20
>> behaviour."
>> I suggest to either remove this paragraph, OR replace it with a=20
>> sentence saying that a peer must/should not send messages bigger than=20
>> the indicated max value.
>> *Q10 (CLARIFICATION):*
>> The text says:
>> "If the parameter is not present, the implementation should provide a=20
>> default, with a suggested value of 64K."
>> I don't understand "suggested". If the default value is 64K the text=20
>> should specify (rather than suggest) that.
>> The same comment applies to the stream parameter, described later in=20
>> the section.
>> *Q11 (CLARIFICATION):*
>> The text says:
>> "It may also provide the stream parameter to specify the initial=20
>> number of incoming streams to be supported by each side of the=20
>> association."
>> First, to be consistent with previous text, I suggest to replace "It=20
>> may also provide" with "The sctpmap line may also contain"
>> Second, I don't understand "incoming streams to be supported by each=20
>> side of the association". Doesn't each peer indicate the number of=20
>> streams it is willing to receive? Then, we can always recommend that=20
>> the peer indicates the same number of streams, if possible.
>> *Q12 (EDITORIAL):*
>> The text that follows the ABNF should be put into a separate "SDP=20
>> Offer/Answer Procedures" section.
>> *Section 7:*
>> *------------*
>> *Q13 (EDITORIAL):*
>> Please don't talk about re-INVITEs, or other SIP messages. Simply=20
>> talk about offers and answers.
>> *Q14 (EDITORIAL):*
>> The text says:
>> "It is worth to underline that when using SCTP on top of DTLS,.."
>> To have consistent terminology, please say "SCTP over DTLS".
>> That's all for now! :)
>> Regards,
>> Christer
>> -----Alkuper=E4inen viesti-----
>> L=E4hett=E4j=E4: mmusic [mailto:mmusic-bounces@ietf.org] Puolesta=20
>> internet-drafts@ietf.org
>> L=E4hetetty: 13. helmikuuta 2014 16:21
>> Vastaanottaja: i-d-announce@ietf.org
>> Kopio: mmusic@ietf.org
>> Aihe: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-06.txt A New=20
>> Internet-Draft is available from the on-line Internet-Drafts=20
>> directories.
>> This draft is a work item of the Multiparty Multimedia Session=20
>> Control Working Group of the IETF.
>> Title : Stream Control Transmission Protocol (SCTP)-Based Media=20
>> Transport in the Session Description Protocol (SDP) Authors :
>> Salvatore Loreto Gonzalo Camarillo Filename :
>> draft-ietf-mmusic-sctp-sdp-06.txt Pages : 14 Date : 2014-02-13
>> Abstract:
>> SCTP (Stream Control Transmission Protocol) is a transport protocol=20
>> used to establish associations between two endpoints. This document=20
>> describes how to express media transport over SCTP in SDP (Session=20
>> Description Protocol). This document defines the 'SCTP', 'SCTP/DTLS'
>> and 'DTLS/SCTP' protocol identifiers for SDP.
>> The IETF datatracker status page for this draft is:
>> https://datatracker.ietf.org/doc/draft-ietf-mmusic-sctp-sdp/
>> There's also a htmlized version available at:
>> http://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-06
>> A diff from the previous version is available at:
>> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sctp-sdp-06
>> Please note that it may take a couple of minutes from the time of=20
>> submission until the htmlized version and diff are available at=20
>> tools.ietf.org.
>> Internet-Drafts are also available by anonymous FTP at:
>> ftp://ftp.ietf.org/internet-drafts/
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org <mailto:mmusic@ietf.org>=20
>> 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
>


From nobody Fri Feb 14 01:36:49 2014
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From: "Pal Martinsen (palmarti)" <palmarti@cisco.com>
To: =?Windows-1252?Q?Ari_Ker=E4nen?= <ari.keranen@ericsson.com>
Thread-Topic: [MMUSIC] New Version Notification for draft-reddy-mmusic-ice-happy-eyeballs-04.txt
Thread-Index: AQHPEFUUlb4N8W3Wl06M5kiJnPk5wZqUXPIAgAZGcwCAAv8DgIABtqYAgBS4XgCAAQMCgA==
Date: Fri, 14 Feb 2014 09:36:41 +0000
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References: <20131209121243.31704.62698.idtracker@ietfa.amsl.com> <540B748E-8B51-4763-B75F-F80ACCFB53D7@cisco.com> <52D01911.8030804@ericsson.com> <647464EE-B8E0-47FA-A375-8C758046D854@cisco.com> <B6CF90AA-44E7-4D38-81CA-8A85370944A7@ericsson.com> <7C0E6D63-D816-4415-B621-73A0309C6AB2@cisco.com> <52EA38C9.4070003@ericsson.com> <37A6AA83-867F-4DD1-A79D-081ECFA23B22@cisco.com> <52FD0A60.2040909@ericsson.com>
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Cc: mmusic <mmusic@ietf.org>, "Tirumaleswar Reddy \(tireddy\)" <tireddy@cisco.com>
Subject: Re: [MMUSIC] New Version Notification for draft-reddy-mmusic-ice-happy-eyeballs-04.txt
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On 13 Feb 2014, at 19:09 pm, Ari Ker=E4nen <ari.keranen@ericsson.com> wrote=
:

> On 31/01/14 15:44, Pal Martinsen (palmarti) wrote:
>> On 30 Jan 2014, at 12:34 pm, Ari Ker=E4nen <ari.keranen@ericsson.com> wr=
ote:
>>> On 28/01/14 15:49, Pal Martinsen (palmarti) wrote:
> [...]
>>> What I would find helpful for discussion is that if you could provide a=
 step-by-step example of how the candidate lists are formed and connectivit=
y checks done in cases where a) both support this with same promo algo b) b=
oth support this but with different promo algo c) only one endpoint support=
s this.
>>>=20
>>=20
>> As I do believe in running code, I wrote a simple implementation of the =
algorithm described below.
>=20
> That's very nice, but unfortunately that didn't compile on my Mac so I=20
> couldn't give it a try.

Runs fine on my mac. (Famous last words..) But I have ports installed with =
gcc and the usual *NIX toolchain.
Have not yet figured out to create a x-code project. If time permits..

I should probably have written it in phyton, but my skills in that language=
 is limited.
=20
>=20
>>> Also maybe a recommended promo algo would make sense so that there is a=
t least a fair chance that both will use the same algo/values and get most =
out of this.
>>>=20
>> First shot an an algorithm:
>>=20
>> 5.  Example Algorithm for Choosing the Local Preference
>>=20
>>    The value space for the local preference is from 0 to 65535
>>    inclusive.  This value space can be divided up in chunks for each IP
>>    address family.
>>=20
>>    An IPv6 and IPv4 start priority must be given.  In this example IPv6
>>    starts at 60000 and IPv4 at 59000.  This leaves enough address space
>>    to further play with the values if pr interface priorities needs to
>>    be added.  The highest value should be given to the address family
>>    that should be prioritized.
>>=20
>>         IPv6    IPv4
>>          Start   Start
>>    65535  60k     59k    58k    57k    56k    55k                    0
>>    +--------+------+------+------+------+------+---------------------+
>>    |        | IPv6 | IPv4 | IPv6 | IPv4 | IPv6 |                     |
>>    |        | (1)  |  (1) |  (2) |  (2) |  (3) |                     |
>>    +--------+------+------+------+------+------+---------------------+
>>              <- N->
>>=20
>>=20
>>    The local preference can be calculated by the given formula:
>>=20
>>=20
>>          local_preference =3D N*2*(Cn/Cmax)
>>=20
>>    Where N is the absolute value of IPv6_start-IPv4_start.  This ensures
>>    a positive number even if IPv4 is the highest priority.  Cn is the
>>    number of current candidates of a specific IP address type and
>>    candidate type (HOST, SRFLX, RELAY).  Cmax is the number of allowed
>>    consecutive candidates of the same IP address type.
>>=20
>=20
> Could you give an example how this algorithm and some priority values,=20
> e.g., from your program, result in the candidate pairs and order of=20
> checks? Especially the pairing part with respect to different types of=20
> candidates and IP families is where I'm afraid things may go wrong.
>=20

I am struggling a bit with the RFC format, trying to create an readable lis=
t that is of any use to the reader.

The generated checklist will depend largely on what values the remote side =
have chosen. If he the remote side for some reason gave all his candidates =
a local_preference in the range 64000-65000 the above formula would have no=
 effect. If he chooses more =93normal=94 values we should see an effect.
But the most important thing is that this does not break anything as the IC=
E spec allows us to play with this value.

My biggest concern now is how the unfrezing works. Reading 7.1.3.2.3 in the=
 ICE spec I realise that some of the sorting of the candidates regarding co=
mponent might not be optimal, but not sure. Will mark this as an open quest=
ion in the draft (Will  post in a few hrs, so will not attempt to solve thi=
s problem now.)

.-.
P=E5l-Erik




>=20
> Cheers,
> Ari
>=20


From nobody Fri Feb 14 04:02:47 2014
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From: "Pal Martinsen (palmarti)" <palmarti@cisco.com>
To: "mmusic (E-mail)" <mmusic@ietf.org>
Thread-Topic: New Version Notification for draft-reddy-mmusic-ice-happy-eyeballs-06.txt
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Subject: [MMUSIC] Fwd: New Version Notification for draft-reddy-mmusic-ice-happy-eyeballs-06.txt
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Hi All,

We have updated the ICE Happy Eyeballs draft to include an example algorith=
m.
(local_preference =3D N*2*(Cn/Cmax), se draft for details)

Minor tweaks and improvements based on feedback on the list.=20

Usual submission details below:
>=20
> A new version of I-D, draft-reddy-mmusic-ice-happy-eyeballs-06.txt
> has been successfully submitted by Paal-Erik Martinsen and posted to the
> IETF repository.
>=20
> Name:		draft-reddy-mmusic-ice-happy-eyeballs
> Revision:	06
> Title:		Happy Eyeballs Extension for ICE
> Document date:	2014-02-14
> Group:		Individual Submission
> Pages:		11
> URL:            http://www.ietf.org/internet-drafts/draft-reddy-mmusic-ic=
e-happy-eyeballs-06.txt
> Status:         https://datatracker.ietf.org/doc/draft-reddy-mmusic-ice-h=
appy-eyeballs/
> Htmlized:       http://tools.ietf.org/html/draft-reddy-mmusic-ice-happy-e=
yeballs-06
> Diff:           http://www.ietf.org/rfcdiff?url2=3Ddraft-reddy-mmusic-ice=
-happy-eyeballs-06
>=20
> Abstract:
>   This document provides guidelines on how to make Interactive
>   Connectivity Establishment (ICE) conclude faster in IPv4/IPv6 dual-
>   stack scenarios where broken paths exist.
>=20
>=20
>=20

.-.
P=E5l-Erik=


From nobody Fri Feb 14 07:04:06 2014
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To: "Pal Martinsen (palmarti)" <palmarti@cisco.com>
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Cc: mmusic <mmusic@ietf.org>, "Tirumaleswar Reddy \(tireddy\)" <tireddy@cisco.com>
Subject: Re: [MMUSIC] New Version Notification for draft-reddy-mmusic-ice-happy-eyeballs-04.txt
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On 14/02/14 11:36, Pal Martinsen (palmarti) wrote:
>
> On 13 Feb 2014, at 19:09 pm, Ari Keränen <ari.keranen@ericsson.com> wrote:
>
>> On 31/01/14 15:44, Pal Martinsen (palmarti) wrote:
>>> On 30 Jan 2014, at 12:34 pm, Ari Keränen <ari.keranen@ericsson.com> wrote:
>>>> On 28/01/14 15:49, Pal Martinsen (palmarti) wrote:
>> [...]
>>>> What I would find helpful for discussion is that if you could provide a step-by-step example of how the candidate lists are formed and connectivity checks done in cases where a) both support this with same promo algo b) both support this but with different promo algo c) only one endpoint supports this.
>>>>
>>>
>>> As I do believe in running code, I wrote a simple implementation of the algorithm described below.
>>
>> That's very nice, but unfortunately that didn't compile on my Mac so I
>> couldn't give it a try.
>
> Runs fine on my mac. (Famous last words..) But I have ports installed with gcc and the usual *NIX toolchain.
> Have not yet figured out to create a x-code project. If time permits..
>
> I should probably have written it in phyton, but my skills in that language is limited.

I have ports and gcc too, but probably missing something then. Will have 
to have a closer look at this later.

>>
>>>> Also maybe a recommended promo algo would make sense so that there is at least a fair chance that both will use the same algo/values and get most out of this.
>>>>
>>> First shot an an algorithm:
>>>
>>> 5.  Example Algorithm for Choosing the Local Preference
>>>
>>>     The value space for the local preference is from 0 to 65535
>>>     inclusive.  This value space can be divided up in chunks for each IP
>>>     address family.
>>>
>>>     An IPv6 and IPv4 start priority must be given.  In this example IPv6
>>>     starts at 60000 and IPv4 at 59000.  This leaves enough address space
>>>     to further play with the values if pr interface priorities needs to
>>>     be added.  The highest value should be given to the address family
>>>     that should be prioritized.
>>>
>>>          IPv6    IPv4
>>>           Start   Start
>>>     65535  60k     59k    58k    57k    56k    55k                    0
>>>     +--------+------+------+------+------+------+---------------------+
>>>     |        | IPv6 | IPv4 | IPv6 | IPv4 | IPv6 |                     |
>>>     |        | (1)  |  (1) |  (2) |  (2) |  (3) |                     |
>>>     +--------+------+------+------+------+------+---------------------+
>>>               <- N->
>>>
>>>
>>>     The local preference can be calculated by the given formula:
>>>
>>>
>>>           local_preference = N*2*(Cn/Cmax)
>>>
>>>     Where N is the absolute value of IPv6_start-IPv4_start.  This ensures
>>>     a positive number even if IPv4 is the highest priority.  Cn is the
>>>     number of current candidates of a specific IP address type and
>>>     candidate type (HOST, SRFLX, RELAY).  Cmax is the number of allowed
>>>     consecutive candidates of the same IP address type.
>>>
>>
>> Could you give an example how this algorithm and some priority values,
>> e.g., from your program, result in the candidate pairs and order of
>> checks? Especially the pairing part with respect to different types of
>> candidates and IP families is where I'm afraid things may go wrong.
>>
>
> I am struggling a bit with the RFC format, trying to create an readable list that is of any use to the reader.

I didn't mean (necessarily) for the draft, but just on an e-mail, or 
what ever works for you, would be sufficient for now.

> The generated checklist will depend largely on what values the remote side have chosen. If he the remote side for some reason gave all his candidates a local_preference in the range 64000-65000 the above formula would have no effect. If he chooses more “normal” values we should see an effect.

Is this something that should be advised about in the draft?

> But the most important thing is that this does not break anything as the ICE spec allows us to play with this value.

That sounds good, but I'm still not 100% sure it works, and hence an 
example of the whole procedure with some realistic values would be nice.

> My biggest concern now is how the unfrezing works. Reading 7.1.3.2.3 in the ICE spec I realise that some of the sorting of the candidates regarding component might not be optimal, but not sure. Will mark this as an open question in the draft (Will  post in a few hrs, so will not attempt to solve this problem now.)

OK.


Cheers,
Ari


From nobody Fri Feb 14 09:34:08 2014
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Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-06.txt - Christer's initial comments
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On 2/14/14 4:05 AM, Christian Groves wrote:
> Hello Christer,
>
> I'm not assuming the use of webrtc-data channel. The issue would apply
> to any "app" being used multiple times in an SCTP association.
>
> Are you suggesting any time that you want to use multiple instances of
> an "app" that you must use "webrtc-datachannel" even if there are no web
> browsers involved?

How do we *know* there won't be browsers involved?

IMO we have a problem because the "webrtc-datachannel" suggests it 
necessarily has something to do with webrtc, when it need not. I'd like 
to see that name changed.

The webrtc data channel just adds another layer of abstraction on top of 
SCTP, building bidirectional channels out of pairs of unidirectional 
streams. It also comes with an (optional) associated in-band protocol 
for dynamically establishing new channels over the sctp association, and 
with a defined javascript API that allows it to be used by browsers.

The a=sctpmap attribute is used to negotiate a particular "usage" for 
the entire SCTP association - including all the streams within it. There 
may indeed be cases where an application has a specialized need for 
that. But in many cases it will be particular channels that the 
application needs, and it can use generic mechanisms, such as come with 
"webrtc-datachannel" to manage the individual channels. Certainly that 
is the vision in rtcweb (where the lower layer mechanisms aren't exposed).

In the case of CLUE, we need one channel. It is fine with clue if the 
sctp association also carries other channels that it doesn't know about. 
If we were to use:

a=sctpmap:5000 clue ...

then we would be claiming the entire association for clue. The other 
channels wouldn't be available for anything else. And the clue channel 
wouldn't be accessible from a browser via webrtc.

	Thanks,
	Paul

> Regards, Christian
>
> On 14/02/2014 5:56 PM, Christer Holmberg wrote:
>> Hi Christian,
>>
>> A comment, based on your example. I don't think we would indicate
>> "clue" in the sctpmap attribute. We would indicate
>> "webrtc-datachannel", in order to be able to interwork with browsers.
>> Then, on that webrtc-datachannel, we would run the CLUE protocol
>> (which we would negotiate e.g. using the DCP, Richard's draft, or some
>> other mechanism).
>>
>> Regards,
>>
>> Christer
>>
>> -----Original Message-----
>> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of Christian
>> Groves
>> Sent: 14. helmikuuta 2014 1:49
>> To: mmusic@ietf.org
>> Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-06.txt -
>> Christer's initial comments
>>
>> Hello,
>>
>> Some further comments:
>>
>> Section 11.1 (Editorial) - 11.1 The first paragraph regarding 'sctpmap'
>> attribute appears to be repeated.
>>
>> Section 4.1 (Technical): I see the draft has removed the using using
>> "bfcp" and "t38" as values for the "App" field under one m-line.
>> However that also removes any hint about multiple a=sctpmaps being
>> used under one m-line. For example should there be one a=sctpmap per
>> 'fmt' or could there be zero or more a=sctpmaps per 'fmt' in the
>> m-line? If think this should be clarified.
>>
>> General: There was a discussion on the CLUE mailing list whether there
>> could be multiple CLUE protocol instances per SCTP association. We
>> couldn't identify a use case but couldn't rule it out either. I assume
>> this would apply to other "apps" also.
>> For example something like:
>> m=application 54111 DTLS/SCTP 5000
>> c=IN IP4 79.97.215.79
>> a=sctpmap:5000 clue max-message-size=100000 streams=1
>> a=sctpmap:5000 clue max-message-size=100000 streams=1
>> a=sctpmap:5000 clue max-message-size=100000 streams=1 Do we need a
>> "label" to uniquely identify a particular 'app' instance?
>> i.e. Similar to the optional label that
>> draft-ietf-rtcweb-data-protocol uses.
>> e.g. a=sctpmap:5000 clue max-message-size=100000 streams=1 label1
>> a=sctpmap:5000 clue max-message-size=100000 streams=1 label2
>> a=sctpmap:5000 clue max-message-size=100000 streams=1 label3
>>
>> Regards, Christian
>>
>>
>> On 14/02/2014 7:57 AM, Christer Holmberg wrote:
>>> Hi,
>>> I gave some comments off-line to Salvatore, which have been addressed
>>> in the new version.
>>> Some additional comments. I do realize some are pure editorial, but
>>> rather to write then down in some notes I put them directly into the
>>> e-mail J *Section 1:*
>>> *------------*
>>> *Q1**(EDITORIAL)**:*
>>> The text says:
>>> "Since
>>> no-one so far has implemented SCTP over TLS, due to some serious
>>> limitations described in [RFC6083], this document does not make use of
>>> TLS over SCTP as described in RFC3436 [RFC3436]."
>>> I suggest to simply say something like:
>>> "As described in [RFC6083], there are limitations associated with
>>> implementing SCTP over TLS. For that reason, SCTP over TLS is outside
>>> the scope of this document."
>>> *Q2**(EDITORIAL)**:*
>>> The text says:
>>> "Additionally, this document specifies the use of the 'setup' and
>>> 'connection' SDP attributes to establish SCTP associations. These
>>> attributes were defined in RFC4145 [RFC4145] for TCP. This document
>>> discusses their use with SCTP."
>>> I suggest to remove the last sentence, as it only repeats what is said
>>> in the first sentence.
>>> *Section 3:*
>>> *------------*
>>> *Q3 (EDITORIAL):*
>>> The text says:
>>> "This document defines three new values for the 'proto' field: 'SCTP',
>>> 'SCTP/DTLS' and 'DTLS/SCTP'."
>>> I would suggest to say:
>>> "This document defines three new *protocol identifier* values for the
>>> 'proto' field: 'SCTP', 'SCTP/DTLS' and 'DTLS/SCTP'."
>>> .because, you use "protocol identifier" terminology later.
>>> *Q4 (CLARIFICATION):*
>>> The text says:
>>> "An 'm' line that specifies 'SCTP' or 'SCTP/DTLS' or 'DTLS/SCTP' MUST
>>> further qualify the application-layer protocol using an fmt
>>> identifier."
>>> I do understand what you are trying to say, but I think the "qualify"
>>> word is a little confusing.
>>> Maybe something like:
>>> "An 'm' line that specifies 'SCTP' or 'SCTP/DTLS' or 'DTLS/SCTP' MUST
>>> further include an fmt value that identifies the application-layer
>>> protocol carried on top of the SCTP association."
>>> *Section 4.1:*
>>> *--------------*
>>> *Q5 (TECHNICAL):*
>>> The text says:
>>> "For the payload type assignments the "a=sctpmap:"
>>> attribute (see Section 5.1) 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."
>>> It is very difficult for me to parse this, and figure out the
>>> semantics of the sctpmap attribute.
>>> First, what do you mean by "media encoding name"?
>>> Second, when you talk about "application protocol", I assume you refer
>>> to the SCTP protocol?
>>> Third, the text say SHOULD. When is it ok to not use the sctpmap
>>> attribute?
>>> *Section **5**:*
>>> *------------*
>>> *Q6 (EDITORIAL):*
>>> I suggest to rename section 5 to "SDP sctpmap attribute", rename
>>> section 5.1 to "General", and add a new section 5.2 named "ABNF".
>>> *Q7 (TECHNICAL):*
>>> The text says:
>>> "The sctpmap MUST include the app parameter indicating the application
>>> running on top of the association."
>>> This is fine if the sctpmap attribute maps to an application. But,
>>> what if the sctpmap attribute maps to a "media encoding" or "payload"?
>>> *Q8 **(EDITORIAL):*
>>> The text says:
>>> "The sctpmap line should also contain the max-message-size parameter
>>> indicating the maximum message size, in bytes, the endpoint is willing
>>> to accept."
>>> I suggest to replace "accept" with "receive on the associated SCTP
>>> association".
>>> *Q9 (EDITORIAL):*
>>> The text says:
>>> "The peer should assume that larger message will be rejected by the
>>> endpoint, though it is up to the endpoint decide the appropriate
>>> behaviour."
>>> I suggest to either remove this paragraph, OR replace it with a
>>> sentence saying that a peer must/should not send messages bigger than
>>> the indicated max value.
>>> *Q10 (CLARIFICATION):*
>>> The text says:
>>> "If the parameter is not present, the implementation should provide a
>>> default, with a suggested value of 64K."
>>> I don't understand "suggested". If the default value is 64K the text
>>> should specify (rather than suggest) that.
>>> The same comment applies to the stream parameter, described later in
>>> the section.
>>> *Q11 (CLARIFICATION):*
>>> The text says:
>>> "It may also provide the stream parameter to specify the initial
>>> number of incoming streams to be supported by each side of the
>>> association."
>>> First, to be consistent with previous text, I suggest to replace "It
>>> may also provide" with "The sctpmap line may also contain"
>>> Second, I don't understand "incoming streams to be supported by each
>>> side of the association". Doesn't each peer indicate the number of
>>> streams it is willing to receive? Then, we can always recommend that
>>> the peer indicates the same number of streams, if possible.
>>> *Q12 (EDITORIAL):*
>>> The text that follows the ABNF should be put into a separate "SDP
>>> Offer/Answer Procedures" section.
>>> *Section 7:*
>>> *------------*
>>> *Q13 (EDITORIAL):*
>>> Please don't talk about re-INVITEs, or other SIP messages. Simply talk
>>> about offers and answers.
>>> *Q14 (EDITORIAL):*
>>> The text says:
>>> "It is worth to underline that when using SCTP on top of DTLS,.."
>>> To have consistent terminology, please say "SCTP over DTLS".
>>> That's all for now! :)
>>> Regards,
>>> Christer
>>> -----Alkuperäinen viesti-----
>>> Lähettäjä: mmusic [mailto:mmusic-bounces@ietf.org] Puolesta
>>> internet-drafts@ietf.org
>>> Lähetetty: 13. helmikuuta 2014 16:21
>>> Vastaanottaja: i-d-announce@ietf.org
>>> Kopio: mmusic@ietf.org
>>> Aihe: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-06.txt A New
>>> Internet-Draft is available from the on-line Internet-Drafts
>>> directories.
>>> This draft is a work item of the Multiparty Multimedia Session Control
>>> Working Group of the IETF.
>>> Title : Stream Control Transmission Protocol (SCTP)-Based Media
>>> Transport in the Session Description Protocol (SDP) Authors :
>>> Salvatore Loreto Gonzalo Camarillo Filename :
>>> draft-ietf-mmusic-sctp-sdp-06.txt Pages : 14 Date : 2014-02-13
>>> Abstract:
>>> SCTP (Stream Control Transmission Protocol) is a transport protocol
>>> used to establish associations between two endpoints. This document
>>> describes how to express media transport over SCTP in SDP (Session
>>> Description Protocol). This document defines the 'SCTP', 'SCTP/DTLS'
>>> and 'DTLS/SCTP' protocol identifiers for SDP.
>>> The IETF datatracker status page for this draft is:
>>> https://datatracker.ietf.org/doc/draft-ietf-mmusic-sctp-sdp/
>>> There's also a htmlized version available at:
>>> http://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-06
>>> A diff from the previous version is available at:
>>> http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-sctp-sdp-06
>>> Please note that it may take a couple of minutes from the time of
>>> submission until the htmlized version and diff are available at
>>> tools.ietf.org.
>>> Internet-Drafts are also available by anonymous FTP at:
>>> ftp://ftp.ietf.org/internet-drafts/
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org <mailto:mmusic@ietf.org>
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>
>>>
>>> _______________________________________________
>>> mmusic mailing list
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>>> https://www.ietf.org/mailman/listinfo/mmusic
>> _______________________________________________
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Christian Groves <Christian.Groves@nteczone.com>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-06.txt - Christer's initial comments
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Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-06.txt - Christer's initial comments
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Would agree to changing webrtc-data channel to something more generic.

Keith

> -----Original Message-----
> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of=20
> Christer Holmberg
> Sent: 14 February 2014 09:21
> To: Christian Groves; mmusic@ietf.org
> Subject: Re: [MMUSIC] I-D Action:=20
> draft-ietf-mmusic-sctp-sdp-06.txt - Christer's initial comments
>=20
> Hi Christian,
>=20
> > I'm not assuming the use of webrtc-data channel. The issue=20
> would apply to any "app" being used multiple times in an SCTP=20
> association.
>=20
> Correct.=20
>=20
> My comment was not on multiple apps per SCTP association=20
> (which I agree was the main focus of your e-mail), but simply=20
> on the example :)
>=20
> > Are you suggesting any time that you want to use multiple=20
> instances of an "app" that you must use "webrtc-datachannel"=20
> even if there are no web browsers involved?
>=20
> No. Only if you want interoperability with a browser.
>=20
> BUT, even if there are no browser involved, you can still use=20
> the webrtc-datachannel if you want. The name should probably=20
> be changed to something more generic, though.
>=20
> Regards,
>=20
> Christer
>=20
>=20
> On 14/02/2014 5:56 PM, Christer Holmberg wrote:
> > Hi Christian,
> >
> > A comment, based on your example. I don't think we would=20
> indicate "clue" in the sctpmap attribute. We would indicate=20
> "webrtc-datachannel", in order to be able to interwork with=20
> browsers. Then, on that webrtc-datachannel, we would run the=20
> CLUE protocol (which we would negotiate e.g. using the DCP,=20
> Richard's draft, or some other mechanism).
> >
> > Regards,
> >
> > Christer
> >
> > -----Original Message-----
> > From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of=20
> Christian=20
> > Groves
> > Sent: 14. helmikuuta 2014 1:49
> > To: mmusic@ietf.org
> > Subject: Re: [MMUSIC] I-D Action:=20
> draft-ietf-mmusic-sctp-sdp-06.txt -=20
> > Christer's initial comments
> >
> > Hello,
> >
> > Some further comments:
> >
> > Section 11.1 (Editorial) - 11.1 The first paragraph=20
> regarding 'sctpmap'
> > attribute appears to be repeated.
> >
> > Section 4.1 (Technical): I see the draft has removed the=20
> using using "bfcp" and "t38" as values for the "App" field=20
> under one m-line. However that also removes any hint about=20
> multiple a=3Dsctpmaps being used under one m-line. For example=20
> should there be one a=3Dsctpmap per 'fmt' or could there be=20
> zero or more a=3Dsctpmaps per 'fmt' in the m-line? If think=20
> this should be clarified.
> >
> > General: There was a discussion on the CLUE mailing list=20
> whether there could be multiple CLUE protocol instances per=20
> SCTP association. We couldn't identify a use case but=20
> couldn't rule it out either. I assume this would apply to=20
> other "apps" also.
> > For example something like:
> > m=3Dapplication 54111 DTLS/SCTP 5000
> > c=3DIN IP4 79.97.215.79
> > a=3Dsctpmap:5000 clue max-message-size=3D100000 streams=3D1=20
> a=3Dsctpmap:5000=20
> > clue max-message-size=3D100000 streams=3D1 a=3Dsctpmap:5000 clue=20
> > max-message-size=3D100000 streams=3D1 Do we need a "label" to=20
> uniquely identify a particular 'app' instance?
> > i.e. Similar to the optional label that=20
> draft-ietf-rtcweb-data-protocol uses.
> > e.g. a=3Dsctpmap:5000 clue max-message-size=3D100000 streams=3D1 label1=
=20
> > a=3Dsctpmap:5000 clue max-message-size=3D100000 streams=3D1 label2=20
> > a=3Dsctpmap:5000 clue max-message-size=3D100000 streams=3D1 label3
> >
> > Regards, Christian
> >
> >
> > On 14/02/2014 7:57 AM, Christer Holmberg wrote:
> >> Hi,
> >> I gave some comments off-line to Salvatore, which have=20
> been addressed=20
> >> in the new version.
> >> Some additional comments. I do realize some are pure=20
> editorial, but=20
> >> rather to write then down in some notes I put them=20
> directly into the=20
> >> e-mail J *Section 1:*
> >> *------------*
> >> *Q1**(EDITORIAL)**:*
> >> The text says:
> >> "Since
> >> no-one so far has implemented SCTP over TLS, due to some serious=20
> >> limitations described in [RFC6083], this document does not=20
> make use=20
> >> of TLS over SCTP as described in RFC3436 [RFC3436]."
> >> I suggest to simply say something like:
> >> "As described in [RFC6083], there are limitations associated with=20
> >> implementing SCTP over TLS. For that reason, SCTP over TLS=20
> is outside=20
> >> the scope of this document."
> >> *Q2**(EDITORIAL)**:*
> >> The text says:
> >> "Additionally, this document specifies the use of the 'setup' and=20
> >> 'connection' SDP attributes to establish SCTP associations. These=20
> >> attributes were defined in RFC4145 [RFC4145] for TCP. This=20
> document=20
> >> discusses their use with SCTP."
> >> I suggest to remove the last sentence, as it only repeats what is=20
> >> said in the first sentence.
> >> *Section 3:*
> >> *------------*
> >> *Q3 (EDITORIAL):*
> >> The text says:
> >> "This document defines three new values for the 'proto' field:=20
> >> 'SCTP', 'SCTP/DTLS' and 'DTLS/SCTP'."
> >> I would suggest to say:
> >> "This document defines three new *protocol identifier*=20
> values for the=20
> >> 'proto' field: 'SCTP', 'SCTP/DTLS' and 'DTLS/SCTP'."
> >> .because, you use "protocol identifier" terminology later.
> >> *Q4 (CLARIFICATION):*
> >> The text says:
> >> "An 'm' line that specifies 'SCTP' or 'SCTP/DTLS' or=20
> 'DTLS/SCTP' MUST=20
> >> further qualify the application-layer protocol using an fmt=20
> >> identifier."
> >> I do understand what you are trying to say, but I think=20
> the "qualify"
> >> word is a little confusing.
> >> Maybe something like:
> >> "An 'm' line that specifies 'SCTP' or 'SCTP/DTLS' or=20
> 'DTLS/SCTP' MUST=20
> >> further include an fmt value that identifies the application-layer=20
> >> protocol carried on top of the SCTP association."
> >> *Section 4.1:*
> >> *--------------*
> >> *Q5 (TECHNICAL):*
> >> The text says:
> >> "For the payload type assignments the "a=3Dsctpmap:"
> >> attribute (see Section 5.1) SHOULD be used to map from a=20
> port number=20
> >> to a media encoding name that identifies the payload format=20
> >> transported by the association or the actual application protocol=20
> >> running on top of it."
> >> It is very difficult for me to parse this, and figure out the=20
> >> semantics of the sctpmap attribute.
> >> First, what do you mean by "media encoding name"?
> >> Second, when you talk about "application protocol", I assume you=20
> >> refer to the SCTP protocol?
> >> Third, the text say SHOULD. When is it ok to not use the sctpmap=20
> >> attribute?
> >> *Section **5**:*
> >> *------------*
> >> *Q6 (EDITORIAL):*
> >> I suggest to rename section 5 to "SDP sctpmap attribute", rename=20
> >> section 5.1 to "General", and add a new section 5.2 named "ABNF".
> >> *Q7 (TECHNICAL):*
> >> The text says:
> >> "The sctpmap MUST include the app parameter indicating the=20
> >> application running on top of the association."
> >> This is fine if the sctpmap attribute maps to an application. But,=20
> >> what if the sctpmap attribute maps to a "media encoding"=20
> or "payload"?
> >> *Q8 **(EDITORIAL):*
> >> The text says:
> >> "The sctpmap line should also contain the max-message-size=20
> parameter=20
> >> indicating the maximum message size, in bytes, the endpoint is=20
> >> willing to accept."
> >> I suggest to replace "accept" with "receive on the associated SCTP=20
> >> association".
> >> *Q9 (EDITORIAL):*
> >> The text says:
> >> "The peer should assume that larger message will be=20
> rejected by the=20
> >> endpoint, though it is up to the endpoint decide the appropriate=20
> >> behaviour."
> >> I suggest to either remove this paragraph, OR replace it with a=20
> >> sentence saying that a peer must/should not send messages=20
> bigger than=20
> >> the indicated max value.
> >> *Q10 (CLARIFICATION):*
> >> The text says:
> >> "If the parameter is not present, the implementation=20
> should provide a=20
> >> default, with a suggested value of 64K."
> >> I don't understand "suggested". If the default value is=20
> 64K the text=20
> >> should specify (rather than suggest) that.
> >> The same comment applies to the stream parameter,=20
> described later in=20
> >> the section.
> >> *Q11 (CLARIFICATION):*
> >> The text says:
> >> "It may also provide the stream parameter to specify the initial=20
> >> number of incoming streams to be supported by each side of the=20
> >> association."
> >> First, to be consistent with previous text, I suggest to=20
> replace "It=20
> >> may also provide" with "The sctpmap line may also contain"
> >> Second, I don't understand "incoming streams to be=20
> supported by each=20
> >> side of the association". Doesn't each peer indicate the number of=20
> >> streams it is willing to receive? Then, we can always=20
> recommend that=20
> >> the peer indicates the same number of streams, if possible.
> >> *Q12 (EDITORIAL):*
> >> The text that follows the ABNF should be put into a separate "SDP=20
> >> Offer/Answer Procedures" section.
> >> *Section 7:*
> >> *------------*
> >> *Q13 (EDITORIAL):*
> >> Please don't talk about re-INVITEs, or other SIP messages. Simply=20
> >> talk about offers and answers.
> >> *Q14 (EDITORIAL):*
> >> The text says:
> >> "It is worth to underline that when using SCTP on top of DTLS,.."
> >> To have consistent terminology, please say "SCTP over DTLS".
> >> That's all for now! :)
> >> Regards,
> >> Christer
> >> -----Alkuper=E4inen viesti-----
> >> L=E4hett=E4j=E4: mmusic [mailto:mmusic-bounces@ietf.org] Puolesta=20
> >> internet-drafts@ietf.org
> >> L=E4hetetty: 13. helmikuuta 2014 16:21
> >> Vastaanottaja: i-d-announce@ietf.org
> >> Kopio: mmusic@ietf.org
> >> Aihe: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-06.txt A New=20
> >> Internet-Draft is available from the on-line Internet-Drafts=20
> >> directories.
> >> This draft is a work item of the Multiparty Multimedia Session=20
> >> Control Working Group of the IETF.
> >> Title : Stream Control Transmission Protocol (SCTP)-Based Media=20
> >> Transport in the Session Description Protocol (SDP) Authors :
> >> Salvatore Loreto Gonzalo Camarillo Filename :
> >> draft-ietf-mmusic-sctp-sdp-06.txt Pages : 14 Date : 2014-02-13
> >> Abstract:
> >> SCTP (Stream Control Transmission Protocol) is a transport=20
> protocol=20
> >> used to establish associations between two endpoints. This=20
> document=20
> >> describes how to express media transport over SCTP in SDP (Session=20
> >> Description Protocol). This document defines the 'SCTP',=20
> 'SCTP/DTLS'
> >> and 'DTLS/SCTP' protocol identifiers for SDP.
> >> The IETF datatracker status page for this draft is:
> >> https://datatracker.ietf.org/doc/draft-ietf-mmusic-sctp-sdp/
> >> There's also a htmlized version available at:
> >> http://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-06
> >> A diff from the previous version is available at:
> >> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sctp-sdp-06
> >> Please note that it may take a couple of minutes from the time of=20
> >> submission until the htmlized version and diff are available at=20
> >> tools.ietf.org.
> >> Internet-Drafts are also available by anonymous FTP at:
> >> ftp://ftp.ietf.org/internet-drafts/
> >> _______________________________________________
> >> mmusic mailing list
> >> mmusic@ietf.org <mailto:mmusic@ietf.org>=20
> >> 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
> >
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
> =


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Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-06.txt - Christer's initial comments
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On 2/14/14 1:40 PM, DRAGE, Keith (Keith) wrote:
> Would agree to changing webrtc-data channel to something more generic.

A couple of possibilities that would WFM:

- sctp-data-channel (or sctp-datachannel)
- data-channel (or datachannel)

The latter is appealing, but may be *too* generic.

	Thanks,
	Paul

> Keith
>
>> -----Original Message-----
>> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of
>> Christer Holmberg
>> Sent: 14 February 2014 09:21
>> To: Christian Groves; mmusic@ietf.org
>> Subject: Re: [MMUSIC] I-D Action:
>> draft-ietf-mmusic-sctp-sdp-06.txt - Christer's initial comments
>>
>> Hi Christian,
>>
>>> I'm not assuming the use of webrtc-data channel. The issue
>> would apply to any "app" being used multiple times in an SCTP
>> association.
>>
>> Correct.
>>
>> My comment was not on multiple apps per SCTP association
>> (which I agree was the main focus of your e-mail), but simply
>> on the example :)
>>
>>> Are you suggesting any time that you want to use multiple
>> instances of an "app" that you must use "webrtc-datachannel"
>> even if there are no web browsers involved?
>>
>> No. Only if you want interoperability with a browser.
>>
>> BUT, even if there are no browser involved, you can still use
>> the webrtc-datachannel if you want. The name should probably
>> be changed to something more generic, though.
>>
>> Regards,
>>
>> Christer
>>
>>
>> On 14/02/2014 5:56 PM, Christer Holmberg wrote:
>>> Hi Christian,
>>>
>>> A comment, based on your example. I don't think we would
>> indicate "clue" in the sctpmap attribute. We would indicate
>> "webrtc-datachannel", in order to be able to interwork with
>> browsers. Then, on that webrtc-datachannel, we would run the
>> CLUE protocol (which we would negotiate e.g. using the DCP,
>> Richard's draft, or some other mechanism).
>>>
>>> Regards,
>>>
>>> Christer
>>>
>>> -----Original Message-----
>>> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of
>> Christian
>>> Groves
>>> Sent: 14. helmikuuta 2014 1:49
>>> To: mmusic@ietf.org
>>> Subject: Re: [MMUSIC] I-D Action:
>> draft-ietf-mmusic-sctp-sdp-06.txt -
>>> Christer's initial comments
>>>
>>> Hello,
>>>
>>> Some further comments:
>>>
>>> Section 11.1 (Editorial) - 11.1 The first paragraph
>> regarding 'sctpmap'
>>> attribute appears to be repeated.
>>>
>>> Section 4.1 (Technical): I see the draft has removed the
>> using using "bfcp" and "t38" as values for the "App" field
>> under one m-line. However that also removes any hint about
>> multiple a=sctpmaps being used under one m-line. For example
>> should there be one a=sctpmap per 'fmt' or could there be
>> zero or more a=sctpmaps per 'fmt' in the m-line? If think
>> this should be clarified.
>>>
>>> General: There was a discussion on the CLUE mailing list
>> whether there could be multiple CLUE protocol instances per
>> SCTP association. We couldn't identify a use case but
>> couldn't rule it out either. I assume this would apply to
>> other "apps" also.
>>> For example something like:
>>> m=application 54111 DTLS/SCTP 5000
>>> c=IN IP4 79.97.215.79
>>> a=sctpmap:5000 clue max-message-size=100000 streams=1
>> a=sctpmap:5000
>>> clue max-message-size=100000 streams=1 a=sctpmap:5000 clue
>>> max-message-size=100000 streams=1 Do we need a "label" to
>> uniquely identify a particular 'app' instance?
>>> i.e. Similar to the optional label that
>> draft-ietf-rtcweb-data-protocol uses.
>>> e.g. a=sctpmap:5000 clue max-message-size=100000 streams=1 label1
>>> a=sctpmap:5000 clue max-message-size=100000 streams=1 label2
>>> a=sctpmap:5000 clue max-message-size=100000 streams=1 label3
>>>
>>> Regards, Christian
>>>
>>>
>>> On 14/02/2014 7:57 AM, Christer Holmberg wrote:
>>>> Hi,
>>>> I gave some comments off-line to Salvatore, which have
>> been addressed
>>>> in the new version.
>>>> Some additional comments. I do realize some are pure
>> editorial, but
>>>> rather to write then down in some notes I put them
>> directly into the
>>>> e-mail J *Section 1:*
>>>> *------------*
>>>> *Q1**(EDITORIAL)**:*
>>>> The text says:
>>>> "Since
>>>> no-one so far has implemented SCTP over TLS, due to some serious
>>>> limitations described in [RFC6083], this document does not
>> make use
>>>> of TLS over SCTP as described in RFC3436 [RFC3436]."
>>>> I suggest to simply say something like:
>>>> "As described in [RFC6083], there are limitations associated with
>>>> implementing SCTP over TLS. For that reason, SCTP over TLS
>> is outside
>>>> the scope of this document."
>>>> *Q2**(EDITORIAL)**:*
>>>> The text says:
>>>> "Additionally, this document specifies the use of the 'setup' and
>>>> 'connection' SDP attributes to establish SCTP associations. These
>>>> attributes were defined in RFC4145 [RFC4145] for TCP. This
>> document
>>>> discusses their use with SCTP."
>>>> I suggest to remove the last sentence, as it only repeats what is
>>>> said in the first sentence.
>>>> *Section 3:*
>>>> *------------*
>>>> *Q3 (EDITORIAL):*
>>>> The text says:
>>>> "This document defines three new values for the 'proto' field:
>>>> 'SCTP', 'SCTP/DTLS' and 'DTLS/SCTP'."
>>>> I would suggest to say:
>>>> "This document defines three new *protocol identifier*
>> values for the
>>>> 'proto' field: 'SCTP', 'SCTP/DTLS' and 'DTLS/SCTP'."
>>>> .because, you use "protocol identifier" terminology later.
>>>> *Q4 (CLARIFICATION):*
>>>> The text says:
>>>> "An 'm' line that specifies 'SCTP' or 'SCTP/DTLS' or
>> 'DTLS/SCTP' MUST
>>>> further qualify the application-layer protocol using an fmt
>>>> identifier."
>>>> I do understand what you are trying to say, but I think
>> the "qualify"
>>>> word is a little confusing.
>>>> Maybe something like:
>>>> "An 'm' line that specifies 'SCTP' or 'SCTP/DTLS' or
>> 'DTLS/SCTP' MUST
>>>> further include an fmt value that identifies the application-layer
>>>> protocol carried on top of the SCTP association."
>>>> *Section 4.1:*
>>>> *--------------*
>>>> *Q5 (TECHNICAL):*
>>>> The text says:
>>>> "For the payload type assignments the "a=sctpmap:"
>>>> attribute (see Section 5.1) 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."
>>>> It is very difficult for me to parse this, and figure out the
>>>> semantics of the sctpmap attribute.
>>>> First, what do you mean by "media encoding name"?
>>>> Second, when you talk about "application protocol", I assume you
>>>> refer to the SCTP protocol?
>>>> Third, the text say SHOULD. When is it ok to not use the sctpmap
>>>> attribute?
>>>> *Section **5**:*
>>>> *------------*
>>>> *Q6 (EDITORIAL):*
>>>> I suggest to rename section 5 to "SDP sctpmap attribute", rename
>>>> section 5.1 to "General", and add a new section 5.2 named "ABNF".
>>>> *Q7 (TECHNICAL):*
>>>> The text says:
>>>> "The sctpmap MUST include the app parameter indicating the
>>>> application running on top of the association."
>>>> This is fine if the sctpmap attribute maps to an application. But,
>>>> what if the sctpmap attribute maps to a "media encoding"
>> or "payload"?
>>>> *Q8 **(EDITORIAL):*
>>>> The text says:
>>>> "The sctpmap line should also contain the max-message-size
>> parameter
>>>> indicating the maximum message size, in bytes, the endpoint is
>>>> willing to accept."
>>>> I suggest to replace "accept" with "receive on the associated SCTP
>>>> association".
>>>> *Q9 (EDITORIAL):*
>>>> The text says:
>>>> "The peer should assume that larger message will be
>> rejected by the
>>>> endpoint, though it is up to the endpoint decide the appropriate
>>>> behaviour."
>>>> I suggest to either remove this paragraph, OR replace it with a
>>>> sentence saying that a peer must/should not send messages
>> bigger than
>>>> the indicated max value.
>>>> *Q10 (CLARIFICATION):*
>>>> The text says:
>>>> "If the parameter is not present, the implementation
>> should provide a
>>>> default, with a suggested value of 64K."
>>>> I don't understand "suggested". If the default value is
>> 64K the text
>>>> should specify (rather than suggest) that.
>>>> The same comment applies to the stream parameter,
>> described later in
>>>> the section.
>>>> *Q11 (CLARIFICATION):*
>>>> The text says:
>>>> "It may also provide the stream parameter to specify the initial
>>>> number of incoming streams to be supported by each side of the
>>>> association."
>>>> First, to be consistent with previous text, I suggest to
>> replace "It
>>>> may also provide" with "The sctpmap line may also contain"
>>>> Second, I don't understand "incoming streams to be
>> supported by each
>>>> side of the association". Doesn't each peer indicate the number of
>>>> streams it is willing to receive? Then, we can always
>> recommend that
>>>> the peer indicates the same number of streams, if possible.
>>>> *Q12 (EDITORIAL):*
>>>> The text that follows the ABNF should be put into a separate "SDP
>>>> Offer/Answer Procedures" section.
>>>> *Section 7:*
>>>> *------------*
>>>> *Q13 (EDITORIAL):*
>>>> Please don't talk about re-INVITEs, or other SIP messages. Simply
>>>> talk about offers and answers.
>>>> *Q14 (EDITORIAL):*
>>>> The text says:
>>>> "It is worth to underline that when using SCTP on top of DTLS,.."
>>>> To have consistent terminology, please say "SCTP over DTLS".
>>>> That's all for now! :)
>>>> Regards,
>>>> Christer
>>>> -----Alkuperäinen viesti-----
>>>> Lähettäjä: mmusic [mailto:mmusic-bounces@ietf.org] Puolesta
>>>> internet-drafts@ietf.org
>>>> Lähetetty: 13. helmikuuta 2014 16:21
>>>> Vastaanottaja: i-d-announce@ietf.org
>>>> Kopio: mmusic@ietf.org
>>>> Aihe: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-06.txt A New
>>>> Internet-Draft is available from the on-line Internet-Drafts
>>>> directories.
>>>> This draft is a work item of the Multiparty Multimedia Session
>>>> Control Working Group of the IETF.
>>>> Title : Stream Control Transmission Protocol (SCTP)-Based Media
>>>> Transport in the Session Description Protocol (SDP) Authors :
>>>> Salvatore Loreto Gonzalo Camarillo Filename :
>>>> draft-ietf-mmusic-sctp-sdp-06.txt Pages : 14 Date : 2014-02-13
>>>> Abstract:
>>>> SCTP (Stream Control Transmission Protocol) is a transport
>> protocol
>>>> used to establish associations between two endpoints. This
>> document
>>>> describes how to express media transport over SCTP in SDP (Session
>>>> Description Protocol). This document defines the 'SCTP',
>> 'SCTP/DTLS'
>>>> and 'DTLS/SCTP' protocol identifiers for SDP.
>>>> The IETF datatracker status page for this draft is:
>>>> https://datatracker.ietf.org/doc/draft-ietf-mmusic-sctp-sdp/
>>>> There's also a htmlized version available at:
>>>> http://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-06
>>>> A diff from the previous version is available at:
>>>> http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-sctp-sdp-06
>>>> Please note that it may take a couple of minutes from the time of
>>>> submission until the htmlized version and diff are available at
>>>> tools.ietf.org.
>>>> Internet-Drafts are also available by anonymous FTP at:
>>>> ftp://ftp.ietf.org/internet-drafts/
>>>> _______________________________________________
>>>> mmusic mailing list
>>>> mmusic@ietf.org <mailto:mmusic@ietf.org>
>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>>
>>>>
>>>> _______________________________________________
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>>> _______________________________________________
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>>
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Multiparty Multimedia Session Control Working Group of the IETF.

        Title           : A Framework for SDP Attributes when Multiplexing
        Author          : Suhas Nandakumar
	Filename        : draft-ietf-mmusic-sdp-mux-attributes-01.txt
	Pages           : 60
	Date            : 2014-02-14

Abstract:
   The Session Description Protocol (SDP) provides mechanisms to
   describe attributes of multimedia sessions and of individual media
   streams (e.g., Real-time Transport Protocol (RTP) sessions) within a
   multimedia session.  In the RTCWeb WG, there is a need to use a
   single 5-tuple for sending and receiving media associated with
   multiple media descriptions ("m=" lines).  Such a requirement has
   raised concerns over the semantic implications of the SDP attributes
   associated with the RTP Media Streams multiplexed over a single
   transport layer flow.

   The scope of this specification is to provide a framework for
   analyzing the multiplexing characteristics of SDP attributes.  The
   specification also categorizes existing attributes based on the
   framework described herein.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-sdp-mux-attributes-01

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-sdp-mux-attributes-01


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
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>4.1        a=3Dsctpmap:5000 webrtc-datachannel max-message-size=3D100000 s=
treams=3D16
>9.3        a=3Dsctpmap:5000 webrtc-DataChannel max-message-size=3D100000 s=
treams=3D16
>11.1   'app' parameter and register as first value "webrtc-datachannel" fo=
r
[Raju]
Above three types of representations of "webrtc-datachannel" should use a s=
ingle value;
The value may be based on the outcome from discussion that is going on a di=
fferent thread.


> a=3Dsctpmap:5000 webrtc-DataChannel max-message-size=3D100000 streams=3D1=
6
[Raju] Instead of putting webrtc data channel attributes on the same line (=
which gets long),=20
how about using existing a=3Dfmtp? as in:
a=3Dfmtp:5000 max-message-size=3D100000 streams=3D16
or how about even further shortening it to?
a=3Dfmtp:5000 mms=3D100000; streams=3D16

Then these attributes of 'app' protocol can be defined either in rtcweb-dat=
a-channel
and then IANA (section 11.1) will have a reference to it. Any such attribut=
es defined=20
for this or any generic 'app' protocol are carried in a=3Dfmtp line transpa=
rently.

Regards
Raju

> -----Original Message-----
> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of internet-
> drafts@ietf.org
> Sent: Thursday, February 13, 2014 8:21 AM
> To: i-d-announce@ietf.org
> Cc: mmusic@ietf.org
> Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-06.txt
>=20
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>  This draft is a work item of the Multiparty Multimedia Session Control
> Working Group of the IETF.
>=20
>         Title           : Stream Control Transmission Protocol (SCTP)-Bas=
ed
> Media Transport in the Session Description Protocol (SDP)
>         Authors         : Salvatore Loreto
>                           Gonzalo Camarillo
> 	Filename        : draft-ietf-mmusic-sctp-sdp-06.txt
> 	Pages           : 14
> 	Date            : 2014-02-13
>=20
> Abstract:
>    SCTP (Stream Control Transmission Protocol) is a transport protocol
>    used to establish associations between two endpoints.  This document
>    describes how to express media transport over SCTP in SDP (Session
>    Description Protocol).  This document defines the 'SCTP', 'SCTP/DTLS'
>    and 'DTLS/SCTP' protocol identifiers for SDP.
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-mmusic-sctp-sdp/
>=20
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-06
>=20
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sctp-sdp-06
>=20
>=20
> Please note that it may take a couple of minutes from the time of submiss=
ion
> until the htmlized version and diff are available at tools.ietf.org.
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From nobody Fri Feb 14 13:07:59 2014
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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-06.txt - Christer's initial comments
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Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-06.txt - Christer's initial comments
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> On 2/14/14 1:40 PM, DRAGE, Keith (Keith) wrote:
> > Would agree to changing webrtc-data channel to something more generic.
>=20
> A couple of possibilities that would WFM:
>=20
> - sctp-data-channel (or sctp-datachannel)
> - data-channel (or datachannel)
>=20
> The latter is appealing, but may be *too* generic.
[Raju] I am ok with either options.=20
However, "sctp-datachannel" probably more appropriate as for external negot=
iation=20
point of view, datachannel provides a bidrectional communication channel wi=
th=20
one SCTP stream, using same stream id, in both the directions. "sctp-datach=
annel"
name captures that well.

Regards
Raju


From nobody Fri Feb 14 15:43:31 2014
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Subject: [MMUSIC] POF/PAN draft update
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This is a multi-part message in MIME format.
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I've updated the partial offer/partial answer draft to reflect the 
conversations we had in Vancouver:
http://tools.ietf.org/html/draft-roach-mmusic-pof-pan-02

Diffs: http://tools.ietf.org/rfcdiff?url2=draft-roach-mmusic-pof-pan-02.txt

Note that this relies on the following draft for group membership 
manipulation:
http://tools.ietf.org/html/draft-roach-mmusic-groupid

Please point out whether you think I've missed any of the Vancouver 
discussions or other feedback. Thanks!

/a

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<html>
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    <meta http-equiv="content-type" content="text/html; charset=UTF-8">
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    I've updated the partial offer/partial answer draft to reflect the
    conversations we had in Vancouver:<br>
    <a class="moz-txt-link-freetext" href="http://tools.ietf.org/html/draft-roach-mmusic-pof-pan-02">http://tools.ietf.org/html/draft-roach-mmusic-pof-pan-02</a><br>
    <br>
    Diffs:
    <a class="moz-txt-link-freetext" href="http://tools.ietf.org/rfcdiff?url2=draft-roach-mmusic-pof-pan-02.txt">http://tools.ietf.org/rfcdiff?url2=draft-roach-mmusic-pof-pan-02.txt</a><br>
    <br>
    Note that this relies on the following draft for group membership
    manipulation:<br>
    <meta http-equiv="content-type" content="text/html; charset=UTF-8">
    <span class="insert"><a class="moz-txt-link-freetext" href="http://tools.ietf.org/html/draft-roach-mmusic-groupid">http://tools.ietf.org/html/draft-roach-mmusic-groupid</a></span><br>
    <br>
    Please point out whether you think I've missed any of the Vancouver
    discussions or other feedback. Thanks!<br>
    <br>
    /a<br>
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--------------080603090805070302040501--


From nobody Fri Feb 14 18:10:17 2014
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-06.txt - Christer's initial comments
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Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-06.txt - Christer's initial comments
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I'd be happy with either.

I've just submitted draft-ejzak-mmusic-data-channel-sdpneg using the term "=
data channel" in the text, but the SDP is still aligned with the sctp-sdp d=
raft.

Keith=20

> -----Original Message-----
> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of=20
> Paul Kyzivat
> Sent: 14 February 2014 19:42
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] I-D Action:=20
> draft-ietf-mmusic-sctp-sdp-06.txt - Christer's initial comments
>=20
> On 2/14/14 1:40 PM, DRAGE, Keith (Keith) wrote:
> > Would agree to changing webrtc-data channel to something=20
> more generic.
>=20
> A couple of possibilities that would WFM:
>=20
> - sctp-data-channel (or sctp-datachannel)
> - data-channel (or datachannel)
>=20
> The latter is appealing, but may be *too* generic.
>=20
> 	Thanks,
> 	Paul
>=20
> > Keith
> >
> >> -----Original Message-----
> >> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of=20
> Christer=20
> >> Holmberg
> >> Sent: 14 February 2014 09:21
> >> To: Christian Groves; mmusic@ietf.org
> >> Subject: Re: [MMUSIC] I-D Action:
> >> draft-ietf-mmusic-sctp-sdp-06.txt - Christer's initial comments
> >>
> >> Hi Christian,
> >>
> >>> I'm not assuming the use of webrtc-data channel. The issue
> >> would apply to any "app" being used multiple times in an SCTP=20
> >> association.
> >>
> >> Correct.
> >>
> >> My comment was not on multiple apps per SCTP association (which I=20
> >> agree was the main focus of your e-mail), but simply on=20
> the example=20
> >> :)
> >>
> >>> Are you suggesting any time that you want to use multiple
> >> instances of an "app" that you must use "webrtc-datachannel"
> >> even if there are no web browsers involved?
> >>
> >> No. Only if you want interoperability with a browser.
> >>
> >> BUT, even if there are no browser involved, you can still use the=20
> >> webrtc-datachannel if you want. The name should probably=20
> be changed=20
> >> to something more generic, though.
> >>
> >> Regards,
> >>
> >> Christer
> >>
> >>
> >> On 14/02/2014 5:56 PM, Christer Holmberg wrote:
> >>> Hi Christian,
> >>>
> >>> A comment, based on your example. I don't think we would
> >> indicate "clue" in the sctpmap attribute. We would indicate=20
> >> "webrtc-datachannel", in order to be able to interwork=20
> with browsers.=20
> >> Then, on that webrtc-datachannel, we would run the CLUE protocol=20
> >> (which we would negotiate e.g. using the DCP, Richard's draft, or=20
> >> some other mechanism).
> >>>
> >>> Regards,
> >>>
> >>> Christer
> >>>
> >>> -----Original Message-----
> >>> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of
> >> Christian
> >>> Groves
> >>> Sent: 14. helmikuuta 2014 1:49
> >>> To: mmusic@ietf.org
> >>> Subject: Re: [MMUSIC] I-D Action:
> >> draft-ietf-mmusic-sctp-sdp-06.txt -
> >>> Christer's initial comments
> >>>
> >>> Hello,
> >>>
> >>> Some further comments:
> >>>
> >>> Section 11.1 (Editorial) - 11.1 The first paragraph
> >> regarding 'sctpmap'
> >>> attribute appears to be repeated.
> >>>
> >>> Section 4.1 (Technical): I see the draft has removed the
> >> using using "bfcp" and "t38" as values for the "App" field=20
> under one=20
> >> m-line. However that also removes any hint about multiple=20
> a=3Dsctpmaps=20
> >> being used under one m-line. For example should there be one=20
> >> a=3Dsctpmap per 'fmt' or could there be zero or more a=3Dsctpmaps per=
=20
> >> 'fmt' in the m-line? If think this should be clarified.
> >>>
> >>> General: There was a discussion on the CLUE mailing list
> >> whether there could be multiple CLUE protocol instances per SCTP=20
> >> association. We couldn't identify a use case but couldn't=20
> rule it out=20
> >> either. I assume this would apply to other "apps" also.
> >>> For example something like:
> >>> m=3Dapplication 54111 DTLS/SCTP 5000
> >>> c=3DIN IP4 79.97.215.79
> >>> a=3Dsctpmap:5000 clue max-message-size=3D100000 streams=3D1
> >> a=3Dsctpmap:5000
> >>> clue max-message-size=3D100000 streams=3D1 a=3Dsctpmap:5000 clue=20
> >>> max-message-size=3D100000 streams=3D1 Do we need a "label" to
> >> uniquely identify a particular 'app' instance?
> >>> i.e. Similar to the optional label that
> >> draft-ietf-rtcweb-data-protocol uses.
> >>> e.g. a=3Dsctpmap:5000 clue max-message-size=3D100000 streams=3D1 labe=
l1=20
> >>> a=3Dsctpmap:5000 clue max-message-size=3D100000 streams=3D1 label2=20
> >>> a=3Dsctpmap:5000 clue max-message-size=3D100000 streams=3D1 label3
> >>>
> >>> Regards, Christian
> >>>
> >>>
> >>> On 14/02/2014 7:57 AM, Christer Holmberg wrote:
> >>>> Hi,
> >>>> I gave some comments off-line to Salvatore, which have
> >> been addressed
> >>>> in the new version.
> >>>> Some additional comments. I do realize some are pure
> >> editorial, but
> >>>> rather to write then down in some notes I put them
> >> directly into the
> >>>> e-mail J *Section 1:*
> >>>> *------------*
> >>>> *Q1**(EDITORIAL)**:*
> >>>> The text says:
> >>>> "Since
> >>>> no-one so far has implemented SCTP over TLS, due to some serious=20
> >>>> limitations described in [RFC6083], this document does not
> >> make use
> >>>> of TLS over SCTP as described in RFC3436 [RFC3436]."
> >>>> I suggest to simply say something like:
> >>>> "As described in [RFC6083], there are limitations=20
> associated with=20
> >>>> implementing SCTP over TLS. For that reason, SCTP over TLS
> >> is outside
> >>>> the scope of this document."
> >>>> *Q2**(EDITORIAL)**:*
> >>>> The text says:
> >>>> "Additionally, this document specifies the use of the=20
> 'setup' and=20
> >>>> 'connection' SDP attributes to establish SCTP=20
> associations. These=20
> >>>> attributes were defined in RFC4145 [RFC4145] for TCP. This
> >> document
> >>>> discusses their use with SCTP."
> >>>> I suggest to remove the last sentence, as it only=20
> repeats what is=20
> >>>> said in the first sentence.
> >>>> *Section 3:*
> >>>> *------------*
> >>>> *Q3 (EDITORIAL):*
> >>>> The text says:
> >>>> "This document defines three new values for the 'proto' field:
> >>>> 'SCTP', 'SCTP/DTLS' and 'DTLS/SCTP'."
> >>>> I would suggest to say:
> >>>> "This document defines three new *protocol identifier*
> >> values for the
> >>>> 'proto' field: 'SCTP', 'SCTP/DTLS' and 'DTLS/SCTP'."
> >>>> .because, you use "protocol identifier" terminology later.
> >>>> *Q4 (CLARIFICATION):*
> >>>> The text says:
> >>>> "An 'm' line that specifies 'SCTP' or 'SCTP/DTLS' or
> >> 'DTLS/SCTP' MUST
> >>>> further qualify the application-layer protocol using an fmt=20
> >>>> identifier."
> >>>> I do understand what you are trying to say, but I think
> >> the "qualify"
> >>>> word is a little confusing.
> >>>> Maybe something like:
> >>>> "An 'm' line that specifies 'SCTP' or 'SCTP/DTLS' or
> >> 'DTLS/SCTP' MUST
> >>>> further include an fmt value that identifies the=20
> application-layer=20
> >>>> protocol carried on top of the SCTP association."
> >>>> *Section 4.1:*
> >>>> *--------------*
> >>>> *Q5 (TECHNICAL):*
> >>>> The text says:
> >>>> "For the payload type assignments the "a=3Dsctpmap:"
> >>>> attribute (see Section 5.1) SHOULD be used to map from a
> >> port number
> >>>> to a media encoding name that identifies the payload format=20
> >>>> transported by the association or the actual application=20
> protocol=20
> >>>> running on top of it."
> >>>> It is very difficult for me to parse this, and figure out the=20
> >>>> semantics of the sctpmap attribute.
> >>>> First, what do you mean by "media encoding name"?
> >>>> Second, when you talk about "application protocol", I assume you=20
> >>>> refer to the SCTP protocol?
> >>>> Third, the text say SHOULD. When is it ok to not use the sctpmap=20
> >>>> attribute?
> >>>> *Section **5**:*
> >>>> *------------*
> >>>> *Q6 (EDITORIAL):*
> >>>> I suggest to rename section 5 to "SDP sctpmap attribute", rename=20
> >>>> section 5.1 to "General", and add a new section 5.2 named "ABNF".
> >>>> *Q7 (TECHNICAL):*
> >>>> The text says:
> >>>> "The sctpmap MUST include the app parameter indicating the=20
> >>>> application running on top of the association."
> >>>> This is fine if the sctpmap attribute maps to an=20
> application. But,=20
> >>>> what if the sctpmap attribute maps to a "media encoding"
> >> or "payload"?
> >>>> *Q8 **(EDITORIAL):*
> >>>> The text says:
> >>>> "The sctpmap line should also contain the max-message-size
> >> parameter
> >>>> indicating the maximum message size, in bytes, the endpoint is=20
> >>>> willing to accept."
> >>>> I suggest to replace "accept" with "receive on the=20
> associated SCTP=20
> >>>> association".
> >>>> *Q9 (EDITORIAL):*
> >>>> The text says:
> >>>> "The peer should assume that larger message will be
> >> rejected by the
> >>>> endpoint, though it is up to the endpoint decide the appropriate=20
> >>>> behaviour."
> >>>> I suggest to either remove this paragraph, OR replace it with a=20
> >>>> sentence saying that a peer must/should not send messages
> >> bigger than
> >>>> the indicated max value.
> >>>> *Q10 (CLARIFICATION):*
> >>>> The text says:
> >>>> "If the parameter is not present, the implementation
> >> should provide a
> >>>> default, with a suggested value of 64K."
> >>>> I don't understand "suggested". If the default value is
> >> 64K the text
> >>>> should specify (rather than suggest) that.
> >>>> The same comment applies to the stream parameter,
> >> described later in
> >>>> the section.
> >>>> *Q11 (CLARIFICATION):*
> >>>> The text says:
> >>>> "It may also provide the stream parameter to specify the initial=20
> >>>> number of incoming streams to be supported by each side of the=20
> >>>> association."
> >>>> First, to be consistent with previous text, I suggest to
> >> replace "It
> >>>> may also provide" with "The sctpmap line may also contain"
> >>>> Second, I don't understand "incoming streams to be
> >> supported by each
> >>>> side of the association". Doesn't each peer indicate the=20
> number of=20
> >>>> streams it is willing to receive? Then, we can always
> >> recommend that
> >>>> the peer indicates the same number of streams, if possible.
> >>>> *Q12 (EDITORIAL):*
> >>>> The text that follows the ABNF should be put into a=20
> separate "SDP=20
> >>>> Offer/Answer Procedures" section.
> >>>> *Section 7:*
> >>>> *------------*
> >>>> *Q13 (EDITORIAL):*
> >>>> Please don't talk about re-INVITEs, or other SIP=20
> messages. Simply=20
> >>>> talk about offers and answers.
> >>>> *Q14 (EDITORIAL):*
> >>>> The text says:
> >>>> "It is worth to underline that when using SCTP on top of DTLS,.."
> >>>> To have consistent terminology, please say "SCTP over DTLS".
> >>>> That's all for now! :)
> >>>> Regards,
> >>>> Christer
> >>>> -----Alkuper=E4inen viesti-----
> >>>> L=E4hett=E4j=E4: mmusic [mailto:mmusic-bounces@ietf.org] Puolesta=20
> >>>> internet-drafts@ietf.org
> >>>> L=E4hetetty: 13. helmikuuta 2014 16:21
> >>>> Vastaanottaja: i-d-announce@ietf.org
> >>>> Kopio: mmusic@ietf.org
> >>>> Aihe: [MMUSIC] I-D Action:=20
> draft-ietf-mmusic-sctp-sdp-06.txt A New=20
> >>>> Internet-Draft is available from the on-line Internet-Drafts=20
> >>>> directories.
> >>>> This draft is a work item of the Multiparty Multimedia Session=20
> >>>> Control Working Group of the IETF.
> >>>> Title : Stream Control Transmission Protocol (SCTP)-Based Media=20
> >>>> Transport in the Session Description Protocol (SDP) Authors :
> >>>> Salvatore Loreto Gonzalo Camarillo Filename :
> >>>> draft-ietf-mmusic-sctp-sdp-06.txt Pages : 14 Date : 2014-02-13
> >>>> Abstract:
> >>>> SCTP (Stream Control Transmission Protocol) is a transport
> >> protocol
> >>>> used to establish associations between two endpoints. This
> >> document
> >>>> describes how to express media transport over SCTP in=20
> SDP (Session=20
> >>>> Description Protocol). This document defines the 'SCTP',
> >> 'SCTP/DTLS'
> >>>> and 'DTLS/SCTP' protocol identifiers for SDP.
> >>>> The IETF datatracker status page for this draft is:
> >>>> https://datatracker.ietf.org/doc/draft-ietf-mmusic-sctp-sdp/
> >>>> There's also a htmlized version available at:
> >>>> http://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-06
> >>>> A diff from the previous version is available at:
> >>>> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sctp-sdp-06
> >>>> Please note that it may take a couple of minutes from=20
> the time of=20
> >>>> submission until the htmlized version and diff are available at=20
> >>>> tools.ietf.org.
> >>>> Internet-Drafts are also available by anonymous FTP at:
> >>>> ftp://ftp.ietf.org/internet-drafts/
> >>>> _______________________________________________
> >>>> mmusic mailing list
> >>>> mmusic@ietf.org <mailto:mmusic@ietf.org>=20
> >>>> 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
> >>>
> >>
> >> _______________________________________________
> >> mmusic mailing list
> >> mmusic@ietf.org
> >> https://www.ietf.org/mailman/listinfo/mmusic
> >>
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www.ietf.org/mailman/listinfo/mmusic
> >
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
> =


From nobody Sun Feb 16 15:35:09 2014
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Date: Sun, 16 Feb 2014 18:35:02 -0500
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [MMUSIC] POF/PAN draft update - what is the state model?
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This is one comment about this draft. I've separated it out to keep 
conversation threads distinct.

Many places in this draft refer to attributes of "the SDP". For 
instance, in section 3.2:

    ...  Assuming the recipient does not
    have an outstanding, unanswered partial offer that also modifies that
    exact same stream, this media section is treated as an independent
    renegotiation of that stream.  The SDP version is increased, and an
    answering media section is created for inclusion in the resulting
    partial answer.  ...

In this paragraph, what does "the SDP version is increased" mean? 
Presumably the recipient is retaining some state about the history of 
O/A exchanges, and this means that the state is to be updated. But AFAIK 
this is not explicitly documented. That state might include:

- the last full local SDP sent in an O/A exchange
- the last full remote SDP received in an O/A exchange
- a "current" full local SDP, including PANs sent and POFs sent that
   received good PANs, since the last full local SDP was sent
- a "current" full remote SDP, including PANs received and POFs
   received for which good PANs were sent, since the last full remote
   SDP was received
- some state of POFs outstanding that have not yet received PANs.

Some saved SDP state is identified in rfc3264. There is "previous SDP" 
(of self), and also "previous SDP of the offerer".  A more complex state 
model is needed to support use of POF/PAN, in the procedures of several 
sections.

(I didn't use "previous" in my list above. In order to dovetail with 
3264 I suspect that what I have called "current full local SDP" should 
correspond to "previous SDP" of self, and "current full remote SDP" 
should correspond to "previous SDP of offerer" when a full offer is 
received.)

I think it is important to make a state model explicit, to properly 
support what the procedures are referring to, though the one I sketch 
may not be quite right.

	Thanks,
	Paul


On 2/14/14 6:43 PM, Adam Roach wrote:
> I've updated the partial offer/partial answer draft to reflect the
> conversations we had in Vancouver:
> http://tools.ietf.org/html/draft-roach-mmusic-pof-pan-02
>
> Diffs: http://tools.ietf.org/rfcdiff?url2=draft-roach-mmusic-pof-pan-02.txt
>
> Note that this relies on the following draft for group membership
> manipulation:
> http://tools.ietf.org/html/draft-roach-mmusic-groupid
>
> Please point out whether you think I've missed any of the Vancouver
> discussions or other feedback. Thanks!
>
> /a
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


From nobody Sun Feb 16 15:38:41 2014
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Subject: Re: [MMUSIC] POF/PAN draft update - move BUNDLE-specific stuff to bundle draft
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Another comment on this draft:

While the mechanism described in this draft is strongly motivated by 
BUNDLE, it is conceptually independent of BUNDLE. IMO all the 
BUNDLE-specific behavior should be removed from this draft, and instead 
placed into the BUNDLE draft.

I realize there is a chicken/egg thing here. It would make sense to keep 
the bundle-specifics here, so they don't get forgotten, until they are 
incorporated into the bundle draft.

	Thanks,
	Paul


On 2/14/14 6:43 PM, Adam Roach wrote:
> I've updated the partial offer/partial answer draft to reflect the
> conversations we had in Vancouver:
> http://tools.ietf.org/html/draft-roach-mmusic-pof-pan-02
>
> Diffs: http://tools.ietf.org/rfcdiff?url2=draft-roach-mmusic-pof-pan-02.txt
>
> Note that this relies on the following draft for group membership
> manipulation:
> http://tools.ietf.org/html/draft-roach-mmusic-groupid
>
> Please point out whether you think I've missed any of the Vancouver
> discussions or other feedback. Thanks!
>
> /a
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


From nobody Sun Feb 16 20:49:22 2014
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Subject: Re: [MMUSIC] POF/PAN draft update - what is the state model?
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On 2/16/14 15:35, Paul Kyzivat wrote:
> what does "the SDP version is increased" mean?

Sorry, I mean the <sess-version> from the "o=" line:

    <sess-version> is a version number for this session description.  Its
       usage is up to the creating tool, so long as <sess-version> is
       increased when a modification is made to the session data.


I'll make this more explicit in the next version.

> I think it is important to make a state model explicit, to properly 
> support what the procedures are referring to

That's a very good point, since it will help us identify where we might 
have some holes in how things need to be handled. I'll make a run at 
defining a state diagram for POF/PAN.

/a

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    <div class="moz-cite-prefix">On 2/16/14 15:35, Paul Kyzivat wrote:<br>
    </div>
    <blockquote cite="mid:53014B26.40203@alum.mit.edu" type="cite">what
      does "the SDP version is increased" mean?</blockquote>
    <br>
    Sorry, I mean the &lt;sess-version&gt; from the "o=" line:<br>
    <br>
    <meta http-equiv="content-type" content="text/html; charset=UTF-8">
    <pre class="newpage">   &lt;sess-version&gt; is a version number for this session description.  Its
      usage is up to the creating tool, so long as &lt;sess-version&gt; is
      increased when a modification is made to the session data.</pre>
    <br>
    I'll make this more explicit in the next version.<br>
    <br>
    <blockquote type="cite">I think it is important to make a state
      model explicit, to properly support what the procedures are
      referring to</blockquote>
    <br>
    That's a very good point, since it will help us identify where we
    might have some holes in how things need to be handled. I'll make a
    run at defining a state diagram for POF/PAN.<br>
    <br>
    /a<br>
  </body>
</html>

--------------040408050708020603080208--


From nobody Mon Feb 17 07:31:23 2014
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From: =?iso-8859-1?Q?Ari_Ker=E4nen?= <ari.keranen@ericsson.com>
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Hi all,

The draft agenda for the MMUSIC meeting at IETF 89 is now available:
http://www.ietf.org/proceedings/89/agenda/agenda-89-mmusic

Please have a look and let us know if you have any questions or comments.


Thanks,
Ari & Flemming


From nobody Mon Feb 17 09:08:48 2014
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: =?iso-8859-1?Q?Ari_Ker=E4nen?= <ari.keranen@ericsson.com>, mmusic <mmusic@ietf.org>
Thread-Topic: MMUSIC draft agenda for IETF 89
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The drafts I will be presenting are:

draft-ejzak-mmusic-data-channel-sdpneg-00
draft-ejzak-dispatch-msrp-data-channel-00

i.e. in this order and the sdpneg draft has changed to be mmusic with the o=
mission of webrtc.

Regards

Keith=20

> -----Original Message-----
> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of Ari Ker=E4nen
> Sent: 17 February 2014 15:31
> To: mmusic
> Cc: Flemming Andreasen
> Subject: [MMUSIC] MMUSIC draft agenda for IETF 89
>=20
> Hi all,
>=20
> The draft agenda for the MMUSIC meeting at IETF 89 is now available:
> http://www.ietf.org/proceedings/89/agenda/agenda-89-mmusic
>=20
> Please have a look and let us know if you have any questions=20
> or comments.
>=20
>=20
> Thanks,
> Ari & Flemming
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
> =


From nobody Mon Feb 17 10:23:30 2014
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From: =?iso-8859-1?Q?Ari_Ker=E4nen?= <ari.keranen@ericsson.com>
To: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
Thread-Topic: MMUSIC draft agenda for IETF 89
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Subject: Re: [MMUSIC] MMUSIC draft agenda for IETF 89
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OK, thanks for the info Keith. Fixed now.


Cheers,
Ari

On Feb 17, 2014, at 7:08 PM, DRAGE, Keith (Keith) wrote:

> The drafts I will be presenting are:
>=20
> draft-ejzak-mmusic-data-channel-sdpneg-00
> draft-ejzak-dispatch-msrp-data-channel-00
>=20
> i.e. in this order and the sdpneg draft has changed to be mmusic with the=
 omission of webrtc.
>=20
> Regards
>=20
> Keith=20
>=20
>> -----Original Message-----
>> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of Ari Ker=E4nen
>> Sent: 17 February 2014 15:31
>> To: mmusic
>> Cc: Flemming Andreasen
>> Subject: [MMUSIC] MMUSIC draft agenda for IETF 89
>>=20
>> Hi all,
>>=20
>> The draft agenda for the MMUSIC meeting at IETF 89 is now available:
>> http://www.ietf.org/proceedings/89/agenda/agenda-89-mmusic
>>=20
>> Please have a look and let us know if you have any questions=20
>> or comments.
>>=20
>>=20
>> Thanks,
>> Ari & Flemming
>>=20
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic


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Thread-Topic: MMUSIC draft agenda for IETF 89
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OK, thanks for the info Keith. Fixed now.


Cheers,
Ari

On Feb 17, 2014, at 7:08 PM, DRAGE, Keith (Keith) wrote:

> The drafts I will be presenting are:
>=20
> draft-ejzak-mmusic-data-channel-sdpneg-00
> draft-ejzak-dispatch-msrp-data-channel-00
>=20
> i.e. in this order and the sdpneg draft has changed to be mmusic with the=
 omission of webrtc.
>=20
> Regards
>=20
> Keith=20
>=20
>> -----Original Message-----
>> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of Ari Ker=E4nen
>> Sent: 17 February 2014 15:31
>> To: mmusic
>> Cc: Flemming Andreasen
>> Subject: [MMUSIC] MMUSIC draft agenda for IETF 89
>>=20
>> Hi all,
>>=20
>> The draft agenda for the MMUSIC meeting at IETF 89 is now available:
>> http://www.ietf.org/proceedings/89/agenda/agenda-89-mmusic
>>=20
>> Please have a look and let us know if you have any questions=20
>> or comments.
>>=20
>>=20
>> Thanks,
>> Ari & Flemming
>>=20
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic


From nobody Mon Feb 17 11:09:10 2014
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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: RE: [MMUSIC] Comments on draft-ietf-mmusic-sctp-sdp-06.txt
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Date: Mon, 17 Feb 2014 19:08:45 +0000
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(Sorry for the duplicate post. Posting this again as this post did not get =
any response probably due to posting issues!)



>4.1        a=3Dsctpmap:5000 webrtc-datachannel max-message-size=3D100000 s=
treams=3D16

>9.3        a=3Dsctpmap:5000 webrtc-DataChannel max-message-size=3D100000 s=
treams=3D16

>11.1   'app' parameter and register as first value "webrtc-datachannel" fo=
r



[Raju]

Above three types of representations of "webrtc-datachannel" should use a s=
ingle value;

The value may be based on the outcome from discussion that is going on a di=
fferent thread.





>4.1 a=3Dsctpmap:5000 webrtc-DataChannel max-message-size=3D100000 streams=
=3D16



[Raju] Instead of putting webrtc data channel attributes on the same line (=
which gets long too), How about reusing existing a=3Dfmtp? as in:

a=3Dfmtp:5000 max-message-size=3D100000 streams=3D16

or how about even further shortening it to?

a=3Dfmtp:5000 mms=3D100000; streams=3D16



Then these attributes of 'app' protocol can be defined either in rtcweb-dat=
a-channel

and then IANA (section 11.1) will have a reference to it. Any such attribut=
es defined

for this or any generic 'app' protocol are carried in a=3Dfmtp line transpa=
rently.



Regards

Raju



> -----Original Message-----

> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of internet-

> drafts@ietf.org<mailto:drafts@ietf.org>

> Sent: Thursday, February 13, 2014 8:21 AM

> To: i-d-announce@ietf.org<mailto:i-d-announce@ietf.org>

> Cc: mmusic@ietf.org<mailto:mmusic@ietf.org>

> Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-06.txt

>

>

> A New Internet-Draft is available from the on-line Internet-Drafts

> directories.

>  This draft is a work item of the Multiparty Multimedia Session Control

> Working Group of the IETF.

>

>         Title           : Stream Control Transmission Protocol (SCTP)-Bas=
ed

> Media Transport in the Session Description Protocol (SDP)

>         Authors         : Salvatore Loreto

>                           Gonzalo Camarillo

>     Filename        : draft-ietf-mmusic-sctp-sdp-06.txt

>     Pages           : 14

>     Date            : 2014-02-13

>

> Abstract:

>    SCTP (Stream Control Transmission Protocol) is a transport protocol

>    used to establish associations between two endpoints.  This document

>    describes how to express media transport over SCTP in SDP (Session

>    Description Protocol).  This document defines the 'SCTP', 'SCTP/DTLS'

>    and 'DTLS/SCTP' protocol identifiers for SDP.

>

>

> The IETF datatracker status page for this draft is:

> https://datatracker.ietf.org/doc/draft-ietf-mmusic-sctp-sdp/

>

> There's also a htmlized version available at:

> http://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-06

>

> A diff from the previous version is available at:

> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sctp-sdp-06

>

>

> Please note that it may take a couple of minutes from the time of submiss=
ion

> until the htmlized version and diff are available at tools.ietf.org.

>

> Internet-Drafts are also available by anonymous FTP at:

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

>

> _______________________________________________

> mmusic mailing list

> mmusic@ietf.org<mailto:mmusic@ietf.org>

> https://www.ietf.org/mailman/listinfo/mmusic


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<div class=3D"WordSection1">
<p class=3D"MsoPlainText">(Sorry for the duplicate post. Posting this again=
 as this post did not get any response probably due to posting issues!)<o:p=
></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">&gt;4.1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
 a=3Dsctpmap:5000 webrtc-datachannel max-message-size=3D100000 streams=3D16=
<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;9.3&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
 a=3Dsctpmap:5000 webrtc-DataChannel max-message-size=3D100000 streams=3D16=
<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;11.1&nbsp;&nbsp; 'app' parameter and register=
 as first value &quot;webrtc-datachannel&quot; for<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">[Raju]<o:p></o:p></p>
<p class=3D"MsoPlainText">Above three types of representations of &quot;web=
rtc-datachannel&quot; should use a single value;<o:p></o:p></p>
<p class=3D"MsoPlainText">The value may be based on the outcome from discus=
sion that is going on a different thread.<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">&gt;4.1 a=3Dsctpmap:5000 webrtc-DataChannel max-m=
essage-size=3D100000 streams=3D16<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">[Raju] Instead of putting webrtc data channel att=
ributes on the same line (which gets long too), How about reusing existing =
a=3Dfmtp? as in:<o:p></o:p></p>
<p class=3D"MsoPlainText">a=3Dfmtp:5000 max-message-size=3D100000 streams=
=3D16<o:p></o:p></p>
<p class=3D"MsoPlainText">or how about even further shortening it to?<o:p><=
/o:p></p>
<p class=3D"MsoPlainText">a=3Dfmtp:5000 mms=3D100000; streams=3D16<o:p></o:=
p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">Then these attributes of 'app' protocol can be de=
fined either in rtcweb-data-channel<o:p></o:p></p>
<p class=3D"MsoPlainText">and then IANA (section 11.1) will have a referenc=
e to it. Any such attributes defined
<o:p></o:p></p>
<p class=3D"MsoPlainText">for this or any generic 'app' protocol are carrie=
d in a=3Dfmtp line transparently.<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">Regards<o:p></o:p></p>
<p class=3D"MsoPlainText">Raju<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">&gt; -----Original Message-----<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; From: mmusic [<a href=3D"mailto:mmusic-bounc=
es@ietf.org">mailto:mmusic-bounces@ietf.org</a>] On Behalf Of internet-<o:p=
></o:p></p>
<p class=3D"MsoPlainText">&gt; <a href=3D"mailto:drafts@ietf.org">drafts@ie=
tf.org</a><o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; Sent: Thursday, February 13, 2014 8:21 AM<o:=
p></o:p></p>
<p class=3D"MsoPlainText">&gt; To: <a href=3D"mailto:i-d-announce@ietf.org"=
>i-d-announce@ietf.org</a><o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; Cc: <a href=3D"mailto:mmusic@ietf.org">mmusi=
c@ietf.org</a><o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; Subject: [MMUSIC] I-D Action: draft-ietf-mmu=
sic-sctp-sdp-06.txt<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; <o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; <o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; A New Internet-Draft is available from the o=
n-line Internet-Drafts<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; directories.<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&nbsp; This draft is a work item of the Multi=
party Multimedia Session Control<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; Working Group of the IETF.<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; <o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp; Title&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : Str=
eam Control Transmission Protocol (SCTP)-Based<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; Media Transport in the Session Description P=
rotocol (SDP)<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp; Authors&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : Salvatore Lor=
eto<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Gonzalo Camarillo<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; &nbsp;&nbsp;&nbsp; Filename&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp; : draft-ietf-mmusic-sctp-sdp-06.txt<o:p></o:p></p=
>
<p class=3D"MsoPlainText">&gt; &nbsp;&nbsp;&nbsp; Pages&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 14<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; &nbsp;&nbsp;&nbsp; Date&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 2014-02-13<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; <o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; Abstract:<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&nbsp;&nbsp;&nbsp; SCTP (Stream Control Trans=
mission Protocol) is a transport protocol<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&nbsp;&nbsp;&nbsp; used to establish associat=
ions between two endpoints.&nbsp; This document<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&nbsp;&nbsp;&nbsp; describes how to express m=
edia transport over SCTP in SDP (Session<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&nbsp;&nbsp;&nbsp; Description Protocol).&nbs=
p; This document defines the 'SCTP', 'SCTP/DTLS'<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&nbsp;&nbsp;&nbsp; and 'DTLS/SCTP' protocol i=
dentifiers for SDP.<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; <o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; <o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; The IETF datatracker status page for this dr=
aft is:<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; <a href=3D"https://datatracker.ietf.org/doc/=
draft-ietf-mmusic-sctp-sdp/">
https://datatracker.ietf.org/doc/draft-ietf-mmusic-sctp-sdp/</a><o:p></o:p>=
</p>
<p class=3D"MsoPlainText">&gt; <o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; There's also a htmlized version available at=
:<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; <a href=3D"http://tools.ietf.org/html/draft-=
ietf-mmusic-sctp-sdp-06">
http://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-06</a><o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; <o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; A diff from the previous version is availabl=
e at:<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; <a href=3D"http://www.ietf.org/rfcdiff?url2=
=3Ddraft-ietf-mmusic-sctp-sdp-06">
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sctp-sdp-06</a><o:p></=
o:p></p>
<p class=3D"MsoPlainText">&gt; <o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; <o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; Please note that it may take a couple of min=
utes from the time of submission<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; until the htmlized version and diff are avai=
lable at tools.ietf.org.<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; <o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; Internet-Drafts are also available by anonym=
ous FTP at:<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; <a href=3D"ftp://ftp.ietf.org/internet-draft=
s/">ftp://ftp.ietf.org/internet-drafts/</a><o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; <o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; ____________________________________________=
___<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; mmusic mailing list<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; <a href=3D"mailto:mmusic@ietf.org">mmusic@ie=
tf.org</a><o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; <a href=3D"https://www.ietf.org/mailman/list=
info/mmusic">
https://www.ietf.org/mailman/listinfo/mmusic</a><o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
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From nobody Mon Feb 17 13:26:21 2014
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Date: Mon, 17 Feb 2014 16:26:15 -0500
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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To: Adam Roach <adam@nostrum.com>, mmusic@ietf.org
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Subject: Re: [MMUSIC] POF/PAN draft update - what is the state model?
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On 2/16/14 11:49 PM, Adam Roach wrote:
> On 2/16/14 15:35, Paul Kyzivat wrote:
>> what does "the SDP version is increased" mean?
>
> Sorry, I mean the <sess-version> from the "o=" line:
>
>     <sess-version> is a version number for this session description.  Its
>        usage is up to the creating tool, so long as <sess-version> is
>        increased when a modification is made to the session data.
>
>
> I'll make this more explicit in the next version.
>
>> I think it is important to make a state model explicit, to properly
>> support what the procedures are referring to
>
> That's a very good point, since it will help us identify where we might
> have some holes in how things need to be handled. I'll make a run at
> defining a state diagram for POF/PAN.

I'm less concerned with a state *diagram* than I am in specifying what 
state must be retained and which of it is used for each step. But both 
may be useful.

	Thanks,
	Paul


From nobody Tue Feb 18 06:11:32 2014
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Date: Tue, 18 Feb 2014 08:11:18 -0600
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From: Richard Ejzak <richard.ejzak@gmail.com>
To: "Liuyan (Scarlett)" <scarlett.liuyan@huawei.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, mmusic@ietf.org
Subject: Re: [MMUSIC] [rtcweb] I-D Action: draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00.txt - the data channel's state and the SCTP association
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--089e013c6ae494471104f2aed8b7
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Hi Scarlett,
Please note that this document has been replaced by
draft-ejzak-mmusic-data-channel-sdpneg-00 (this is noted in datatracker)
and that discussion of the topic should be directed to the mmusic list
(although there is some obvious overlap with the data channel discussion
regarding external negotiation).  I copied both groups for this email only
- please respond only to mmusic.

" It is not clear that if the SCTP association is failed to establish, how
to we deal with the newly created data channels.
  Furthermore, how long the valid duration of the connecting state of the
newly created data channels are before the SCTP association has not yet
established ?"

This issue is common to all forms of data channel negotiation and should be
addressed by the core documents rather than this one.  It appears that a
data channel can be in the connecting state indefinitely until the peer
connection is closed.

"When there are multiple data channels with different sub-protocol per a
SCTP association,
Maybe it needs to identify how to close a single data channel, how to close
all data channels with the same sub-protocol and how to close a SCTP
association.
Of course, closing a SCTP association can just set SCTP port number zero.:-)
"

The procedure describes how to close a single data channel.  We did not see
a need to have a special way to close all data channels with the same
sub-protocol since they can just be closed individually.  I don't see a
compelling use case for this, but perhaps you have one?  The document
should point out that data channels terminate with the SCTP association and
peer connection.  Sometimes we forget to state the most obvious things!

BR, Richard


On Mon, Feb 17, 2014 at 5:30 AM, Liuyan (Scarlett) <
scarlett.liuyan@huawei.com> wrote:

> Hi,
>
> Section 5.5.2 does say:
>
>         "o  As described in Section 4.2.2, if the SCTP association is not
> yet
>       established, then the newly created data channels are in the
>       Connecting state, else if the SCTP association is already
>       established, then the newly created data channels are in the Open
>       state."
>   .....
>   It is not clear that if the SCTP association is failed to establish, how
> to we deal with the newly created data channels.
>   Furthermore, how long the valid duration of the connecting state of the
> newly created data channels are before the SCTP association has not yet
> established ?
>
> Section 5.5.3 does say:
>        "5.2.3.  Closing a data channel
>
>       When the application requests the closing of a data channel that was
>       externally negotiated, the browser always performs an in-band SSN
>       reset for this channel.
>
>       It is specific to the sub-protocol whether this closing must in
>       addition be signaled to the peer via a new SDP offer/answer exchange.
>
>       The application must also close any data channel that was externally
>       negotiated, for which the stream identifiers are not listed in an
>       incoming SDP offer."
>    ...
>    When there are multiple data channels with different sub-protocol per a
> SCTP association,
> Maybe it needs to identify how to close a single data channel, how to
> close all data channels with the same sub-protocol and how to close a SCTP
> association.
> Of course, closing a SCTP association can just set SCTP port number
> zero.:-)
>
> Regards,
> Scarlett
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--089e013c6ae494471104f2aed8b7
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<div dir=3D"ltr">Hi Scarlett,<div>Please note that this document has been r=
eplaced by draft-ejzak-mmusic-data-channel-sdpneg-00 (this is noted in data=
tracker) and that discussion of the topic should be directed to the mmusic =
list (although there is some obvious overlap with the data channel discussi=
on regarding external negotiation). =A0I copied both groups for this email =
only - please respond only to mmusic.</div>
<div><br></div><div>&quot;<span style=3D"font-family:arial,sans-serif;font-=
size:13px">=A0</span><span style=3D"font-family:arial,sans-serif;font-size:=
13px">It is not clear that if the SCTP association is failed to establish, =
how to we deal with the newly created data channels.</span></div>
<div><span style=3D"font-family:arial,sans-serif;font-size:13px">=A0 Furthe=
rmore, how long the valid duration of the connecting state of the newly cre=
ated data channels are before the SCTP association has not yet established =
?</span>&quot;</div>
<div><br></div><div>This issue is common to all forms of data channel negot=
iation and should be addressed by the core documents rather than this one. =
=A0It appears that a data channel can be in the connecting state indefinite=
ly until the peer connection is closed.</div>
<div><br></div><div>&quot;<span style=3D"font-family:arial,sans-serif;font-=
size:13px">When there are multiple data channels with different sub-protoco=
l per a SCTP association,</span></div><span style=3D"font-family:arial,sans=
-serif;font-size:13px">Maybe it needs to identify how to close a single dat=
a channel, how to close all data channels with the same sub-protocol and ho=
w to close a SCTP association.</span><br style=3D"font-family:arial,sans-se=
rif;font-size:13px">
<div><span style=3D"font-family:arial,sans-serif;font-size:13px">Of course,=
 closing a SCTP association can just set SCTP port number zero.:-)</span>&q=
uot;</div><div><br></div><div>The procedure describes how to close a single=
 data channel. =A0We did not see a need to have a special way to close all =
data channels with the same sub-protocol since they can just be closed indi=
vidually. =A0I don&#39;t see a compelling use case for this, but perhaps yo=
u have one? =A0The document should point out that data channels terminate w=
ith the SCTP association and peer connection. =A0Sometimes we forget to sta=
te the most obvious things!</div>
<div><br></div><div>BR, Richard=A0</div></div><div class=3D"gmail_extra"><b=
r><br><div class=3D"gmail_quote">On Mon, Feb 17, 2014 at 5:30 AM, Liuyan (S=
carlett) <span dir=3D"ltr">&lt;<a href=3D"mailto:scarlett.liuyan@huawei.com=
" target=3D"_blank">scarlett.liuyan@huawei.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi,<br>
<br>
Section 5.5.2 does say:<br>
<br>
=A0 =A0 =A0 =A0 &quot;o =A0As described in Section 4.2.2, if the SCTP assoc=
iation is not yet<br>
=A0 =A0 =A0 established, then the newly created data channels are in the<br=
>
=A0 =A0 =A0 Connecting state, else if the SCTP association is already<br>
=A0 =A0 =A0 established, then the newly created data channels are in the Op=
en<br>
=A0 =A0 =A0 state.&quot;<br>
=A0 .....<br>
=A0 It is not clear that if the SCTP association is failed to establish, ho=
w to we deal with the newly created data channels.<br>
=A0 Furthermore, how long the valid duration of the connecting state of the=
 newly created data channels are before the SCTP association has not yet es=
tablished ?<br>
<br>
Section 5.5.3 does say:<br>
=A0 =A0 =A0 =A0&quot;5.2.3. =A0Closing a data channel<br>
<br>
=A0 =A0 =A0 When the application requests the closing of a data channel tha=
t was<br>
=A0 =A0 =A0 externally negotiated, the browser always performs an in-band S=
SN<br>
=A0 =A0 =A0 reset for this channel.<br>
<br>
=A0 =A0 =A0 It is specific to the sub-protocol whether this closing must in=
<br>
=A0 =A0 =A0 addition be signaled to the peer via a new SDP offer/answer exc=
hange.<br>
<br>
=A0 =A0 =A0 The application must also close any data channel that was exter=
nally<br>
=A0 =A0 =A0 negotiated, for which the stream identifiers are not listed in =
an<br>
=A0 =A0 =A0 incoming SDP offer.&quot;<br>
=A0 =A0...<br>
=A0 =A0When there are multiple data channels with different sub-protocol pe=
r a SCTP association,<br>
Maybe it needs to identify how to close a single data channel, how to close=
 all data channels with the same sub-protocol and how to close a SCTP assoc=
iation.<br>
Of course, closing a SCTP association can just set SCTP port number zero.:-=
)<br>
<br>
Regards,<br>
Scarlett<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div>

--089e013c6ae494471104f2aed8b7--


From nobody Tue Feb 18 08:23:08 2014
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Jonathan Lennox'" <jonathan@vidyo.com>, "'mmusic'" <mmusic@ietf.org>
References: <10C8325A-F234-4034-BD05-D0D24C0B562B@vidyo.com>
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Date: Tue, 18 Feb 2014 18:22:58 +0200
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Subject: Re: [MMUSIC] MSID and AppID: solving different problems
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Hi,
I agree with Jonathan description about where appId will be used and the
difference from MSID.
I think that a major advantage of appId is not only when the SSRC is not
known in the SDP but it is useful even if can be known in order to define
the SSRC in at the RTP level and not in the SDP level. 

For the SSRC collision case it is simpler to change the SSRC without a need
for re-invite to update the SDP since the appId does not hange but points to
the new SSRC

Roni Even

> -----Original Message-----
> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of Jonathan
> Lennox
> Sent: 13 February, 2014 8:09 PM
> To: mmusic
> Subject: [MMUSIC] MSID and AppID: solving different problems
> 
> The chairs asked that the authors of the MSID and AppID drafts discuss the
> relationship between the two documents.
> 
> My personal opinion (which I mentioned in the RTCWeb Rejecting
> MediaStreamTracks thread, but folks might not have seen) is that they're
> solving different problems, and so are non-conflicting.
> 
> MSID provides an identifier for SDP m= lines.  Its primary use case is
providing
> a Javascript-level handle in WebRTC.
> 
> AppID -- which needs a better name, suggestions welcome -- provides an
> indirection label for identifying individual RTP packet streams in SDP, in
> scenarios where the SSRC is unavailable to the signaling.
> 
> The scenarios we've identified so far for AppID where an SSRC might be
> unavailable are: "early media" race conditions where media outraces an SDP
> answer; SSRC collisions; applications which wish to dynamically switch
which
> SSRC is fulfilling a higher-level application role; and architectures
where the
> signaling stack can't control/doesn't have visibility onto the SSRC
selection.
> 
> AppID should be applicable to anything using BUNDLE, I think, and
certainly
> anything using it with a vaguely Unified-style semantic.  (I think it
would apply
> to a "Plan B"-style semantic, as well, which I know some people are still
> advocating for some non-WebRTC contexts; however, I'm not positive all the
> syntax necessary for that is currently complete.)
> 
> The difference between AppID and MSID should be most clear in scenarios
> where an m= line group has more than one packet stream (my canonical
> example is a primary stream and a retransmission stream).  In this case,
the
> m= line group has one msid attribute, but multiple appid attributes.
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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Subject: Re: [MMUSIC] MSID and AppID: solving different problems
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On 13 February 2014 14:43, Suhas Nandakumar <suhasietf@gmail.com> wrote:
> [Suhas] How about the name SSID (Source Stream ID) to identify the
> individual RTP packet streams

I was thinking that the really useful feature was the label applied by
the receiver, not the source.  Maybe something like "receiver SSRC ID"
(or "receiver SSRC label").  I think that hits the mark.


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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: Tom Kristensen <2mkristensen@gmail.com>
Thread-Topic: [MMUSIC] SDP Directorate: Review of draft-ietf-bfcpbis-rfc4583bis-08
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Subject: Re: [MMUSIC] SDP Directorate: Review of draft-ietf-bfcpbis-rfc4583bis-08
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--_000_CF2A162F1F4C5eckelcuciscocom_
Content-Type: text/plain; charset="Windows-1252"
Content-Transfer-Encoding: quoted-printable

Hi Tom,

The proposed changes look good. One question inline with the rest of the te=
xt omitted.

From: Tom Kristensen <2mkristensen@gmail.com<mailto:2mkristensen@gmail.com>=
>
Date: Thursday, February 13, 2014 at 5:31 AM
To: Charles Eckel <eckelcu@cisco.com<mailto:eckelcu@cisco.com>>
Cc: "Ali C. Begen (abegen)" <abegen@cisco.com<mailto:abegen@cisco.com>>, To=
m Kristensen <tomkrist@cisco.com<mailto:tomkrist@cisco.com>>, "bfcpbis-chai=
rs@tools.ietf.org<mailto:bfcpbis-chairs@tools.ietf.org>" <bfcpbis-chairs@to=
ols.ietf.org<mailto:bfcpbis-chairs@tools.ietf.org>>, "draft-ietf-bfcpbis-rf=
c4583bis@tools.ietf.org<mailto:draft-ietf-bfcpbis-rfc4583bis@tools.ietf.org=
>" <draft-ietf-bfcpbis-rfc4583bis@tools.ietf.org<mailto:draft-ietf-bfcpbis-=
rfc4583bis@tools.ietf.org>>, "mmusic@ietf.org<mailto:mmusic@ietf.org>" <mmu=
sic@ietf.org<mailto:mmusic@ietf.org>>
Subject: Re: [MMUSIC] SDP Directorate: Review of draft-ietf-bfcpbis-rfc4583=
bis-08

On 14 January 2014 03:05, Charles Eckel (eckelcu) <eckelcu@cisco.com<mailto=
:eckelcu@cisco.com>> wrote:
I am afraid Ali is right. Let's take the SHOULDs 1 by 1 and see how to
deal with them.

<snip>

5) There are several SHOULDs in section 7, which defines the new bfcpver
attribute. As this attribute did not exist in RFC 4583, the SHOULDs are
warranted for backward compatibility. However, they can be made mandatory
for implementations of BFCP/UDP as well as those compliant with this
update to RFC 4583.

True, and as proposed in the accompanying BFCPbis WG PROTO review thread me=
rging in this requirement for implementations of the upcoming RFCXXX of thi=
s draft. The paragraph will then read:

  "Endpoints that use the offer/answer model to establish BFCP
   connections SHOULD support the 'bfcpver' attribute. A floor control
   server acting as an offerer or as an answerer SHOULD include
   this attribute in its session descriptions.   However, endpoints that
   support RFCXXXX, and not only the RFC 4583 subset, are
   REQUIRED to support and, when acting as a floor control server
   to use the 'bfcpver' attribute.  [...]"

Why is the condition ", when acting as a floor control server=94 relevant h=
ere?

Cheers,
Charles

--_000_CF2A162F1F4C5eckelcuciscocom_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3DWindows-1=
252">
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space;">
<div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-s=
ize: 14px;">
Hi Tom,</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-s=
ize: 14px;">
<br>
</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-s=
ize: 14px;">
The proposed changes look good. One question inline with the rest of the te=
xt omitted.</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-s=
ize: 14px;">
<br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION" style=3D"color: rgb(0, 0, 0); font-family=
: Calibri, sans-serif; font-size: 14px;">
<div style=3D"font-family:Calibri; font-size:11pt; text-align:left; color:b=
lack; BORDER-BOTTOM: medium none; BORDER-LEFT: medium none; PADDING-BOTTOM:=
 0in; PADDING-LEFT: 0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt solid;=
 BORDER-RIGHT: medium none; PADDING-TOP: 3pt">
<span style=3D"font-weight:bold">From: </span>Tom Kristensen &lt;<a href=3D=
"mailto:2mkristensen@gmail.com">2mkristensen@gmail.com</a>&gt;<br>
<span style=3D"font-weight:bold">Date: </span>Thursday, February 13, 2014 a=
t 5:31 AM<br>
<span style=3D"font-weight:bold">To: </span>Charles Eckel &lt;<a href=3D"ma=
ilto:eckelcu@cisco.com">eckelcu@cisco.com</a>&gt;<br>
<span style=3D"font-weight:bold">Cc: </span>&quot;Ali C. Begen (abegen)&quo=
t; &lt;<a href=3D"mailto:abegen@cisco.com">abegen@cisco.com</a>&gt;, Tom Kr=
istensen &lt;<a href=3D"mailto:tomkrist@cisco.com">tomkrist@cisco.com</a>&g=
t;, &quot;<a href=3D"mailto:bfcpbis-chairs@tools.ietf.org">bfcpbis-chairs@t=
ools.ietf.org</a>&quot;
 &lt;<a href=3D"mailto:bfcpbis-chairs@tools.ietf.org">bfcpbis-chairs@tools.=
ietf.org</a>&gt;, &quot;<a href=3D"mailto:draft-ietf-bfcpbis-rfc4583bis@too=
ls.ietf.org">draft-ietf-bfcpbis-rfc4583bis@tools.ietf.org</a>&quot; &lt;<a =
href=3D"mailto:draft-ietf-bfcpbis-rfc4583bis@tools.ietf.org">draft-ietf-bfc=
pbis-rfc4583bis@tools.ietf.org</a>&gt;,
 &quot;<a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a>&quot; &lt;<a =
href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a>&gt;<br>
<span style=3D"font-weight:bold">Subject: </span>Re: [MMUSIC] SDP Directora=
te: Review of draft-ietf-bfcpbis-rfc4583bis-08<br>
</div>
<div><br>
</div>
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div>
<div>
<div dir=3D"ltr">On 14 January 2014 03:05, Charles Eckel (eckelcu) <span di=
r=3D"ltr">
&lt;<a href=3D"mailto:eckelcu@cisco.com" target=3D"_blank">eckelcu@cisco.co=
m</a>&gt;</span> wrote:<br>
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
I am afraid Ali is right. Let's take the SHOULDs 1 by 1 and see how to<br>
deal with them.<br>
</blockquote>
<div><br>
</div>
<div style=3D"">&lt;snip&gt;</div>
<div>&nbsp;</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
5) There are several SHOULDs in section 7, which defines the new bfcpver<br=
>
attribute. As this attribute did not exist in RFC 4583, the SHOULDs are<br>
warranted for backward compatibility. However, they can be made mandatory<b=
r>
for implementations of BFCP/UDP as well as those compliant with this<br>
update to RFC 4583.<br>
</blockquote>
<div><br>
</div>
<div style=3D"">True, and as proposed in the accompanying BFCPbis WG PROTO =
review thread merging in this requirement for implementations of the upcomi=
ng RFCXXX of this draft. The paragraph will then read:</div>
<div style=3D""><br>
</div>
<div style=3D"">&nbsp; &quot;Endpoints that use the offer/answer model to e=
stablish BFCP</div>
<div style=3D"">&nbsp; &nbsp;connections SHOULD support the 'bfcpver' attri=
bute. A floor control</div>
<div style=3D"">&nbsp; &nbsp;server acting as an offerer or as an answerer =
SHOULD include</div>
<div style=3D"">&nbsp; &nbsp;this attribute in its session descriptions. &n=
bsp; However, endpoints that</div>
<div style=3D"">&nbsp; &nbsp;support RFCXXXX, and not only the RFC 4583 sub=
set, are</div>
<div style=3D"">&nbsp; &nbsp;REQUIRED to support and, when acting as a floo=
r control server</div>
<div style=3D"">&nbsp; &nbsp;to use the 'bfcpver' attribute. &nbsp;[...]&qu=
ot;</div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
</span>
<div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-s=
ize: 14px;">
<br>
</div>
<div><font face=3D"Calibri,sans-serif">Why is the condition &quot;</font>, =
when acting as a floor control server=94 relevant here?</div>
<div><br>
</div>
<div>Cheers,</div>
<div>Charles</div>
</body>
</html>

--_000_CF2A162F1F4C5eckelcuciscocom_--


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Date: Sat, 22 Feb 2014 15:05:12 -0500
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [MMUSIC] Comments on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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Long ago I made some comments on an earlier version. At least some of 
those were adopted in this version. Now I'm reviewing it again. Here are 
new thoughts:

* General comment about document organization:

IMO there is a muddling of responsibilities among several drafts. I 
suggest that it could be improved, as follows. (This isn't a big change 
from the current situation, but rationalizes the layering.):

- draft-ietf-mmusic-sctp-sdp

SDP for declaration/negotiation of SCTP associations.
Defines a=sctpmap and establishes registry for "app" values.

Defines what the registration of an app must specify. (This is currently 
deficient.)

(Shouldn't register any - the registration of "webrtc-datachannel should 
be removed.)

- draft-ietf-rtcweb-data-channel

Should register an "app" name for data channels. (Preferably a more 
generic name than "webrtc-datachannel".)

Should define (or reference an existing) IANA registry for data channel 
protocols. (This currently isn't done anywhere I can find.)

As now, defines internal and external negotiation of data channels. 
(Should probably have what is now section 4 of this (ejzak) draft.)

As now, specifies that "webrtc-datachannel" implies support for internal 
negotiation via rtcweb-data-protocol.

- this draft (draft-ejzak-dispatch-webrtc-data-channel-sdpneg)

Make it clear that it just specifies a way to do external negotiation of 
data channels for "webrtc-datachannel" SCTP associations.

Shouldn't cover any other kinds of SCTP associations. (Except to explain 
how external negotiation of channels can replace use of multiple sctp 
associations.)

* Section 4

This section should probably be moved to draft-ietf-rtcweb-data-channel.

Section 5.1.1:

The 'maxretr', 'maxtime', and 'unordered' parameters are presumably 
intended to correspond to the Channel Type and Reliability Parameter 
elements in section 5.1 of draft-ietf-rtcweb-data-protocol. This should 
be made clear. (And are all Channel Types covered?)

The 'subprotocol' parameter is presumably intended to be the same thing 
that is called "protocol" in sections 5.1 and 8.4 of 
draft-ietf-rtcweb-data-protocol. These ought to be linked and be 
consistent about naming.

Also, then syntax is:

    subprotocol-attr       =  "subprotocol=" labelstring
    labelstring            =  text
    text                   =  byte-string

(Where labelstring and text are the same as in the syntax of the label 
parameter.) This is misleading. It should probably be:

    subprotocol-attr       =  "subprotocol=" protocol-string
    protocol-string        =  text
    text                   =  byte-string

	Thanks,
	Paul







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Subject: [MMUSIC] Comments on draft-ietf-mmusic-sctp-sdp-06
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I've commented earlier on some threads about this draft.
Now I'm going through it carefully again, and have new comments:

* Section 4:

This covers media descriptions using three different <proto> fields: 
'SCTP', 'SCTP/DTLS', and 'DTLS/SCTP'.

For both 'SCTP' and 'SCTP/DTLS' the "bottom" of the protocol stack is 
SCTP, and the identifier of one end for an association consists of an IP 
address and an SCTP port. This fits with m-line syntax - the SCTP port 
number can be carried in the <port> field.

For 'DTLS/SCTP', the "bottom" of the protocol stack is UDP, with SCTP 
over it. In this case the identifier for one end of an association 
consists of an IP address, an SCTP port, and a UDP port. This doesn't 
fit well into the syntax of an m-line.

Because the *UDP* port is carried in the <port> field, for a single 
m-line there can be multiple DTLS/SCTP associations, one per SCTP port 
number. The <fmt> field is repurposed to carry the SCTP port numbers.

Then, the problems caused by DTLS/SCTP are propagated to 'SCTP' and 
'SCTP/DTLS' usages, by also placing the SCTP port number in the <fmt> 
field in these usages. (For these protos it is redundant.)

Because <fmt> is used this way, it was necessary to introduce a=sctpmap 
to specify what otherwise could have been specified using the <fmt> field.

IMO this could be made much clearer by making the following changes:

- an m-line with <proto> of 'SCTP', 'SCTP/DTLS', or 'DTLS/SCTP' always 
describes a *single* SCTP association.

- define the usage of <fmt> for these protos so that it can be used to 
carry the info currently in sctpmap, and in the DTLS/SCTP the SCTP port 
number.

- use BUNDLE for the (unlikely) case of multiple SCTP associations over 
a single UDP port.

It is a little difficult, but possible, to fit they necessary info into 
the restricted syntax of <fmt>. Here is an example of one possibility:

   m=application 12345 DTLS/SCTP webrtc-datachannel \
           sctp-port#1 streams#16 max-message-size#100000

The following is a syntax that does does that:

sctp-m-line = %x6d "="
    ("application" SP sctp-port SP "SCTP"      SP sctp-fmt CRLF) /
    ("application" SP sctp-port SP "SCTP/DTLS" SP sctp-fmt CRLF) /
    ("application" SP udp-port  SP "DTLS/SCTP" SP sctp-fmt CRLF)

sctp-port = port

udp-port = port

sctp-fmt = sctp-app *(SP sctp-param)

sctp-app = token

sctp-param = maxmsg-param / streams-param / sctp-port-param

maxmsg-param = "max-message-size#" port

streams-param = "streams#" 1*DIGIT

sctp-port-param = "sctp-port#" sctp-port

(The use of "#" as separator isn't ideal, but it is the best choice of 
what is permitted in a token, and it seems to read reasonably well. If 
people hate this there are other ways to get the same effect.)

Of course sctp-port-param is only meaningful with DTLS/SCTP. It and the 
other sctp-params can be optional, with suitable defaults defined. (For 
normal use, with only one association per UDP port everyone would 
probably take the default sctp port, and so the parameter could be omitted.)

This then also allows a=fmtp to be used with these - most likely only 
with sctp-app. E.g.,

   m=application 12345 DTLS/SCTP webrtc-datachannel \
           sctp-port#1 streams#16 max-message-size#100000
   c=IN IP4 example.com
   a=fmtp:webrtc-datachannel <whatever>

At the moment nothing webrtc-datachannel specific has been defined that 
needs to be declared in SDP, so this may not be needed. It *could* be 
used as a replacement for "a=dcmap" in 
draft-ejzak-mmusic-data-channel-sdpneg.

Bundling would apply in a case such as in section 10.1 of 
draft-ietf-mmusic-sdp-bundle-negotiation-05. That example would be 
rewritten as follows:

   a=group:BUNDLE dc bfcp t38
   m=application 12345 DTLS/SCTP webrtc-datachannel \
           sctp-port#5000 streams#16 max-message-size#100000
   c=IN IP4 example.com
   a=mid:dc
   m=application 12345 DTLS/SCTP bfcp \
           sctp-port#5001 streams#1 max-message-size#16384
   c=IN IP4 example.com
   a=mid:bfcp

   m=application 12345 DTLS/SCTP t38 \
           sctp-port#5002 streams#1 max-message-size#65536
   c=IN IP4 example.com
   a=mid:t38

* Section 5.1:

This section starts out with:

    The sctpmap attribute maps from a port number (as used in an "m="
    line) to an encoding name denoting the payload format to be used on
    top of the SCTP association or the actual protocol running on top of
    it.

This is just wrong. It doesn't really map to an encoding name denoted a 
payload format. Immediately following the above is:

    The sctpmap MUST include the app parameter indicating the application
    running on top of the association.

And the syntax defines an app parameter, instead of an encoding name. 
And as best I can tell the app parameter doesn't denote a payload 
format. (Payload formats can vary from stream to stream.) The app seems 
to denote the overall set of conventions that are used to ascribe 
meaning and format to the streams. (This needs better definition. 
Section 11.1 has an open issue to define a new IANA registry for this. 
But there is also need to say more about the semantics - what a new 
value must specify.)

Later in this section:

    Any offered association MAY be rejected in the answer, for any
    reason.  If an association offer is rejected, the offerer and
    answerer MUST NOT establish an SCTP association for it.  To reject an
    SCTP association, the SCTP port number in the "a=sctpmap:" attribute
    line in the answer MUST be set to zero.

In the case of DTLS/SCTP there are two ports - the udp port and the sctp 
port. Either can be set to zero, with different consequences. Something 
should be said about the behavior when the udp port is set to zero. (Of 
course my suggestion to revamp the m-line renders this moot.)

* Section 7:

What if the c= line contains a DNS name rather than an IP address?
Can that cause the initialization of more than one IP address?

* Section 8:

This contains the following statements:

    ... Therefore, this
    specification does not define how to describe SCTP-over-UDP streams
    in SDP or how to establish and maintain SCTP associations using ICE.
    Should these features be specified for SCTP in the future, there will
    be a need to specify how to use them in an SDP environment as well.

Doesn't the definition of DTLS/SCTP render the above incorrect?

	Thanks,
	Paul


From nobody Sat Feb 22 19:03:26 2014
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Subject: Re: [MMUSIC] Comments on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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On 2/22/14 3:05 PM, Paul Kyzivat wrote:
> Long ago I made some comments on an earlier version. At least some of
> those were adopted in this version. Now I'm reviewing it again. Here are
> new thoughts:
>
> * General comment about document organization:
>
> IMO there is a muddling of responsibilities among several drafts. I
> suggest that it could be improved, as follows. (This isn't a big change
> from the current situation, but rationalizes the layering.):
>
> - draft-ietf-mmusic-sctp-sdp
>
> SDP for declaration/negotiation of SCTP associations.
> Defines a=sctpmap and establishes registry for "app" values.
>
> Defines what the registration of an app must specify. (This is currently
> deficient.)
>
> (Shouldn't register any - the registration of "webrtc-datachannel should
> be removed.)
>
> - draft-ietf-rtcweb-data-channel
>
> Should register an "app" name for data channels. (Preferably a more
> generic name than "webrtc-datachannel".)
>
> Should define (or reference an existing) IANA registry for data channel
> protocols. (This currently isn't done anywhere I can find.)

My mistake - is currently done by draft-ietf-rtcweb-data-protocol-03. 
But I think it would make more sense to define the registry in 
draft-ietf-rtcweb-data-channel so that it is known to also be applicable 
to externally negotiated channels.

> As now, defines internal and external negotiation of data channels.
> (Should probably have what is now section 4 of this (ejzak) draft.)
>
> As now, specifies that "webrtc-datachannel" implies support for internal
> negotiation via rtcweb-data-protocol.
>
> - this draft (draft-ejzak-dispatch-webrtc-data-channel-sdpneg)
>
> Make it clear that it just specifies a way to do external negotiation of
> data channels for "webrtc-datachannel" SCTP associations.
>
> Shouldn't cover any other kinds of SCTP associations. (Except to explain
> how external negotiation of channels can replace use of multiple sctp
> associations.)
>
> * Section 4
>
> This section should probably be moved to draft-ietf-rtcweb-data-channel.
>
> Section 5.1.1:
>
> The 'maxretr', 'maxtime', and 'unordered' parameters are presumably
> intended to correspond to the Channel Type and Reliability Parameter
> elements in section 5.1 of draft-ietf-rtcweb-data-protocol. This should
> be made clear. (And are all Channel Types covered?)
>
> The 'subprotocol' parameter is presumably intended to be the same thing
> that is called "protocol" in sections 5.1 and 8.4 of
> draft-ietf-rtcweb-data-protocol. These ought to be linked and be
> consistent about naming.
>
> Also, then syntax is:
>
>     subprotocol-attr       =  "subprotocol=" labelstring
>     labelstring            =  text
>     text                   =  byte-string
>
> (Where labelstring and text are the same as in the syntax of the label
> parameter.) This is misleading. It should probably be:
>
>     subprotocol-attr       =  "subprotocol=" protocol-string
>     protocol-string        =  text
>     text                   =  byte-string
>
>      Thanks,
>      Paul
>
>
>
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


From nobody Sun Feb 23 15:13:33 2014
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Date: Sun, 23 Feb 2014 18:13:28 -0500
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: [MMUSIC] Comments on draft-ejzak-dispatch-msrp-data-channel-00
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A few comments/questions on this draft:

Section 5.1.1.2 says:

    The msrp-cema attribute shall be assumed to be present for every MSRP
    session using data channel transport, so the inclusion of the msrp-
    cema attribute is optional.  This ensures that the data channel
    transport for the MSRP session is established without using the path
    attribute.

Why is use of cema mandatory?

RGC6714 has fallback rules for an endpoint that supports cema to 
interoperate with one that does not. That requires removing the 
indication of cema support. How does an msrp data channel endpoint do that?

Section 5.1.3 says:

    Each text-based MSRP message is sent on the corresponding SCTP stream
    using standard MSRP framing and chunking procedures, as defined in
    [RFC4975], with each MSRP chunk delivered in a single SCTP user
    message.

I think it would be better to say:

    Each text-based MSRP message is sent on the corresponding SCTP stream
    using standard MSRP framing and chunking procedures, as defined in
    [RFC4975], with each MSRP chunk delivered in a single Data Channel
    user message.

While data channel user messages map directly onto SCTP user messages, 
they are conceptually distinct.

Section 6 says:

    o  The endpoint establishing an MSRP session using data channel
       transport shall not request inclusion of any relays, although it
       may interoperate with a peer that signals the use of relays.

Why must relays not be requested? That seems the most natural thing to 
do if the goal is to talk to a peer that doesn't support data channels.

In any case, some explanation of how things work with relays would be 
appreciated.

	Thanks,
	Paul


From nobody Sun Feb 23 17:13:13 2014
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--089e0160c16ca41ee304f31cacf7
Content-Type: text/plain; charset=ISO-8859-1

On Sun, Feb 23, 2014 at 5:13 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> A few comments/questions on this draft:
>
> Section 5.1.1.2 says:
>
>    The msrp-cema attribute shall be assumed to be present for every MSRP
>    session using data channel transport, so the inclusion of the msrp-
>    cema attribute is optional.  This ensures that the data channel
>    transport for the MSRP session is established without using the path
>    attribute.
>
> Why is use of cema mandatory?
>

Existing MSRP procedures (without CEMA) are based on using the Path
attribute to establish the transport connection (and essentially ignore the
connection information associated with the m line).  If you are going to
use a data channel, then you have no choice but to use the one established
by the peer connection, so the existing MSRP procedure will NOT WORK.  CEMA
allows you to set up a tls or tcp connection based on the connection
information in the m line rather than in the path attribute, but this is
still not everything needed to make it work with data channels.  But it's a
start.


> RGC6714 has fallback rules for an endpoint that supports cema to
> interoperate with one that does not. That requires removing the indication
> of cema support. How does an msrp data channel endpoint do that?
>

Interoperating with an endpoint that does not support data channel will
require going through a gateway that does.  This gateway will have to do
something to provide a tls or tcp transport connection suitable for the
other peer.  If the gateway can't do the enhanced CEMA procedures described
here, it will have to perform as an MSRP B2BUA.


>
> Section 5.1.3 says:
>
>    Each text-based MSRP message is sent on the corresponding SCTP stream
>    using standard MSRP framing and chunking procedures, as defined in
>    [RFC4975], with each MSRP chunk delivered in a single SCTP user
>    message.
>
> I think it would be better to say:
>
>    Each text-based MSRP message is sent on the corresponding SCTP stream
>    using standard MSRP framing and chunking procedures, as defined in
>    [RFC4975], with each MSRP chunk delivered in a single Data Channel
>    user message.
>
> While data channel user messages map directly onto SCTP user messages,
> they are conceptually distinct.
>

Agreed.


>
> Section 6 says:
>
>    o  The endpoint establishing an MSRP session using data channel
>       transport shall not request inclusion of any relays, although it
>       may interoperate with a peer that signals the use of relays.
>
> Why must relays not be requested? That seems the most natural thing to do
> if the goal is to talk to a peer that doesn't support data channels.
>

The MSRP relay is not typically on the signaling path, so is an unlikely
data channel endpoint unless a separate peer connection and data channel is
established just for MSRP.  This would be silly to do.  We want a gateway
that is on the signaling path to perform any necessary conversion.  The
CEMA gateway model is a much better fit and makes the use of relays
unnecessary (at least on the side of the user using a data channel for
MSRP).


>
> In any case, some explanation of how things work with relays would be
> appreciated.
>

I must say that CEMA is confusing, and I agree that the CEMA parts of the
document are not easy to follow without a thorough understanding of CEMA.
 Since the CEMA RFC already explains all the confusing bits, I'm reluctant
to repeat all of that here...  Perhaps a note that a thorough understanding
of CEMA is necessary background for the CEMA stuff here?

Richard



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

--089e0160c16ca41ee304f31cacf7
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Sun, Feb 23, 2014 at 5:13 PM, Paul Kyzivat <span dir=3D"ltr">&lt=
;<a href=3D"mailto:pkyzivat@alum.mit.edu" target=3D"_blank">pkyzivat@alum.m=
it.edu</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">A few comments/questions on this draft:<br>
<br>
Section 5.1.1.2 says:<br>
<br>
=A0 =A0The msrp-cema attribute shall be assumed to be present for every MSR=
P<br>
=A0 =A0session using data channel transport, so the inclusion of the msrp-<=
br>
=A0 =A0cema attribute is optional. =A0This ensures that the data channel<br=
>
=A0 =A0transport for the MSRP session is established without using the path=
<br>
=A0 =A0attribute.<br>
<br>
Why is use of cema mandatory?<br></blockquote><div><br></div><div>Existing =
MSRP procedures (without CEMA) are based on using the Path attribute to est=
ablish the transport connection (and essentially ignore the connection info=
rmation associated with the m line). =A0If you are going to use a data chan=
nel, then you have no choice but to use the one established by the peer con=
nection, so the existing MSRP procedure will NOT WORK. =A0CEMA allows you t=
o set up a tls or tcp connection based on the connection information in the=
 m line rather than in the path attribute, but this is still not everything=
 needed to make it work with data channels. =A0But it&#39;s a start.</div>
<div><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex=
;border-left:1px #ccc solid;padding-left:1ex">
<br>
RGC6714 has fallback rules for an endpoint that supports cema to interopera=
te with one that does not. That requires removing the indication of cema su=
pport. How does an msrp data channel endpoint do that?<br></blockquote>
<div><br></div><div>Interoperating with an endpoint that does not support d=
ata channel will require going through a gateway that does. =A0This gateway=
 will have to do something to provide a tls or tcp transport connection sui=
table for the other peer. =A0If the gateway can&#39;t do the enhanced CEMA =
procedures described here, it will have to perform as an MSRP B2BUA.</div>
<div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;=
border-left:1px #ccc solid;padding-left:1ex">
<br>
Section 5.1.3 says:<br>
<br>
=A0 =A0Each text-based MSRP message is sent on the corresponding SCTP strea=
m<br>
=A0 =A0using standard MSRP framing and chunking procedures, as defined in<b=
r>
=A0 =A0[RFC4975], with each MSRP chunk delivered in a single SCTP user<br>
=A0 =A0message.<br>
<br>
I think it would be better to say:<br>
<br>
=A0 =A0Each text-based MSRP message is sent on the corresponding SCTP strea=
m<br>
=A0 =A0using standard MSRP framing and chunking procedures, as defined in<b=
r>
=A0 =A0[RFC4975], with each MSRP chunk delivered in a single Data Channel<b=
r>
=A0 =A0user message.<br>
<br>
While data channel user messages map directly onto SCTP user messages, they=
 are conceptually distinct.<br></blockquote><div><br></div><div>Agreed.</di=
v><div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8e=
x;border-left:1px #ccc solid;padding-left:1ex">

<br>
Section 6 says:<br>
<br>
=A0 =A0o =A0The endpoint establishing an MSRP session using data channel<br=
>
=A0 =A0 =A0 transport shall not request inclusion of any relays, although i=
t<br>
=A0 =A0 =A0 may interoperate with a peer that signals the use of relays.<br=
>
<br>
Why must relays not be requested? That seems the most natural thing to do i=
f the goal is to talk to a peer that doesn&#39;t support data channels.<br>=
</blockquote><div><br></div><div>The MSRP relay is not typically on the sig=
naling path, so is an unlikely data channel endpoint unless a separate peer=
 connection and data channel is established just for MSRP. =A0This would be=
 silly to do. =A0We want a gateway that is on the signaling path to perform=
 any necessary conversion. =A0The CEMA gateway model is a much better fit a=
nd makes the use of relays unnecessary (at least on the side of the user us=
ing a data channel for MSRP).</div>
<div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;=
border-left:1px #ccc solid;padding-left:1ex">
<br>
In any case, some explanation of how things work with relays would be appre=
ciated.<br></blockquote><div><br></div><div>I must say that CEMA is confusi=
ng, and I agree that the CEMA parts of the document are not easy to follow =
without a thorough understanding of CEMA. =A0Since the CEMA RFC already exp=
lains all the confusing bits, I&#39;m reluctant to repeat all of that here.=
.. =A0Perhaps a note that a thorough understanding of CEMA is necessary bac=
kground for the CEMA stuff here?</div>
<div><br></div><div>Richard</div><div><br></div><div><br></div><blockquote =
class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid=
;padding-left:1ex">
<br>
=A0 =A0 =A0 =A0 Thanks,<br>
=A0 =A0 =A0 =A0 Paul<br>
<br>
______________________________<u></u>_________________<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/<u></u>listinfo/mmusic</a><br>
</blockquote></div><br></div></div>

--089e0160c16ca41ee304f31cacf7--


From nobody Mon Feb 24 09:36:16 2014
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Subject: [MMUSIC] Protocol Action: 'Session Description Protocol (SDP) Extension For Setting Audio and Video Media Streams Over Circuit-Switched Bearers In The Public Switched Telephone Network (PSTN)' to Proposed Standard (draft-ietf-mmusic-sdp-cs-23.txt)
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The IESG has approved the following document:
- 'Session Description Protocol (SDP) Extension For Setting Audio and
   Video Media Streams Over Circuit-Switched Bearers In The Public
   Switched Telephone Network (PSTN)'
  (draft-ietf-mmusic-sdp-cs-23.txt) as Proposed Standard

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

The IESG contact persons are Gonzalo Camarillo and Richard Barnes.

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




*Technical Summary
*

The document 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 Publich Switched Telephone Network (PSTN)

*Working Group Summary
*

The WG had some discussion around the format to use for E.164 numbers
and whether to align this with the existing definition in RFC 3108. The
RFC 3108 definition was seen as deficient and the WG agreed it was
better to align with relevant parts of the tel URI format defined in RFC
3966, not least since SDP address types are defined in the context of a
particular network type, and hence RFC 3108 compatibility is not a
concern (the implication is that the "E164" address type may differ
between network types in SDP).

*Document Quality
*

There are currently no known implementations of the draft, however the
draft is a dependency for 3GPP, so future implementations are expected.

The document has received good overall review in the WG, some of which
resulted in changes to the detailed specification. The document has been
reviewed in detail several times, including of the last few versions.
The major contributors to these as well as earlier discussions are
listed in the Acknowledgements section of the document.


*Personnel
*

Document Shepherd: Flemming Andreasen
Responsible AD: Gonzalo Camarillo 


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From: Salvatore Loreto <salvatore.loreto@ericsson.com>
To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
Thread-Topic: [MMUSIC] Comments on draft-ietf-mmusic-sctp-sdp-06.txt
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Date: Mon, 24 Feb 2014 19:51:14 +0000
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On Feb 17, 2014, at 9:08 PM, "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@=
alcatel-lucent.com<mailto:Raju.Makaraju@alcatel-lucent.com>> wrote:

(Sorry for the duplicate post. Posting this again as this post did not get =
any response probably due to posting issues!)

>4.1        a=3Dsctpmap:5000 webrtc-datachannel max-message-size=3D100000 s=
treams=3D16
>9.3        a=3Dsctpmap:5000 webrtc-DataChannel max-message-size=3D100000 s=
treams=3D16
>11.1   'app' parameter and register as first value "webrtc-datachannel" fo=
r

[Raju]
Above three types of representations of "webrtc-datachannel" should use a s=
ingle value;
The value may be based on the outcome from discussion that is going on a di=
fferent thread.

you are absolutely right,
those are editorial typo I will address in the next version of the document



>4.1 a=3Dsctpmap:5000 webrtc-DataChannel max-message-size=3D100000 streams=
=3D16

[Raju] Instead of putting webrtc data channel attributes on the same line (=
which gets long too), How about reusing existing a=3Dfmtp? as in:
a=3Dfmtp:5000 max-message-size=3D100000 streams=3D16
or how about even further shortening it to?
a=3Dfmtp:5000 mms=3D100000; streams=3D16

I have receive a similar comment before
however my reading of RFC4566 is that a=3Dfmtp is defined only for specifyi=
ng format parameter for a=3Drtpmap:

if we want to use the same a=3Dfmtp for the a=3Dsctpmap it is necessary to =
to specify this usage within the draft-ietf-mmusic-sctp-sdp-05,
I don't have any problem to do so=85 if I see consensus on doing so.

br
Salvatore


Then these attributes of 'app' protocol can be defined either in rtcweb-dat=
a-channel
and then IANA (section 11.1) will have a reference to it. Any such attribut=
es defined
for this or any generic 'app' protocol are carried in a=3Dfmtp line transpa=
rently.

Regards
Raju

> -----Original Message-----
> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of internet-
> drafts@ietf.org<mailto:drafts@ietf.org>
> Sent: Thursday, February 13, 2014 8:21 AM
> To: i-d-announce@ietf.org<mailto:i-d-announce@ietf.org>
> Cc: mmusic@ietf.org<mailto:mmusic@ietf.org>
> Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-06.txt
>
>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>  This draft is a work item of the Multiparty Multimedia Session Control
> Working Group of the IETF.
>
>         Title           : Stream Control Transmission Protocol (SCTP)-Bas=
ed
> Media Transport in the Session Description Protocol (SDP)
>         Authors         : Salvatore Loreto
>                           Gonzalo Camarillo
>     Filename        : draft-ietf-mmusic-sctp-sdp-06.txt
>     Pages           : 14
>     Date            : 2014-02-13
>
> Abstract:
>    SCTP (Stream Control Transmission Protocol) is a transport protocol
>    used to establish associations between two endpoints.  This document
>    describes how to express media transport over SCTP in SDP (Session
>    Description Protocol).  This document defines the 'SCTP', 'SCTP/DTLS'
>    and 'DTLS/SCTP' protocol identifiers for SDP.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-mmusic-sctp-sdp/
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-06
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sctp-sdp-06
>
>
> Please note that it may take a couple of minutes from the time of submiss=
ion
> until the htmlized version and diff are available at tools.ietf.org<http:=
//tools.ietf.org>.
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org<mailto:mmusic@ietf.org>
> https://www.ietf.org/mailman/listinfo/mmusic

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--_000_FC95563C0BB6487892625D4E8AA15F8Dericssoncom_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3DWindows-1=
252">
<base href=3D"x-msg://65/">
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; ">
<br>
<div>
<div>On Feb 17, 2014, at 9:08 PM, &quot;Makaraju, Maridi Raju (Raju)&quot; =
&lt;<a href=3D"mailto:Raju.Makaraju@alcatel-lucent.com">Raju.Makaraju@alcat=
el-lucent.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: He=
lvetica; font-size: medium; font-style: normal; font-variant: normal; font-=
weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; te=
xt-align: -webkit-auto; text-indent: 0px; text-transform: none; white-space=
: normal; widows: 2; word-spacing: 0px; -webkit-text-size-adjust: auto; -we=
bkit-text-stroke-width: 0px; ">
<div class=3D"WordSection1" style=3D"page: WordSection1; ">
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
(Sorry for the duplicate post. Posting this again as this post did not get =
any response probably due to posting issues!)<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
<o:p>&nbsp;</o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt;4.1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=3Dsctpmap:5000 webrtc-d=
atachannel max-message-size=3D100000 streams=3D16<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt;9.3&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=3Dsctpmap:5000 webrtc-D=
ataChannel max-message-size=3D100000 streams=3D16<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt;11.1&nbsp;&nbsp; 'app' parameter and register as first value &quot;webr=
tc-datachannel&quot; for<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
<o:p>&nbsp;</o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
[Raju]<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
Above three types of representations of &quot;webrtc-datachannel&quot; shou=
ld use a single value;<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
The value may be based on the outcome from discussion that is going on a di=
fferent thread.</div>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>you are absolutely right,&nbsp;</div>
<div>those are editorial typo I will address in the next version of the doc=
ument</div>
<br>
<blockquote type=3D"cite">
<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple" style=3D"font-family: He=
lvetica; font-size: medium; font-style: normal; font-variant: normal; font-=
weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; te=
xt-align: -webkit-auto; text-indent: 0px; text-transform: none; white-space=
: normal; widows: 2; word-spacing: 0px; -webkit-text-size-adjust: auto; -we=
bkit-text-stroke-width: 0px; ">
<div class=3D"WordSection1" style=3D"page: WordSection1; ">
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
<o:p>&nbsp;</o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
<o:p>&nbsp;</o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt;4.1 a=3Dsctpmap:5000 webrtc-DataChannel max-message-size=3D100000 strea=
ms=3D16<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
<o:p>&nbsp;</o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
[Raju] Instead of putting webrtc data channel attributes on the same line (=
which gets long too), How about reusing existing a=3Dfmtp? as in:<o:p></o:p=
></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
a=3Dfmtp:5000 max-message-size=3D100000 streams=3D16<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
or how about even further shortening it to?<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
a=3Dfmtp:5000 mms=3D100000; streams=3D16</div>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>I have receive a similar comment before</div>
<div>however my reading of RFC4566 is that a=3Dfmtp is defined only for spe=
cifying format parameter for&nbsp;a=3Drtpmap:
<div><br>
</div>
<div>if we want to use the same a=3Dfmtp for the a=3Dsctpmap it is necessar=
y to&nbsp;to specify this usage within&nbsp;the&nbsp;draft-ietf-mmusic-sctp=
-sdp-05,</div>
<div>I don't have any problem to do so=85 if I see consensus on doing so.</=
div>
<div><br>
</div>
<div>br</div>
<div>Salvatore</div>
</div>
<br>
<blockquote type=3D"cite">
<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple" style=3D"font-family: He=
lvetica; font-size: medium; font-style: normal; font-variant: normal; font-=
weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; te=
xt-align: -webkit-auto; text-indent: 0px; text-transform: none; white-space=
: normal; widows: 2; word-spacing: 0px; -webkit-text-size-adjust: auto; -we=
bkit-text-stroke-width: 0px; ">
<div class=3D"WordSection1" style=3D"page: WordSection1; ">
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
<o:p>&nbsp;</o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
Then these attributes of 'app' protocol can be defined either in rtcweb-dat=
a-channel<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
and then IANA (section 11.1) will have a reference to it. Any such attribut=
es defined<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
for this or any generic 'app' protocol are carried in a=3Dfmtp line transpa=
rently.<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
<o:p>&nbsp;</o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
Regards<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
Raju<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
<o:p>&nbsp;</o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt; -----Original Message-----<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt; From: mmusic [<a href=3D"mailto:mmusic-bounces@ietf.org" style=3D"colo=
r: purple; text-decoration: underline; ">mailto:mmusic-bounces@ietf.org</a>=
] On Behalf Of internet-<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt;<span class=3D"Apple-converted-space">&nbsp;</span><a href=3D"mailto:dr=
afts@ietf.org" style=3D"color: purple; text-decoration: underline; ">drafts=
@ietf.org</a><o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt; Sent: Thursday, February 13, 2014 8:21 AM<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt; To:<span class=3D"Apple-converted-space">&nbsp;</span><a href=3D"mailt=
o:i-d-announce@ietf.org" style=3D"color: purple; text-decoration: underline=
; ">i-d-announce@ietf.org</a><o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt; Cc:<span class=3D"Apple-converted-space">&nbsp;</span><a href=3D"mailt=
o:mmusic@ietf.org" style=3D"color: purple; text-decoration: underline; ">mm=
usic@ietf.org</a><o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt; Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-06.txt<o:p></=
o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt;<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt;<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt; A New Internet-Draft is available from the on-line Internet-Drafts<o:p=
></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt; directories.<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt;&nbsp; This draft is a work item of the Multiparty Multimedia Session C=
ontrol<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt; Working Group of the IETF.<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt;<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Title&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : Stream Control Transmission P=
rotocol (SCTP)-Based<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt; Media Transport in the Session Description Protocol (SDP)<o:p></o:p></=
div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Authors&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : Salvatore Loreto<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp; Gonzalo Camarillo<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt; &nbsp;&nbsp;&nbsp; Filename&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
: draft-ietf-mmusic-sctp-sdp-06.txt<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt; &nbsp;&nbsp;&nbsp; Pages&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp; : 14<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt; &nbsp;&nbsp;&nbsp; Date&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp; : 2014-02-13<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt;<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt; Abstract:<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt;&nbsp;&nbsp;&nbsp; SCTP (Stream Control Transmission Protocol) is a tra=
nsport protocol<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt;&nbsp;&nbsp;&nbsp; used to establish associations between two endpoints=
.&nbsp; This document<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt;&nbsp;&nbsp;&nbsp; describes how to express media transport over SCTP i=
n SDP (Session<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt;&nbsp;&nbsp;&nbsp; Description Protocol).&nbsp; This document defines t=
he 'SCTP', 'SCTP/DTLS'<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt;&nbsp;&nbsp;&nbsp; and 'DTLS/SCTP' protocol identifiers for SDP.<o:p></=
o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt;<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt;<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt; The IETF datatracker status page for this draft is:<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt;<span class=3D"Apple-converted-space">&nbsp;</span><a href=3D"https://d=
atatracker.ietf.org/doc/draft-ietf-mmusic-sctp-sdp/" style=3D"color: purple=
; text-decoration: underline; ">https://datatracker.ietf.org/doc/draft-ietf=
-mmusic-sctp-sdp/</a><o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt;<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt; There's also a htmlized version available at:<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt;<span class=3D"Apple-converted-space">&nbsp;</span><a href=3D"http://to=
ols.ietf.org/html/draft-ietf-mmusic-sctp-sdp-06" style=3D"color: purple; te=
xt-decoration: underline; ">http://tools.ietf.org/html/draft-ietf-mmusic-sc=
tp-sdp-06</a><o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt;<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt; A diff from the previous version is available at:<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt;<span class=3D"Apple-converted-space">&nbsp;</span><a href=3D"http://ww=
w.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sctp-sdp-06" style=3D"color: pu=
rple; text-decoration: underline; ">http://www.ietf.org/rfcdiff?url2=3Ddraf=
t-ietf-mmusic-sctp-sdp-06</a><o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt;<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt;<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt; Please note that it may take a couple of minutes from the time of subm=
ission<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt; until the htmlized version and diff are available at<span class=3D"App=
le-converted-space">&nbsp;</span><a href=3D"http://tools.ietf.org" style=3D=
"color: purple; text-decoration: underline; ">tools.ietf.org</a>.<o:p></o:p=
></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt;<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt; Internet-Drafts are also available by anonymous FTP at:<o:p></o:p></di=
v>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt;<span class=3D"Apple-converted-space">&nbsp;</span><a href=3D"ftp://ftp=
.ietf.org/internet-drafts/" style=3D"color: purple; text-decoration: underl=
ine; ">ftp://ftp.ietf.org/internet-drafts/</a><o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt;<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt; _______________________________________________<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt; mmusic mailing list<o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt;<span class=3D"Apple-converted-space">&nbsp;</span><a href=3D"mailto:mm=
usic@ietf.org" style=3D"color: purple; text-decoration: underline; ">mmusic=
@ietf.org</a><o:p></o:p></div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 10.5pt; font-family: Con=
solas; ">
&gt;<span class=3D"Apple-converted-space">&nbsp;</span><a href=3D"https://w=
ww.ietf.org/mailman/listinfo/mmusic" style=3D"color: purple; text-decoratio=
n: underline; ">https://www.ietf.org/mailman/listinfo/mmusic</a><o:p></o:p>=
</div>
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 11pt; font-family: Calib=
ri, sans-serif; ">
<o:p>&nbsp;</o:p></div>
</div>
_______________________________________________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org" style=3D"color: purple; text-decoration:=
 underline; ">mmusic@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" style=3D"color: pu=
rple; text-decoration: underline; ">https://www.ietf.org/mailman/listinfo/m=
music</a><br>
</div>
</blockquote>
</div>
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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Salvatore Loreto <salvatore.loreto@ericsson.com>
Thread-Topic: [MMUSIC] Comments on draft-ietf-mmusic-sctp-sdp-06.txt
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>4.1 a=3Dsctpmap:5000 webrtc-DataChannel max-message-size=3D100000 streams=
=3D16

[Raju] Instead of putting webrtc data channel attributes on the same line (=
which gets long too), How about reusing existing a=3Dfmtp? as in:
a=3Dfmtp:5000 max-message-size=3D100000 streams=3D16
or how about even further shortening it to?
a=3Dfmtp:5000 mms=3D100000; streams=3D16

I have receive a similar comment before
however my reading of RFC4566 is that a=3Dfmtp is defined only for specifyi=
ng format parameter for a=3Drtpmap:

if we want to use the same a=3Dfmtp for the a=3Dsctpmap it is necessary to =
to specify this usage within the draft-ietf-mmusic-sctp-sdp-05,
I don't have any problem to do so... if I see consensus on doing so.
[Raju] This draft followed same m=3D line syntax for SCTP. Similr to RTP co=
dec formats, SCTP has "port #s" as formats. SDP RFC 4566 Section 6<http://t=
ools.ietf.org/html/rfc4566#section-6> http://tools.ietf.org/html/rfc4566#se=
ction-6 does allow additional attributes to be specified for these formats.=
 The text there does not say they are for RTP only. So, IMHO, we can use a=
=3Dfmtp for data channel attributes.

<RFC 4566 Section 6<http://tools.ietf.org/html/rfc4566#section-6> http://to=
ols.ietf.org/html/rfc4566#section-6>


    m=3D<media> <port> <proto> <fmt> ...
...
      a=3Dfmtp:<format> <format specific parameters>

         This attribute allows parameters that are specific to a
         particular format to be conveyed in a way that SDP does not
         have to understand them.  The format must be one of the formats
         specified for the media.  Format-specific parameters may be any
      set of parameters required to be conveyed by SDP and given
   unchanged to the media tool that will use this format.  At most
      one instance of this attribute is allowed for each format.

</RFC 4566 Section 6<http://tools.ietf.org/html/rfc4566#section-6> http://t=
ools.ietf.org/html/rfc4566#section-6>

Resulting example look like this; which seem to fit very naturally and same=
 look and feel as RTP case with the only difference being "rtpmap" is repla=
ced, appropriately, with "sctpmap".

m=3Dapplication 54111 DTLS/SCTP 5000
c=3DIN IP4 79.97.215.79
a=3Dsctpmap:5000 webrtc-datachannel
a=3Dfmtp:5000 max-message-size=3D100000 streams=3D16

-Raju




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<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:Consolas=
;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:Consolas=
">&nbsp;<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:Consolas=
">&gt;4.1 a=3Dsctpmap:5000 webrtc-DataChannel max-message-size=3D100000 str=
eams=3D16<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:Consolas=
">&nbsp;<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:Consolas=
">[Raju] Instead of putting webrtc data channel attributes on the same line=
 (which gets long too), How about reusing existing a=3Dfmtp? as in:<o:p></o=
:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:Consolas=
">a=3Dfmtp:5000 max-message-size=3D100000 streams=3D16<o:p></o:p></span></p=
>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:Consolas=
">or how about even further shortening it to?<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:Consolas=
">a=3Dfmtp:5000 mms=3D100000; streams=3D16<o:p></o:p></span></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">I have receive a similar comment before<o:p></o:p></=
p>
</div>
<div>
<p class=3D"MsoNormal">however my reading of RFC4566 is that a=3Dfmtp is de=
fined only for specifying format parameter for&nbsp;a=3Drtpmap:
<o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">if we want to use the same a=3Dfmtp for the a=3Dsctp=
map it is necessary to&nbsp;to specify this usage within&nbsp;the&nbsp;draf=
t-ietf-mmusic-sctp-sdp-05,<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">I don't have any problem to do so&#8230; if I see co=
nsensus on doing so.<o:p></o:p></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">[Raju] This draft f=
ollowed same m=3D line syntax for SCTP. Similr to RTP codec formats, SCTP h=
as &#8220;port #s&#8221; as formats. SDP
<a href=3D"http://tools.ietf.org/html/rfc4566#section-6">RFC 4566 Section 6=
</a> <a href=3D"http://tools.ietf.org/html/rfc4566#section-6">
http://tools.ietf.org/html/rfc4566#section-6</a> does allow additional attr=
ibutes to be specified for these formats. The text there does not say they =
are for RTP only. So, IMHO, we can use a=3Dfmtp for data channel attributes=
.<o:p></o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></=
span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&lt;<a href=3D"http=
://tools.ietf.org/html/rfc4566#section-6">RFC 4566 Section 6</a>
<a href=3D"http://tools.ietf.org/html/rfc4566#section-6">http://tools.ietf.=
org/html/rfc4566#section-6</a>&gt;<o:p></o:p></span></i></b></p>
<pre style=3D"page-break-before:always"><span style=3D"font-size:12.0pt;col=
or:black"><o:p>&nbsp;</o:p></span></pre>
<p class=3D"MsoNormal" style=3D"margin-left:10.3pt"><b><i><span style=3D"fo=
nt-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color=
:#1F497D">&nbsp; &nbsp;&nbsp;m=3D&lt;media&gt; &lt;port&gt; &lt;proto&gt; &=
lt;</span></i></b><b><i><span style=3D"font-size:11.0pt;font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;;color:red">fmt</span></i></b><b><i><spa=
n style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-seri=
f&quot;;color:#1F497D">&gt;
 ...<o:p></o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&#8230;<o:p></o:p><=
/span></i></b></p>
<p class=3D"MsoNormal" style=3D"margin-left:5.15pt"><b><i><span style=3D"fo=
nt-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color=
:#1F497D">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=3Dfmtp:&lt;</span></i></b><b><i>=
<span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-=
serif&quot;;color:red">format</span></i></b><b><i><span style=3D"font-size:=
11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D=
">&gt;
 &lt;format specific parameters&gt;<o:p></o:p></span></i></b></p>
<p class=3D"MsoNormal" style=3D"margin-left:5.15pt"><b><i><span style=3D"fo=
nt-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color=
:#1F497D"><o:p>&nbsp;</o:p></span></i></b></p>
<p class=3D"MsoNormal" style=3D"margin-left:5.15pt"><b><i><span style=3D"fo=
nt-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color=
:#1F497D">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; This attribute a=
llows parameters that are specific to a<o:p></o:p></span></i></b></p>
<p class=3D"MsoNormal" style=3D"margin-left:5.15pt"><b><i><span style=3D"fo=
nt-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color=
:#1F497D">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; particular forma=
t to be conveyed in a way that SDP does not<o:p></o:p></span></i></b></p>
<p class=3D"MsoNormal" style=3D"margin-left:5.15pt"><b><i><span style=3D"fo=
nt-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color=
:#1F497D">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; have to understa=
nd them.&nbsp; The format must be one of the formats<o:p></o:p></span></i><=
/b></p>
<p class=3D"MsoNormal" style=3D"margin-left:5.15pt"><b><i><span style=3D"fo=
nt-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color=
:#1F497D">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; specified for th=
e media.&nbsp; Format-specific parameters may be any<o:p></o:p></span></i><=
/b></p>
<p class=3D"MsoNormal" style=3D"margin-left:10.3pt"><b><i><span style=3D"fo=
nt-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color=
:#1F497D">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; set of parameters required to be c=
onveyed by SDP and given</span></i></b><span style=3D"color:black">
<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-left:10.5pt;text-indent:5.05pt"><b><=
i><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;san=
s-serif&quot;;color:#1F497D">&nbsp;&nbsp;&nbsp;unchanged to the media tool =
that will use this format.&nbsp; At most<o:p></o:p></span></i></b></p>
<p class=3D"MsoNormal" style=3D"margin-left:10.3pt"><b><i><span style=3D"fo=
nt-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color=
:#1F497D">&nbsp;&nbsp;&nbsp;&nbsp; &nbsp;one instance of this attribute is =
allowed for each format.<o:p></o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></=
span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&lt;/<a href=3D"htt=
p://tools.ietf.org/html/rfc4566#section-6">RFC 4566 Section 6</a>
<a href=3D"http://tools.ietf.org/html/rfc4566#section-6">http://tools.ietf.=
org/html/rfc4566#section-6</a>&gt;<o:p></o:p></span></i></b></p>
<p class=3D"MsoNormal" style=3D"margin-left:15.55pt"><b><i><span style=3D"f=
ont-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;colo=
r:#1F497D"><o:p>&nbsp;</o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Resulting example l=
ook like this; which seem to fit very naturally and same look and feel as R=
TP case with the only difference being &#8220;rtpmap&#8221; is replaced,
 appropriately, with &#8220;sctpmap&#8221;.<o:p></o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></=
span></i></b></p>
<p class=3D"MsoNormal" style=3D"margin-left:5.15pt"><b><i><span style=3D"fo=
nt-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color=
:#1F497D">m=3Dapplication 54111 DTLS/SCTP 5000<o:p></o:p></span></i></b></p=
>
<p class=3D"MsoNormal" style=3D"margin-left:5.15pt"><b><i><span style=3D"fo=
nt-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color=
:#1F497D">c=3DIN IP4 79.97.215.79<o:p></o:p></span></i></b></p>
<p class=3D"MsoNormal" style=3D"margin-left:5.15pt"><b><i><span style=3D"fo=
nt-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color=
:#1F497D">a=3Dsctpmap:5000 webrtc-datachannel
<o:p></o:p></span></i></b></p>
<p class=3D"MsoNormal" style=3D"margin-left:5.15pt"><b><i><span style=3D"fo=
nt-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color=
:#1F497D">a=3Dfmtp:5000 max-message-size=3D100000 streams=3D16<o:p></o:p></=
span></i></b></p>
<p class=3D"MsoNormal" style=3D"page-break-before:always"><span style=3D"fo=
nt-family:&quot;Courier New&quot;,&quot;serif&quot;;color:black"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span class=3D"MsoHyperlink"><b><i><span style=3D"fo=
nt-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">-Raj=
u<o:p></o:p></span></i></b></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>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</body>
</html>

--_000_E1FE4C082A89A246A11D7F32A95A17826DFF5C02US70UWXCHMBA02z_--


From nobody Mon Feb 24 15:02:04 2014
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Date: Mon, 24 Feb 2014 18:01:59 -0500
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: [MMUSIC] Comments on draft-even-mmusic-application-token-02
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This is much improved from prior versions I reviewed!!!

General comments:

I think this would be easier to read if in examples mnemonic values were 
used for appID tokes, rather than numbers.

Now that recv-appId is being proposed, I think it would read better if 
there was symmetry in the names for the two directions. Also this mixing 
of hyphenated names with camelCase names is pretty ugly. So how about: 
app-id-send, app-id-recv?

Section 3.1:

    The definition is

    a=appId:token

    a=appId:token <attribute>

    a=appId:token <attribute>:<value>

    The formal representation of the appId token is:
       appId-attribute = "appId:" token *[WSP attribute]
       attribute =/ appId-attr
       ; The base definition of "attribute" is in [RFC4566].
       ; (It is the content of "a=" lines.)
       ; WSP and DIGIT defined in [RFC5234]
       ; token defined in [RFC4566]

The examples above are a bit misleading because the definition of 
'attribute' includes the value. Also the above defines appId-attribute 
but then uses appId-attr.

I think the following would be clearer:

    The definition is

    a=appId:token

    a=appId:token <att-field>

    a=appId:token <att-field>:<att-value>

    The formal representation of the appId token is:
       appId-attribute = "appId:" token *[WSP attribute]
       attribute =/ appId-attribute
       ; The base definition of "attribute" is in [RFC4566].
       ; (It is the content of "a=" lines.)
       ; WSP and DIGIT defined in [RFC5234]
       ; token defined in [RFC4566]
       ; att-field and att-value defined in [RFC4566]

Section 3.1.3:

The syntax here is messed up. I think you mean:

    The formal representation of recv-appId is:
       recv-appId-attribute = "recv-appId:" token
       attribute =/ appId-attr
       ; The base definition of "attribute" is in [RFC4566].
       ; (It is the content of "a=" lines.)
       ; token defined in [RFC4566]

Also, if there will be multiple appIds per m-line, with different roles, 
then won't the receiver care which role is assigned to each of the 
appIds it requests? How can that be specified? (Maybe the pt attribute 
specified later, but then this needs to be extended to allow that.)

Section 3.2:

    Editor note: Since the appId is unique in an SDP session the app-Id
    group can be used also at the session level - do we want it?

You say that an appID is unique to an SDP session. But must it not also 
be unique to an RTP session? If so, that seems like it could present 
problems - the two constraints might conflict for certain types of RTP 
middle boxes.

Section 3.3.1:

I'm confused. IIUC the appId attribute in an offer or answer defines the 
appIds that will be *sent* by the end sending that offer or answer. But 
the pt numbers in the corresponding m-line define the PTs that it will 
*receive*.

Section 4:

This is just an example of O/A. More is expected. Clear normative 
specifications, e.g.:

Generating the Initial Offer
Receiving the Initial Offer
Generating the Initial Answer
Receiving the Initial Answer
Subsequent Offer/Answer Exchanges:
   Generating the Offer
   Receiving the Offer
   Generating the Answer
   Receiving the Answer

Some things that need explanation:

If recv-appId is used in an offer, that can lead to an answer that uses 
the same values in an appId. But can recv-appId be used in an answer? If 
so, what does that mean? What about the *next* O/A? Should the 
recv-appId from the prior O/A be repeated in the new one?

For that matter, what is the meaning if the the token in an appId 
attribute changes from one O/A to the next?

	Thanks,
	Paul


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Subject: Re: [MMUSIC] Comments on draft-ejzak-dispatch-msrp-data-channel-00
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On 2/23/14 8:13 PM, Richard Ejzak wrote:
>
>
>
> On Sun, Feb 23, 2014 at 5:13 PM, Paul Kyzivat <pkyzivat@alum.mit.edu
> <mailto:pkyzivat@alum.mit.edu>> wrote:
>
>     A few comments/questions on this draft:
>
>     Section 5.1.1.2 says:
>
>         The msrp-cema attribute shall be assumed to be present for every
>     MSRP
>         session using data channel transport, so the inclusion of the msrp-
>         cema attribute is optional.  This ensures that the data channel
>         transport for the MSRP session is established without using the path
>         attribute.
>
>     Why is use of cema mandatory?
>
>
> Existing MSRP procedures (without CEMA) are based on using the Path
> attribute to establish the transport connection (and essentially ignore
> the connection information associated with the m line).  If you are
> going to use a data channel, then you have no choice but to use the one
> established by the peer connection, so the existing MSRP procedure will
> NOT WORK.

Ahh! Of course.

> CEMA allows you to set up a tls or tcp connection based on
> the connection information in the m line rather than in the path
> attribute, but this is still not everything needed to make it work with
> data channels.  But it's a start.
>
>
>     RGC6714 has fallback rules for an endpoint that supports cema to
>     interoperate with one that does not. That requires removing the
>     indication of cema support. How does an msrp data channel endpoint
>     do that?
>
>
> Interoperating with an endpoint that does not support data channel will
> require going through a gateway that does.  This gateway will have to do
> something to provide a tls or tcp transport connection suitable for the
> other peer.  If the gateway can't do the enhanced CEMA procedures
> described here, it will have to perform as an MSRP B2BUA.

I gather you are thinking that this gateway won't be an MSRP relay?

>     Section 5.1.3 says:
>
>         Each text-based MSRP message is sent on the corresponding SCTP
>     stream
>         using standard MSRP framing and chunking procedures, as defined in
>         [RFC4975], with each MSRP chunk delivered in a single SCTP user
>         message.
>
>     I think it would be better to say:
>
>         Each text-based MSRP message is sent on the corresponding SCTP
>     stream
>         using standard MSRP framing and chunking procedures, as defined in
>         [RFC4975], with each MSRP chunk delivered in a single Data Channel
>         user message.
>
>     While data channel user messages map directly onto SCTP user
>     messages, they are conceptually distinct.
>
>
> Agreed.
>
>
>     Section 6 says:
>
>         o  The endpoint establishing an MSRP session using data channel
>            transport shall not request inclusion of any relays, although it
>            may interoperate with a peer that signals the use of relays.
>
>     Why must relays not be requested? That seems the most natural thing
>     to do if the goal is to talk to a peer that doesn't support data
>     channels.
>
>
> The MSRP relay is not typically on the signaling path, so is an unlikely
> data channel endpoint unless a separate peer connection and data channel
> is established just for MSRP.  This would be silly to do.  We want a
> gateway that is on the signaling path to perform any necessary
> conversion.  The CEMA gateway model is a much better fit and makes the
> use of relays unnecessary (at least on the side of the user using a data
> channel for MSRP).

ISTM there are lots of layers here. With webrtc the "signaling path" is 
pretty much out of scope, so it is unclear exactly what you mean in that 
case. (Usually there will be a web server on the signaling path between 
the two peers. Potentially that web server could have a relationship 
with an MSRP gw, that could have relay capabilities.)

I guess I'm mostly just confused on how this all hangs together.

>     In any case, some explanation of how things work with relays would
>     be appreciated.
>
>
> I must say that CEMA is confusing, and I agree that the CEMA parts of
> the document are not easy to follow without a thorough understanding of
> CEMA.  Since the CEMA RFC already explains all the confusing bits, I'm
> reluctant to repeat all of that here...  Perhaps a note that a thorough
> understanding of CEMA is necessary background for the CEMA stuff here?

Maybe. Obviously *I* need to study CEMA more carefully.

	Thanks,
	Paul

> Richard
>
>
>
>              Thanks,
>              Paul
>
>     _________________________________________________
>     mmusic mailing list
>     mmusic@ietf.org <mailto:mmusic@ietf.org>
>     https://www.ietf.org/mailman/__listinfo/mmusic
>     <https://www.ietf.org/mailman/listinfo/mmusic>
>
>


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: draft-ejzak-mmusic-data-channel-sdpneg: Allignment with draft-ietf-mmusic-sctp-sdp-06
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Hi,

In the latest version (-06) of draft-ietf-mmusic-sctp-sdp it is no more all=
owed (at least that is my reading when comparing the 05->06 diff) to assign=
 multiple SDP sctpmap attributes to a single m- line.

I guess that doesn't affect the mechanism in draft-ejzak-mmusic-data-channe=
l-sdpneg as such (eventhough some text may not be needed anymore). But, at =
least the example/note in section 4 should be aligned.

Example:

m=3Dapplication 54111 DTLS/SCTP 5000 5001 5002
c=3DIN IP4 79.97.215.79
                a=3Dsctpmap:5000 webrtc-datachannel
                a=3Dsctpmap:5001 bfcp
                a=3Dsctpmap:5002 t38

...should be:

m=3Dapplication 54111 DTLS/SCTP 5000
c=3DIN IP4 79.97.215.79
                a=3Dsctpmap:5000 webrtc-datachannel

(If you want to use multiple sctpmap attributes, I guess you would need to =
assign one m- line for each).

In addition, perhaps the SCTP port number is not needed in the extension at=
tributes (dcmap, dcsa) defined in the draft?

Regards,

Christer





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<p class=3D"MsoNormal">Hi,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">In the latest version (-06) of draft-ietf-mmusic-sct=
p-sdp it is no more allowed (at least that is my reading when comparing the=
 05-&gt;06 diff) to assign multiple SDP sctpmap attributes to a single m- l=
ine.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">I guess that doesn&#8217;t affect the mechanism in d=
raft-ejzak-mmusic-data-channel-sdpneg as such (eventhough some text may not=
 be needed anymore). But, at least the example/note in section 4 should be =
aligned.
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Example:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">m=3Dapplication 54111 D=
TLS/SCTP 5000 5001 5002<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">c=3DIN IP4 79.97.215.79=
<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=3Dsctpmap:5000 webrtc-datachannel<o:p><=
/o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=3Dsctpmap:5001 bfcp<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=3Dsctpmap:5002 t38<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">&#8230;should be:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">m=3Dapplication 54111 D=
TLS/SCTP 5000<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">c=3DIN IP4 79.97.215.79=
<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=3Dsctpmap:5000 webrtc-datachannel<o:p><=
/o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">(If you want to use multiple sctpmap attributes, I g=
uess you would need to assign one m- line for each).<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">In addition, perhaps the SCTP port number is not nee=
ded in the extension attributes (dcmap, dcsa) defined in the draft?<o:p></o=
:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Christer<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: draft-ejzak-mmusic-data-channel-sdpneg: external closing of channel
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Hi,

Section 5.2.3 says:

"It is specific to the sub-protocol whether this closing must in
                addition be signaled to the peer via a new SDP offer/answer=
 exchange."

Shouldn't the draft define how it is done using an SDP offer/answer exchang=
e?

Regards,

Christer

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<p class=3D"MsoNormal"><span lang=3D"FI">Hi,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"FI"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"FI">Section 5.2.3 says:<o:p></o:p></sp=
an></p>
<p class=3D"MsoNormal"><span lang=3D"FI"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">&#8221;It is specific t=
o the sub-protocol whether this closing must in<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; addition be signaled to the peer via a ne=
w SDP offer/answer exchange.&#8221;<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Shouldn&#8217;t the draft define how it is done usin=
g an SDP offer/answer exchange?<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Christer<o:p></o:p></p>
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: draft-ejzak-mmusic-data-channel-sdpneg: external rejection of channel
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Subject: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: external rejection of channel
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Hi,

Section 5.2.2 describes what an answerer does when accepting a data channel=
:

  "The peer receiving such an SDP offer performs the following:

                ...

   o  For accepted data channels, creates peer instances for the data
      channels with the browser using the channel parameters described
      in the SDP offer.  Note that the browser is asked to create data
      channels with stream identifiers not "owned" by the agent."

But, there is no text on rejecting a data channel in the SDP answer.

The answer can of course reject the whole m- line, using port zero in the A=
nswer m- line. But, how does the answerer reject an individual sub-protocol=
 data channel?

Section 5.1.1 says, regarding the SDP dcmap attribute, for accepted data ch=
annels:

"This line MUST be replicated without changes in the SDP answer, if
                the answerer accepts the offered data channel."

I guess one way to reject would be to simply not include the dcmap attribut=
e in the Answer, and the Offerer then needs to figure out what was rejected=
 based on the stream values not present in the Answer.

Regards,

Christer


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<p class=3D"MsoNormal"><span lang=3D"FI">Hi,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"FI"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal">Section 5.2.2 describes what an answerer does when a=
ccepting a data channel:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">&nbsp; &#8220;The peer receiving such an SDP offer p=
erforms the following:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &#8230;<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; o&nbsp; For accepted data channels, cre=
ates peer instances for the data<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; channels with the bro=
wser using the channel parameters described<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; in the SDP offer.&nbs=
p; Note that the browser is asked to create data<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; channels with stream =
identifiers not &quot;owned&quot; by the agent.&#8221;<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">But, there is no text on rejecting a data channel in=
 the SDP answer.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The answer can of course reject the whole m- line, u=
sing port zero in the Answer m- line. But, how does the answerer reject an =
individual sub-protocol data channel?<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Section 5.1.1 says, regarding the SDP dcmap attribut=
e, for accepted data channels:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">&#8220;This line MUST b=
e replicated without changes in the SDP answer, if<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; the answerer accepts the offered data cha=
nnel.&#8221;<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">I guess one way to reject would be to simply not inc=
lude the dcmap attribute in the Answer, and the Offerer then needs to figur=
e out what was rejected based on the stream values not present in the Answe=
r.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Christer<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
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From nobody Wed Feb 26 05:59:52 2014
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From: Richard Ejzak <richard.ejzak@gmail.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: Allignment with draft-ietf-mmusic-sctp-sdp-06
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Hi Christer,
Thanks for this input.  Clearly the example needs to be updated.  If the
group is certain that we don't want the flexibility to establish multiple
SCTP associations on the same 5-tuple, then we should definitely make the
change to simplify the dcmap and dcsa attributes.

BR, RIchard


On Wed, Feb 26, 2014 at 7:04 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>  Hi,
>
>
>
> In the latest version (-06) of draft-ietf-mmusic-sctp-sdp it is no more
> allowed (at least that is my reading when comparing the 05->06 diff) to
> assign multiple SDP sctpmap attributes to a single m- line.
>
>
>
> I guess that doesn't affect the mechanism in
> draft-ejzak-mmusic-data-channel-sdpneg as such (eventhough some text may
> not be needed anymore). But, at least the example/note in section 4 should
> be aligned.
>
>
>
> Example:
>
>
>
> m=application 54111 DTLS/SCTP 5000 5001 5002
>
> c=IN IP4 79.97.215.79
>
>                 a=sctpmap:5000 webrtc-datachannel
>
>                 a=sctpmap:5001 bfcp
>
>                 a=sctpmap:5002 t38
>
>
>
> ...should be:
>
>
>
> m=application 54111 DTLS/SCTP 5000
>
> c=IN IP4 79.97.215.79
>
>                 a=sctpmap:5000 webrtc-datachannel
>
>
>
> (If you want to use multiple sctpmap attributes, I guess you would need to
> assign one m- line for each).
>
>
>
> In addition, perhaps the SCTP port number is not needed in the extension
> attributes (dcmap, dcsa) defined in the draft?
>
>
>
> Regards,
>
>
>
> Christer
>
>
>
>
>
>
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>
>

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<div dir=3D"ltr">Hi Christer,<div>Thanks for this input. &nbsp;Clearly the =
example needs to be updated. &nbsp;If the group is certain that we don&#39;=
t want the flexibility to establish multiple SCTP associations on the same =
5-tuple, then we should definitely make the change to simplify the dcmap an=
d dcsa attributes.</div>
<div><br></div><div>BR, RIchard</div></div><div class=3D"gmail_extra"><br><=
br><div class=3D"gmail_quote">On Wed, Feb 26, 2014 at 7:04 AM, Christer Hol=
mberg <span dir=3D"ltr">&lt;<a href=3D"mailto:christer.holmberg@ericsson.co=
m" target=3D"_blank">christer.holmberg@ericsson.com</a>&gt;</span> wrote:<b=
r>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">





<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal">Hi,<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">In the latest version (-06) of draft-ietf-mmusic-sct=
p-sdp it is no more allowed (at least that is my reading when comparing the=
 05-&gt;06 diff) to assign multiple SDP sctpmap attributes to a single m- l=
ine.<u></u><u></u></p>

<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">I guess that doesn&rsquo;t affect the mechanism in d=
raft-ejzak-mmusic-data-channel-sdpneg as such (eventhough some text may not=
 be needed anymore). But, at least the example/note in section 4 should be =
aligned.
<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">Example:<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">m=3Dapplication 54111 D=
TLS/SCTP 5000 5001 5002<u></u><u></u></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">c=3DIN IP4 79.97.215.79=
<u></u><u></u></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=3Dsctpmap:5000 webrtc-datachannel<u></u=
><u></u></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=3Dsctpmap:5001 bfcp<u></u><u></u></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=3Dsctpmap:5002 t38<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">&hellip;should be:<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">m=3Dapplication 54111 D=
TLS/SCTP 5000<u></u><u></u></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">c=3DIN IP4 79.97.215.79=
<u></u><u></u></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=3Dsctpmap:5000 webrtc-datachannel<u></u=
><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">(If you want to use multiple sctpmap attributes, I g=
uess you would need to assign one m- line for each).<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">In addition, perhaps the SCTP port number is not nee=
ded in the extension attributes (dcmap, dcsa) defined in the draft?<u></u><=
u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">Regards,<span class=3D"HOEnZb"><font color=3D"#88888=
8"><u></u><u></u></font></span></p><span class=3D"HOEnZb"><font color=3D"#8=
88888">
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">Christer<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
</font></span></div>
</div>

<br>_______________________________________________<br>
mmusic mailing list<br>
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<br></blockquote></div><br></div>

--089e0158c486558b3b04f34f9d40--


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From: Richard Ejzak <richard.ejzak@gmail.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: external closing of channel
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Hi Christer,
Another good catch.  Although it is implied that closed data channels are
removed from subsequent offers, this should be stated explicitly.

BR, RIchard


On Wed, Feb 26, 2014 at 7:19 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>  Hi,
>
>
>
> Section 5.2.3 says:
>
>
>
> "It is specific to the sub-protocol whether this closing must in
>
>                 addition be signaled to the peer via a new SDP
> offer/answer exchange."
>
>
>
> Shouldn't the draft define how it is done using an SDP offer/answer
> exchange?
>
>
>
> Regards,
>
>
>
> Christer
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>
>

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<div dir="ltr">Hi Christer,<div>Another good catch. &nbsp;Although it is implied that closed data channels are removed from subsequent offers, this should be stated explicitly.</div><div><br></div><div>BR, RIchard</div></div><div class="gmail_extra">
<br><br><div class="gmail_quote">On Wed, Feb 26, 2014 at 7:19 AM, Christer Holmberg <span dir="ltr">&lt;<a href="mailto:christer.holmberg@ericsson.com" target="_blank">christer.holmberg@ericsson.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 lang="EN-US" link="blue" vlink="purple">
<div>
<p class="MsoNormal"><span lang="FI">Hi,<u></u><u></u></span></p>
<p class="MsoNormal"><span lang="FI"><u></u>&nbsp;<u></u></span></p>
<p class="MsoNormal"><span lang="FI">Section 5.2.3 says:<u></u><u></u></span></p>
<p class="MsoNormal"><span lang="FI"><u></u>&nbsp;<u></u></span></p>
<p class="MsoNormal" style="text-indent:36.0pt">&rdquo;It is specific to the sub-protocol whether this closing must in<u></u><u></u></p>
<p class="MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; addition be signaled to the peer via a new SDP offer/answer exchange.&rdquo;<u></u><u></u></p>
<p class="MsoNormal"><u></u>&nbsp;<u></u></p>
<p class="MsoNormal">Shouldn&rsquo;t the draft define how it is done using an SDP offer/answer exchange?<u></u><u></u></p>
<p class="MsoNormal"><u></u>&nbsp;<u></u></p>
<p class="MsoNormal">Regards,<span class="HOEnZb"><font color="#888888"><u></u><u></u></font></span></p><span class="HOEnZb"><font color="#888888">
<p class="MsoNormal"><u></u>&nbsp;<u></u></p>
<p class="MsoNormal">Christer<u></u><u></u></p>
</font></span></div>
</div>

<br>_______________________________________________<br>
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<br></blockquote></div><br></div>

--001a11c385e03df6ff04f34fbbba--


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On Wed, Feb 26, 2014 at 7:30 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>  Hi,
>
>
>
> Section 5.2.2 describes what an answerer does when accepting a data
> channel:
>
>
>
>   "The peer receiving such an SDP offer performs the following:
>
>
>
>                 ...
>
>
>
>    o  For accepted data channels, creates peer instances for the data
>
>       channels with the browser using the channel parameters described
>
>       in the SDP offer.  Note that the browser is asked to create data
>
>       channels with stream identifiers not "owned" by the agent."
>
>
>
> But, there is no text on rejecting a data channel in the SDP answer.
>
>
>
> The answer can of course reject the whole m- line, using port zero in the
> Answer m- line. But, how does the answerer reject an individual
> sub-protocol data channel?
>

Only attributes for "accepted data channels" are included in the SDP
answer, which indicates back to the offerer which ones are rejected (they
just aren't there).  This should be made clearer in the text.


>
>
> Section 5.1.1 says, regarding the SDP dcmap attribute, for accepted data
> channels:
>
>
>
> "This line MUST be replicated without changes in the SDP answer, if
>
>                 the answerer accepts the offered data channel."
>
>
>
> I guess one way to reject would be to simply not include the dcmap
> attribute in the Answer, and the Offerer then needs to figure out what was
> rejected based on the stream values not present in the Answer.
>

Yes.  That is what was intended.

BR, RIchard


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

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Wed, Feb 26, 2014 at 7:30 AM, Christer Holmberg <span dir=3D"ltr=
">&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">c=
hrister.holmberg@ericsson.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 lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span lang=3D"FI">Hi,<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"FI"><u></u>&nbsp;<u></u></span></p>
<p class=3D"MsoNormal">Section 5.2.2 describes what an answerer does when a=
ccepting a data channel:<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">&nbsp; &ldquo;The peer receiving such an SDP offer p=
erforms the following:<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &hellip;<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; o&nbsp; For accepted data channels, cre=
ates peer instances for the data<u></u><u></u></p>
<p class=3D"MsoNormal">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; channels with the bro=
wser using the channel parameters described<u></u><u></u></p>
<p class=3D"MsoNormal">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; in the SDP offer.&nbs=
p; Note that the browser is asked to create data<u></u><u></u></p>
<p class=3D"MsoNormal">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; channels with stream =
identifiers not &quot;owned&quot; by the agent.&rdquo;<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">But, there is no text on rejecting a data channel in=
 the SDP answer.<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">The answer can of course reject the whole m- line, u=
sing port zero in the Answer m- line. But, how does the answerer reject an =
individual sub-protocol data channel?</p></div></div></blockquote><div><br>
</div><div>Only attributes for &quot;accepted data channels&quot; are inclu=
ded in the SDP answer, which indicates back to the offerer which ones are r=
ejected (they just aren&#39;t there). &nbsp;This should be made clearer in =
the text.</div>
<div>&nbsp;</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8=
ex;border-left:1px #ccc solid;padding-left:1ex"><div lang=3D"EN-US" link=3D=
"blue" vlink=3D"purple"><div><p class=3D"MsoNormal"><u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">Section 5.1.1 says, regarding the SDP dcmap attribut=
e, for accepted data channels:<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">&ldquo;This line MUST b=
e replicated without changes in the SDP answer, if<u></u><u></u></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; the answerer accepts the offered data cha=
nnel.&rdquo;<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">I guess one way to reject would be to simply not inc=
lude the dcmap attribute in the Answer, and the Offerer then needs to figur=
e out what was rejected based on the stream values not present in the Answe=
r.</p>
</div></div></blockquote><div><br></div><div>Yes. &nbsp;That is what was in=
tended.</div><div><br></div><div>BR, RIchard</div><div>&nbsp;</div><blockqu=
ote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc s=
olid;padding-left:1ex">
<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple"><div><p class=3D"MsoNorm=
al"><u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">Regards,<span class=3D"HOEnZb"><font color=3D"#88888=
8"><u></u><u></u></font></span></p><span class=3D"HOEnZb"><font color=3D"#8=
88888">
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">Christer<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
</font></span></div>
</div>

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<br></blockquote></div><br></div></div>

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Richard Ejzak <richard.ejzak@gmail.com>
Thread-Topic: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: Allignment with draft-ietf-mmusic-sctp-sdp-06
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Subject: Re: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: Allignment with draft-ietf-mmusic-sctp-sdp-06
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Hi,

> Thanks for this input. =A0Clearly the example needs to be updated. =A0If =
the group is certain that we don't want the flexibility
> to establish multiple SCTP associations on the same 5-tuple

It MAY (we need to look into that) still be possible to do that with BUNDLE=
, but you would use separate m- lines for each SCTP association.

Regards,

Christer





On Wed, Feb 26, 2014 at 7:04 AM, Christer Holmberg <christer.holmberg@erics=
son.com> wrote:
Hi,
=A0
In the latest version (-06) of draft-ietf-mmusic-sctp-sdp it is no more all=
owed (at least that is my reading when comparing the 05->06 diff) to assign=
 multiple SDP sctpmap attributes to a single m- line.
=A0
I guess that doesn't affect the mechanism in draft-ejzak-mmusic-data-channe=
l-sdpneg as such (eventhough some text may not be needed anymore). But, at =
least the example/note in section 4 should be aligned.=20
=A0
Example:
=A0
      m=3Dapplication 54111 DTLS/SCTP 5000 5001 5002
      c=3DIN IP4 79.97.215.79
=A0=A0 =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 a=3Dsctpmap:5000 webrtc-datacha=
nnel
=A0=A0 =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 a=3Dsctpmap:5001 bfcp
=A0=A0 =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 a=3Dsctpmap:5002 t38
=A0
.should be:
=A0
      m=3Dapplication 54111 DTLS/SCTP 5000
      c=3DIN IP4 79.97.215.79
=A0=A0 =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 a=3Dsctpmap:5000 webrtc-datacha=
nnel
=A0
(If you want to use multiple sctpmap attributes, I guess you would need to =
assign one m- line for each).
=A0
In addition, perhaps the SCTP port number is not needed in the extension at=
tributes (dcmap, dcsa) defined in the draft?
=A0
Regards,
=A0
Christer
=A0
=A0=A0 =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=20
=A0
=A0

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Richard Ejzak <richard.ejzak@gmail.com>
Thread-Topic: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: external closing of channel
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Subject: Re: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: external closing of channel
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Hi,

There are no channels in the offers - there are SDP attributes associated w=
ith the channels :)

You also need to be more clear on whether sending of SSN reset is mandatory=
 when using your draft (you only say that "browsers will send it").

Regards,

Christer

From: Richard Ejzak [mailto:richard.ejzak@gmail.com]
Sent: 26. helmikuuta 2014 16:08
To: Christer Holmberg
Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: external clos=
ing of channel

Hi Christer,
Another good catch.  Although it is implied that closed data channels are r=
emoved from subsequent offers, this should be stated explicitly.

BR, RIchard

On Wed, Feb 26, 2014 at 7:19 AM, Christer Holmberg <christer.holmberg@erics=
son.com<mailto:christer.holmberg@ericsson.com>> wrote:
Hi,

Section 5.2.3 says:

"It is specific to the sub-protocol whether this closing must in
                addition be signaled to the peer via a new SDP offer/answer=
 exchange."

Shouldn't the draft define how it is done using an SDP offer/answer exchang=
e?

Regards,

Christer

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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">There are no channels in =
the offers &#8211; there are SDP attributes associated with the channels
</span><span style=3D"font-size:11.0pt;font-family:Wingdings;color:#1F497D"=
>J</span><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D"><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">You also need to be more =
clear on whether sending of SSN reset is mandatory when using your draft (y=
ou only say that &#8220;browsers will send it&#8221;).<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>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Richard =
Ejzak [mailto:richard.ejzak@gmail.com]
<br>
<b>Sent:</b> 26. helmikuuta 2014 16:08<br>
<b>To:</b> Christer Holmberg<br>
<b>Cc:</b> mmusic@ietf.org<br>
<b>Subject:</b> Re: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: extern=
al closing of channel<o:p></o:p></span></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">Hi Christer,<o:p></o:p></p>
<div>
<p class=3D"MsoNormal">Another good catch. &nbsp;Although it is implied tha=
t closed data channels are removed from subsequent offers, this should be s=
tated explicitly.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">BR, RIchard<o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On Wed, Feb 26, 2014 at 7:19 AM, Christer Holmberg &=
lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chri=
ster.holmberg@ericsson.com</a>&gt; wrote:<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"FI">Hi,</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"FI">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"FI">Section 5.2.3 says:</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"FI">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;text-indent:36.0pt">
&#8221;It is specific to the sub-protocol whether this closing must in<o:p>=
</o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp; addition be signaled to the peer via a new SDP offer/an=
swer exchange.&#8221;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">Shouldn&#8217;t the draft define how it is done using an SDP offer=
/answer exchange?<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#888888">&nbsp;<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#888888">Christer<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
_______________________________________________<br>
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From nobody Wed Feb 26 07:42:59 2014
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Date: Wed, 26 Feb 2014 09:42:41 -0600
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From: Richard Ejzak <richard.ejzak@gmail.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: external closing of channel
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--089e011836be2049f604f3510e47
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On Wed, Feb 26, 2014 at 9:13 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>  Hi,
>
>
>
> There are no channels in the offers - there are SDP attributes associated
> with the channels J
>

Fair enough.


>
>
> You also need to be more clear on whether sending of SSN reset is
> mandatory when using your draft (you only say that "browsers will send it").
>

The text should refer to protocol stack behavior rather than to the browser
since it's supposed to also apply to data channel usage outside the context
of WebRTC.  (We intended to fix all th browser references in this draft and
missed this one.)  So, yes, SSN reset is mandatory.

Thanks again,
Richard


>
> Regards,
>
>
>
> Christer
>
>
>
> *From:* Richard Ejzak [mailto:richard.ejzak@gmail.com]
> *Sent:* 26. helmikuuta 2014 16:08
> *To:* Christer Holmberg
> *Cc:* mmusic@ietf.org
> *Subject:* Re: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: external
> closing of channel
>
>
>
> Hi Christer,
>
> Another good catch.  Although it is implied that closed data channels are
> removed from subsequent offers, this should be stated explicitly.
>
>
>
> BR, RIchard
>
>
>
> On Wed, Feb 26, 2014 at 7:19 AM, Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
>
> Hi,
>
>
>
> Section 5.2.3 says:
>
>
>
> "It is specific to the sub-protocol whether this closing must in
>
>                 addition be signaled to the peer via a new SDP
> offer/answer exchange."
>
>
>
> Shouldn't the draft define how it is done using an SDP offer/answer
> exchange?
>
>
>
> Regards,
>
>
>
> Christer
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>
>
>

--089e011836be2049f604f3510e47
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Wed, Feb 26, 2014 at 9:13 AM, Christer Holmberg <span dir=3D"ltr=
">&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">c=
hrister.holmberg@ericsson.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 lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Hi,<u></u><u></u></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"><u></u>&nbsp;<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">There are no channels in =
the offers &ndash; there are SDP attributes associated with the channels
</span><span style=3D"font-size:11.0pt;font-family:Wingdings;color:#1f497d"=
>J</span></p></div></div></blockquote><div><br></div><div>Fair enough.</div=
><div>&nbsp;<br></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0=
 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple"><div><p class=3D"MsoNorm=
al"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;s=
ans-serif&quot;;color:#1f497d"><u></u><u></u></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"><u></u>&nbsp;<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">You also need to be more =
clear on whether sending of SSN reset is mandatory when using your draft (y=
ou only say that &ldquo;browsers will send it&rdquo;).</span></p>
</div></div></blockquote><div><br></div><div>The text should refer to proto=
col stack behavior rather than to the browser since it&#39;s supposed to al=
so apply to data channel usage outside the context of WebRTC. &nbsp;(We int=
ended to fix all th browser references in this draft and missed this one.) =
&nbsp;So, yes, SSN reset is mandatory. &nbsp; &nbsp;</div>
<div><br></div><div>Thanks again,</div><div>Richard</div><div><br></div><bl=
ockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #=
ccc solid;padding-left:1ex"><div lang=3D"EN-US" link=3D"blue" vlink=3D"purp=
le"><div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u><u></u></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"><u></u>&nbsp;<u></u></spa=
n></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,<u></u><u></u></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>&nbsp;<u></u></spa=
n></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<u></u><u></u></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>&nbsp;<u></u></spa=
n></p>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Richard =
Ejzak [mailto:<a href=3D"mailto:richard.ejzak@gmail.com" target=3D"_blank">=
richard.ejzak@gmail.com</a>]
<br>
<b>Sent:</b> 26. helmikuuta 2014 16:08<br>
<b>To:</b> Christer Holmberg<br>
<b>Cc:</b> <a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf=
.org</a><br>
<b>Subject:</b> Re: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: extern=
al closing of channel<u></u><u></u></span></p><div><div class=3D"h5">
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<div>
<p class=3D"MsoNormal">Hi Christer,<u></u><u></u></p>
<div>
<p class=3D"MsoNormal">Another good catch. &nbsp;Although it is implied tha=
t closed data channels are removed from subsequent offers, this should be s=
tated explicitly.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">BR, RIchard<u></u><u></u></p>
</div>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><u></u>&nbsp;<u></u><=
/p>
<div>
<p class=3D"MsoNormal">On Wed, Feb 26, 2014 at 7:19 AM, Christer Holmberg &=
lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chri=
ster.holmberg@ericsson.com</a>&gt; wrote:<u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"FI">Hi,</span><u></u><u></u></p>
<p class=3D"MsoNormal"><span lang=3D"FI">&nbsp;</span><u></u><u></u></p>
<p class=3D"MsoNormal"><span lang=3D"FI">Section 5.2.3 says:</span><u></u><=
u></u></p>
<p class=3D"MsoNormal"><span lang=3D"FI">&nbsp;</span><u></u><u></u></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">
&rdquo;It is specific to the sub-protocol whether this closing must in<u></=
u><u></u></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; addition be signaled to the peer via a ne=
w SDP offer/answer exchange.&rdquo;<u></u><u></u></p>
<p class=3D"MsoNormal">&nbsp;<u></u><u></u></p>
<p class=3D"MsoNormal">Shouldn&rsquo;t the draft define how it is done usin=
g an SDP offer/answer exchange?<u></u><u></u></p>
<p class=3D"MsoNormal">&nbsp;<u></u><u></u></p>
<p class=3D"MsoNormal">Regards,<u></u><u></u></p>
<p class=3D"MsoNormal"><span style=3D"color:#888888">&nbsp;<u></u><u></u></=
span></p>
<p class=3D"MsoNormal"><span style=3D"color:#888888">Christer<u></u><u></u>=
</span></p>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><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><u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
</div>
</div></div></div>
</div>

</blockquote></div><br></div></div>

--089e011836be2049f604f3510e47--


From nobody Wed Feb 26 10:59:54 2014
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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "mmusic@ietf.org" <mmusic@ietf.org>
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Subject: Re: [MMUSIC] [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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+ mmusic
- rtcweb (in bcc)

Hi Paul,
I wanted to close this email loop. Please see my inline comments.

> >
> > [Raju] If I understand correctly, these parameters simply define the
> > data channel transport properties. Yes, they may have some correletion
> with
> > subprotocol, but that's not a one-to-one mapping.
> > For example, a given subprotocol could run on a reliable or
> > unreliable transport; depending on app needs it may use unreliable
> > transport and deal with reliablity at subprotocol stack (or ignore it).
>=20
> In some cases. But in general it is easy and common to define a protocol
> that needs a reliable transport, and it won't work without one.
>=20
> What if every application in the world that uses TCP had the option to
> specify if it was reliable or not? How many would screw it up?
>=20
[Raju]=20
Application subprotocol users need to know the transport needs for the prot=
ocol and select a single transport if multiple options exist. This is simil=
ar to how users select TCP or UDP when implementing any Layer5 protocol (e.=
g. SIP) on top of Layer 4.

If APIs abstract this subprotocol then the transport selection can be hidde=
n under the API (if only one choice exists) and API user won't have to spec=
ify a transport.=20

We really appreciate your time in providing valuable comments on these draf=
ts!

Best Regards!
Raju


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>, "Paul Kyzivat" <pkyzivat@alum.mit.edu>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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Subject: Re: [MMUSIC] [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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Hi,

>I wanted to close this email loop. Please see my inline comments.

I am ok closing it for now - but I still have a different opinion :)

>> [Raju] If I understand correctly, these parameters simply define the=20
>> data channel transport properties. Yes, they may have some=20
>> correletion with
>> subprotocol, but that's not a one-to-one mapping.
>> For example, a given subprotocol could run on a reliable or=20
>> unreliable transport; depending on app needs it may use unreliable=20
>> transport and deal with reliablity at subprotocol stack (or ignore it).
>>=20
>> In some cases. But in general it is easy and common to define a=20
>> protocol that needs a reliable transport, and it won't work without one.
>>=20
>> What if every application in the world that uses TCP had the option to=20
>> specify if it was reliable or not? How many would screw it up?
>>=20
>[Raju]
>Application subprotocol users need to know the transport needs for the pro=
tocol and select a single >transport if multiple options exist. This is sim=
ilar to how users select TCP or UDP when >implementing any Layer5 protocol =
(e.g. SIP) on top of Layer 4.

RFC 3261, i.e. the "sub-protocol specification", defines the characteristic=
s when using TCP or UDP for SIP. There is no mechanism in UDP or TCP to neg=
otiate SIP timers.

I DO buy the argument that, when using DCP, there may not be a registered s=
ub-protocol to begin with, so this type of low-level information needs to b=
e negotiated.

BUT, when using draft-ejzak my assumption is that people will indicate the =
sup-protocol.

Regards,

Christer


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Richard Ejzak <richard.ejzak@gmail.com>
Thread-Topic: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: external closing of channel
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Subject: Re: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: external closing of channel
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Hi,
 You also need to be more clear on whether sending of SSN reset is mandator=
y when using your draft (you only say that "browsers will send it").

The text should refer to protocol stack behavior rather than to the browser=
 since it's supposed to also apply to data channel usage outside the contex=
t of WebRTC.  (We intended to fix all th browser references in this draft a=
nd missed this one.)  So, yes, SSN reset is mandatory.

Then you also need to cover that in the procedures.

For example, when an entity receives an Offer removing a channel, it MUST s=
end a SSN reset.

Also, what to do if an entity receives an SSN reset, but not an associated =
Offer.

These kind of error situations often cause troubles at interop testings, so=
 let's try to cover them from the beginning :)

Regards,

Christer


From: Richard Ejzak [mailto:richard.ejzak@gmail.com<mailto:richard.ejzak@gm=
ail.com>]
Sent: 26. helmikuuta 2014 16:08
To: Christer Holmberg
Cc: mmusic@ietf.org<mailto:mmusic@ietf.org>
Subject: Re: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: external clos=
ing of channel

Hi Christer,
Another good catch.  Although it is implied that closed data channels are r=
emoved from subsequent offers, this should be stated explicitly.

BR, RIchard

On Wed, Feb 26, 2014 at 7:19 AM, Christer Holmberg <christer.holmberg@erics=
son.com<mailto:christer.holmberg@ericsson.com>> wrote:
Hi,

Section 5.2.3 says:

"It is specific to the sub-protocol whether this closing must in
                addition be signaled to the peer via a new SDP offer/answer=
 exchange."

Shouldn't the draft define how it is done using an SDP offer/answer exchang=
e?

Regards,

Christer

_______________________________________________
mmusic mailing list
mmusic@ietf.org<mailto:mmusic@ietf.org>
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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;mso-fareast-language:EN-US=
">Hi,<o:p></o:p></span></p>
<div>
<div>
<div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;You also need to b=
e more clear on whether sending of SSN reset is mandatory when using
 your draft (you only say that &#8220;browsers will send it&#8221;).</span>=
<span lang=3D"EN-US"><o:p></o:p></span></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal">The text should refer to protocol stack behavior rat=
her than to the browser since it's supposed to also apply to data channel u=
sage outside the context of WebRTC. &nbsp;(We intended to fix all th browse=
r references in this draft and missed this
 one.) &nbsp;So, yes, SSN reset is mandatory. &nbsp; &nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Then you also need to cov=
er that in the procedures.<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">For example, when an enti=
ty receives an Offer removing a channel, it MUST send a SSN reset.<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">Also, what to do if an en=
tity receives an SSN reset, but not an associated Offer.<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">These kind of error situa=
tions often cause troubles at interop testings, so let&#8217;s try to cover=
 them from the beginning
</span><span style=3D"font-size:11.0pt;font-family:Wingdings;color:#1F497D"=
>J</span><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D"><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>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><span lang=
=3D"EN-US"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><b><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quo=
t;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span lang=3D"EN-US"=
 style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&=
quot;"> Richard
 Ejzak [mailto:<a href=3D"mailto:richard.ejzak@gmail.com" target=3D"_blank"=
>richard.ejzak@gmail.com</a>]
<br>
<b>Sent:</b> 26. helmikuuta 2014 16:08<br>
<b>To:</b> Christer Holmberg<br>
<b>Cc:</b> <a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf=
.org</a><br>
<b>Subject:</b> Re: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: extern=
al closing of channel</span><span lang=3D"EN-US"><o:p></o:p></span></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"EN-US">&nbsp;<o:p></o:p></span></p>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"EN-US">Hi Christer,<o:p></o:p></span></p>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"EN-US">Another good catch. &nbsp;Although it is impl=
ied that closed data channels are removed from subsequent offers, this shou=
ld be stated explicitly.<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"EN-US">&nbsp;<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"EN-US">BR, RIchard<o:p></o:p></span></p>
</div>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;margin-bottom:12.0p=
t"><span lang=3D"EN-US">&nbsp;<o:p></o:p></span></p>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"EN-US">On Wed, Feb 26, 2014 at 7:19 AM, Christer Hol=
mberg &lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blan=
k">christer.holmberg@ericsson.com</a>&gt; wrote:<o:p></o:p></span></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"FI">Hi,</span><span lang=3D"EN-US"><o:p></o:p></span=
></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"FI">&nbsp;</span><span lang=3D"EN-US"><o:p></o:p></s=
pan></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"FI">Section 5.2.3 says:</span><span lang=3D"EN-US"><=
o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"FI">&nbsp;</span><span lang=3D"EN-US"><o:p></o:p></s=
pan></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;text-indent:36.0pt">
<span lang=3D"EN-US">&#8221;It is specific to the sub-protocol whether this=
 closing must in<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"EN-US">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; addition be signaled to the peer v=
ia a new SDP offer/answer exchange.&#8221;<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"EN-US">&nbsp;<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"EN-US">Shouldn&#8217;t the draft define how it is do=
ne using an SDP offer/answer exchange?<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"EN-US">&nbsp;<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"EN-US">Regards,<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"EN-US" style=3D"color:#888888">&nbsp;</span><span la=
ng=3D"EN-US"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"EN-US" style=3D"color:#888888">Christer</span><span =
lang=3D"EN-US"><o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;margin-bottom:12.0p=
t"><span lang=3D"EN-US"><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><o:p></o:p></span></p>
</div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"EN-US">&nbsp;<o:p></o:p></span></p>
</div>
</div>
</div>
</div>
</div>
</blockquote>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</div>
</body>
</html>

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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Richard Ejzak <richard.ejzak@gmail.com>, Christer Holmberg <christer.holmberg@ericsson.com>
Thread-Topic: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: external rejection of channel
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Please find my comments inline.
But, there is no text on rejecting a data channel in the SDP answer.

The answer can of course reject the whole m- line, using port zero in the A=
nswer m- line. But, how does the answerer reject an individual sub-protocol=
 data channel?

Only attributes for "accepted data channels" are included in the SDP answer=
, which indicates back to the offerer which ones are rejected (they just ar=
en't there).  This should be made clearer in the text.


Section 5.1.1 says, regarding the SDP dcmap attribute, for accepted data ch=
annels:

"This line MUST be replicated without changes in the SDP answer, if
                the answerer accepts the offered data channel."

I guess one way to reject would be to simply not include the dcmap attribut=
e in the Answer, and the Offerer then needs to figure out what was rejected=
 based on the stream values not present in the Answer.

Yes.  That is what was intended.
[Raju] In addition to the above mentiooed cases, if we all agree we need to=
 add the following cases in the next draft:

1.       SDP offer has no a=3Ddcmap. This has two use cases:

a.       Initial SDP offer: no data channel streams (DCS) are open yet.

b.      Subsequent SDP offer: Remove all previously opened data channels/st=
reams.
In either case, the SCTP association is kept open (and ICE procedures are p=
erformed) even when no data channels are in use.

2.       Creation of new DCS(s) using SDP offer (initial or subsequent)

a.       Offerer has to create the data channels locally first by doing cre=
ateDataChannel or equivalent API into data channel stack before sending off=
er.

3.       SDP offer was rejected

a.       DCSs are kept per previous SDP offer/answer. This means DCSs that =
are to be deleted has to be deleted (i.e. calling createDataChannel or equi=
valent API) only after successful SDP answer.

b.      If any new DCSs are attempted to be created per new SDP offer, thes=
e have to be closed now.

4.       Per http://dev.w3.org/2011/webrtc/editor/webrtc.html#dom-peerconne=
ction-createdatachannel createDataChannel() API the 'external' negotiation =
field is per data channel.  But this draft only allows external negotiation=
 to be used for all or no streams; it can't be selective. This restriction =
is also needed so that the SDP offer/answer odd/even stream id allocation w=
on't conflict DCP's TLS role based odd/even mechanism.

Best Regards!
Raju

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4.0pt">
<div>
<div>
<div>
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#1F497D">Please find my comments inline.<o:p>=
</o:p></span></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">But, there is no text on rejecting a data channel in the SDP answe=
r.<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">The answer can of course reject the whole m- line, using port zero=
 in the Answer m- line. But, how does the answerer reject an individual sub=
-protocol data channel?<o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Only attributes for &quot;accepted data channels&quo=
t; are included in the SDP answer, which indicates back to the offerer whic=
h ones are rejected (they just aren't there). &nbsp;This should be made cle=
arer in the text.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0i=
n 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in">
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">Section 5.1.1 says, regarding the SDP dcmap attribute, for accepte=
d data channels:<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;text-indent:.5in">
&#8220;This line MUST be replicated without changes in the SDP answer, if<o=
:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp; the answerer accepts the offered data channel.&#8221;<o=
:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">I guess one way to reject would be to simply not include the dcmap=
 attribute in the Answer, and the Offerer then needs to figure out what was=
 rejected based on the stream values
 not present in the Answer.<o:p></o:p></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Yes. &nbsp;That is what was intended.<o:p></o:p></p>
<p class=3D"MsoNormal"><i><span style=3D"font-size:11.0pt;font-family:&quot=
;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">[Raju] In addition to =
the above mentiooed cases, if we all agree we need to add the following cas=
es in the next draft:<o:p></o:p></span></i></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-.25in;mso-list:l0 level=
1 lfo1"><![if !supportLists]><b><i><span style=3D"font-size:11.0pt;font-fam=
ily:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><span style=
=3D"mso-list:Ignore">1.<span style=3D"font:7.0pt &quot;Times New Roman&quot=
;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span></i></b><![endif]><span style=3D"font-size:11.0pt;font=
-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">SDP offer=
 has no a=3Ddcmap. This has two use cases:<o:p></o:p></span></p>
<p class=3D"MsoListParagraph" style=3D"margin-left:1.0in;text-indent:-.25in=
;mso-list:l0 level2 lfo1">
<![if !supportLists]><span style=3D"font-size:11.0pt;font-family:&quot;Cali=
bri&quot;,&quot;sans-serif&quot;;color:#1F497D"><span style=3D"mso-list:Ign=
ore">a.<span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Initial SDP offer=
: no data channel streams (DCS) are open yet.<o:p></o:p></span></p>
<p class=3D"MsoListParagraph" style=3D"margin-left:1.0in;text-indent:-.25in=
;mso-list:l0 level2 lfo1">
<![if !supportLists]><span style=3D"font-size:11.0pt;font-family:&quot;Cali=
bri&quot;,&quot;sans-serif&quot;;color:#1F497D"><span style=3D"mso-list:Ign=
ore">b.<span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Subsequent SDP of=
fer: Remove all previously opened data channels/streams.<o:p></o:p></span><=
/p>
<p class=3D"MsoNormal" style=3D"margin-left:.5in"><span style=3D"font-size:=
11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D=
">In either case, the SCTP association is kept open (and ICE procedures are=
 performed) even when no data channels are in use.<o:p></o:p></span></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-.25in;mso-list:l0 level=
1 lfo1"><![if !supportLists]><b><i><span style=3D"font-size:11.0pt;font-fam=
ily:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><span style=
=3D"mso-list:Ignore">2.<span style=3D"font:7.0pt &quot;Times New Roman&quot=
;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span></i></b><![endif]><span style=3D"font-size:11.0pt;font=
-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Creation =
of new DCS(s) using SDP offer (initial or subsequent)<o:p></o:p></span></p>
<p class=3D"MsoListParagraph" style=3D"margin-left:1.0in;text-indent:-.25in=
;mso-list:l0 level2 lfo1">
<![if !supportLists]><span style=3D"font-size:11.0pt;font-family:&quot;Cali=
bri&quot;,&quot;sans-serif&quot;;color:#1F497D"><span style=3D"mso-list:Ign=
ore">a.<span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Offerer has to cr=
eate the data channels locally first by doing createDataChannel or equivale=
nt API into data channel stack before sending offer.<o:p></o:p></span></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-.25in;mso-list:l0 level=
1 lfo1"><![if !supportLists]><b><i><span style=3D"font-size:11.0pt;font-fam=
ily:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><span style=
=3D"mso-list:Ignore">3.<span style=3D"font:7.0pt &quot;Times New Roman&quot=
;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span></i></b><![endif]><span style=3D"font-size:11.0pt;font=
-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">SDP offer=
 was rejected<o:p></o:p></span></p>
<p class=3D"MsoListParagraph" style=3D"margin-left:1.0in;text-indent:-.25in=
;mso-list:l0 level2 lfo1">
<![if !supportLists]><span style=3D"font-size:11.0pt;font-family:&quot;Cali=
bri&quot;,&quot;sans-serif&quot;;color:#1F497D"><span style=3D"mso-list:Ign=
ore">a.<span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">DCSs are kept per=
 previous SDP offer/answer. This means DCSs that are to be deleted has to b=
e deleted (i.e. calling createDataChannel or equivalent
 API) only after successful SDP answer.<o:p></o:p></span></p>
<p class=3D"MsoListParagraph" style=3D"margin-left:1.0in;text-indent:-.25in=
;mso-list:l0 level2 lfo1">
<![if !supportLists]><span style=3D"font-size:11.0pt;font-family:&quot;Cali=
bri&quot;,&quot;sans-serif&quot;;color:#1F497D"><span style=3D"mso-list:Ign=
ore">b.<span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">If any new DCSs a=
re attempted to be created per new SDP offer, these have to be closed now.<=
o:p></o:p></span></p>
<p class=3D"MsoListParagraph" style=3D"margin-left:41.25pt;text-indent:-.25=
in;mso-list:l0 level1 lfo1">
<![if !supportLists]><b><i><span style=3D"font-size:11.0pt;font-family:&quo=
t;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><span style=3D"mso-li=
st:Ignore">4.<span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span></i></b><![endif]><span style=3D"font-size:11.0pt;font=
-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Per
<a href=3D"http://dev.w3.org/2011/webrtc/editor/webrtc.html#dom-peerconnect=
ion-createdatachannel">
http://dev.w3.org/2011/webrtc/editor/webrtc.html#dom-peerconnection-created=
atachannel</a> createDataChannel() API the &#8216;external&#8217; negotiati=
on field is per data channel. &nbsp;But this draft only allows external neg=
otiation to be used for all or no streams; it can&#8217;t
 be selective. This restriction is also needed so that the SDP offer/answer=
 odd/even stream id allocation won&#8217;t conflict DCP&#8217;s TLS role ba=
sed odd/even mechanism.<o:p></o:p></span></p>
<div>
<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">Best 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">Raju<o:p></o:p></span></p=
>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</body>
</html>

--_000_E1FE4C082A89A246A11D7F32A95A17826DFFB545US70UWXCHMBA02z_--


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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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Date: Wed, 26 Feb 2014 20:18:46 +0000
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References: <7594FB04B1934943A5C02806D1A2204B1D1B4DA3@ESESSMB209.ericsson.se> <CAJuyhsz4ZG_ReNEqmu+fTcSDfXxCnKWaVBhYvjf4XsWxSXB1mQ@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D1B5097@ESESSMB209.ericsson.se> <CAJuyhsxz4X2aDNx2Y1Y2Gi9G0D12Ort95QFD=cQ6s-zM5GK9Uw@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D1B5305@ESESSMB209.ericsson.se> <E1FE4C082A89A246A11D7F32A95A17826DFF58DB@US70UWXCHMBA02.zam.alcatel-lucent.com> <7594FB04B1934943A5C02806D1A2204B1D1B59FA@ESESSMB209.ericsson.se> <E1FE4C082A89A246A11D7F32A95A17826DFF5D26@US70UWXCHMBA02.zam.alcatel-lucent.com> <7594FB04B1934943A5C02806D1A2204B1D1B839A@ESESSMB209.ericsson.se> <530CCB54.4000106@alum.mit.edu> <E1FE4C082A89A246A11D7F32A95A17826DFF7AFD@US70UWXCHMBA02.zam.alcatel-lucent.com> <530D189F.8020804@alum.mit.edu> <E1FE4C082A89A246A11D7F32A95A17826DFFB2C3@US70UWXCHMBA02.zam.alcatel-lucent.com> <7594FB04B1934943A5C02806D1A2204B1D1BCCB6@ESESSMB209.ericsson.se>
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Subject: Re: [MMUSIC] [rtcweb] A few questions on draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00
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Hi,
Please see my comments inline.

> Hi,
>=20
> >I wanted to close this email loop. Please see my inline comments.
>=20
> I am ok closing it for now - but I still have a different opinion :)

[Raju] There won't be much fun if we all agree on everything. Isn't it?! :-=
)

>=20
> >> [Raju] If I understand correctly, these parameters simply define the
> >> data channel transport properties. Yes, they may have some
> >> correletion with
> >> subprotocol, but that's not a one-to-one mapping.
> >> For example, a given subprotocol could run on a reliable or
> >> unreliable transport; depending on app needs it may use unreliable
> >> transport and deal with reliablity at subprotocol stack (or ignore it)=
.
> >>
> >> In some cases. But in general it is easy and common to define a
> >> protocol that needs a reliable transport, and it won't work without on=
e.
> >>
> >> What if every application in the world that uses TCP had the option to
> >> specify if it was reliable or not? How many would screw it up?
> >>
> >[Raju]
> >Application subprotocol users need to know the transport needs for the
> protocol and select a single >transport if multiple options exist. This i=
s
> similar to how users select TCP or UDP when >implementing any Layer5
> protocol (e.g. SIP) on top of Layer 4.
>=20
> RFC 3261, i.e. the "sub-protocol specification", defines the characterist=
ics
> when using TCP or UDP for SIP. There is no mechanism in UDP or TCP to
> negotiate SIP timers.

[Raju] What you said is correct! However, int his case DCP does negotiate t=
hese timers inband. The external negotiation is simply providing similar me=
chanism so that both peers have same data channel properties (similar to DC=
P).

Thanks
Raju

>=20
> I DO buy the argument that, when using DCP, there may not be a registered
> sub-protocol to begin with, so this type of low-level information needs t=
o
> be negotiated.
>=20
> BUT, when using draft-ejzak my assumption is that people will indicate th=
e
> sup-protocol.


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Subject: Re: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: external rejection of channel
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I would like some clarity about how this interacts with DCEP.

IMO these are not mutually exclusive. But they must be used with care.

So, when a channel that was in the SDP previously then disappears from 
the SDP it is necessary to close the corresponding channel. But channels 
that were opened via DCEP should not be closed.

That also means that one should never mention a DCEP channel in SDP.

	Thanks,
	Paul

On 2/26/14 3:12 PM, Makaraju, Maridi Raju (Raju) wrote:
> Please find my comments inline.
>
> But, there is no text on rejecting a data channel in the SDP answer.
>
> The answer can of course reject the whole m- line, using port zero in
> the Answer m- line. But, how does the answerer reject an individual
> sub-protocol data channel?
>
> Only attributes for "accepted data channels" are included in the SDP
> answer, which indicates back to the offerer which ones are rejected
> (they just aren't there).  This should be made clearer in the text.
>
>     Section 5.1.1 says, regarding the SDP dcmap attribute, for accepted
>     data channels:
>
>     “This line MUST be replicated without changes in the SDP answer, if
>
>                      the answerer accepts the offered data channel.”
>
>     I guess one way to reject would be to simply not include the dcmap
>     attribute in the Answer, and the Offerer then needs to figure out
>     what was rejected based on the stream values not present in the Answer.
>
> Yes.  That is what was intended.
>
> /[Raju] In addition to the above mentiooed cases, if we all agree we
> need to add the following cases in the next draft:/
>
> */1./*SDP offer has no a=dcmap. This has two use cases:
>
> a.Initial SDP offer: no data channel streams (DCS) are open yet.
>
> b.Subsequent SDP offer: Remove all previously opened data channels/streams.
>
> In either case, the SCTP association is kept open (and ICE procedures
> are performed) even when no data channels are in use.
>
> */2./*Creation of new DCS(s) using SDP offer (initial or subsequent)
>
> a.Offerer has to create the data channels locally first by doing
> createDataChannel or equivalent API into data channel stack before
> sending offer.
>
> */3./*SDP offer was rejected
>
> a.DCSs are kept per previous SDP offer/answer. This means DCSs that are
> to be deleted has to be deleted (i.e. calling createDataChannel or
> equivalent API) only after successful SDP answer.
>
> b.If any new DCSs are attempted to be created per new SDP offer, these
> have to be closed now.
>
> */4./*Per
> http://dev.w3.org/2011/webrtc/editor/webrtc.html#dom-peerconnection-createdatachannel
> createDataChannel() API the ‘external’ negotiation field is per data
> channel.  But this draft only allows external negotiation to be used for
> all or no streams; it can’t be selective. This restriction is also
> needed so that the SDP offer/answer odd/even stream id allocation won’t
> conflict DCP’s TLS role based odd/even mechanism.
>
> Best Regards!
>
> Raju
>
>
>
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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Richard Ejzak <richard.ejzak@gmail.com>
Thread-Topic: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: external closing of channel
Thread-Index: Ac8y9K3pwn95tj5YT7SqspOdgSm1jf///isA///d/LCAADx9gP//qDPA//9J3TA=
Date: Wed, 26 Feb 2014 20:31:38 +0000
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References: <7594FB04B1934943A5C02806D1A2204B1D1BC4D7@ESESSMB209.ericsson.se> <CAJuyhsyQ6i9jj+T=x=fSJnEAAqjHoTa98X5X-Boyk+EZGbFycw@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D1BC7FC@ESESSMB209.ericsson.se> <CAJuyhswb4WZKHk-2FkC5YVP0pBuP4Zs4XozAXi7c50vrXk-uzw@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D1BCD0C@ESESSMB209.ericsson.se>
In-Reply-To: <7594FB04B1934943A5C02806D1A2204B1D1BCD0C@ESESSMB209.ericsson.se>
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Subject: Re: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: external closing of channel
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Hi,
Comments inline.



Hi,
 You also need to be more clear on whether sending of SSN reset is mandator=
y when using your draft (you only say that "browsers will send it").

The text should refer to protocol stack behavior rather than to the browser=
 since it's supposed to also apply to data channel usage outside the contex=
t of WebRTC.  (We intended to fix all th browser references in this draft a=
nd missed this one.)  So, yes, SSN reset is mandatory.

Then you also need to cover that in the procedures.

For example, when an entity receives an Offer removing a channel, it MUST s=
end a SSN reset.
[Raju] Sure, we will make that clear. Thanks. As I already commented on oth=
er thread, the section "4.3.2 Closing data channel" needs more description.

Also, what to do if an entity receives an SSN reset, but not an associated =
Offer.
[Raju] Also as noted in other email thread, SDP offerer who wants to close =
a data channel should close the data channel ONLY when successful SDP answe=
r is received. The above mentioned SSN reset case happens when SDP answerer=
 starts closing channel right after sending SDP answer. SDP offer side data=
 channel stack needs to handle such a case anyway (we will have this scenar=
io with inband DCP too) and informs the application via onClose() or equiva=
lent API. Then at SDP answer, offerer do not have to explictly close the da=
ta channel. Lastly, SSN reset

Thanks
Raju


These kind of error situations often cause troubles at interop testings, so=
 let's try to cover them from the beginning :)

Regards,

Christer


From: Richard Ejzak [mailto:richard.ejzak@gmail.com<mailto:richard.ejzak@gm=
ail.com>]
Sent: 26. helmikuuta 2014 16:08
To: Christer Holmberg
Cc: mmusic@ietf.org<mailto:mmusic@ietf.org>
Subject: Re: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: external clos=
ing of channel

Hi Christer,
Another good catch.  Although it is implied that closed data channels are r=
emoved from subsequent offers, this should be stated explicitly.

BR, RIchard

On Wed, Feb 26, 2014 at 7:19 AM, Christer Holmberg <christer.holmberg@erics=
son.com<mailto:christer.holmberg@ericsson.com>> wrote:
Hi,

Section 5.2.3 says:

"It is specific to the sub-protocol whether this closing must in
                addition be signaled to the peer via a new SDP offer/answer=
 exchange."

Shouldn't the draft define how it is done using an SDP offer/answer exchang=
e?

Regards,

Christer

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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:red">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:red">Comments inline.<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 style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ta=
homa&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi,<o:p></=
o:p></span></p>
<div>
<div>
<div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0i=
n 0in 0in 6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0in;margin-=
bottom:5.0pt">
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">&nbsp;You also need to be more clear on=
 whether sending of SSN reset is mandatory when using your draft
 (you only say that &#8220;browsers will send it&#8221;).</span><o:p></o:p>=
</p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-GB">The text should refer to protoc=
ol stack behavior rather than to the browser since it's supposed to also ap=
ply to data channel usage outside the context of WebRTC. &nbsp;(We intended=
 to fix all th browser references in this
 draft and missed this one.) &nbsp;So, yes, SSN reset is mandatory. &nbsp; =
&nbsp;<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"color:#1F497D"><o:p>&n=
bsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Then you a=
lso need to cover that in the procedures.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">For exampl=
e, when an entity receives an Offer removing a channel, it MUST send a SSN =
reset.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><b><i><span lang=3D"EN-GB" style=3D"font-size:11.0pt=
;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:red">[Raju] S=
ure, we will make that clear. Thanks. As I already commented on other threa=
d, the section &#8220;4.3.2 Closing data channel&#8221; needs more descript=
ion.</span></i></b><span lang=3D"EN-GB" style=3D"font-size:11.0pt;font-fami=
ly:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:red"><o:p></o:p></span>=
</p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Also, what=
 to do if an entity receives an SSN reset, but not an associated Offer.<o:p=
></o:p></span></p>
<p class=3D"MsoNormal"><b><i><span lang=3D"EN-GB" style=3D"font-size:11.0pt=
;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:red">[Raju] A=
lso as noted in other email thread, SDP offerer who wants to close a data c=
hannel should close the data channel ONLY when successful
 SDP answer is received. The above mentioned SSN reset case happens when SD=
P answerer starts closing channel right after sending SDP answer. SDP offer=
 side data channel stack needs to handle such a case anyway (we will have t=
his scenario with inband DCP too)
 and informs the application via onClose() or equivalent API. Then at SDP a=
nswer, offerer do not have to explictly close the data channel. Lastly, SSN=
 reset<o:p></o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span lang=3D"EN-GB" style=3D"font-size:11.0pt=
;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:red"><o:p>&nb=
sp;</o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span lang=3D"EN-GB" style=3D"font-size:11.0pt=
;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:red">Thanks<o=
:p></o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span lang=3D"EN-GB" style=3D"font-size:11.0pt=
;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:red">Raju<o:p=
></o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span lang=3D"EN-GB" style=3D"font-size:11.0pt=
;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:red"><o:p>&nb=
sp;</o:p></span></i></b></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">These kind=
 of error situations often cause troubles at interop testings, so let&#8217=
;s try to cover them from the beginning
</span><span lang=3D"EN-GB" style=3D"font-size:11.0pt;font-family:Wingdings=
;color:#1F497D">J</span><span lang=3D"EN-GB" style=3D"font-size:11.0pt;font=
-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p></o:=
p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Regards,<o=
:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Christer<o=
:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
</div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0i=
n 0in 0in 6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0in;margin-=
bottom:5.0pt">
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><b><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,=
&quot;sans-serif&quot;">From:</span></b><span style=3D"font-size:10.0pt;fon=
t-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Richard Ejzak [mailto:=
<a href=3D"mailto:richard.ejzak@gmail.com" target=3D"_blank">richard.ejzak@=
gmail.com</a>]
<br>
<b>Sent:</b> 26. helmikuuta 2014 16:08<br>
<b>To:</b> Christer Holmberg<br>
<b>Cc:</b> <a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf=
.org</a><br>
<b>Subject:</b> Re: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: extern=
al closing of channel</span><o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">Hi Christer,<o:p></o:p></p>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">Another good catch. &nbsp;Although it is implied that closed data =
channels are removed from subsequent offers, this should be stated explicit=
ly.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">BR, RIchard<o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;margin-bottom:12.0p=
t">&nbsp;<o:p></o:p></p>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">On Wed, Feb 26, 2014 at 7:19 AM, Christer Holmberg &lt;<a href=3D"=
mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.holmberg@=
ericsson.com</a>&gt; wrote:<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"FI">Hi,</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"FI">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"FI">Section 5.2.3 says:</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"FI">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;text-indent:.5in">
&#8221;It is specific to the sub-protocol whether this closing must in<o:p>=
</o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp; addition be signaled to the peer via a new SDP offer/an=
swer exchange.&#8221;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">Shouldn&#8217;t the draft define how it is done using an SDP offer=
/answer exchange?<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#888888">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#888888">Christer</span><o:p></o:p></p>
</div>
</div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;margin-bottom:12.0p=
t"><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><o:p></o:p></p>
</div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
</div>
</div>
</div>
</div>
</div>
</blockquote>
</div>
<p class=3D"MsoNormal"><span lang=3D"EN-GB"><o:p>&nbsp;</o:p></span></p>
</div>
</div>
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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Richard Ejzak <richard.ejzak@gmail.com>
Thread-Topic: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: external closing of channel
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Subject: Re: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: external closing of channel
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Sorry, hit the send button by accident.  Comments inline.

From: Makaraju, Maridi Raju (Raju)
Sent: Wednesday, February 26, 2014 2:32 PM
To: Christer Holmberg; Richard Ejzak
Cc: mmusic@ietf.org
Subject: RE: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: external clos=
ing of channel

Hi,
Comments inline.



Hi,
 You also need to be more clear on whether sending of SSN reset is mandator=
y when using your draft (you only say that "browsers will send it").

The text should refer to protocol stack behavior rather than to the browser=
 since it's supposed to also apply to data channel usage outside the contex=
t of WebRTC.  (We intended to fix all th browser references in this draft a=
nd missed this one.)  So, yes, SSN reset is mandatory.

Then you also need to cover that in the procedures.

For example, when an entity receives an Offer removing a channel, it MUST s=
end a SSN reset.
[Raju] Sure, we will make that clear. Thanks. As I already commented on oth=
er thread, the section "4.3.2 Closing data channel" needs more description.

Also, what to do if an entity receives an SSN reset, but not an associated =
Offer.
[Raju] Also as noted in other email thread, SDP offerer who wants to close =
a data channel should close the data channel ONLY when successful SDP answe=
r is received. The above mentioned SSN reset case happens when SDP answerer=
 starts closing channel right after sending SDP answer. SDP offer side data=
 channel stack needs to handle such a case anyway (we will have this scenar=
io with inband DCP too) and informs the application via onClose() or equiva=
lent API. Then at SDP answer, offerer do not have to explictly close the da=
ta channel. Lastly, SSN reset
[Raju] If SSN reset is received on a side which did not having a pending SD=
P answer (to remove the stream) or the stream was not closed earlier via SD=
P offer/answer then a new SDP offer has to be generated to remove the strea=
m via external negotiation. Otherwise no new SDP offer is needed as the str=
eam is being removed or was removed already.
We will add this text to section 4.3.2.

Thanks
Raju

Thanks
Raju


These kind of error situations often cause troubles at interop testings, so=
 let's try to cover them from the beginning :)

Regards,

Christer


From: Richard Ejzak [mailto:richard.ejzak@gmail.com<mailto:richard.ejzak@gm=
ail.com>]
Sent: 26. helmikuuta 2014 16:08
To: Christer Holmberg
Cc: mmusic@ietf.org<mailto:mmusic@ietf.org>
Subject: Re: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: external clos=
ing of channel

Hi Christer,
Another good catch.  Although it is implied that closed data channels are r=
emoved from subsequent offers, this should be stated explicitly.

BR, RIchard

On Wed, Feb 26, 2014 at 7:19 AM, Christer Holmberg <christer.holmberg@erics=
son.com<mailto:christer.holmberg@ericsson.com>> wrote:
Hi,

Section 5.2.3 says:

"It is specific to the sub-protocol whether this closing must in
                addition be signaled to the peer via a new SDP offer/answer=
 exchange."

Shouldn't the draft define how it is done using an SDP offer/answer exchang=
e?

Regards,

Christer

_______________________________________________
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mmusic@ietf.org<mailto:mmusic@ietf.org>
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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:#00B050">Sorry, hit the send butto=
n by accident. &nbsp;Comments inline.<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 style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Makaraju=
, Maridi Raju (Raju)
<br>
<b>Sent:</b> Wednesday, February 26, 2014 2:32 PM<br>
<b>To:</b> Christer Holmberg; Richard Ejzak<br>
<b>Cc:</b> mmusic@ietf.org<br>
<b>Subject:</b> RE: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: extern=
al closing of channel<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</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:red">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:red">Comments inline.<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 style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ta=
homa&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi,<o:p></=
o:p></span></p>
<div>
<div>
<div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0i=
n 0in 0in 6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0in;margin-=
bottom:5.0pt">
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">&nbsp;You also need to be more clear on=
 whether sending of SSN reset is mandatory when using your draft
 (you only say that &#8220;browsers will send it&#8221;).</span><o:p></o:p>=
</p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-GB">The text should refer to protoc=
ol stack behavior rather than to the browser since it's supposed to also ap=
ply to data channel usage outside the context of WebRTC. &nbsp;(We intended=
 to fix all th browser references in this
 draft and missed this one.) &nbsp;So, yes, SSN reset is mandatory. &nbsp; =
&nbsp;<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"color:#1F497D"><o:p>&n=
bsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Then you a=
lso need to cover that in the procedures.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">For exampl=
e, when an entity receives an Offer removing a channel, it MUST send a SSN =
reset.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><b><i><span lang=3D"EN-GB" style=3D"font-size:11.0pt=
;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:red">[Raju] S=
ure, we will make that clear. Thanks. As I already commented on other threa=
d, the section &#8220;4.3.2 Closing data channel&#8221; needs more descript=
ion.</span></i></b><span lang=3D"EN-GB" style=3D"font-size:11.0pt;font-fami=
ly:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:red"><o:p></o:p></span>=
</p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Also, what=
 to do if an entity receives an SSN reset, but not an associated Offer.<o:p=
></o:p></span></p>
<p class=3D"MsoNormal"><b><i><span lang=3D"EN-GB" style=3D"font-size:11.0pt=
;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:red">[Raju] A=
lso as noted in other email thread, SDP offerer who wants to close a data c=
hannel should close the data channel ONLY when successful
 SDP answer is received. The above mentioned SSN reset case happens when SD=
P answerer starts closing channel right after sending SDP answer. SDP offer=
 side data channel stack needs to handle such a case anyway (we will have t=
his scenario with inband DCP too)
 and informs the application via onClose() or equivalent API. Then at SDP a=
nswer, offerer do not have to explictly close the data channel. Lastly, SSN=
 reset<o:p></o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span lang=3D"EN-GB" style=3D"font-size:11.0pt=
;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#00B050">[Raj=
u] If SSN reset is received on a side which did not having a pending SDP an=
swer (to remove the stream) or the stream was not closed earlier
 via SDP offer/answer then a new SDP offer has to be generated to remove th=
e stream via external negotiation. Otherwise no new SDP offer is needed as =
the stream is being removed or was removed already.<o:p></o:p></span></i></=
b></p>
<p class=3D"MsoNormal"><b><i><span lang=3D"EN-GB" style=3D"font-size:11.0pt=
;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#00B050">We w=
ill add this text to section 4.3.2.<o:p></o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span lang=3D"EN-GB" style=3D"font-size:11.0pt=
;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#00B050"><o:p=
>&nbsp;</o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span lang=3D"EN-GB" style=3D"font-size:11.0pt=
;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#00B050">Than=
ks<o:p></o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span lang=3D"EN-GB" style=3D"font-size:11.0pt=
;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#00B050">Raju=
<o:p></o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span lang=3D"EN-GB" style=3D"font-size:11.0pt=
;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:red"><o:p>&nb=
sp;</o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span lang=3D"EN-GB" style=3D"font-size:11.0pt=
;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:red">Thanks<o=
:p></o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span lang=3D"EN-GB" style=3D"font-size:11.0pt=
;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:red">Raju<o:p=
></o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span lang=3D"EN-GB" style=3D"font-size:11.0pt=
;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:red"><o:p>&nb=
sp;</o:p></span></i></b></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">These kind=
 of error situations often cause troubles at interop testings, so let&#8217=
;s try to cover them from the beginning
</span><span lang=3D"EN-GB" style=3D"font-size:11.0pt;font-family:Wingdings=
;color:#1F497D">J</span><span lang=3D"EN-GB" style=3D"font-size:11.0pt;font=
-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p></o:=
p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Regards,<o=
:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Christer<o=
:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
</div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0i=
n 0in 0in 6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0in;margin-=
bottom:5.0pt">
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><b><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,=
&quot;sans-serif&quot;">From:</span></b><span style=3D"font-size:10.0pt;fon=
t-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Richard Ejzak [mailto:=
<a href=3D"mailto:richard.ejzak@gmail.com" target=3D"_blank">richard.ejzak@=
gmail.com</a>]
<br>
<b>Sent:</b> 26. helmikuuta 2014 16:08<br>
<b>To:</b> Christer Holmberg<br>
<b>Cc:</b> <a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf=
.org</a><br>
<b>Subject:</b> Re: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: extern=
al closing of channel</span><o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">Hi Christer,<o:p></o:p></p>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">Another good catch. &nbsp;Although it is implied that closed data =
channels are removed from subsequent offers, this should be stated explicit=
ly.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">BR, RIchard<o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;margin-bottom:12.0p=
t">&nbsp;<o:p></o:p></p>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">On Wed, Feb 26, 2014 at 7:19 AM, Christer Holmberg &lt;<a href=3D"=
mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.holmberg@=
ericsson.com</a>&gt; wrote:<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"FI">Hi,</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"FI">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"FI">Section 5.2.3 says:</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"FI">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;text-indent:.5in">
&#8221;It is specific to the sub-protocol whether this closing must in<o:p>=
</o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp; addition be signaled to the peer via a new SDP offer/an=
swer exchange.&#8221;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">Shouldn&#8217;t the draft define how it is done using an SDP offer=
/answer exchange?<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#888888">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#888888">Christer</span><o:p></o:p></p>
</div>
</div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;margin-bottom:12.0p=
t"><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><o:p></o:p></p>
</div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
</div>
</div>
</div>
</div>
</div>
</blockquote>
</div>
<p class=3D"MsoNormal"><span lang=3D"EN-GB"><o:p>&nbsp;</o:p></span></p>
</div>
</div>
</div>
</div>
</div>
</body>
</html>

--_000_E1FE4C082A89A246A11D7F32A95A17826DFFB693US70UWXCHMBA02z_--


From nobody Wed Feb 26 21:45:34 2014
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Subject: Re: [MMUSIC] POF/PAN draft update - move BUNDLE-specific stuff to bundle draft
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On 02/17/2014 12:38 AM, Paul Kyzivat wrote:
> Another comment on this draft:
>
> While the mechanism described in this draft is strongly motivated by
> BUNDLE, it is conceptually independent of BUNDLE. IMO all the
> BUNDLE-specific behavior should be removed from this draft, and
> instead placed into the BUNDLE draft.
>
> I realize there is a chicken/egg thing here. It would make sense to
> keep the bundle-specifics here, so they don't get forgotten, until
> they are incorporated into the bundle draft.

If moving the BUNDLE-specific stuff to the BUNDLE draft means that
BUNDLE is dependent on completion of POF/PAN before considered
completed, I'm strongly opposed to this.

BUNDLE should not depend on POF/PAN. If BUNDLE-specific aspects of
POF/PAN need to be broken out, there needs to be a third draft that
references both.


>
>     Thanks,
>     Paul
>
>
> On 2/14/14 6:43 PM, Adam Roach wrote:
>> I've updated the partial offer/partial answer draft to reflect the
>> conversations we had in Vancouver:
>> http://tools.ietf.org/html/draft-roach-mmusic-pof-pan-02
>>
>> Diffs:
>> http://tools.ietf.org/rfcdiff?url2=draft-roach-mmusic-pof-pan-02.txt
>>
>> Note that this relies on the following draft for group membership
>> manipulation:
>> http://tools.ietf.org/html/draft-roach-mmusic-groupid
>>
>> Please point out whether you think I've missed any of the Vancouver
>> discussions or other feedback. Thanks!
>>
>> /a
>>
>>
>> _______________________________________________
>> 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: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
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+ rtcweb as it impacts DCP draft.

Please see comments inline.

> I would like some clarity about how this interacts with DCEP.
>=20
> IMO these are not mutually exclusive. But they must be used with care.

[Raju] I debated on this with myself before concluding in my earlier
reply that they are mutually exclusive. My mistake, instead of=20
concluding I should have given options. To make this internal or=20
external per data channel granuality we need to add more=20
restrictions/notes to these draft.=20

http://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol-03#section-4 say=
s:
   "To avoid glare in opening Channels, each side MUST use either even or
   odd Streams when sending a DATA_CHANNEL_OPEN message.  The method
   used to determine which side uses odd or even is based on the
   underlying DTLS connection role when used in WebRTC, with the side
   acting as the DTLS client using even stream identifiers."

Where as draft-ejzak-mmusic-data-channel-sdpneg determines this=20
based on SDP offerer (even) and answerer (odd).

What if the SDP answerer becomes TLS client? The stream #id assignments
conflict with SDP offer/answer.
So, to avoid such conflict we need to add the following clarifications:
1. draft-ejzak-mmusic-data-channel-sdpneg
   If DCP is used along with external negotiation, when creating
   data channel the client should always specify the stream id=20
   to use. Leaving selection to data channel stack (DCSK) will=20
   result in conflicts as DCSK (browser) will assign stream ids=20
   per DTLS role.

2. draft-ietf-rtcweb-data-protocol-03
   Item (1) above conflicts with DCP's odd/even rule, for which
   the draft says "MUST" ("each side MUST use either even or=20
   odd Streams").
   This section 4 (and other sections) text has to be updated
   to "SHOULD" or "MAY".

If we all agree to these updates then I am ok to make=20
internal or external negotiation granularity per data channel.

Best Regards
Raju

>=20
> So, when a channel that was in the SDP previously then disappears from
> the SDP it is necessary to close the corresponding channel. But channels
> that were opened via DCEP should not be closed.
>=20
> That also means that one should never mention a DCEP channel in SDP.
>=20
> 	Thanks,
> 	Paul
>=20
> On 2/26/14 3:12 PM, Makaraju, Maridi Raju (Raju) wrote:
> > Please find my comments inline.
> >
> > But, there is no text on rejecting a data channel in the SDP answer.
> >
> > The answer can of course reject the whole m- line, using port zero in
> > the Answer m- line. But, how does the answerer reject an individual
> > sub-protocol data channel?
> >
> > Only attributes for "accepted data channels" are included in the SDP
> > answer, which indicates back to the offerer which ones are rejected
> > (they just aren't there).  This should be made clearer in the text.
> >
> >     Section 5.1.1 says, regarding the SDP dcmap attribute, for accepted
> >     data channels:
> >
> >     "This line MUST be replicated without changes in the SDP answer, if
> >
> >                      the answerer accepts the offered data channel."
> >
> >     I guess one way to reject would be to simply not include the dcmap
> >     attribute in the Answer, and the Offerer then needs to figure out
> >     what was rejected based on the stream values not present in the
> Answer.
> >
> > Yes.  That is what was intended.
> >
> > /[Raju] In addition to the above mentiooed cases, if we all agree we
> > need to add the following cases in the next draft:/
> >
> > */1./*SDP offer has no a=3Ddcmap. This has two use cases:
> >
> > a.Initial SDP offer: no data channel streams (DCS) are open yet.
> >
> > b.Subsequent SDP offer: Remove all previously opened data
> channels/streams.
> >
> > In either case, the SCTP association is kept open (and ICE procedures
> > are performed) even when no data channels are in use.
> >
> > */2./*Creation of new DCS(s) using SDP offer (initial or subsequent)
> >
> > a.Offerer has to create the data channels locally first by doing
> > createDataChannel or equivalent API into data channel stack before
> > sending offer.
> >
> > */3./*SDP offer was rejected
> >
> > a.DCSs are kept per previous SDP offer/answer. This means DCSs that are
> > to be deleted has to be deleted (i.e. calling createDataChannel or
> > equivalent API) only after successful SDP answer.
> >
> > b.If any new DCSs are attempted to be created per new SDP offer, these
> > have to be closed now.
> >
> > */4./*Per
> > http://dev.w3.org/2011/webrtc/editor/webrtc.html#dom-peerconnection-
> createdatachannel
> > createDataChannel() API the 'external' negotiation field is per data
> > channel.  But this draft only allows external negotiation to be used fo=
r
> > all or no streams; it can't be selective. This restriction is also
> > needed so that the SDP offer/answer odd/even stream id allocation won't
> > conflict DCP's TLS role based odd/even mechanism.
> >
> > Best Regards!
> >
> > Raju
> >
> >
> >
> > _______________________________________________
> > 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


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>, "Paul Kyzivat" <pkyzivat@alum.mit.edu>, "mmusic@ietf.org" <mmusic@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: external rejection of channel
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Subject: Re: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: external rejection of channel
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Hi,

Isn't it enough to, in the data channel protocol spec, say that the odd/eve=
n rule applies unless the stream id is explicitly negotiated using some oth=
er mechanism?

Or something like that...

Regards,

Christer




-----Alkuper=E4inen viesti-----
L=E4hett=E4j=E4: mmusic [mailto:mmusic-bounces@ietf.org] Puolesta Makaraju,=
 Maridi Raju (Raju)
L=E4hetetty: 27. helmikuuta 2014 21:55
Vastaanottaja: Paul Kyzivat; mmusic@ietf.org; rtcweb@ietf.org
Aihe: Re: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: external rejecti=
on of channel

+ rtcweb as it impacts DCP draft.

Please see comments inline.

> I would like some clarity about how this interacts with DCEP.
>=20
> IMO these are not mutually exclusive. But they must be used with care.

[Raju] I debated on this with myself before concluding in my earlier reply =
that they are mutually exclusive. My mistake, instead of concluding I shoul=
d have given options. To make this internal or external per data channel gr=
anuality we need to add more restrictions/notes to these draft.=20

http://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol-03#section-4 say=
s:
   "To avoid glare in opening Channels, each side MUST use either even or
   odd Streams when sending a DATA_CHANNEL_OPEN message.  The method
   used to determine which side uses odd or even is based on the
   underlying DTLS connection role when used in WebRTC, with the side
   acting as the DTLS client using even stream identifiers."

Where as draft-ejzak-mmusic-data-channel-sdpneg determines this based on SD=
P offerer (even) and answerer (odd).

What if the SDP answerer becomes TLS client? The stream #id assignments con=
flict with SDP offer/answer.
So, to avoid such conflict we need to add the following clarifications:
1. draft-ejzak-mmusic-data-channel-sdpneg
   If DCP is used along with external negotiation, when creating
   data channel the client should always specify the stream id=20
   to use. Leaving selection to data channel stack (DCSK) will=20
   result in conflicts as DCSK (browser) will assign stream ids=20
   per DTLS role.

2. draft-ietf-rtcweb-data-protocol-03
   Item (1) above conflicts with DCP's odd/even rule, for which
   the draft says "MUST" ("each side MUST use either even or=20
   odd Streams").
   This section 4 (and other sections) text has to be updated
   to "SHOULD" or "MAY".

If we all agree to these updates then I am ok to make internal or external =
negotiation granularity per data channel.

Best Regards
Raju

>=20
> So, when a channel that was in the SDP previously then disappears from=20
> the SDP it is necessary to close the corresponding channel. But=20
> channels that were opened via DCEP should not be closed.
>=20
> That also means that one should never mention a DCEP channel in SDP.
>=20
> 	Thanks,
> 	Paul
>=20
> On 2/26/14 3:12 PM, Makaraju, Maridi Raju (Raju) wrote:
> > Please find my comments inline.
> >
> > But, there is no text on rejecting a data channel in the SDP answer.
> >
> > The answer can of course reject the whole m- line, using port zero=20
> > in the Answer m- line. But, how does the answerer reject an=20
> > individual sub-protocol data channel?
> >
> > Only attributes for "accepted data channels" are included in the SDP=20
> > answer, which indicates back to the offerer which ones are rejected=20
> > (they just aren't there).  This should be made clearer in the text.
> >
> >     Section 5.1.1 says, regarding the SDP dcmap attribute, for accepted
> >     data channels:
> >
> >     "This line MUST be replicated without changes in the SDP answer,=20
> > if
> >
> >                      the answerer accepts the offered data channel."
> >
> >     I guess one way to reject would be to simply not include the dcmap
> >     attribute in the Answer, and the Offerer then needs to figure out
> >     what was rejected based on the stream values not present in the
> Answer.
> >
> > Yes.  That is what was intended.
> >
> > /[Raju] In addition to the above mentiooed cases, if we all agree we=20
> > need to add the following cases in the next draft:/
> >
> > */1./*SDP offer has no a=3Ddcmap. This has two use cases:
> >
> > a.Initial SDP offer: no data channel streams (DCS) are open yet.
> >
> > b.Subsequent SDP offer: Remove all previously opened data
> channels/streams.
> >
> > In either case, the SCTP association is kept open (and ICE=20
> > procedures are performed) even when no data channels are in use.
> >
> > */2./*Creation of new DCS(s) using SDP offer (initial or subsequent)
> >
> > a.Offerer has to create the data channels locally first by doing=20
> > createDataChannel or equivalent API into data channel stack before=20
> > sending offer.
> >
> > */3./*SDP offer was rejected
> >
> > a.DCSs are kept per previous SDP offer/answer. This means DCSs that=20
> > are to be deleted has to be deleted (i.e. calling createDataChannel=20
> > or equivalent API) only after successful SDP answer.
> >
> > b.If any new DCSs are attempted to be created per new SDP offer,=20
> > these have to be closed now.
> >
> > */4./*Per
> > http://dev.w3.org/2011/webrtc/editor/webrtc.html#dom-peerconnection-
> createdatachannel
> > createDataChannel() API the 'external' negotiation field is per data=20
> > channel.  But this draft only allows external negotiation to be used=20
> > for all or no streams; it can't be selective. This restriction is=20
> > also needed so that the SDP offer/answer odd/even stream id=20
> > allocation won't conflict DCP's TLS role based odd/even mechanism.
> >
> > Best Regards!
> >
> > Raju
> >
> >
> >
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www.ietf.org/mailman/listinfo/mmusic
> >
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

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


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To: Christer Holmberg <christer.holmberg@ericsson.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>, "mmusic@ietf.org" <mmusic@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: external rejection of channel
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Hi Christer,

> Hi,
>=20
> Isn't it enough to, in the data channel protocol spec, say that the odd/e=
ven
> rule applies unless the stream id is explicitly negotiated using some oth=
er
> mechanism?
>=20
[Raju] Yes, that is sufficient from DCP draft. draft-ejzak-mmusic-data-chan=
nel-sdpneg should still need to say that "when both external and DCP are us=
ed, even for DCP created streams the stream ids have to be specified and ma=
naged by the application (else DCP stack will default to odd/even rule per =
DTLS role which may conflict with SDP offer/answer rule)".

Thanks for agreeing to add this to the draft!

Raju


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A new Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Multiparty Multimedia Session Control Working Group of the IETF.

    Title         : Offer/Answer Considerations for G723 Annex A and G729 Annex B
    Author(s)     : M. Perumal, et al
    Filename      : draft-ietf-mmusic-sdp-g723-g729
    Pages         : 8 
    Date          : 2014-02-28 
    
   This document provides the offer/answer considerations for the annexa
   parameter of G723 and the annexb parameter of G729, G729D and G729E
   when the value of the annexa or annexb parameter does not match in
   the Session Description protocol (SDP) offer and answer.


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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-g723-g729-06.txt
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This revision addresses the comments received during IESG evaluation.

Muthu

|-----Original Message-----
|From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of Internet-Drafts=
@ietf.org
|Sent: Friday, February 28, 2014 4:04 PM
|To: i-d-announce@ietf.org
|Cc: mmusic@ietf.org
|Subject: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-g723-g729-06.txt
|
|A new Internet-Draft is available from the on-line Internet-Drafts directo=
ries.
|This draft is a work item of the Multiparty Multimedia Session Control Wor=
king Group of the IETF.
|
|    Title         : Offer/Answer Considerations for G723 Annex A and G729 =
Annex B
|    Author(s)     : M. Perumal, et al
|    Filename      : draft-ietf-mmusic-sdp-g723-g729
|    Pages         : 8
|    Date          : 2014-02-28
|
|   This document provides the offer/answer considerations for the annexa
|   parameter of G723 and the annexb parameter of G729, G729D and G729E
|   when the value of the annexa or annexb parameter does not match in
|   the Session Description protocol (SDP) offer and answer.
|
|
|A URL for this Internet-Draft is:
|http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-g723-g729-06.txt
|
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|Internet-Draft.


From nobody Fri Feb 28 04:36:31 2014
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Subject: [MMUSIC] Protocol Action: 'Offer/Answer Considerations for G723 Annex A and G729 Annex B' to Proposed Standard (draft-ietf-mmusic-sdp-g723-g729-06.txt)
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The IESG has approved the following document:
- 'Offer/Answer Considerations for G723 Annex A and G729 Annex B'
  (draft-ietf-mmusic-sdp-g723-g729-06.txt) as Proposed Standard

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

The IESG contact persons are Gonzalo Camarillo and Richard Barnes.

A URL of this Internet Draft is:
http://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-g723-g729/




Technical Summary

   RFC4856 describes the annexa parameter for G723 and the annexb
   parameter for G729, G729D and G729E. However, the specification does
   not describe the offerer and answerer behavior when the value of the
   annexa or annexb parameter does not match in the Session Description
   protocol (SDP) offer and answer.  This document provides the offer/
   answer considerations for these parameters and updates RFC4856.


Working Group Summary

   There is WG consensus behind this draft. All issues raised during
   and before WGLC have been addressed.  


Document Quality

  This document clarifies behavior with the discussed SDP parameters
  to be in line with existing specifications and pragmatic
  implementations.  The document has been discussed in MMUSIC,
  PAYLOAD, and AVTEXT working groups and has been revised according to
  the feedback.


Personnel

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

  Ari Keranen is the Document Shepherd, and the Responsible Area
  Director is Gonzalo Camarillo.


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Subject: Re: [MMUSIC] ICE-bis: pacing of ICE connectivity checks
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Hi Justin,

We discussed this issue at the previous IETF meeting and back then you 
were doing large scale measurements with Chrome on the sensible 
connectivity check values. Did you come into some conclusions with those 
studies? Anything new we could recommend based on the results?


Cheers,
Ari

On 18/10/13 21:34, Justin Uberti wrote:
> FWIW, we are now doing trials to understand what is the absolute lower
> bound for ICE checks on today's internet (i.e. what is the minimum time
> to allow for a NAT to create a new UDP binding).
>
>
> On Fri, Oct 18, 2013 at 10:30 AM, Adam Roach <adam@nostrum.com
> <mailto:adam@nostrum.com>> wrote:
>
>     On 10/18/13 09:44, Emil Ivov wrote:
>>     It might be worth reminding exactly what these 50ms would be
>>     protecting us against
>
>     They're protecting us against RTP arriving at a rate of 20ms per packet.
>
>
>>     and why is it that we don't need to be
>>     protected against it in the case of RTP where 20ms is accepted as a
>>     fine default value.
>
>     Well, it's not. This is exactly why we ended up requiring ICE: it
>     ensures explicit consent before you start blowing media at someone.
>
>     See also
>     http://www.ietf.org/mail-archive/web/mmusic/current/msg04589.html
>
>     /a
>
>     _______________________________________________
>     mmusic mailing list
>     mmusic@ietf.org <mailto:mmusic@ietf.org>
>     https://www.ietf.org/mailman/listinfo/mmusic
>
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


From nobody Fri Feb 28 07:57:18 2014
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To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>,  Christer Holmberg <christer.holmberg@ericsson.com>, "mmusic@ietf.org" <mmusic@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: Re: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: external rejection of channel
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Raju,

On 2/27/14 5:59 PM, Makaraju, Maridi Raju (Raju) wrote:
> Hi Christer,
>
>> Hi,
>>
>> Isn't it enough to, in the data channel protocol spec, say that the odd/even
>> rule applies unless the stream id is explicitly negotiated using some other
>> mechanism?
>>
> [Raju] Yes, that is sufficient from DCP draft. draft-ejzak-mmusic-data-channel-sdpneg should still need to say that "when both external and DCP are used, even for DCP created streams the stream ids have to be specified and managed by the application (else DCP stack will default to odd/even rule per DTLS role which may conflict with SDP offer/answer rule)".

Can we assume there must be a mechanism for using DCEP and still 
controlling which channels are used?

It might help if the O/A negotiation also followed the even/odd rule.
But there are difficulties in knowing who is even and who is odd for the 
first O/A.

If the offerer always creates the channel before sending the offer, then 
that will prevent the two mechanisms from stepping on one another's 
stream ids.

	Thanks,
	Paul

> Thanks for agreeing to add this to the draft!
>
> Raju
>
>


From nobody Fri Feb 28 07:58:52 2014
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Date: Fri, 28 Feb 2014 17:58:46 +0200
From: =?ISO-8859-1?Q?Ari_Ker=E4nen?= <ari.keranen@ericsson.com>
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Subject: [MMUSIC] ICE options: need new optional and/or required attributes?
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Hi all,

One open issue regarding ICE-bis has been the semantics of the 
ice-options attribute. Currently there is no way of saying "do 
aggressive ICE even if you don't understand this option" or "reject my 
offer if you don't understand this option".

One way of solving this would be to introduce a new attribute like:

a=ice-options-required:foo

That is, if "foo" is unknown option to the answerer, the offer is 
rejected. In addition to solving the second case, it could also be used 
to test for the first and if option was not understood, simply leave it 
out of the new offer. Of course this would result in one signaling 
exchange of delay, and hence for the first case (do still aggressive) we 
could also introduce yet another option:

a=ice-options-optional:bar

Opinions?


Cheers,
Ari


From nobody Fri Feb 28 08:46:54 2014
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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, Christer Holmberg <christer.holmberg@ericsson.com>, "mmusic@ietf.org" <mmusic@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: external rejection of channel
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Subject: Re: [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: external rejection of channel
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Hi Paul,


> >> Isn't it enough to, in the data channel protocol spec, say that the
> odd/even
> >> rule applies unless the stream id is explicitly negotiated using some
> other
> >> mechanism?
> >>
> > [Raju] Yes, that is sufficient from DCP draft. draft-ejzak-mmusic-data-
> channel-sdpneg should still need to say that "when both external and DCP =
are
> used, even for DCP created streams the stream ids have to be specified an=
d
> managed by the application (else DCP stack will default to odd/even rule =
per
> DTLS role which may conflict with SDP offer/answer rule)".
>=20
> Can we assume there must be a mechanism for using DCEP and still
> controlling which channels are used?

[Raju] Yes, we can assume that and it is already supported by :
http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-07#section-6.5
and
http://dev.w3.org/2011/webrtc/editor/webrtc.html#widl-RTCDataChannel-id
I believe it is safe to assume non-browser implementations of=20
data channel stack gives an option for application to select=20
stream ids independent of DCEP or external negotiation.


>=20
> It might help if the O/A negotiation also followed the even/odd rule.
[Raju] May be I am missing something here. Richard's draft already has
a specific even/odd rule for O/A. But the issue is TLS based rule may
conflict with the O/A rule as TLS roles may change depending on a=3Dsetup.

> But there are difficulties in knowing who is even and who is odd for the
> first O/A.

[Raju] Not related to first or subsequent O/A, but rather a conflict=20
between TLS roles vs. O/A.

>=20
> If the offerer always creates the channel before sending the offer, then
> that will prevent the two mechanisms from stepping on one another's
> stream ids.
[Raju]=20
In some cases, initial offer may not have a data channel stream created.=20
I don't know how this can prevent stepping on each other?!
May be I am missing something here?! In most cases TLS role is=20
determined after O/A is completed, but offerer creates a data=20
channel before answer, then how does it know even/odd rule?=20
So, trying to use TLS role for external negotiation is problematic.

Thanks
Raju

