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From: Martin Thomson <martin.thomson@gmail.com>
Date: Sun, 1 Jan 2017 21:49:15 +1100
Message-ID: <CABkgnnWLw7QPLd6qtgN1C-Pg+UHim6s=QK0EFgkYViQy8Ad2oQ@mail.gmail.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Archived-At: <https://mailarchive.ietf.org/arch/msg/mmusic/jKvFP_Y5SqwOLN2Et97a5KFYdPg>
Cc: "Jonathan Lennox \(jonathan@vidyo.com\)" <jonathan@vidyo.com>, "mmusic@ietf.org" <mmusic@ietf.org>, "Cullen Jennings \(fluffy@iii.ca\)" <fluffy@iii.ca>
Subject: Re: [MMUSIC] draft-4572-update: Spec contains references to a number of obsoleted RFCs
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We can remove MD2.  MD5 is dead, SHA-1 is in its death throes, but MD2
is merely a (bad) memory.

On 30 December 2016 at 22:27, Christer Holmberg
<christer.holmberg@ericsson.com> wrote:
> Hi,
>
>
>
> Please note the following:
>
>
>
> RFC 6149, which obsoletes RFC 1319, makes MD2 historic. Do people have a
> problem with that? I assume we=E2=80=99ll end up in trouble with the secu=
rity folks
> if we keep the old RFC=E2=80=A6
>
>
>
> And, considering MD2 is historic, do we even need to mention it in
> draft-4572-update anymore?
>
>
>
> Regards,
>
>
>
> Christer
>
>
>
>
>
> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of Christer Holmb=
erg
> Sent: 30 December 2016 11:42
> To: mmusic@ietf.org
> Cc: Jonathan Lennox (jonathan@vidyo.com) <jonathan@vidyo.com>; Cullen
> Jennings (fluffy@iii.ca) <fluffy@iii.ca>
> Subject: [MMUSIC] draft-4572-update: Spec contains references to a number=
 of
> obsoleted RFCs
>
>
>
> Hi,
>
>
>
> The idnits check returns the following for draft-4572-update.
>
>
>
> ** Obsolete normative reference: RFC 1319 (ref. '3') (Obsoleted by RFC 61=
49)
>
>
>
>   ** Downref: Normative reference to an Informational RFC: RFC 1321 (ref.
> '4')
>
>
>
>   ** Obsolete normative reference: RFC 3280 (ref. '8') (Obsoleted by RFC
> 5280)
>
>
>
>   ** Obsolete normative reference: RFC 4234 (ref. '11') (Obsoleted by RFC
>
>      5234)
>
>
>
>   ** Obsolete normative reference: RFC 4288 (ref. '12') (Obsoleted by RFC
>
>      6838)
>
>
>
>   ** Obsolete normative reference: RFC 4346 (ref. '13') (Obsoleted by RFC
>
>      5246)
>
>
>
>   -- Obsolete informational reference (is this intentional?): RFC 2617 (r=
ef.
>
>      '15') (Obsoleted by RFC 7235, RFC 7615, RFC 7616, RFC 7617)
>
>
>
>   -- Obsolete informational reference (is this intentional?): RFC 3525 (r=
ef.
>
>      '20') (Obsoleted by RFC 5125)
>
>
>
>   -- Obsolete informational reference (is this intentional?): RFC 3851 (r=
ef.
>
>      '22') (Obsoleted by RFC 5751)
>
>
>
> The reason for this is that we used RFC 4572 as base, and did not
> change/update the references.
>
>
>
> I had a look, and I don=E2=80=99t think there should be any issues in rep=
lacing the
> current RFCs with the new ones. But, please indicate if you see any issue=
s.
>
>
>
> Regards,
>
>
>
> Christer


From nobody Mon Jan  2 01:48:42 2017
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Adam Roach <adam@nostrum.com>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] BUNDLE: meaning of "unspecified" when describing "bundle-only"
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Subject: Re: [MMUSIC] BUNDLE: meaning of "unspecified" when describing "bundle-only"
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Hi Adam,

Thanks for your comment! I'll get back to you asap.

Regards,

Christer

-----Original Message-----
From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of Adam Roach
Sent: 29 December 2016 23:16
To: mmusic@ietf.org
Subject: [MMUSIC] BUNDLE: meaning of "unspecified" when describing "bundle-=
only"

We've recently come across an issue with the way the "bundle-only"=20
attribute is described in the current document. The current language regard=
ing port handling reads:

    The usage of the 'bundle-only' attribute is only defined for a
    bundled "m=3D" line with a zero port value, within an offer. Other
    usage is unspecified.

Usually, when we have this kind of language, we still ensure that behavior =
is well defined, to help avoid unnecessary interop failures. I see a couple=
 of different options here:

 1. Remove the final sentence and add language saying that creators of
    SDP MUST NOT include a "bundle-only" attribute in an m-section that
    has a non-zero port, and that recipients of such SDP {SHOULD,MUST}
    reject it; or

 2. Retain language saying that including a "bundle-only" attribute in a
    non-zero m-section is unspecified, but add normative language along
    the lines of: "implementations that receive an m-section with a
    non-zero port that also contains a 'bundle-only' attribute MUST
    ignore the {attribute,port}."

I don't have a preference between these choices, but I think we do need cla=
rity. To be absolutely clear, this feedback is based on actual implementati=
on interop failures in the field. This problem is not theoretical.

/a

_______________________________________________
mmusic mailing list
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Martin Thomson <martin.thomson@gmail.com>
Thread-Topic: draft-4572-update: Spec contains references to a number of obsoleted RFCs
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Cc: "Jonathan Lennox \(jonathan@vidyo.com\)" <jonathan@vidyo.com>, "mmusic@ietf.org" <mmusic@ietf.org>, "Cullen Jennings \(fluffy@iii.ca\)" <fluffy@iii.ca>
Subject: Re: [MMUSIC] draft-4572-update: Spec contains references to a number of obsoleted RFCs
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From nobody Mon Jan  2 04:25:58 2017
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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 of the IETF.

        Title           : Connection-Oriented Media Transport over TLS in SDP
        Authors         : Jonathan Lennox
                          Christer Holmberg
	Filename        : draft-ietf-mmusic-4572-update-09.txt
	Pages           : 17
	Date            : 2017-01-02

Abstract:
   This document specifies how to establish secure connection-oriented
   media transport sessions over the Transport Layer Security (TLS)
   protocol using the Session Description Protocol (SDP).  It defines a
   new SDP protocol identifier, 'TCP/TLS'.  It also defines the syntax
   and semantics for an SDP 'fingerprint' attribute that identifies the
   certificate that will be presented for the TLS session.  This
   mechanism allows media transport over TLS connections to be
   established securely, so long as the integrity of session
   descriptions is assured.

   This document obsoletes RFC 4572, by clarifying the usage of multiple
   fingerprints.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-4572-update/

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-mmusic-4572-update-09

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-4572-update-09


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 Mon Jan  2 05:42:34 2017
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Subject: Re: [MMUSIC] JSEP Issue #394: What appears in m= lines.
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Den 22. des. 2016 18:30, skrev Eric Rescorla:
>      __
> 
>     Okay, I agree that having any rules for what you should offer here
>     based on what the actual outcome is not the best idea here. I think
>     the rules should be based on capability and intent. So if you only
>     are going offer TCP candidates, then you clearly should use TCP/…,
> 
> 
> I'm going to push on this. If we've already agreed that mismatches are
> normal, why should we do that? Wouldn't it be better to just effectively
> deprecate this field?
> 

Concur with EKR.

This field (the TCP/UDP part of the protocol field) was defined in a way
that makes its original purpose (to negotiate profiles) not only
impossible while using ICE, but actively harmful, because you end up
telling lies in very many cases, so people will have to accept things
that don't match reality.

Make the world as simple for implementors as possible: Send a fixed
string, and accept all strings.




From nobody Mon Jan  2 10:04:21 2017
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Cc: "Jonathan Lennox \(jonathan@vidyo.com\)" <jonathan@vidyo.com>, "mmusic@ietf.org" <mmusic@ietf.org>, "Cullen Jennings \(fluffy@iii.ca\)" <fluffy@iii.ca>
Subject: Re: [MMUSIC] draft-4572-update: Spec contains references to a number of obsoleted RFCs
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On Mon, Jan 2, 2017 at 2:41 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> >We can remove MD2.  MD5 is dead, SHA-1 is in its death throes, but MD2 i=
s
> merely a >(bad) memory.
>
> So, my suggestion is to remove all references to MD2 (including the ABNF)
> for now, and we'll then see what the security folks say about MD5 and SHA=
-1.
>

Given the threat model here, I think we want to tell people to ignore MD*
(i.e., treat it as an unknown hash) and to accept SHA-1 (though perhaps
only temporarily). Accordingly, I propose removing MD2 and MD5 from this
grammar, but leave SHA-1.

-Ekr


Regards,
>
> Christer
>
>
> On 30 December 2016 at 22:27, Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
> > Hi,
> >
> >
> >
> > Please note the following:
> >
> >
> >
> > RFC 6149, which obsoletes RFC 1319, makes MD2 historic. Do people have
> > a problem with that? I assume we=E2=80=99ll end up in trouble with the
> > security folks if we keep the old RFC=E2=80=A6
> >
> >
> >
> > And, considering MD2 is historic, do we even need to mention it in
> > draft-4572-update anymore?
> >
> >
> >
> > Regards,
> >
> >
> >
> > Christer
> >
> >
> >
> >
> >
> > From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of Christer
> > Holmberg
> > Sent: 30 December 2016 11:42
> > To: mmusic@ietf.org
> > Cc: Jonathan Lennox (jonathan@vidyo.com) <jonathan@vidyo.com>; Cullen
> > Jennings (fluffy@iii.ca) <fluffy@iii.ca>
> > Subject: [MMUSIC] draft-4572-update: Spec contains references to a
> > number of obsoleted RFCs
> >
> >
> >
> > Hi,
> >
> >
> >
> > The idnits check returns the following for draft-4572-update.
> >
> >
> >
> > ** Obsolete normative reference: RFC 1319 (ref. '3') (Obsoleted by RFC
> > 6149)
> >
> >
> >
> >   ** Downref: Normative reference to an Informational RFC: RFC 1321 (re=
f.
> > '4')
> >
> >
> >
> >   ** Obsolete normative reference: RFC 3280 (ref. '8') (Obsoleted by
> > RFC
> > 5280)
> >
> >
> >
> >   ** Obsolete normative reference: RFC 4234 (ref. '11') (Obsoleted by
> > RFC
> >
> >      5234)
> >
> >
> >
> >   ** Obsolete normative reference: RFC 4288 (ref. '12') (Obsoleted by
> > RFC
> >
> >      6838)
> >
> >
> >
> >   ** Obsolete normative reference: RFC 4346 (ref. '13') (Obsoleted by
> > RFC
> >
> >      5246)
> >
> >
> >
> >   -- Obsolete informational reference (is this intentional?): RFC 2617
> (ref.
> >
> >      '15') (Obsoleted by RFC 7235, RFC 7615, RFC 7616, RFC 7617)
> >
> >
> >
> >   -- Obsolete informational reference (is this intentional?): RFC 3525
> (ref.
> >
> >      '20') (Obsoleted by RFC 5125)
> >
> >
> >
> >   -- Obsolete informational reference (is this intentional?): RFC 3851
> (ref.
> >
> >      '22') (Obsoleted by RFC 5751)
> >
> >
> >
> > The reason for this is that we used RFC 4572 as base, and did not
> > change/update the references.
> >
> >
> >
> > I had a look, and I don=E2=80=99t think there should be any issues in
> > replacing the current RFCs with the new ones. But, please indicate if
> you see any issues.
> >
> >
> >
> > Regards,
> >
> >
> >
> > Christer
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>

--001a11492dfa53575105452061a6
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On M=
on, Jan 2, 2017 at 2:41 AM, Christer Holmberg <span dir=3D"ltr">&lt;<a href=
=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.holmb=
erg@ericsson.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote"=
 style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi=
,<br>
<span class=3D""><br>
&gt;We can remove MD2.=C2=A0 MD5 is dead, SHA-1 is in its death throes, but=
 MD2 is merely a &gt;(bad) memory.<br>
<br>
</span>So, my suggestion is to remove all references to MD2 (including the =
ABNF) for now, and we&#39;ll then see what the security folks say about MD5=
 and SHA-1.<br></blockquote><div><br></div><div>Given the threat model here=
, I think we want to tell people to ignore MD* (i.e., treat it as an unknow=
n hash) and to accept SHA-1 (though perhaps only temporarily). Accordingly,=
 I propose removing MD2 and MD5 from this grammar, but leave SHA-1.</div><d=
iv><br></div><div>-Ekr</div><div><br></div><div><br></div><blockquote class=
=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rg=
b(204,204,204);padding-left:1ex">
Regards,<br>
<br>
Christer<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
On 30 December 2016 at 22:27, Christer Holmberg &lt;<a href=3D"mailto:chris=
ter.holmberg@ericsson.com">christer.holmberg@ericsson.<wbr>com</a>&gt; wrot=
e:<br>
&gt; Hi,<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; Please note the following:<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; RFC 6149, which obsoletes RFC 1319, makes MD2 historic. Do people have=
<br>
&gt; a problem with that? I assume we=E2=80=99ll end up in trouble with the=
<br>
&gt; security folks if we keep the old RFC=E2=80=A6<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; And, considering MD2 is historic, do we even need to mention it in<br>
&gt; draft-4572-update anymore?<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; Regards,<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; Christer<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; From: mmusic [mailto:<a href=3D"mailto:mmusic-bounces@ietf.org">mmusic=
-bounces@ietf.<wbr>org</a>] On Behalf Of Christer<br>
&gt; Holmberg<br>
&gt; Sent: 30 December 2016 11:42<br>
&gt; To: <a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
&gt; Cc: Jonathan Lennox (<a href=3D"mailto:jonathan@vidyo.com">jonathan@vi=
dyo.com</a>) &lt;<a href=3D"mailto:jonathan@vidyo.com">jonathan@vidyo.com</=
a>&gt;; Cullen<br>
&gt; Jennings (<a href=3D"mailto:fluffy@iii.ca">fluffy@iii.ca</a>) &lt;<a h=
ref=3D"mailto:fluffy@iii.ca">fluffy@iii.ca</a>&gt;<br>
&gt; Subject: [MMUSIC] draft-4572-update: Spec contains references to a<br>
&gt; number of obsoleted RFCs<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; Hi,<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; The idnits check returns the following for draft-4572-update.<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; ** Obsolete normative reference: RFC 1319 (ref. &#39;3&#39;) (Obsolete=
d by RFC<br>
&gt; 6149)<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0** Downref: Normative reference to an Informational RFC: R=
FC 1321 (ref.<br>
&gt; &#39;4&#39;)<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0** Obsolete normative reference: RFC 3280 (ref. &#39;8&#39=
;) (Obsoleted by<br>
&gt; RFC<br>
&gt; 5280)<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0** Obsolete normative reference: RFC 4234 (ref. &#39;11&#3=
9;) (Obsoleted by<br>
&gt; RFC<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 5234)<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0** Obsolete normative reference: RFC 4288 (ref. &#39;12&#3=
9;) (Obsoleted by<br>
&gt; RFC<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 6838)<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0** Obsolete normative reference: RFC 4346 (ref. &#39;13&#3=
9;) (Obsoleted by<br>
&gt; RFC<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 5246)<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0-- Obsolete informational reference (is this intentional?)=
: RFC 2617 (ref.<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 &#39;15&#39;) (Obsoleted by RFC 7235, RFC 7615, RF=
C 7616, RFC 7617)<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0-- Obsolete informational reference (is this intentional?)=
: RFC 3525 (ref.<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 &#39;20&#39;) (Obsoleted by RFC 5125)<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0-- Obsolete informational reference (is this intentional?)=
: RFC 3851 (ref.<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 &#39;22&#39;) (Obsoleted by RFC 5751)<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; The reason for this is that we used RFC 4572 as base, and did not<br>
&gt; change/update the references.<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; I had a look, and I don=E2=80=99t think there should be any issues in<=
br>
&gt; replacing the current RFCs with the new ones. But, please indicate if =
you see any issues.<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; Regards,<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; Christer<br>
______________________________<wbr>_________________<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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/mmusic</a><br=
>
</div></div></blockquote></div><br></div></div>

--001a11492dfa53575105452061a6--


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From: Martin Thomson <martin.thomson@gmail.com>
Date: Tue, 3 Jan 2017 13:28:15 +1100
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To: Eric Rescorla <ekr@rtfm.com>
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Cc: "Jonathan Lennox \(jonathan@vidyo.com\)" <jonathan@vidyo.com>, "mmusic@ietf.org" <mmusic@ietf.org>, "Cullen Jennings \(fluffy@iii.ca\)" <fluffy@iii.ca>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] draft-4572-update: Spec contains references to a number of obsoleted RFCs
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On 3 January 2017 at 05:03, Eric Rescorla <ekr@rtfm.com> wrote:
> Given the threat model here, I think we want to tell people to ignore MD*
> (i.e., treat it as an unknown hash) and to accept SHA-1 (though perhaps only
> temporarily). Accordingly, I propose removing MD2 and MD5 from this grammar,
> but leave SHA-1.


That makes sense.  These busted hashes will then appear to be
unsupported extensions.


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Martin Thomson <martin.thomson@gmail.com>
Thread-Topic: [MMUSIC] draft-4572-update: Spec contains references to a number of obsoleted RFCs
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References: <7594FB04B1934943A5C02806D1A2204B4BF50A9B@ESESSMB209.ericsson.se> <7594FB04B1934943A5C02806D1A2204B4BF50DF5@ESESSMB209.ericsson.se> <CABkgnnWLw7QPLd6qtgN1C-Pg+UHim6s=QK0EFgkYViQy8Ad2oQ@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B4BF53260@ESESSMB209.ericsson.se> <CABcZeBNGm27Hf4mrGosjpAMOYSc2_O-4q72-HNpC5g0D_mhKzQ@mail.gmail.com>, <CABkgnnUBJ7VxRPpsVAebp4cLS6pgk6osWCV3AF-5JyTTZKvLrg@mail.gmail.com>
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Subject: Re: [MMUSIC] draft-4572-update: Spec contains references to a number of obsoleted RFCs
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Hi,

I am fine to also remove MD5. However, I want to make sure we don't bring b=
ack Cullen's backward compatibility issues.

Regards,

Christer


Sent from my iPhone

> On 3 Jan 2017, at 4.28, Martin Thomson <martin.thomson@gmail.com> wrote:
>=20
>> On 3 January 2017 at 05:03, Eric Rescorla <ekr@rtfm.com> wrote:
>> Given the threat model here, I think we want to tell people to ignore MD=
*
>> (i.e., treat it as an unknown hash) and to accept SHA-1 (though perhaps =
only
>> temporarily). Accordingly, I propose removing MD2 and MD5 from this gram=
mar,
>> but leave SHA-1.
>=20
>=20
> That makes sense.  These busted hashes will then appear to be
> unsupported extensions.


From nobody Tue Jan  3 09:34:43 2017
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From: Cullen Jennings <fluffy@iii.ca>
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References: <7594FB04B1934943A5C02806D1A2204B4BF50A9B@ESESSMB209.ericsson.se> <7594FB04B1934943A5C02806D1A2204B4BF50DF5@ESESSMB209.ericsson.se> <CABkgnnWLw7QPLd6qtgN1C-Pg+UHim6s=QK0EFgkYViQy8Ad2oQ@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B4BF53260@ESESSMB209.ericsson.se> <CABcZeBNGm27Hf4mrGosjpAMOYSc2_O-4q72-HNpC5g0D_mhKzQ@mail.gmail.com>
To: Eric Rescorla <ekr@rtfm.com>
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Cc: Jonathan Lennox <jonathan@vidyo.com>, "mmusic@ietf.org" <mmusic@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] draft-4572-update: Spec contains references to a number of obsoleted RFCs
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Trivial nit but ...

Actually I don't think you want MD5 treated as a unknown hash, you want =
treated as a known but don't use. If A offers old and bad crypto to B, =
we have B log that such that we can track down and upgrade A. If we got =
a a new unknown cipher call SSHHAA we would not log that as bad crypto =
because we would assume it was new and good.=20


> On Jan 2, 2017, at 11:03 AM, Eric Rescorla <ekr@rtfm.com> wrote:
>=20
> On Mon, Jan 2, 2017 at 2:41 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
> Hi,
>=20
> >We can remove MD2.  MD5 is dead, SHA-1 is in its death throes, but =
MD2 is merely a >(bad) memory.
>=20
> So, my suggestion is to remove all references to MD2 (including the =
ABNF) for now, and we'll then see what the security folks say about MD5 =
and SHA-1.
>=20
> Given the threat model here, I think we want to tell people to ignore =
MD* (i.e., treat it as an unknown hash) and to accept SHA-1 (though =
perhaps only temporarily). Accordingly, I propose removing MD2 and MD5 =
from this grammar, but leave SHA-1.
>=20
> -Ekr
>=20
>=20
> Regards,
>=20
> Christer
>=20
>=20
> On 30 December 2016 at 22:27, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
> > Hi,
> >
> >
> >
> > Please note the following:
> >
> >
> >
> > RFC 6149, which obsoletes RFC 1319, makes MD2 historic. Do people =
have
> > a problem with that? I assume we=E2=80=99ll end up in trouble with =
the
> > security folks if we keep the old RFC=E2=80=A6
> >
> >
> >
> > And, considering MD2 is historic, do we even need to mention it in
> > draft-4572-update anymore?
> >
> >
> >
> > Regards,
> >
> >
> >
> > Christer
> >
> >
> >
> >
> >
> > From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of Christer
> > Holmberg
> > Sent: 30 December 2016 11:42
> > To: mmusic@ietf.org
> > Cc: Jonathan Lennox (jonathan@vidyo.com) <jonathan@vidyo.com>; =
Cullen
> > Jennings (fluffy@iii.ca) <fluffy@iii.ca>
> > Subject: [MMUSIC] draft-4572-update: Spec contains references to a
> > number of obsoleted RFCs
> >
> >
> >
> > Hi,
> >
> >
> >
> > The idnits check returns the following for draft-4572-update.
> >
> >
> >
> > ** Obsolete normative reference: RFC 1319 (ref. '3') (Obsoleted by =
RFC
> > 6149)
> >
> >
> >
> >   ** Downref: Normative reference to an Informational RFC: RFC 1321 =
(ref.
> > '4')
> >
> >
> >
> >   ** Obsolete normative reference: RFC 3280 (ref. '8') (Obsoleted by
> > RFC
> > 5280)
> >
> >
> >
> >   ** Obsolete normative reference: RFC 4234 (ref. '11') (Obsoleted =
by
> > RFC
> >
> >      5234)
> >
> >
> >
> >   ** Obsolete normative reference: RFC 4288 (ref. '12') (Obsoleted =
by
> > RFC
> >
> >      6838)
> >
> >
> >
> >   ** Obsolete normative reference: RFC 4346 (ref. '13') (Obsoleted =
by
> > RFC
> >
> >      5246)
> >
> >
> >
> >   -- Obsolete informational reference (is this intentional?): RFC =
2617 (ref.
> >
> >      '15') (Obsoleted by RFC 7235, RFC 7615, RFC 7616, RFC 7617)
> >
> >
> >
> >   -- Obsolete informational reference (is this intentional?): RFC =
3525 (ref.
> >
> >      '20') (Obsoleted by RFC 5125)
> >
> >
> >
> >   -- Obsolete informational reference (is this intentional?): RFC =
3851 (ref.
> >
> >      '22') (Obsoleted by RFC 5751)
> >
> >
> >
> > The reason for this is that we used RFC 4572 as base, and did not
> > change/update the references.
> >
> >
> >
> > I had a look, and I don=E2=80=99t think there should be any issues =
in
> > replacing the current RFCs with the new ones. But, please indicate =
if you see any issues.
> >
> >
> >
> > Regards,
> >
> >
> >
> > Christer
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>=20


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From: Cullen Jennings <fluffy@iii.ca>
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Subject: Re: [MMUSIC] draft-4572-update: Spec contains references to a number of obsoleted RFCs
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> On Jan 3, 2017, at 1:04 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
>=20
> Hi,
>=20
> I am fine to also remove MD5. However, I want to make sure we don't =
bring back Cullen's backward compatibility issues.
>=20
> Regards,

Everything I know that offered MD5 also offered SHA1 so I don't see any =
problem with not support MD5. And if there is a problem, I'm not very =
sympathetic.=20


From nobody Tue Jan  3 09:44:38 2017
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From: Roman Shpount <roman@telurix.com>
Date: Tue, 3 Jan 2017 12:44:32 -0500
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Subject: Re: [MMUSIC] draft-4572-update: Spec contains references to a number of obsoleted RFCs
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I agree, I think MD2 and MD5 should be defined in the grammar but
specification should state that they MUST NOT be used. This way there are
no potential backwards interop problems.

Regards,

_____________
Roman Shpount

On Tue, Jan 3, 2017 at 12:34 PM, Cullen Jennings <fluffy@iii.ca> wrote:

>
> Trivial nit but ...
>
> Actually I don't think you want MD5 treated as a unknown hash, you want
> treated as a known but don't use. If A offers old and bad crypto to B, we
> have B log that such that we can track down and upgrade A. If we got a a
> new unknown cipher call SSHHAA we would not log that as bad crypto becaus=
e
> we would assume it was new and good.
>
>
> > On Jan 2, 2017, at 11:03 AM, Eric Rescorla <ekr@rtfm.com> wrote:
> >
> > On Mon, Jan 2, 2017 at 2:41 AM, Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
> > Hi,
> >
> > >We can remove MD2.  MD5 is dead, SHA-1 is in its death throes, but MD2
> is merely a >(bad) memory.
> >
> > So, my suggestion is to remove all references to MD2 (including the
> ABNF) for now, and we'll then see what the security folks say about MD5 a=
nd
> SHA-1.
> >
> > Given the threat model here, I think we want to tell people to ignore
> MD* (i.e., treat it as an unknown hash) and to accept SHA-1 (though perha=
ps
> only temporarily). Accordingly, I propose removing MD2 and MD5 from this
> grammar, but leave SHA-1.
> >
> > -Ekr
> >
> >
> > Regards,
> >
> > Christer
> >
> >
> > On 30 December 2016 at 22:27, Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
> > > Hi,
> > >
> > >
> > >
> > > Please note the following:
> > >
> > >
> > >
> > > RFC 6149, which obsoletes RFC 1319, makes MD2 historic. Do people hav=
e
> > > a problem with that? I assume we=E2=80=99ll end up in trouble with th=
e
> > > security folks if we keep the old RFC=E2=80=A6
> > >
> > >
> > >
> > > And, considering MD2 is historic, do we even need to mention it in
> > > draft-4572-update anymore?
> > >
> > >
> > >
> > > Regards,
> > >
> > >
> > >
> > > Christer
> > >
> > >
> > >
> > >
> > >
> > > From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of Christer
> > > Holmberg
> > > Sent: 30 December 2016 11:42
> > > To: mmusic@ietf.org
> > > Cc: Jonathan Lennox (jonathan@vidyo.com) <jonathan@vidyo.com>; Cullen
> > > Jennings (fluffy@iii.ca) <fluffy@iii.ca>
> > > Subject: [MMUSIC] draft-4572-update: Spec contains references to a
> > > number of obsoleted RFCs
> > >
> > >
> > >
> > > Hi,
> > >
> > >
> > >
> > > The idnits check returns the following for draft-4572-update.
> > >
> > >
> > >
> > > ** Obsolete normative reference: RFC 1319 (ref. '3') (Obsoleted by RF=
C
> > > 6149)
> > >
> > >
> > >
> > >   ** Downref: Normative reference to an Informational RFC: RFC 1321
> (ref.
> > > '4')
> > >
> > >
> > >
> > >   ** Obsolete normative reference: RFC 3280 (ref. '8') (Obsoleted by
> > > RFC
> > > 5280)
> > >
> > >
> > >
> > >   ** Obsolete normative reference: RFC 4234 (ref. '11') (Obsoleted by
> > > RFC
> > >
> > >      5234)
> > >
> > >
> > >
> > >   ** Obsolete normative reference: RFC 4288 (ref. '12') (Obsoleted by
> > > RFC
> > >
> > >      6838)
> > >
> > >
> > >
> > >   ** Obsolete normative reference: RFC 4346 (ref. '13') (Obsoleted by
> > > RFC
> > >
> > >      5246)
> > >
> > >
> > >
> > >   -- Obsolete informational reference (is this intentional?): RFC 261=
7
> (ref.
> > >
> > >      '15') (Obsoleted by RFC 7235, RFC 7615, RFC 7616, RFC 7617)
> > >
> > >
> > >
> > >   -- Obsolete informational reference (is this intentional?): RFC 352=
5
> (ref.
> > >
> > >      '20') (Obsoleted by RFC 5125)
> > >
> > >
> > >
> > >   -- Obsolete informational reference (is this intentional?): RFC 385=
1
> (ref.
> > >
> > >      '22') (Obsoleted by RFC 5751)
> > >
> > >
> > >
> > > The reason for this is that we used RFC 4572 as base, and did not
> > > change/update the references.
> > >
> > >
> > >
> > > I had a look, and I don=E2=80=99t think there should be any issues in
> > > replacing the current RFCs with the new ones. But, please indicate if
> you see any issues.
> > >
> > >
> > >
> > > Regards,
> > >
> > >
> > >
> > > Christer
> > _______________________________________________
> > 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
>

--001a114a6f028aca4505453438fa
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">I agree, I think MD2 and MD5 should be defined in the gram=
mar but specification should state that they MUST NOT be used. This way the=
re are no potential backwards interop problems.<div><br></div><div>Regards,=
</div></div><div class=3D"gmail_extra"><br clear=3D"all"><div><div class=3D=
"gmail_signature" data-smartmail=3D"gmail_signature">_____________<br>Roman=
 Shpount</div></div>
<br><div class=3D"gmail_quote">On Tue, Jan 3, 2017 at 12:34 PM, Cullen Jenn=
ings <span dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@iii.ca" target=3D"_blan=
k">fluffy@iii.ca</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote"=
 style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><b=
r>
Trivial nit but ...<br>
<br>
Actually I don&#39;t think you want MD5 treated as a unknown hash, you want=
 treated as a known but don&#39;t use. If A offers old and bad crypto to B,=
 we have B log that such that we can track down and upgrade A. If we got a =
a new unknown cipher call SSHHAA we would not log that as bad crypto becaus=
e we would assume it was new and good.<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
&gt; On Jan 2, 2017, at 11:03 AM, Eric Rescorla &lt;<a href=3D"mailto:ekr@r=
tfm.com">ekr@rtfm.com</a>&gt; wrote:<br>
&gt;<br>
&gt; On Mon, Jan 2, 2017 at 2:41 AM, Christer Holmberg &lt;<a href=3D"mailt=
o:christer.holmberg@ericsson.com">christer.holmberg@ericsson.<wbr>com</a>&g=
t; wrote:<br>
&gt; Hi,<br>
&gt;<br>
&gt; &gt;We can remove MD2.=C2=A0 MD5 is dead, SHA-1 is in its death throes=
, but MD2 is merely a &gt;(bad) memory.<br>
&gt;<br>
&gt; So, my suggestion is to remove all references to MD2 (including the AB=
NF) for now, and we&#39;ll then see what the security folks say about MD5 a=
nd SHA-1.<br>
&gt;<br>
&gt; Given the threat model here, I think we want to tell people to ignore =
MD* (i.e., treat it as an unknown hash) and to accept SHA-1 (though perhaps=
 only temporarily). Accordingly, I propose removing MD2 and MD5 from this g=
rammar, but leave SHA-1.<br>
&gt;<br>
&gt; -Ekr<br>
&gt;<br>
&gt;<br>
&gt; Regards,<br>
&gt;<br>
&gt; Christer<br>
&gt;<br>
&gt;<br>
&gt; On 30 December 2016 at 22:27, Christer Holmberg &lt;<a href=3D"mailto:=
christer.holmberg@ericsson.com">christer.holmberg@ericsson.<wbr>com</a>&gt;=
 wrote:<br>
&gt; &gt; Hi,<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; Please note the following:<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; RFC 6149, which obsoletes RFC 1319, makes MD2 historic. Do people=
 have<br>
&gt; &gt; a problem with that? I assume we=E2=80=99ll end up in trouble wit=
h the<br>
&gt; &gt; security folks if we keep the old RFC=E2=80=A6<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; And, considering MD2 is historic, do we even need to mention it i=
n<br>
&gt; &gt; draft-4572-update anymore?<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; Regards,<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; Christer<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; From: mmusic [mailto:<a href=3D"mailto:mmusic-bounces@ietf.org">m=
music-bounces@ietf.<wbr>org</a>] On Behalf Of Christer<br>
&gt; &gt; Holmberg<br>
&gt; &gt; Sent: 30 December 2016 11:42<br>
&gt; &gt; To: <a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
&gt; &gt; Cc: Jonathan Lennox (<a href=3D"mailto:jonathan@vidyo.com">jonath=
an@vidyo.com</a>) &lt;<a href=3D"mailto:jonathan@vidyo.com">jonathan@vidyo.=
com</a>&gt;; Cullen<br>
&gt; &gt; Jennings (<a href=3D"mailto:fluffy@iii.ca">fluffy@iii.ca</a>) &lt=
;<a href=3D"mailto:fluffy@iii.ca">fluffy@iii.ca</a>&gt;<br>
&gt; &gt; Subject: [MMUSIC] draft-4572-update: Spec contains references to =
a<br>
&gt; &gt; number of obsoleted RFCs<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; Hi,<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; The idnits check returns the following for draft-4572-update.<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; ** Obsolete normative reference: RFC 1319 (ref. &#39;3&#39;) (Obs=
oleted by RFC<br>
&gt; &gt; 6149)<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;=C2=A0 =C2=A0** Downref: Normative reference to an Informational R=
FC: RFC 1321 (ref.<br>
&gt; &gt; &#39;4&#39;)<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;=C2=A0 =C2=A0** Obsolete normative reference: RFC 3280 (ref. &#39;=
8&#39;) (Obsoleted by<br>
&gt; &gt; RFC<br>
&gt; &gt; 5280)<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;=C2=A0 =C2=A0** Obsolete normative reference: RFC 4234 (ref. &#39;=
11&#39;) (Obsoleted by<br>
&gt; &gt; RFC<br>
&gt; &gt;<br>
&gt; &gt;=C2=A0 =C2=A0 =C2=A0 5234)<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;=C2=A0 =C2=A0** Obsolete normative reference: RFC 4288 (ref. &#39;=
12&#39;) (Obsoleted by<br>
&gt; &gt; RFC<br>
&gt; &gt;<br>
&gt; &gt;=C2=A0 =C2=A0 =C2=A0 6838)<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;=C2=A0 =C2=A0** Obsolete normative reference: RFC 4346 (ref. &#39;=
13&#39;) (Obsoleted by<br>
&gt; &gt; RFC<br>
&gt; &gt;<br>
&gt; &gt;=C2=A0 =C2=A0 =C2=A0 5246)<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;=C2=A0 =C2=A0-- Obsolete informational reference (is this intentio=
nal?): RFC 2617 (ref.<br>
&gt; &gt;<br>
&gt; &gt;=C2=A0 =C2=A0 =C2=A0 &#39;15&#39;) (Obsoleted by RFC 7235, RFC 761=
5, RFC 7616, RFC 7617)<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;=C2=A0 =C2=A0-- Obsolete informational reference (is this intentio=
nal?): RFC 3525 (ref.<br>
&gt; &gt;<br>
&gt; &gt;=C2=A0 =C2=A0 =C2=A0 &#39;20&#39;) (Obsoleted by RFC 5125)<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;=C2=A0 =C2=A0-- Obsolete informational reference (is this intentio=
nal?): RFC 3851 (ref.<br>
&gt; &gt;<br>
&gt; &gt;=C2=A0 =C2=A0 =C2=A0 &#39;22&#39;) (Obsoleted by RFC 5751)<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; The reason for this is that we used RFC 4572 as base, and did not=
<br>
&gt; &gt; change/update the references.<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; I had a look, and I don=E2=80=99t think there should be any issue=
s in<br>
&gt; &gt; replacing the current RFCs with the new ones. But, please indicat=
e if you see any issues.<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; Regards,<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; Christer<br>
&gt; ______________________________<wbr>_________________<br>
&gt; mmusic mailing list<br>
&gt; <a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" rel=3D"norefe=
rrer" target=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/mmusic</=
a><br>
&gt;<br>
<br>
______________________________<wbr>_________________<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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/mmusic</a><br=
>
</div></div></blockquote></div><br></div>

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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: Roman Shpount <roman@telurix.com>
Thread-Topic: [MMUSIC] [bfcpbis]  m= line protocol in case of ICE
Thread-Index: AQHSZfm46LjmSy7T7UmDYerMTbGdsg==
Date: Tue, 3 Jan 2017 19:43:55 +0000
Message-ID: <633D3491-83B1-421F-B48C-0A61B1314E99@cisco.com>
References: <CAD5OKxuhvCz82+7JK8QrArtrYcjV9+b7vWMpWRnCjNbrL++srA@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B4BE3AE83@ESESSMB209.ericsson.se> <CAD5OKxu15YgYO0xyWMYXv7VTAVVQ71iJhH_txt31BV0CvCSjqg@mail.gmail.com> <F96AC385-2721-4652-98F5-1BF92F06214A@gmail.com> <D0210B5A-138A-4C86-8D14-6E1FEC011E33@cisco.com> <CAD5OKxuzpVRsR0cMeUyhe35sA9W6bL=p1=0RUpTqwpQDyinwDA@mail.gmail.com> <16B5D8FF-F132-4B09-84D6-AE964CA7858D@cisco.com> <CAD5OKxsAHCykObDwZ2_n+XH7brkCz9yLbZFr9-MCQwzkn4uUmg@mail.gmail.com> <64E8A5CF-89ED-4673-AF23-2C960395C3EF@cisco.com>
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Subject: Re: [MMUSIC] [bfcpbis]  m= line protocol in case of ICE
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From: Roman Shpount <roman@telurix.com>
Date: Tue, 3 Jan 2017 15:08:04 -0500
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To: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
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Archived-At: <https://mailarchive.ietf.org/arch/msg/mmusic/3FtFwp0uuZksf0aheBdB8y8ElJw>
Cc: "bfcpbis@ietf.org" <bfcpbis@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>, "mmusic@ietf.org" <mmusic@ietf.org>, "ice@ietf.org" <ice@ietf.org>
Subject: Re: [MMUSIC] [bfcpbis] m= line protocol in case of ICE
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Charles,

I do not think not supporting ICE tcp candidates will quite work since it
will reduce the usability considerably. The simplest way to move forward is
to define a different transport tag for BFCP  with RFC 4571 over TCP not to
confuse it with TCP/BFCP. It can be TCP/UDP/BFCP (I know this looks strange
and I am open to other suggestions, such as calling all packet based
protocols DBFCP).

In general what I am proposing is:

TCP/BFCP -- existing BFCP over TCP
TLS/BFCP -- exisitng BFCP over TLS
UDP/BFCP -- BFCP over UDP or ICE udp candidates
TCP/UDP/BFCP -- BFCP with RFC 4571 framing over TCP or over ICE tcp
candidates
UDP/DTLS/BFCP -- BFCP over DTLS or over ICE udp candidates
TCP/DTLS/BFCP -- BFCP over DTLS with RFC 4571 framing over TCP or over ICE
tcp candidates

Legacy BFCP over TCP or TLS cannot work with ICE or NAT. Other protocols
can work with NAT or ICE using normal ICE procedures.

If we call all packet based protocols DBFCP then transport tags will be:

TCP/BFCP -- existing BFCP over TCP
TLS/BFCP -- exisitng BFCP over TLS
UDP/DBFCP -- DBFCP over UDP or ICE udp candidates
TCP/DBFCP -- DBFCP with RFC 4571 framing over TCP or over ICE tcp candidate=
s
UDP/DTLS/DBFCP -- DBFCP over DTLS or over ICE udp candidates
TCP/DTLS/DBFCP -- DBFCP over DTLS with RFC 4571 framing over TCP or over
ICE tcp candidates

Since BFCP over UDP (or other packet based protocols) is quite different
due to timers and transmission restrictions, it can have a different
transport tag and even be defined in a separate RFC.

Regards,

_____________
Roman Shpount

On Tue, Jan 3, 2017 at 2:43 PM, Charles Eckel (eckelcu) <eckelcu@cisco.com>
wrote:

> Crickets=E2=80=A6
>
> If no one is or has plans for using ICE with TCP/BFCP, perhaps it is best
> to state that as of this rev of the BFCP spec, BFCP with TCP candidates i=
s
> not defined. Future updates to the spec may define this usage.
>
>
>
> Cheers,
>
> Charles
>
>
>
> *From: *mmusic <mmusic-bounces@ietf.org> on behalf of Charles Eckel <
> eckelcu@cisco.com>
> *Date: *Friday, December 2, 2016 at 4:01 PM
> *To: *Roman Shpount <roman@telurix.com>
> *Cc: *"ice@ietf.org" <ice@ietf.org>, "bfcpbis@ietf.org" <bfcpbis@ietf.org=
>,
> Christer Holmberg <christer.holmberg@ericsson.com>, "mmusic@ietf.org" <
> mmusic@ietf.org>
> *Subject: *Re: [MMUSIC] [bfcpbis] m=3D line protocol in case of ICE
>
>
>
> I have no experience with ICE with TCP candidates so hopefully others can
> chime in as to what they think is a workable solution.
>
>
>
> Cheers,
>
> Charles
>
>
>
> *From: *Roman Shpount <roman@telurix.com>
> *Date: *Thursday, December 1, 2016 at 12:34 PM
> *To: *Charles Eckel <eckelcu@cisco.com>
> *Cc: *Alan Ford <alan.ford@gmail.com>, "bfcpbis@ietf.org" <
> bfcpbis@ietf.org>, "ice@ietf.org" <ice@ietf.org>, "mmusic@ietf.org" <
> mmusic@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
> *Subject: *Re: [bfcpbis] [MMUSIC] m=3D line protocol in case of ICE
>
>
>
> Charles,
>
>
>
> RFC 6544 Sending Media (https://tools.ietf.org/html/rfc6544#section-10.1)=
 says
> that "The framing defined in RFC 4571
> <https://tools.ietf.org/html/rfc4571> MUST be used when sending media."
> This means the protocol used is not TCP/BFCP which is using application
> level framing. I believe that STUN/Media demultiplexing requirements woul=
d
> prevent using TCP/BFCP directly with ice tcp candidates without redesign =
of
> either ICE TCP or TCP/BFCP.
>
>
>
> Furthermore there are other implied ICE requirements that I outlined
> before (switching between udp and tpc candidates, existence of SBC which
> terminate ICE only but do not support the embedded protocol) because of
> which ice tcp is considered unreliable transport and will require
> fragmentation support and re-transmit timers that are not part of TCP/BFC=
P.
>
>
>
> Regards,
>
>
> _____________
> Roman Shpount
>
>
>
> On Thu, Dec 1, 2016 at 3:17 PM, Charles Eckel (eckelcu) <eckelcu@cisco.co=
m>
> wrote:
>
> Roman,
>
>
>
> Why would selecting TCP/BFCP as transport violate RFC 6544? Perhaps it
> does, but after a quick scan I am not sure why.
>
>
>
> Cheers,
>
> Charles
>
>
>
> *From: *Roman Shpount <roman@telurix.com>
> *Date: *Tuesday, November 29, 2016 at 10:38 AM
> *To: *Charles Eckel <eckelcu@cisco.com>
> *Cc: *Alan Ford <alan.ford@gmail.com>, "bfcpbis@ietf.org" <
> bfcpbis@ietf.org>, "ice@ietf.org" <ice@ietf.org>, "mmusic@ietf.org" <
> mmusic@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
> *Subject: *Re: [bfcpbis] [MMUSIC] m=3D line protocol in case of ICE
>
>
>
> On Tue, Nov 29, 2016 at 12:48 PM, Charles Eckel (eckelcu) <
> eckelcu@cisco.com> wrote:
>
> It seems to me that the most straightforward approach would be to mandate
> support for BFCP over UDP when using ICE, use UDP as the default candidat=
e,
> and signal the BFCP m-line as if it is BFCP over UDP. If we can mandate t=
he
> use of DTLS, that would be even better.
>
> Thoughts?
>
>
>
>
>
> I agree.
>
>
>
> The only issue that I still have, if DTLS is not used, what protocol is
> used when ICE tcp candidate is selected for transport. Is this TCP/BFCP
> (which goes against RFC6544)  or is it UDP/BFCP with RFC4571 framing? If =
it
> is UDP/BFCP with RFC4571 framing, what transport tag should be used in th=
e
> re-INVITE which is sent after ICE nomination with only selected candidate=
?
> Should it be TCP/UDP/BFCP or something similar?
>
>
>
> Regards,
>
> _____________
> Roman Shpount
>
>
>
>
>
>

--001a1140396ce390540545363931
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Charles,<div><br></div><div>I do not think not supporting =
ICE tcp candidates will quite work since it will reduce the usability consi=
derably. The simplest way to move forward is to define a different transpor=
t tag for BFCP =C2=A0with RFC 4571 over TCP not to confuse it with TCP/BFCP=
. It can be TCP/UDP/BFCP (I know this looks strange and I am open to other =
suggestions, such as calling all packet based protocols DBFCP).=C2=A0</div>=
<div><br></div><div>In general what I am proposing is:</div><div><br></div>=
<div>TCP/BFCP -- existing BFCP over TCP</div><div>TLS/BFCP -- exisitng BFCP=
 over TLS</div><div>UDP/BFCP -- BFCP over UDP or ICE udp candidates</div><d=
iv>TCP/UDP/BFCP -- BFCP with RFC 4571 framing=C2=A0over TCP or over ICE tcp=
 candidates<br></div><div>UDP/DTLS/BFCP -- BFCP over DTLS or over ICE udp c=
andidates</div><div>TCP/DTLS/BFCP -- BFCP over DTLS with RFC 4571 framing=
=C2=A0over TCP or over ICE tcp candidates</div><div><br></div><div>Legacy B=
FCP over TCP or TLS cannot work with ICE or NAT. Other protocols can work w=
ith NAT or ICE using normal ICE procedures.</div><div><br></div><div>If we =
call all packet based protocols DBFCP then transport tags will be:</div><di=
v><br></div><div><div>TCP/BFCP -- existing BFCP over TCP</div><div>TLS/BFCP=
 -- exisitng BFCP over TLS</div><div>UDP/DBFCP -- DBFCP over UDP or ICE udp=
 candidates</div><div>TCP/DBFCP -- DBFCP with RFC 4571 framing=C2=A0over TC=
P or over ICE tcp candidates<br></div><div>UDP/DTLS/DBFCP -- DBFCP over DTL=
S or over ICE udp candidates</div><div>TCP/DTLS/DBFCP -- DBFCP over DTLS wi=
th RFC 4571 framing=C2=A0over TCP or over ICE tcp candidates</div><div><br>=
</div></div><div>Since BFCP over UDP (or other packet based protocols) is q=
uite different due to timers and transmission restrictions, it can have a d=
ifferent transport tag and even be defined in a separate RFC.</div><div><br=
></div><div>Regards,</div></div><div class=3D"gmail_extra"><br clear=3D"all=
"><div><div class=3D"gmail_signature" data-smartmail=3D"gmail_signature">__=
___________<br>Roman Shpount</div></div>
<br><div class=3D"gmail_quote">On Tue, Jan 3, 2017 at 2:43 PM, Charles Ecke=
l (eckelcu) <span dir=3D"ltr">&lt;<a href=3D"mailto:eckelcu@cisco.com" targ=
et=3D"_blank">eckelcu@cisco.com</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex">







<div bgcolor=3D"white" lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div class=3D"m_6178081229878505394WordSection1">
<p class=3D"MsoNormal">Crickets=E2=80=A6<u></u><u></u></p>
<p class=3D"MsoNormal">If no one is or has plans for using ICE with TCP/BFC=
P, perhaps it is best to state that as of this rev of the BFCP spec, BFCP w=
ith TCP candidates is not defined. Future updates to the spec may define th=
is usage.<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">Cheers,<u></u><u></u></p>
<p class=3D"MsoNormal">Charles<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<blockquote style=3D"border:none;border-left:solid #b5c4df 4.5pt;padding:0i=
n 0in 0in 4.0pt;margin-left:3.75pt;margin-right:0in">
<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-family:Calibri;color:black">F=
rom: </span>
</b><span style=3D"font-family:Calibri;color:black">mmusic &lt;<a href=3D"m=
ailto:mmusic-bounces@ietf.org" target=3D"_blank">mmusic-bounces@ietf.org</a=
>&gt; on behalf of Charles Eckel &lt;<a href=3D"mailto:eckelcu@cisco.com" t=
arget=3D"_blank">eckelcu@cisco.com</a>&gt;<br>
<b>Date: </b>Friday, December 2, 2016 at 4:01 PM<br>
<b>To: </b>Roman Shpount &lt;<a href=3D"mailto:roman@telurix.com" target=3D=
"_blank">roman@telurix.com</a>&gt;<br>
<b>Cc: </b>&quot;<a href=3D"mailto:ice@ietf.org" target=3D"_blank">ice@ietf=
.org</a>&quot; &lt;<a href=3D"mailto:ice@ietf.org" target=3D"_blank">ice@ie=
tf.org</a>&gt;, &quot;<a href=3D"mailto:bfcpbis@ietf.org" target=3D"_blank"=
>bfcpbis@ietf.org</a>&quot; &lt;<a href=3D"mailto:bfcpbis@ietf.org" target=
=3D"_blank">bfcpbis@ietf.org</a>&gt;, Christer Holmberg &lt;<a href=3D"mail=
to:christer.holmberg@ericsson.com" target=3D"_blank">christer.holmberg@eric=
sson.<wbr>com</a>&gt;, &quot;<a href=3D"mailto:mmusic@ietf.org" target=3D"_=
blank">mmusic@ietf.org</a>&quot; &lt;<a href=3D"mailto:mmusic@ietf.org" tar=
get=3D"_blank">mmusic@ietf.org</a>&gt;<br>
<b>Subject: </b>Re: [MMUSIC] [bfcpbis] m=3D line protocol in case of ICE<u>=
</u><u></u></span></p>
</div><div><div class=3D"h5">
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<p class=3D"MsoNormal">I have no experience with ICE with TCP candidates so=
 hopefully others can chime in as to what they think is a workable solution=
.
<u></u><u></u></p>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
<p class=3D"MsoNormal">Cheers,<u></u><u></u></p>
<p class=3D"MsoNormal">Charles<u></u><u></u></p>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
<blockquote style=3D"border:none;border-left:solid #b5c4df 4.5pt;padding:0i=
n 0in 0in 4.0pt;margin-left:3.75pt;margin-top:5.0pt;margin-right:0in;margin=
-bottom:5.0pt">
<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-family:Calibri;color:black">F=
rom: </span>
</b><span style=3D"font-family:Calibri;color:black">Roman Shpount &lt;<a hr=
ef=3D"mailto:roman@telurix.com" target=3D"_blank">roman@telurix.com</a>&gt;=
<br>
<b>Date: </b>Thursday, December 1, 2016 at 12:34 PM<br>
<b>To: </b>Charles Eckel &lt;<a href=3D"mailto:eckelcu@cisco.com" target=3D=
"_blank">eckelcu@cisco.com</a>&gt;<br>
<b>Cc: </b>Alan Ford &lt;<a href=3D"mailto:alan.ford@gmail.com" target=3D"_=
blank">alan.ford@gmail.com</a>&gt;, &quot;<a href=3D"mailto:bfcpbis@ietf.or=
g" target=3D"_blank">bfcpbis@ietf.org</a>&quot; &lt;<a href=3D"mailto:bfcpb=
is@ietf.org" target=3D"_blank">bfcpbis@ietf.org</a>&gt;, &quot;<a href=3D"m=
ailto:ice@ietf.org" target=3D"_blank">ice@ietf.org</a>&quot; &lt;<a href=3D=
"mailto:ice@ietf.org" target=3D"_blank">ice@ietf.org</a>&gt;, &quot;<a href=
=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.org</a>&quot; &lt=
;<a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.org</a>&g=
t;, Christer Holmberg &lt;<a href=3D"mailto:christer.holmberg@ericsson.com"=
 target=3D"_blank">christer.holmberg@ericsson.<wbr>com</a>&gt;<br>
<b>Subject: </b>Re: [bfcpbis] [MMUSIC] m=3D line protocol in case of ICE</s=
pan><u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Charles, <u></u><u></u></p>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">RFC 6544 Sending Media (<a href=3D"https://tools.iet=
f.org/html/rfc6544#section-10.1" target=3D"_blank">https://tools.ietf.org/h=
tml/<wbr>rfc6544#section-10.1</a>)=C2=A0says that &quot;<span style=3D"font=
-size:10.0pt;color:black">The framing defined in
</span><a href=3D"https://tools.ietf.org/html/rfc4571" target=3D"_blank"><s=
pan style=3D"font-size:10.0pt">RFC 4571</span></a><span style=3D"font-size:=
10.0pt;color:black"> MUST be used when sending media.&quot; This means the =
protocol used is not TCP/BFCP which is using application level
 framing. I believe that STUN/Media demultiplexing requirements would preve=
nt using TCP/BFCP directly with ice tcp candidates without redesign of eith=
er ICE TCP or TCP/BFCP.</span><u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;color:black">Further=
more=C2=A0there are other implied ICE requirements that I outlined before (=
switching between udp and tpc candidates, existence of SBC which terminate =
ICE only but do not support the embedded
 protocol) because of which ice tcp is considered unreliable transport and =
will require fragmentation support and re-transmit timers that are not part=
 of TCP/BFCP.</span><u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;color:black">Regards=
,</span><u></u><u></u></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><br clear=3D"all">
<u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal">_____________<br>
Roman Shpount<u></u><u></u></p>
</div>
</div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
<div>
<p class=3D"MsoNormal">On Thu, Dec 1, 2016 at 3:17 PM, Charles Eckel (eckel=
cu) &lt;<a href=3D"mailto:eckelcu@cisco.com" target=3D"_blank">eckelcu@cisc=
o.com</a>&gt; wrote:<u></u><u></u></p>
<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">Roman,<u></u><u></u></p>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
<p class=3D"MsoNormal">Why would selecting TCP/BFCP as transport violate RF=
C 6544? Perhaps it does, but after a quick scan I am not sure why.<u></u><u=
></u></p>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
<p class=3D"MsoNormal">Cheers,<u></u><u></u></p>
<p class=3D"MsoNormal">Charles<u></u><u></u></p>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
<blockquote style=3D"border:none;border-left:solid #b5c4df 4.5pt;padding:0i=
n 0in 0in 4.0pt;margin-left:3.75pt;margin-top:5.0pt;margin-right:0in;margin=
-bottom:5.0pt">
<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-family:Calibri;color:black">F=
rom:
</span></b><span style=3D"font-family:Calibri;color:black">Roman Shpount &l=
t;<a href=3D"mailto:roman@telurix.com" target=3D"_blank">roman@telurix.com<=
/a>&gt;<br>
<b>Date: </b>Tuesday, November 29, 2016 at 10:38 AM<br>
<b>To: </b>Charles Eckel &lt;<a href=3D"mailto:eckelcu@cisco.com" target=3D=
"_blank">eckelcu@cisco.com</a>&gt;<br>
<b>Cc: </b>Alan Ford &lt;<a href=3D"mailto:alan.ford@gmail.com" target=3D"_=
blank">alan.ford@gmail.com</a>&gt;, &quot;<a href=3D"mailto:bfcpbis@ietf.or=
g" target=3D"_blank">bfcpbis@ietf.org</a>&quot; &lt;<a href=3D"mailto:bfcpb=
is@ietf.org" target=3D"_blank">bfcpbis@ietf.org</a>&gt;, &quot;<a href=3D"m=
ailto:ice@ietf.org" target=3D"_blank">ice@ietf.org</a>&quot;
 &lt;<a href=3D"mailto:ice@ietf.org" target=3D"_blank">ice@ietf.org</a>&gt;=
, &quot;<a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.or=
g</a>&quot; &lt;<a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic=
@ietf.org</a>&gt;, Christer Holmberg &lt;<a href=3D"mailto:christer.holmber=
g@ericsson.com" target=3D"_blank">christer.holmberg@ericsson.<wbr>com</a>&g=
t;<br>
<b>Subject: </b>Re: [bfcpbis] [MMUSIC] m=3D line protocol in case of ICE</s=
pan><u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<div>
<div>
<div>
<div>
<p class=3D"MsoNormal">On Tue, Nov 29, 2016 at 12:48 PM, Charles Eckel (eck=
elcu) &lt;<a href=3D"mailto:eckelcu@cisco.com" target=3D"_blank">eckelcu@ci=
sco.com</a>&gt; wrote:<u></u><u></u></p>
</div>
</div>
<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">It seems to me that the most straightforward approac=
h would be to mandate support for BFCP over UDP when using ICE, use UDP as =
the default candidate, and signal the BFCP m-line
 as if it is BFCP over UDP. If we can mandate the use of DTLS, that would b=
e even better.<u></u><u></u></p>
<p class=3D"MsoNormal">Thoughts?<u></u><u></u></p>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">I agree.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">The only issue that I still have, if DTLS is not use=
d, what protocol is used when ICE tcp candidate is=C2=A0selected for transp=
ort. Is this TCP/BFCP (which goes against RFC6544) =C2=A0or
 is it UDP/BFCP with RFC4571 framing? If it is UDP/BFCP with RFC4571 framin=
g, what transport tag should be used in the re-INVITE which is sent after I=
CE nomination with only selected candidate? Should it be TCP/UDP/BFCP or so=
mething similar?<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Regards,<u></u><u></u></p>
</div>
<div>
<div>
<p class=3D"MsoNormal">_____________<br>
Roman Shpount<u></u><u></u></p>
</div>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
</div>
</div>
</blockquote>
</div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
</blockquote>
</div></div></blockquote>
</div>
</div>

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

--001a1140396ce390540545363931--


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From: Martin Thomson <martin.thomson@gmail.com>
Date: Wed, 4 Jan 2017 09:06:32 +1100
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To: Roman Shpount <roman@telurix.com>
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Subject: Re: [MMUSIC] draft-4572-update: Spec contains references to a number of obsoleted RFCs
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On 4 January 2017 at 04:44, Roman Shpount <roman@telurix.com> wrote:
> I agree, I think MD2 and MD5 should be defined in the grammar but
> specification should state that they MUST NOT be used. This way there are no
> potential backwards interop problems.

To address Christer's original concern, I would say that you don't
need a reference to the algorithms to achieve that.  An informative
reference is as far as you might go.


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To: Roman Shpount <roman@telurix.com>
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Subject: Re: [MMUSIC] [bfcpbis]   m= line protocol in case of ICE
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	charset=utf-8

Roman,

Do any of those work for having both tcp and udp ICE candidates? I =
assume any with ICE (+ 4571 framing) would be acceptable for that case?

Ideally we need to support the case where an answer=E2=80=99s m-line =
protocol does not match any of the candidates, so that an answer could =
only support TCP (say) when the offerer supported both (and used a UDP =
m-line protocol). If Christer=E2=80=99s proposed change which permitted =
an answer not to match the m-line protocol in the ICE candidates was =
accepted, that would resolve this case. Do you see any problems here?

Regards,
Alan

> On 3 Jan 2017, at 20:08, Roman Shpount <roman@telurix.com> wrote:
>=20
> Charles,
>=20
> I do not think not supporting ICE tcp candidates will quite work since =
it will reduce the usability considerably. The simplest way to move =
forward is to define a different transport tag for BFCP  with RFC 4571 =
over TCP not to confuse it with TCP/BFCP. It can be TCP/UDP/BFCP (I know =
this looks strange and I am open to other suggestions, such as calling =
all packet based protocols DBFCP).=20
>=20
> In general what I am proposing is:
>=20
> TCP/BFCP -- existing BFCP over TCP
> TLS/BFCP -- exisitng BFCP over TLS
> UDP/BFCP -- BFCP over UDP or ICE udp candidates
> TCP/UDP/BFCP -- BFCP with RFC 4571 framing over TCP or over ICE tcp =
candidates
> UDP/DTLS/BFCP -- BFCP over DTLS or over ICE udp candidates
> TCP/DTLS/BFCP -- BFCP over DTLS with RFC 4571 framing over TCP or over =
ICE tcp candidates
>=20
> Legacy BFCP over TCP or TLS cannot work with ICE or NAT. Other =
protocols can work with NAT or ICE using normal ICE procedures.
>=20
> If we call all packet based protocols DBFCP then transport tags will =
be:
>=20
> TCP/BFCP -- existing BFCP over TCP
> TLS/BFCP -- exisitng BFCP over TLS
> UDP/DBFCP -- DBFCP over UDP or ICE udp candidates
> TCP/DBFCP -- DBFCP with RFC 4571 framing over TCP or over ICE tcp =
candidates
> UDP/DTLS/DBFCP -- DBFCP over DTLS or over ICE udp candidates
> TCP/DTLS/DBFCP -- DBFCP over DTLS with RFC 4571 framing over TCP or =
over ICE tcp candidates
>=20
> Since BFCP over UDP (or other packet based protocols) is quite =
different due to timers and transmission restrictions, it can have a =
different transport tag and even be defined in a separate RFC.
>=20
> Regards,
>=20
> _____________
> Roman Shpount
>=20
> On Tue, Jan 3, 2017 at 2:43 PM, Charles Eckel (eckelcu) =
<eckelcu@cisco.com <mailto:eckelcu@cisco.com>> wrote:
> Crickets=E2=80=A6
>=20
> If no one is or has plans for using ICE with TCP/BFCP, perhaps it is =
best to state that as of this rev of the BFCP spec, BFCP with TCP =
candidates is not defined. Future updates to the spec may define this =
usage.
>=20
> =20
>=20
> Cheers,
>=20
> Charles
>=20
> =20
>=20
> From: mmusic <mmusic-bounces@ietf.org =
<mailto:mmusic-bounces@ietf.org>> on behalf of Charles Eckel =
<eckelcu@cisco.com <mailto:eckelcu@cisco.com>>
> Date: Friday, December 2, 2016 at 4:01 PM
> To: Roman Shpount <roman@telurix.com <mailto:roman@telurix.com>>
> Cc: "ice@ietf.org <mailto:ice@ietf.org>" <ice@ietf.org =
<mailto:ice@ietf.org>>, "bfcpbis@ietf.org <mailto:bfcpbis@ietf.org>" =
<bfcpbis@ietf.org <mailto:bfcpbis@ietf.org>>, Christer Holmberg =
<christer.holmberg@ericsson.com =
<mailto:christer.holmberg@ericsson.com>>, "mmusic@ietf.org =
<mailto:mmusic@ietf.org>" <mmusic@ietf.org <mailto:mmusic@ietf.org>>
> Subject: Re: [MMUSIC] [bfcpbis] m=3D line protocol in case of ICE
>=20
> =20
>=20
> I have no experience with ICE with TCP candidates so hopefully others =
can chime in as to what they think is a workable solution.
>=20
> =20
>=20
> Cheers,
>=20
> Charles
>=20
> =20
>=20
> From: Roman Shpount <roman@telurix.com <mailto:roman@telurix.com>>
> Date: Thursday, December 1, 2016 at 12:34 PM
> To: Charles Eckel <eckelcu@cisco.com <mailto:eckelcu@cisco.com>>
> Cc: Alan Ford <alan.ford@gmail.com <mailto:alan.ford@gmail.com>>, =
"bfcpbis@ietf.org <mailto:bfcpbis@ietf.org>" <bfcpbis@ietf.org =
<mailto:bfcpbis@ietf.org>>, "ice@ietf.org <mailto:ice@ietf.org>" =
<ice@ietf.org <mailto:ice@ietf.org>>, "mmusic@ietf.org =
<mailto:mmusic@ietf.org>" <mmusic@ietf.org <mailto:mmusic@ietf.org>>, =
Christer Holmberg <christer.holmberg@ericsson.com =
<mailto:christer.holmberg@ericsson.com>>
> Subject: Re: [bfcpbis] [MMUSIC] m=3D line protocol in case of ICE
>=20
> =20
>=20
> Charles,
>=20
> =20
>=20
> RFC 6544 Sending Media =
(https://tools.ietf.org/html/rfc6544#section-10.1 =
<https://tools.ietf.org/html/rfc6544#section-10.1>) says that "The =
framing defined in RFC 4571 <https://tools.ietf.org/html/rfc4571> MUST =
be used when sending media." This means the protocol used is not =
TCP/BFCP which is using application level framing. I believe that =
STUN/Media demultiplexing requirements would prevent using TCP/BFCP =
directly with ice tcp candidates without redesign of either ICE TCP or =
TCP/BFCP.
>=20
> =20
>=20
> Furthermore there are other implied ICE requirements that I outlined =
before (switching between udp and tpc candidates, existence of SBC which =
terminate ICE only but do not support the embedded protocol) because of =
which ice tcp is considered unreliable transport and will require =
fragmentation support and re-transmit timers that are not part of =
TCP/BFCP.
>=20
> =20
>=20
> Regards,
>=20
>=20
>=20
> _____________
> Roman Shpount
>=20
> =20
>=20
> On Thu, Dec 1, 2016 at 3:17 PM, Charles Eckel (eckelcu) =
<eckelcu@cisco.com <mailto:eckelcu@cisco.com>> wrote:
>=20
> Roman,
>=20
> =20
>=20
> Why would selecting TCP/BFCP as transport violate RFC 6544? Perhaps it =
does, but after a quick scan I am not sure why.
>=20
> =20
>=20
> Cheers,
>=20
> Charles
>=20
> =20
>=20
> From: Roman Shpount <roman@telurix.com <mailto:roman@telurix.com>>
> Date: Tuesday, November 29, 2016 at 10:38 AM
> To: Charles Eckel <eckelcu@cisco.com <mailto:eckelcu@cisco.com>>
> Cc: Alan Ford <alan.ford@gmail.com <mailto:alan.ford@gmail.com>>, =
"bfcpbis@ietf.org <mailto:bfcpbis@ietf.org>" <bfcpbis@ietf.org =
<mailto:bfcpbis@ietf.org>>, "ice@ietf.org <mailto:ice@ietf.org>" =
<ice@ietf.org <mailto:ice@ietf.org>>, "mmusic@ietf.org =
<mailto:mmusic@ietf.org>" <mmusic@ietf.org <mailto:mmusic@ietf.org>>, =
Christer Holmberg <christer.holmberg@ericsson.com =
<mailto:christer.holmberg@ericsson.com>>
> Subject: Re: [bfcpbis] [MMUSIC] m=3D line protocol in case of ICE
>=20
> =20
>=20
> On Tue, Nov 29, 2016 at 12:48 PM, Charles Eckel (eckelcu) =
<eckelcu@cisco.com <mailto:eckelcu@cisco.com>> wrote:
>=20
> It seems to me that the most straightforward approach would be to =
mandate support for BFCP over UDP when using ICE, use UDP as the default =
candidate, and signal the BFCP m-line as if it is BFCP over UDP. If we =
can mandate the use of DTLS, that would be even better.
>=20
> Thoughts?
>=20
> =20
>=20
> =20
>=20
> I agree.
>=20
> =20
>=20
> The only issue that I still have, if DTLS is not used, what protocol =
is used when ICE tcp candidate is selected for transport. Is this =
TCP/BFCP (which goes against RFC6544)  or is it UDP/BFCP with RFC4571 =
framing? If it is UDP/BFCP with RFC4571 framing, what transport tag =
should be used in the re-INVITE which is sent after ICE nomination with =
only selected candidate? Should it be TCP/UDP/BFCP or something similar?
>=20
> =20
>=20
> Regards,
>=20
> _____________
> Roman Shpount
>=20
> =20
>=20
> =20
>=20
>=20
> _______________________________________________
> bfcpbis mailing list
> bfcpbis@ietf.org
> https://www.ietf.org/mailman/listinfo/bfcpbis


--Apple-Mail=_08AEB35B-C7EA-4835-939D-0F9C88F9998B
Content-Transfer-Encoding: quoted-printable
Content-Type: text/html;
	charset=utf-8

<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dutf-8"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" =
class=3D"">Roman,<div class=3D""><br class=3D""></div><div class=3D"">Do =
any of those work for having both tcp and udp ICE candidates? I assume =
any with ICE (+ 4571 framing) would be acceptable for that =
case?</div><div class=3D""><br class=3D""></div><div class=3D"">Ideally =
we need to support the case where an answer=E2=80=99s m-line protocol =
does not match any of the candidates, so that an answer could only =
support TCP (say) when the offerer supported both (and used a UDP m-line =
protocol). If Christer=E2=80=99s proposed change which permitted an =
answer not to match the m-line protocol in the ICE candidates was =
accepted, that would resolve this case. Do you see any problems =
here?</div><div class=3D""><br class=3D""></div><div =
class=3D"">Regards,</div><div class=3D"">Alan</div><div class=3D""><br =
class=3D""><div><blockquote type=3D"cite" class=3D""><div class=3D"">On =
3 Jan 2017, at 20:08, Roman Shpount &lt;<a =
href=3D"mailto:roman@telurix.com" class=3D"">roman@telurix.com</a>&gt; =
wrote:</div><br class=3D"Apple-interchange-newline"><div class=3D""><div =
dir=3D"ltr" class=3D"">Charles,<div class=3D""><br class=3D""></div><div =
class=3D"">I do not think not supporting ICE tcp candidates will quite =
work since it will reduce the usability considerably. The simplest way =
to move forward is to define a different transport tag for BFCP =
&nbsp;with RFC 4571 over TCP not to confuse it with TCP/BFCP. It can be =
TCP/UDP/BFCP (I know this looks strange and I am open to other =
suggestions, such as calling all packet based protocols =
DBFCP).&nbsp;</div><div class=3D""><br class=3D""></div><div class=3D"">In=
 general what I am proposing is:</div><div class=3D""><br =
class=3D""></div><div class=3D"">TCP/BFCP -- existing BFCP over =
TCP</div><div class=3D"">TLS/BFCP -- exisitng BFCP over TLS</div><div =
class=3D"">UDP/BFCP -- BFCP over UDP or ICE udp candidates</div><div =
class=3D"">TCP/UDP/BFCP -- BFCP with RFC 4571 framing&nbsp;over TCP or =
over ICE tcp candidates<br class=3D""></div><div class=3D"">UDP/DTLS/BFCP =
-- BFCP over DTLS or over ICE udp candidates</div><div =
class=3D"">TCP/DTLS/BFCP -- BFCP over DTLS with RFC 4571 =
framing&nbsp;over TCP or over ICE tcp candidates</div><div class=3D""><br =
class=3D""></div><div class=3D"">Legacy BFCP over TCP or TLS cannot work =
with ICE or NAT. Other protocols can work with NAT or ICE using normal =
ICE procedures.</div><div class=3D""><br class=3D""></div><div =
class=3D"">If we call all packet based protocols DBFCP then transport =
tags will be:</div><div class=3D""><br class=3D""></div><div =
class=3D""><div class=3D"">TCP/BFCP -- existing BFCP over TCP</div><div =
class=3D"">TLS/BFCP -- exisitng BFCP over TLS</div><div =
class=3D"">UDP/DBFCP -- DBFCP over UDP or ICE udp candidates</div><div =
class=3D"">TCP/DBFCP -- DBFCP with RFC 4571 framing&nbsp;over TCP or =
over ICE tcp candidates<br class=3D""></div><div class=3D"">UDP/DTLS/DBFCP=
 -- DBFCP over DTLS or over ICE udp candidates</div><div =
class=3D"">TCP/DTLS/DBFCP -- DBFCP over DTLS with RFC 4571 =
framing&nbsp;over TCP or over ICE tcp candidates</div><div class=3D""><br =
class=3D""></div></div><div class=3D"">Since BFCP over UDP (or other =
packet based protocols) is quite different due to timers and =
transmission restrictions, it can have a different transport tag and =
even be defined in a separate RFC.</div><div class=3D""><br =
class=3D""></div><div class=3D"">Regards,</div></div><div =
class=3D"gmail_extra"><br clear=3D"all" class=3D""><div class=3D""><div =
class=3D"gmail_signature" =
data-smartmail=3D"gmail_signature">_____________<br class=3D"">Roman =
Shpount</div></div>
<br class=3D""><div class=3D"gmail_quote">On Tue, Jan 3, 2017 at 2:43 =
PM, Charles Eckel (eckelcu) <span dir=3D"ltr" class=3D"">&lt;<a =
href=3D"mailto:eckelcu@cisco.com" target=3D"_blank" =
class=3D"">eckelcu@cisco.com</a>&gt;</span> wrote:<br =
class=3D""><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 =
.8ex;border-left:1px #ccc solid;padding-left:1ex">







<div bgcolor=3D"white" lang=3D"EN-US" link=3D"blue" vlink=3D"purple" =
class=3D"">
<div class=3D"m_6178081229878505394WordSection1"><p =
class=3D"MsoNormal">Crickets=E2=80=A6<u class=3D""></u><u =
class=3D""></u></p><p class=3D"MsoNormal">If no one is or has plans for =
using ICE with TCP/BFCP, perhaps it is best to state that as of this rev =
of the BFCP spec, BFCP with TCP candidates is not defined. Future =
updates to the spec may define this usage.<u class=3D""></u><u =
class=3D""></u></p><p class=3D"MsoNormal"><u class=3D""></u>&nbsp;<u =
class=3D""></u></p><p class=3D"MsoNormal">Cheers,<u class=3D""></u><u =
class=3D""></u></p><p class=3D"MsoNormal">Charles<u class=3D""></u><u =
class=3D""></u></p><p class=3D"MsoNormal"><u class=3D""></u>&nbsp;<u =
class=3D""></u></p>
<blockquote style=3D"border:none;border-left:solid #b5c4df =
4.5pt;padding:0in 0in 0in 4.0pt;margin-left:3.75pt;margin-right:0in" =
class=3D"">
<div style=3D"border:none;border-top:solid #b5c4df 1.0pt;padding:3.0pt =
0in 0in 0in" class=3D""><p class=3D"MsoNormal"><b class=3D""><span =
style=3D"font-family: Calibri;" class=3D"">From: </span>
</b><span style=3D"font-family: Calibri;" class=3D"">mmusic &lt;<a =
href=3D"mailto:mmusic-bounces@ietf.org" target=3D"_blank" =
class=3D"">mmusic-bounces@ietf.org</a>&gt; on behalf of Charles Eckel =
&lt;<a href=3D"mailto:eckelcu@cisco.com" target=3D"_blank" =
class=3D"">eckelcu@cisco.com</a>&gt;<br class=3D"">
<b class=3D"">Date: </b>Friday, December 2, 2016 at 4:01 PM<br class=3D"">=

<b class=3D"">To: </b>Roman Shpount &lt;<a =
href=3D"mailto:roman@telurix.com" target=3D"_blank" =
class=3D"">roman@telurix.com</a>&gt;<br class=3D"">
<b class=3D"">Cc: </b>"<a href=3D"mailto:ice@ietf.org" target=3D"_blank" =
class=3D"">ice@ietf.org</a>" &lt;<a href=3D"mailto:ice@ietf.org" =
target=3D"_blank" class=3D"">ice@ietf.org</a>&gt;, "<a =
href=3D"mailto:bfcpbis@ietf.org" target=3D"_blank" =
class=3D"">bfcpbis@ietf.org</a>" &lt;<a href=3D"mailto:bfcpbis@ietf.org" =
target=3D"_blank" class=3D"">bfcpbis@ietf.org</a>&gt;, Christer Holmberg =
&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank" =
class=3D"">christer.holmberg@ericsson.<wbr class=3D"">com</a>&gt;, "<a =
href=3D"mailto:mmusic@ietf.org" target=3D"_blank" =
class=3D"">mmusic@ietf.org</a>" &lt;<a href=3D"mailto:mmusic@ietf.org" =
target=3D"_blank" class=3D"">mmusic@ietf.org</a>&gt;<br class=3D"">
<b class=3D"">Subject: </b>Re: [MMUSIC] [bfcpbis] m=3D line protocol in =
case of ICE<u class=3D""></u><u class=3D""></u></span></p>
</div><div class=3D""><div class=3D"h5">
<div class=3D""><p class=3D"MsoNormal"><u class=3D""></u>&nbsp;<u =
class=3D""></u></p>
</div><p class=3D"MsoNormal">I have no experience with ICE with TCP =
candidates so hopefully others can chime in as to what they think is a =
workable solution.
<u class=3D""></u><u class=3D""></u></p><p class=3D"MsoNormal">&nbsp;<u =
class=3D""></u><u class=3D""></u></p><p class=3D"MsoNormal">Cheers,<u =
class=3D""></u><u class=3D""></u></p><p class=3D"MsoNormal">Charles<u =
class=3D""></u><u class=3D""></u></p><p class=3D"MsoNormal">&nbsp;<u =
class=3D""></u><u class=3D""></u></p>
<blockquote style=3D"border:none;border-left:solid #b5c4df =
4.5pt;padding:0in 0in 0in =
4.0pt;margin-left:3.75pt;margin-top:5.0pt;margin-right:0in;margin-bottom:5=
.0pt" class=3D"">
<div style=3D"border:none;border-top:solid #b5c4df 1.0pt;padding:3.0pt =
0in 0in 0in" class=3D""><p class=3D"MsoNormal"><b class=3D""><span =
style=3D"font-family: Calibri;" class=3D"">From: </span>
</b><span style=3D"font-family: Calibri;" class=3D"">Roman Shpount =
&lt;<a href=3D"mailto:roman@telurix.com" target=3D"_blank" =
class=3D"">roman@telurix.com</a>&gt;<br class=3D"">
<b class=3D"">Date: </b>Thursday, December 1, 2016 at 12:34 PM<br =
class=3D"">
<b class=3D"">To: </b>Charles Eckel &lt;<a =
href=3D"mailto:eckelcu@cisco.com" target=3D"_blank" =
class=3D"">eckelcu@cisco.com</a>&gt;<br class=3D"">
<b class=3D"">Cc: </b>Alan Ford &lt;<a href=3D"mailto:alan.ford@gmail.com"=
 target=3D"_blank" class=3D"">alan.ford@gmail.com</a>&gt;, "<a =
href=3D"mailto:bfcpbis@ietf.org" target=3D"_blank" =
class=3D"">bfcpbis@ietf.org</a>" &lt;<a href=3D"mailto:bfcpbis@ietf.org" =
target=3D"_blank" class=3D"">bfcpbis@ietf.org</a>&gt;, "<a =
href=3D"mailto:ice@ietf.org" target=3D"_blank" =
class=3D"">ice@ietf.org</a>" &lt;<a href=3D"mailto:ice@ietf.org" =
target=3D"_blank" class=3D"">ice@ietf.org</a>&gt;, "<a =
href=3D"mailto:mmusic@ietf.org" target=3D"_blank" =
class=3D"">mmusic@ietf.org</a>" &lt;<a href=3D"mailto:mmusic@ietf.org" =
target=3D"_blank" class=3D"">mmusic@ietf.org</a>&gt;, Christer Holmberg =
&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank" =
class=3D"">christer.holmberg@ericsson.<wbr class=3D"">com</a>&gt;<br =
class=3D"">
<b class=3D"">Subject: </b>Re: [bfcpbis] [MMUSIC] m=3D line protocol in =
case of ICE</span><u class=3D""></u><u class=3D""></u></p>
</div>
<div class=3D""><p class=3D"MsoNormal">&nbsp;<u class=3D""></u><u =
class=3D""></u></p>
</div>
<div class=3D""><p class=3D"MsoNormal">Charles, <u class=3D""></u><u =
class=3D""></u></p>
<div class=3D""><p class=3D"MsoNormal">&nbsp;<u class=3D""></u><u =
class=3D""></u></p>
</div>
<div class=3D""><p class=3D"MsoNormal">RFC 6544 Sending Media (<a =
href=3D"https://tools.ietf.org/html/rfc6544#section-10.1" =
target=3D"_blank" class=3D"">https://tools.ietf.org/html/<wbr =
class=3D"">rfc6544#section-10.1</a>)&nbsp;says that "<span =
style=3D"font-size: 10pt;" class=3D"">The framing defined in
</span><a href=3D"https://tools.ietf.org/html/rfc4571" target=3D"_blank" =
class=3D""><span style=3D"font-size:10.0pt" class=3D"">RFC =
4571</span></a><span style=3D"font-size: 10pt;" class=3D""> MUST be used =
when sending media." This means the protocol used is not TCP/BFCP which =
is using application level
 framing. I believe that STUN/Media demultiplexing requirements would =
prevent using TCP/BFCP directly with ice tcp candidates without redesign =
of either ICE TCP or TCP/BFCP.</span><u class=3D""></u><u =
class=3D""></u></p>
</div>
<div class=3D""><p class=3D"MsoNormal">&nbsp;<u class=3D""></u><u =
class=3D""></u></p>
</div>
<div class=3D""><p class=3D"MsoNormal"><span style=3D"font-size: 10pt;" =
class=3D"">Furthermore&nbsp;there are other implied ICE requirements =
that I outlined before (switching between udp and tpc candidates, =
existence of SBC which terminate ICE only but do not support the =
embedded
 protocol) because of which ice tcp is considered unreliable transport =
and will require fragmentation support and re-transmit timers that are =
not part of TCP/BFCP.</span><u class=3D""></u><u class=3D""></u></p>
</div>
<div class=3D""><p class=3D"MsoNormal">&nbsp;<u class=3D""></u><u =
class=3D""></u></p>
</div>
<div class=3D""><p class=3D"MsoNormal"><span style=3D"font-size: 10pt;" =
class=3D"">Regards,</span><u class=3D""></u><u class=3D""></u></p>
</div>
</div>
<div class=3D""><p class=3D"MsoNormal"><br clear=3D"all" class=3D"">
<u class=3D""></u><u class=3D""></u></p>
<div class=3D"">
<div class=3D""><p class=3D"MsoNormal">_____________<br class=3D"">
Roman Shpount<u class=3D""></u><u class=3D""></u></p>
</div>
</div><p class=3D"MsoNormal">&nbsp;<u class=3D""></u><u =
class=3D""></u></p>
<div class=3D""><p class=3D"MsoNormal">On Thu, Dec 1, 2016 at 3:17 PM, =
Charles Eckel (eckelcu) &lt;<a href=3D"mailto:eckelcu@cisco.com" =
target=3D"_blank" class=3D"">eckelcu@cisco.com</a>&gt; wrote:<u =
class=3D""></u><u class=3D""></u></p>
<blockquote style=3D"border:none;border-left:solid #cccccc =
1.0pt;padding:0in 0in 0in =
6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0in;margin-bottom:5.=
0pt" class=3D"">
<div class=3D"">
<div class=3D""><p class=3D"MsoNormal">Roman,<u class=3D""></u><u =
class=3D""></u></p><p class=3D"MsoNormal">&nbsp;<u class=3D""></u><u =
class=3D""></u></p><p class=3D"MsoNormal">Why would selecting TCP/BFCP =
as transport violate RFC 6544? Perhaps it does, but after a quick scan I =
am not sure why.<u class=3D""></u><u class=3D""></u></p><p =
class=3D"MsoNormal">&nbsp;<u class=3D""></u><u class=3D""></u></p><p =
class=3D"MsoNormal">Cheers,<u class=3D""></u><u class=3D""></u></p><p =
class=3D"MsoNormal">Charles<u class=3D""></u><u class=3D""></u></p><p =
class=3D"MsoNormal">&nbsp;<u class=3D""></u><u class=3D""></u></p>
<blockquote style=3D"border:none;border-left:solid #b5c4df =
4.5pt;padding:0in 0in 0in =
4.0pt;margin-left:3.75pt;margin-top:5.0pt;margin-right:0in;margin-bottom:5=
.0pt" class=3D"">
<div style=3D"border:none;border-top:solid #b5c4df 1.0pt;padding:3.0pt =
0in 0in 0in" class=3D""><p class=3D"MsoNormal"><b class=3D""><span =
style=3D"font-family: Calibri;" class=3D"">From:
</span></b><span style=3D"font-family: Calibri;" class=3D"">Roman =
Shpount &lt;<a href=3D"mailto:roman@telurix.com" target=3D"_blank" =
class=3D"">roman@telurix.com</a>&gt;<br class=3D"">
<b class=3D"">Date: </b>Tuesday, November 29, 2016 at 10:38 AM<br =
class=3D"">
<b class=3D"">To: </b>Charles Eckel &lt;<a =
href=3D"mailto:eckelcu@cisco.com" target=3D"_blank" =
class=3D"">eckelcu@cisco.com</a>&gt;<br class=3D"">
<b class=3D"">Cc: </b>Alan Ford &lt;<a href=3D"mailto:alan.ford@gmail.com"=
 target=3D"_blank" class=3D"">alan.ford@gmail.com</a>&gt;, "<a =
href=3D"mailto:bfcpbis@ietf.org" target=3D"_blank" =
class=3D"">bfcpbis@ietf.org</a>" &lt;<a href=3D"mailto:bfcpbis@ietf.org" =
target=3D"_blank" class=3D"">bfcpbis@ietf.org</a>&gt;, "<a =
href=3D"mailto:ice@ietf.org" target=3D"_blank" =
class=3D"">ice@ietf.org</a>"
 &lt;<a href=3D"mailto:ice@ietf.org" target=3D"_blank" =
class=3D"">ice@ietf.org</a>&gt;, "<a href=3D"mailto:mmusic@ietf.org" =
target=3D"_blank" class=3D"">mmusic@ietf.org</a>" &lt;<a =
href=3D"mailto:mmusic@ietf.org" target=3D"_blank" =
class=3D"">mmusic@ietf.org</a>&gt;, Christer Holmberg &lt;<a =
href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank" =
class=3D"">christer.holmberg@ericsson.<wbr class=3D"">com</a>&gt;<br =
class=3D"">
<b class=3D"">Subject: </b>Re: [bfcpbis] [MMUSIC] m=3D line protocol in =
case of ICE</span><u class=3D""></u><u class=3D""></u></p>
</div>
<div class=3D""><p class=3D"MsoNormal">&nbsp;<u class=3D""></u><u =
class=3D""></u></p>
</div>
<div class=3D"">
<div class=3D"">
<div class=3D"">
<div class=3D""><p class=3D"MsoNormal">On Tue, Nov 29, 2016 at 12:48 PM, =
Charles Eckel (eckelcu) &lt;<a href=3D"mailto:eckelcu@cisco.com" =
target=3D"_blank" class=3D"">eckelcu@cisco.com</a>&gt; wrote:<u =
class=3D""></u><u class=3D""></u></p>
</div>
</div>
<div class=3D"">
<div class=3D"">
<div class=3D"">
<blockquote style=3D"border:none;border-left:solid #cccccc =
1.0pt;padding:0in 0in 0in =
6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0in;margin-bottom:5.=
0pt" class=3D"">
<div class=3D"">
<div class=3D""><p class=3D"MsoNormal">It seems to me that the most =
straightforward approach would be to mandate support for BFCP over UDP =
when using ICE, use UDP as the default candidate, and signal the BFCP =
m-line
 as if it is BFCP over UDP. If we can mandate the use of DTLS, that =
would be even better.<u class=3D""></u><u class=3D""></u></p><p =
class=3D"MsoNormal">Thoughts?<u class=3D""></u><u class=3D""></u></p><p =
class=3D"MsoNormal">&nbsp;<u class=3D""></u><u class=3D""></u></p>
</div>
</div>
</blockquote>
<div class=3D""><p class=3D"MsoNormal">&nbsp;<u class=3D""></u><u =
class=3D""></u></p>
</div>
<div class=3D""><p class=3D"MsoNormal">I agree.<u class=3D""></u><u =
class=3D""></u></p>
</div>
<div class=3D""><p class=3D"MsoNormal">&nbsp;<u class=3D""></u><u =
class=3D""></u></p>
</div>
<div class=3D""><p class=3D"MsoNormal">The only issue that I still have, =
if DTLS is not used, what protocol is used when ICE tcp candidate =
is&nbsp;selected for transport. Is this TCP/BFCP (which goes against =
RFC6544) &nbsp;or
 is it UDP/BFCP with RFC4571 framing? If it is UDP/BFCP with RFC4571 =
framing, what transport tag should be used in the re-INVITE which is =
sent after ICE nomination with only selected candidate? Should it be =
TCP/UDP/BFCP or something similar?<u class=3D""></u><u class=3D""></u></p>=

</div>
<div class=3D""><p class=3D"MsoNormal">&nbsp;<u class=3D""></u><u =
class=3D""></u></p>
</div>
<div class=3D""><p class=3D"MsoNormal">Regards,<u class=3D""></u><u =
class=3D""></u></p>
</div>
<div class=3D"">
<div class=3D""><p class=3D"MsoNormal">_____________<br class=3D"">
Roman Shpount<u class=3D""></u><u class=3D""></u></p>
</div>
</div>
<div class=3D""><p class=3D"MsoNormal">&nbsp;<u class=3D""></u><u =
class=3D""></u></p>
</div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
</div>
</div>
</blockquote>
</div><p class=3D"MsoNormal">&nbsp;<u class=3D""></u><u =
class=3D""></u></p>
</div>
</blockquote>
</div></div></blockquote>
</div>
</div>

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

--Apple-Mail=_08AEB35B-C7EA-4835-939D-0F9C88F9998B--


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From: Roman Shpount <roman@telurix.com>
Date: Tue, 3 Jan 2017 18:44:54 -0500
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To: Alan Ford <alan.ford@gmail.com>
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Cc: "ice@ietf.org" <ice@ietf.org>, "mmusic@ietf.org" <mmusic@ietf.org>, "bfcpbis@ietf.org" <bfcpbis@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] [bfcpbis]  m= line protocol in case of ICE
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Alan,

TCP/BFCP and TLS/BFCP will not work with ICE. Or, to be more precise, will
not work without some specification changes either in ICE tcp or in BFCP.

One pair of protocols which will work with ICE is UDP/DTLS/BFCP and
TCP/DTLS/BFCP.
It will support both tcp and udp candidates. I think this pair should be
recommended since it provides encryption.

Another pair that will work with ICE is UDP/BFCP and TCP/UDP/BFCP. It will
also support tcp and udp candidates. It is not something I would use over
public internet since it will transmit data in the clear text.

Finally, I prefer that any end point which implement BFCP and supports ICE
and DTLS, MUST support UDP/DTLS/BFCP and use it for default candidate. Any
endpoint which implements BFCP and ICE, but does not support DTLS, MUST
support UDP/BFCP and use it as a default candidate.

Once the ICE candidate pair is selected, the transport matching the
selected candidate pair should be used in the SDP.

There is no practical need for the asymmetric transport tags. I think the
whole thing is coming from  the confusion that TCP/BFCP can be used with
ICE tcp candidates. It cannot without some specification changes. Also,
there is no reason to use TCP/DTLS/BFCP or TCP/UDP/BFCP, unless ICE is
used. Because of this, there is no reason to use either of those transports
for default candidate, since default candidate is selected for interop with
end points not supporting ICE.

Regards,

_____________
Roman Shpount

On Tue, Jan 3, 2017 at 6:23 PM, Alan Ford <alan.ford@gmail.com> wrote:

> Roman,
>
> Do any of those work for having both tcp and udp ICE candidates? I assume
> any with ICE (+ 4571 framing) would be acceptable for that case?
>
> Ideally we need to support the case where an answer=E2=80=99s m-line prot=
ocol does
> not match any of the candidates, so that an answer could only support TCP
> (say) when the offerer supported both (and used a UDP m-line protocol). I=
f
> Christer=E2=80=99s proposed change which permitted an answer not to match=
 the
> m-line protocol in the ICE candidates was accepted, that would resolve th=
is
> case. Do you see any problems here?
>
> Regards,
> Alan
>
> On 3 Jan 2017, at 20:08, Roman Shpount <roman@telurix.com> wrote:
>
> Charles,
>
> I do not think not supporting ICE tcp candidates will quite work since it
> will reduce the usability considerably. The simplest way to move forward =
is
> to define a different transport tag for BFCP  with RFC 4571 over TCP not =
to
> confuse it with TCP/BFCP. It can be TCP/UDP/BFCP (I know this looks stran=
ge
> and I am open to other suggestions, such as calling all packet based
> protocols DBFCP).
>
> In general what I am proposing is:
>
> TCP/BFCP -- existing BFCP over TCP
> TLS/BFCP -- exisitng BFCP over TLS
> UDP/BFCP -- BFCP over UDP or ICE udp candidates
> TCP/UDP/BFCP -- BFCP with RFC 4571 framing over TCP or over ICE tcp
> candidates
> UDP/DTLS/BFCP -- BFCP over DTLS or over ICE udp candidates
> TCP/DTLS/BFCP -- BFCP over DTLS with RFC 4571 framing over TCP or over IC=
E
> tcp candidates
>
> Legacy BFCP over TCP or TLS cannot work with ICE or NAT. Other protocols
> can work with NAT or ICE using normal ICE procedures.
>
> If we call all packet based protocols DBFCP then transport tags will be:
>
> TCP/BFCP -- existing BFCP over TCP
> TLS/BFCP -- exisitng BFCP over TLS
> UDP/DBFCP -- DBFCP over UDP or ICE udp candidates
> TCP/DBFCP -- DBFCP with RFC 4571 framing over TCP or over ICE tcp
> candidates
> UDP/DTLS/DBFCP -- DBFCP over DTLS or over ICE udp candidates
> TCP/DTLS/DBFCP -- DBFCP over DTLS with RFC 4571 framing over TCP or over
> ICE tcp candidates
>
> Since BFCP over UDP (or other packet based protocols) is quite different
> due to timers and transmission restrictions, it can have a different
> transport tag and even be defined in a separate RFC.
>
> Regards,
>
> _____________
> Roman Shpount
>
> On Tue, Jan 3, 2017 at 2:43 PM, Charles Eckel (eckelcu) <eckelcu@cisco.co=
m
> > wrote:
>
>> Crickets=E2=80=A6
>>
>> If no one is or has plans for using ICE with TCP/BFCP, perhaps it is bes=
t
>> to state that as of this rev of the BFCP spec, BFCP with TCP candidates =
is
>> not defined. Future updates to the spec may define this usage.
>>
>>
>>
>> Cheers,
>>
>> Charles
>>
>>
>>
>> *From: *mmusic <mmusic-bounces@ietf.org> on behalf of Charles Eckel <
>> eckelcu@cisco.com>
>> *Date: *Friday, December 2, 2016 at 4:01 PM
>> *To: *Roman Shpount <roman@telurix.com>
>> *Cc: *"ice@ietf.org" <ice@ietf.org>, "bfcpbis@ietf.org" <bfcpbis@ietf.or=
g>,
>> Christer Holmberg <christer.holmberg@ericsson.com>, "mmusic@ietf.org" <
>> mmusic@ietf.org>
>> *Subject: *Re: [MMUSIC] [bfcpbis] m=3D line protocol in case of ICE
>>
>>
>>
>> I have no experience with ICE with TCP candidates so hopefully others ca=
n
>> chime in as to what they think is a workable solution.
>>
>>
>>
>> Cheers,
>>
>> Charles
>>
>>
>>
>> *From: *Roman Shpount <roman@telurix.com>
>> *Date: *Thursday, December 1, 2016 at 12:34 PM
>> *To: *Charles Eckel <eckelcu@cisco.com>
>> *Cc: *Alan Ford <alan.ford@gmail.com>, "bfcpbis@ietf.org" <
>> bfcpbis@ietf.org>, "ice@ietf.org" <ice@ietf.org>, "mmusic@ietf.org" <
>> mmusic@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
>> *Subject: *Re: [bfcpbis] [MMUSIC] m=3D line protocol in case of ICE
>>
>>
>>
>> Charles,
>>
>>
>>
>> RFC 6544 Sending Media (https://tools.ietf.org/html/rfc6544#section-10.1=
) says
>> that "The framing defined in RFC 4571
>> <https://tools.ietf.org/html/rfc4571> MUST be used when sending media."
>> This means the protocol used is not TCP/BFCP which is using application
>> level framing. I believe that STUN/Media demultiplexing requirements wou=
ld
>> prevent using TCP/BFCP directly with ice tcp candidates without redesign=
 of
>> either ICE TCP or TCP/BFCP.
>>
>>
>>
>> Furthermore there are other implied ICE requirements that I outlined
>> before (switching between udp and tpc candidates, existence of SBC which
>> terminate ICE only but do not support the embedded protocol) because of
>> which ice tcp is considered unreliable transport and will require
>> fragmentation support and re-transmit timers that are not part of TCP/BF=
CP.
>>
>>
>>
>> Regards,
>>
>>
>> _____________
>> Roman Shpount
>>
>>
>>
>> On Thu, Dec 1, 2016 at 3:17 PM, Charles Eckel (eckelcu) <
>> eckelcu@cisco.com> wrote:
>>
>> Roman,
>>
>>
>>
>> Why would selecting TCP/BFCP as transport violate RFC 6544? Perhaps it
>> does, but after a quick scan I am not sure why.
>>
>>
>>
>> Cheers,
>>
>> Charles
>>
>>
>>
>> *From: *Roman Shpount <roman@telurix.com>
>> *Date: *Tuesday, November 29, 2016 at 10:38 AM
>> *To: *Charles Eckel <eckelcu@cisco.com>
>> *Cc: *Alan Ford <alan.ford@gmail.com>, "bfcpbis@ietf.org" <
>> bfcpbis@ietf.org>, "ice@ietf.org" <ice@ietf.org>, "mmusic@ietf.org" <
>> mmusic@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
>> *Subject: *Re: [bfcpbis] [MMUSIC] m=3D line protocol in case of ICE
>>
>>
>>
>> On Tue, Nov 29, 2016 at 12:48 PM, Charles Eckel (eckelcu) <
>> eckelcu@cisco.com> wrote:
>>
>> It seems to me that the most straightforward approach would be to mandat=
e
>> support for BFCP over UDP when using ICE, use UDP as the default candida=
te,
>> and signal the BFCP m-line as if it is BFCP over UDP. If we can mandate =
the
>> use of DTLS, that would be even better.
>>
>> Thoughts?
>>
>>
>>
>>
>>
>> I agree.
>>
>>
>>
>> The only issue that I still have, if DTLS is not used, what protocol is
>> used when ICE tcp candidate is selected for transport. Is this TCP/BFCP
>> (which goes against RFC6544)  or is it UDP/BFCP with RFC4571 framing? If=
 it
>> is UDP/BFCP with RFC4571 framing, what transport tag should be used in t=
he
>> re-INVITE which is sent after ICE nomination with only selected candidat=
e?
>> Should it be TCP/UDP/BFCP or something similar?
>>
>>
>>
>> Regards,
>>
>> _____________
>> Roman Shpount
>>
>>
>>
>>
>>
>>
> _______________________________________________
> bfcpbis mailing list
> bfcpbis@ietf.org
> https://www.ietf.org/mailman/listinfo/bfcpbis
>
>
>

--94eb2c0830de5493180545394135
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Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Alan,<div><br></div><div><span style=3D"color:rgb(0,0,0);f=
ont-size:12.8px">TCP/BFCP and=C2=A0</span><span style=3D"color:rgb(0,0,0);f=
ont-size:12.8px">TLS/BFCP will not work with ICE. Or, to be more precise, w=
ill not work without some specification changes either in ICE tcp or in BFC=
P.</span></div><div><span style=3D"color:rgb(0,0,0);font-size:12.8px"><br><=
/span></div><div><span style=3D"color:rgb(0,0,0);font-size:12.8px">One pair=
 of protocols which will work with ICE is UDP/DTLS/BFCP and=C2=A0</span><fo=
nt color=3D"#000000"><span style=3D"font-size:12.8px">TCP/DTLS/BFCP. It wil=
l support both tcp and udp candidates. I think this pair should be recommen=
ded since it provides encryption.=C2=A0</span></font></div><div><font color=
=3D"#000000"><span style=3D"font-size:12.8px"><br></span></font></div><div>=
<font color=3D"#000000"><span style=3D"font-size:12.8px">Another pair that =
will work with ICE is=C2=A0</span></font><span style=3D"color:rgb(0,0,0);fo=
nt-size:12.8px">UDP/BFCP and=C2=A0</span><span style=3D"color:rgb(0,0,0);fo=
nt-size:12.8px">TCP/UDP/BFCP. It will also support tcp and udp candidates. =
It is not something I would use over public internet since it will transmit=
 data in the clear text.</span><br></div><div><span style=3D"color:rgb(0,0,=
0);font-size:12.8px"><br></span></div><div><span style=3D"color:rgb(0,0,0);=
font-size:12.8px">Finally, I prefer that any end point which implement BFCP=
 and supports ICE and DTLS, MUST support=C2=A0</span><span style=3D"color:r=
gb(0,0,0);font-size:12.8px">UDP/DTLS/BFCP and use it</span><font color=3D"#=
000000"><span style=3D"font-size:12.8px">=C2=A0for default candidate. Any e=
ndpoint which implements=C2=A0BFCP and ICE, but does not support DTLS, MUST=
 support UDP/BFCP and use it as a default candidate.=C2=A0</span></font></d=
iv><div><font color=3D"#000000"><span style=3D"font-size:12.8px"><br></span=
></font></div><div><font color=3D"#000000"><span style=3D"font-size:12.8px"=
>Once the ICE candidate pair is selected, the transport matching the select=
ed candidate pair should be used in the SDP.</span></font></div><div><font =
color=3D"#000000"><span style=3D"font-size:12.8px"><br></span></font></div>=
<div><font color=3D"#000000"><span style=3D"font-size:12.8px">There is no p=
ractical need for the asymmetric transport tags. I think the whole thing is=
 coming from =C2=A0the confusion that TCP/BFCP can be used with ICE tcp can=
didates. It cannot without some specification changes. Also, there is no re=
ason to use=C2=A0</span></font><span style=3D"color:rgb(0,0,0);font-size:12=
.8px">TCP/DTLS/BFCP or=C2=A0</span><span style=3D"color:rgb(0,0,0);font-siz=
e:12.8px">TCP/UDP/BFCP, unless ICE is used. Because of this, there is no re=
ason to use either of those transports for default candidate, since default=
 candidate is selected for interop with end points not supporting ICE.</spa=
n></div><div><font color=3D"#000000"><span style=3D"font-size:12.8px"><br><=
/span></font></div><div><font color=3D"#000000"><span style=3D"font-size:12=
.8px">Regards,</span></font></div></div><div class=3D"gmail_extra"><br clea=
r=3D"all"><div><div class=3D"gmail_signature" data-smartmail=3D"gmail_signa=
ture">_____________<br>Roman Shpount</div></div>
<br><div class=3D"gmail_quote">On Tue, Jan 3, 2017 at 6:23 PM, Alan Ford <s=
pan dir=3D"ltr">&lt;<a href=3D"mailto:alan.ford@gmail.com" target=3D"_blank=
">alan.ford@gmail.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_q=
uote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1e=
x"><div style=3D"word-wrap:break-word">Roman,<div><br></div><div>Do any of =
those work for having both tcp and udp ICE candidates? I assume any with IC=
E (+ 4571 framing) would be acceptable for that case?</div><div><br></div><=
div>Ideally we need to support the case where an answer=E2=80=99s m-line pr=
otocol does not match any of the candidates, so that an answer could only s=
upport TCP (say) when the offerer supported both (and used a UDP m-line pro=
tocol). If Christer=E2=80=99s proposed change which permitted an answer not=
 to match the m-line protocol in the ICE candidates was accepted, that woul=
d resolve this case. Do you see any problems here?</div><div><br></div><div=
>Regards,</div><div>Alan</div><div><br><div><blockquote type=3D"cite"><div>=
<div class=3D"h5"><div>On 3 Jan 2017, at 20:08, Roman Shpount &lt;<a href=
=3D"mailto:roman@telurix.com" target=3D"_blank">roman@telurix.com</a>&gt; w=
rote:</div><br class=3D"m_-4860766982300894496Apple-interchange-newline"></=
div></div><div><div><div class=3D"h5"><div dir=3D"ltr">Charles,<div><br></d=
iv><div>I do not think not supporting ICE tcp candidates will quite work si=
nce it will reduce the usability considerably. The simplest way to move for=
ward is to define a different transport tag for BFCP =C2=A0with RFC 4571 ov=
er TCP not to confuse it with TCP/BFCP. It can be TCP/UDP/BFCP (I know this=
 looks strange and I am open to other suggestions, such as calling all pack=
et based protocols DBFCP).=C2=A0</div><div><br></div><div>In general what I=
 am proposing is:</div><div><br></div><div>TCP/BFCP -- existing BFCP over T=
CP</div><div>TLS/BFCP -- exisitng BFCP over TLS</div><div>UDP/BFCP -- BFCP =
over UDP or ICE udp candidates</div><div>TCP/UDP/BFCP -- BFCP with RFC 4571=
 framing=C2=A0over TCP or over ICE tcp candidates<br></div><div>UDP/DTLS/BF=
CP -- BFCP over DTLS or over ICE udp candidates</div><div>TCP/DTLS/BFCP -- =
BFCP over DTLS with RFC 4571 framing=C2=A0over TCP or over ICE tcp candidat=
es</div><div><br></div><div>Legacy BFCP over TCP or TLS cannot work with IC=
E or NAT. Other protocols can work with NAT or ICE using normal ICE procedu=
res.</div><div><br></div><div>If we call all packet based protocols DBFCP t=
hen transport tags will be:</div><div><br></div><div><div>TCP/BFCP -- exist=
ing BFCP over TCP</div><div>TLS/BFCP -- exisitng BFCP over TLS</div><div>UD=
P/DBFCP -- DBFCP over UDP or ICE udp candidates</div><div>TCP/DBFCP -- DBFC=
P with RFC 4571 framing=C2=A0over TCP or over ICE tcp candidates<br></div><=
div>UDP/DTLS/DBFCP -- DBFCP over DTLS or over ICE udp candidates</div><div>=
TCP/DTLS/DBFCP -- DBFCP over DTLS with RFC 4571 framing=C2=A0over TCP or ov=
er ICE tcp candidates</div><div><br></div></div><div>Since BFCP over UDP (o=
r other packet based protocols) is quite different due to timers and transm=
ission restrictions, it can have a different transport tag and even be defi=
ned in a separate RFC.</div><div><br></div><div>Regards,</div></div><div cl=
ass=3D"gmail_extra"><br clear=3D"all"><div><div class=3D"m_-486076698230089=
4496gmail_signature" data-smartmail=3D"gmail_signature">_____________<br>Ro=
man Shpount</div></div>
<br><div class=3D"gmail_quote">On Tue, Jan 3, 2017 at 2:43 PM, Charles Ecke=
l (eckelcu) <span dir=3D"ltr">&lt;<a href=3D"mailto:eckelcu@cisco.com" targ=
et=3D"_blank">eckelcu@cisco.com</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex">







<div bgcolor=3D"white" lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div class=3D"m_-4860766982300894496m_6178081229878505394WordSection1"><p c=
lass=3D"MsoNormal">Crickets=E2=80=A6<u></u><u></u></p><p class=3D"MsoNormal=
">If no one is or has plans for using ICE with TCP/BFCP, perhaps it is best=
 to state that as of this rev of the BFCP spec, BFCP with TCP candidates is=
 not defined. Future updates to the spec may define this usage.<u></u><u></=
u></p><p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p><p class=3D"MsoNormal"=
>Cheers,<u></u><u></u></p><p class=3D"MsoNormal">Charles<u></u><u></u></p><=
p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<blockquote style=3D"border:none;border-left:solid #b5c4df 4.5pt;padding:0i=
n 0in 0in 4.0pt;margin-left:3.75pt;margin-right:0in">
<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-family:Calibri">From=
: </span>
</b><span style=3D"font-family:Calibri">mmusic &lt;<a href=3D"mailto:mmusic=
-bounces@ietf.org" target=3D"_blank">mmusic-bounces@ietf.org</a>&gt; on beh=
alf of Charles Eckel &lt;<a href=3D"mailto:eckelcu@cisco.com" target=3D"_bl=
ank">eckelcu@cisco.com</a>&gt;<br>
<b>Date: </b>Friday, December 2, 2016 at 4:01 PM<br>
<b>To: </b>Roman Shpount &lt;<a href=3D"mailto:roman@telurix.com" target=3D=
"_blank">roman@telurix.com</a>&gt;<br>
<b>Cc: </b>&quot;<a href=3D"mailto:ice@ietf.org" target=3D"_blank">ice@ietf=
.org</a>&quot; &lt;<a href=3D"mailto:ice@ietf.org" target=3D"_blank">ice@ie=
tf.org</a>&gt;, &quot;<a href=3D"mailto:bfcpbis@ietf.org" target=3D"_blank"=
>bfcpbis@ietf.org</a>&quot; &lt;<a href=3D"mailto:bfcpbis@ietf.org" target=
=3D"_blank">bfcpbis@ietf.org</a>&gt;, Christer Holmberg &lt;<a href=3D"mail=
to:christer.holmberg@ericsson.com" target=3D"_blank">christer.holmberg@eric=
sson.co<wbr>m</a>&gt;, &quot;<a href=3D"mailto:mmusic@ietf.org" target=3D"_=
blank">mmusic@ietf.org</a>&quot; &lt;<a href=3D"mailto:mmusic@ietf.org" tar=
get=3D"_blank">mmusic@ietf.org</a>&gt;<br>
<b>Subject: </b>Re: [MMUSIC] [bfcpbis] m=3D line protocol in case of ICE<u>=
</u><u></u></span></p>
</div><div><div class=3D"m_-4860766982300894496h5">
<div><p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div><p class=3D"MsoNormal">I have no experience with ICE with TCP candida=
tes so hopefully others can chime in as to what they think is a workable so=
lution.
<u></u><u></u></p><p class=3D"MsoNormal">=C2=A0<u></u><u></u></p><p class=
=3D"MsoNormal">Cheers,<u></u><u></u></p><p class=3D"MsoNormal">Charles<u></=
u><u></u></p><p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
<blockquote style=3D"border:none;border-left:solid #b5c4df 4.5pt;padding:0i=
n 0in 0in 4.0pt;margin-left:3.75pt;margin-top:5.0pt;margin-right:0in;margin=
-bottom:5.0pt">
<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-family:Calibri">From=
: </span>
</b><span style=3D"font-family:Calibri">Roman Shpount &lt;<a href=3D"mailto=
:roman@telurix.com" target=3D"_blank">roman@telurix.com</a>&gt;<br>
<b>Date: </b>Thursday, December 1, 2016 at 12:34 PM<br>
<b>To: </b>Charles Eckel &lt;<a href=3D"mailto:eckelcu@cisco.com" target=3D=
"_blank">eckelcu@cisco.com</a>&gt;<br>
<b>Cc: </b>Alan Ford &lt;<a href=3D"mailto:alan.ford@gmail.com" target=3D"_=
blank">alan.ford@gmail.com</a>&gt;, &quot;<a href=3D"mailto:bfcpbis@ietf.or=
g" target=3D"_blank">bfcpbis@ietf.org</a>&quot; &lt;<a href=3D"mailto:bfcpb=
is@ietf.org" target=3D"_blank">bfcpbis@ietf.org</a>&gt;, &quot;<a href=3D"m=
ailto:ice@ietf.org" target=3D"_blank">ice@ietf.org</a>&quot; &lt;<a href=3D=
"mailto:ice@ietf.org" target=3D"_blank">ice@ietf.org</a>&gt;, &quot;<a href=
=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.org</a>&quot; &lt=
;<a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.org</a>&g=
t;, Christer Holmberg &lt;<a href=3D"mailto:christer.holmberg@ericsson.com"=
 target=3D"_blank">christer.holmberg@ericsson.co<wbr>m</a>&gt;<br>
<b>Subject: </b>Re: [bfcpbis] [MMUSIC] m=3D line protocol in case of ICE</s=
pan><u></u><u></u></p>
</div>
<div><p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<div><p class=3D"MsoNormal">Charles, <u></u><u></u></p>
<div><p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<div><p class=3D"MsoNormal">RFC 6544 Sending Media (<a href=3D"https://tool=
s.ietf.org/html/rfc6544#section-10.1" target=3D"_blank">https://tools.ietf.=
org/html/r<wbr>fc6544#section-10.1</a>)=C2=A0says that &quot;<span style=3D=
"font-size:10pt">The framing defined in
</span><a href=3D"https://tools.ietf.org/html/rfc4571" target=3D"_blank"><s=
pan style=3D"font-size:10.0pt">RFC 4571</span></a><span style=3D"font-size:=
10pt"> MUST be used when sending media.&quot; This means the protocol used =
is not TCP/BFCP which is using application level
 framing. I believe that STUN/Media demultiplexing requirements would preve=
nt using TCP/BFCP directly with ice tcp candidates without redesign of eith=
er ICE TCP or TCP/BFCP.</span><u></u><u></u></p>
</div>
<div><p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<div><p class=3D"MsoNormal"><span style=3D"font-size:10pt">Furthermore=C2=
=A0there are other implied ICE requirements that I outlined before (switchi=
ng between udp and tpc candidates, existence of SBC which terminate ICE onl=
y but do not support the embedded
 protocol) because of which ice tcp is considered unreliable transport and =
will require fragmentation support and re-transmit timers that are not part=
 of TCP/BFCP.</span><u></u><u></u></p>
</div>
<div><p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<div><p class=3D"MsoNormal"><span style=3D"font-size:10pt">Regards,</span><=
u></u><u></u></p>
</div>
</div>
<div><p class=3D"MsoNormal"><br clear=3D"all">
<u></u><u></u></p>
<div>
<div><p class=3D"MsoNormal">_____________<br>
Roman Shpount<u></u><u></u></p>
</div>
</div><p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
<div><p class=3D"MsoNormal">On Thu, Dec 1, 2016 at 3:17 PM, Charles Eckel (=
eckelcu) &lt;<a href=3D"mailto:eckelcu@cisco.com" target=3D"_blank">eckelcu=
@cisco.com</a>&gt; wrote:<u></u><u></u></p>
<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">Roman,<u></u><u></u></p><p class=3D"MsoNormal">=
=C2=A0<u></u><u></u></p><p class=3D"MsoNormal">Why would selecting TCP/BFCP=
 as transport violate RFC 6544? Perhaps it does, but after a quick scan I a=
m not sure why.<u></u><u></u></p><p class=3D"MsoNormal">=C2=A0<u></u><u></u=
></p><p class=3D"MsoNormal">Cheers,<u></u><u></u></p><p class=3D"MsoNormal"=
>Charles<u></u><u></u></p><p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
<blockquote style=3D"border:none;border-left:solid #b5c4df 4.5pt;padding:0i=
n 0in 0in 4.0pt;margin-left:3.75pt;margin-top:5.0pt;margin-right:0in;margin=
-bottom:5.0pt">
<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-family:Calibri">From=
:
</span></b><span style=3D"font-family:Calibri">Roman Shpount &lt;<a href=3D=
"mailto:roman@telurix.com" target=3D"_blank">roman@telurix.com</a>&gt;<br>
<b>Date: </b>Tuesday, November 29, 2016 at 10:38 AM<br>
<b>To: </b>Charles Eckel &lt;<a href=3D"mailto:eckelcu@cisco.com" target=3D=
"_blank">eckelcu@cisco.com</a>&gt;<br>
<b>Cc: </b>Alan Ford &lt;<a href=3D"mailto:alan.ford@gmail.com" target=3D"_=
blank">alan.ford@gmail.com</a>&gt;, &quot;<a href=3D"mailto:bfcpbis@ietf.or=
g" target=3D"_blank">bfcpbis@ietf.org</a>&quot; &lt;<a href=3D"mailto:bfcpb=
is@ietf.org" target=3D"_blank">bfcpbis@ietf.org</a>&gt;, &quot;<a href=3D"m=
ailto:ice@ietf.org" target=3D"_blank">ice@ietf.org</a>&quot;
 &lt;<a href=3D"mailto:ice@ietf.org" target=3D"_blank">ice@ietf.org</a>&gt;=
, &quot;<a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.or=
g</a>&quot; &lt;<a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic=
@ietf.org</a>&gt;, Christer Holmberg &lt;<a href=3D"mailto:christer.holmber=
g@ericsson.com" target=3D"_blank">christer.holmberg@ericsson.co<wbr>m</a>&g=
t;<br>
<b>Subject: </b>Re: [bfcpbis] [MMUSIC] m=3D line protocol in case of ICE</s=
pan><u></u><u></u></p>
</div>
<div><p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<div>
<div>
<div>
<div><p class=3D"MsoNormal">On Tue, Nov 29, 2016 at 12:48 PM, Charles Eckel=
 (eckelcu) &lt;<a href=3D"mailto:eckelcu@cisco.com" target=3D"_blank">eckel=
cu@cisco.com</a>&gt; wrote:<u></u><u></u></p>
</div>
</div>
<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">It seems to me that the most straightforward ap=
proach would be to mandate support for BFCP over UDP when using ICE, use UD=
P as the default candidate, and signal the BFCP m-line
 as if it is BFCP over UDP. If we can mandate the use of DTLS, that would b=
e even better.<u></u><u></u></p><p class=3D"MsoNormal">Thoughts?<u></u><u><=
/u></p><p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
</div>
</blockquote>
<div><p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<div><p class=3D"MsoNormal">I agree.<u></u><u></u></p>
</div>
<div><p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<div><p class=3D"MsoNormal">The only issue that I still have, if DTLS is no=
t used, what protocol is used when ICE tcp candidate is=C2=A0selected for t=
ransport. Is this TCP/BFCP (which goes against RFC6544) =C2=A0or
 is it UDP/BFCP with RFC4571 framing? If it is UDP/BFCP with RFC4571 framin=
g, what transport tag should be used in the re-INVITE which is sent after I=
CE nomination with only selected candidate? Should it be TCP/UDP/BFCP or so=
mething similar?<u></u><u></u></p>
</div>
<div><p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<div><p class=3D"MsoNormal">Regards,<u></u><u></u></p>
</div>
<div>
<div><p class=3D"MsoNormal">_____________<br>
Roman Shpount<u></u><u></u></p>
</div>
</div>
<div><p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
</div>
</div>
</blockquote>
</div><p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
</blockquote>
</div></div></blockquote>
</div>
</div>

</blockquote></div><br></div></div></div>
______________________________<wbr>_________________<br>bfcpbis mailing lis=
t<br><a href=3D"mailto:bfcpbis@ietf.org" target=3D"_blank">bfcpbis@ietf.org=
</a><br><a href=3D"https://www.ietf.org/mailman/listinfo/bfcpbis" target=3D=
"_blank">https://www.ietf.org/mailman/<wbr>listinfo/bfcpbis</a><br></div></=
blockquote></div><br></div></div></blockquote></div><br></div>

--94eb2c0830de5493180545394135--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Martin Thomson <martin.thomson@gmail.com>, Roman Shpount <roman@telurix.com>
Thread-Topic: [MMUSIC] draft-4572-update: Spec contains references to a number of obsoleted RFCs
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Cc: Jonathan Lennox <jonathan@vidyo.com>, "mmusic@ietf.org" <mmusic@ietf.org>, Cullen Jennings <fluffy@iii.ca>
Subject: Re: [MMUSIC] draft-4572-update: Spec contains references to a number of obsoleted RFCs
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From: Roman Shpount <roman@telurix.com>
Date: Wed, 4 Jan 2017 09:31:19 -0500
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [MMUSIC] draft-4572-update: Spec contains references to a number of obsoleted RFCs
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This looks good to me

_____________
Roman Shpount

On Wed, Jan 4, 2017 at 6:22 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Ok, to make sure we are all on the same page, below are the places in the
> previous version (-08) of the draft where MD2 and MD5 are mentioned, and my
> suggestion on what/if to do:
>
>
> Section 5:
> -------------
>
> "hash-func              =  "sha-1" / "sha-224" / "sha-256" /
>                              "sha-384" / "sha-512" /
>                              "md5" / "md2" / token
>                              ; Additional hash functions can only come
>                              ; from updates to RFC 3279"
>
> Christer's suggestion: Keep the text as it is.
>
> -------------
>
>    "Following RFC 3279 [7] as updated by RFC 4055 [9], therefore, the
>    defined hash functions are 'SHA-1' [1] [18], 'SHA-224' [1], 'SHA-256'
>    [1], 'SHA-384'[1], 'SHA-512' [1], 'MD5' [4], and 'MD2' [3], with
>    'SHA-256' preferred.  A new IANA registry of Hash Function Textual
>    Names, specified in Section 8, allows for addition of future tokens,
>    but they may only be added if they are included in RFCs that update
>    or obsolete RFC 3279 [7]."
>
> Christer's suggestion: Keep the text, but update the MD2 and MD5
> references ([3] and [4]), and add the following new paragraph text:
>
> "For backward compatibility with implementations compliant with RFC 4572,
> the MD2 and MD5 cipher suite are still listed in the syntax. However,
> implementations compliant to this specification MUST NOT use them."
>
>
> Section 8:
> -------------
>
> "Table 1 contains the initial values of this registry.
>
>         +--------------------+------------------------+-----------+
>         | Hash Function Name |          OID           | Reference |
>         +--------------------+------------------------+-----------+
>         |       "md2"        |   1.2.840.113549.2.2   |  RFC 3279 |
>         |       "md5"        |   1.2.840.113549.2.5   |  RFC 3279 |
>         |      "sha-1"       |     1.3.14.3.2.26      |  RFC 3279 |
>         |     "sha-224"      | 2.16.840.1.101.3.4.2.4 |  RFC 4055 |
>         |     "sha-256"      | 2.16.840.1.101.3.4.2.1 |  RFC 4055 |
>         |     "sha-384"      | 2.16.840.1.101.3.4.2.2 |  RFC 4055 |
>         |     "sha-512"      | 2.16.840.1.101.3.4.2.3 |  RFC 4055 |
>         +--------------------+------------------------+-----------+"
>
> Christer's suggestion: Keep the above text, without changes. This is the
> initial IANA registration, and we don't change that.
>
>
> Regards,
>
> Christer
>
>
>
>
> -----Original Message-----
> From: Martin Thomson [mailto:martin.thomson@gmail.com]
> Sent: 04 January 2017 00:07
> To: Roman Shpount <roman@telurix.com>
> Cc: Cullen Jennings <fluffy@iii.ca>; Jonathan Lennox <jonathan@vidyo.com>;
> mmusic@ietf.org; Christer Holmberg <christer.holmberg@ericsson.com>
> Subject: Re: [MMUSIC] draft-4572-update: Spec contains references to a
> number of obsoleted RFCs
>
> On 4 January 2017 at 04:44, Roman Shpount <roman@telurix.com> wrote:
> > I agree, I think MD2 and MD5 should be defined in the grammar but
> > specification should state that they MUST NOT be used. This way there
> > are no potential backwards interop problems.
>
> To address Christer's original concern, I would say that you don't need a
> reference to the algorithms to achieve that.  An informative reference is
> as far as you might go.
>

--94eb2c06e7945fd2ba054545a3ee
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr">This looks good to me</div><div class=3D"gmail_extra"><br =
clear=3D"all"><div><div class=3D"gmail_signature" data-smartmail=3D"gmail_s=
ignature">_____________<br>Roman Shpount</div></div>
<br><div class=3D"gmail_quote">On Wed, Jan 4, 2017 at 6:22 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:=
1px #ccc solid;padding-left:1ex">Ok, to make sure we are all on the same pa=
ge, below are the places in the previous version (-08) of the draft where M=
D2 and MD5 are mentioned, and my suggestion on what/if to do:<br>
<br>
<br>
Section 5:<br>
-------------<br>
<br>
&quot;hash-func=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =3D=C2=A0 &=
quot;sha-1&quot; / &quot;sha-224&quot; / &quot;sha-256&quot; /<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0&quot;sha-384&quot; / &quot;sha-512&quot; /<=
br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0&quot;md5&quot; / &quot;md2&quot; / token<br=
>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0; Additional hash functions can only come<br=
>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0; from updates to RFC 3279&quot;<br>
<br>
Christer&#39;s suggestion: Keep the text as it is.<br>
<br>
-------------<br>
<br>
=C2=A0 =C2=A0&quot;Following RFC 3279 [7] as updated by RFC 4055 [9], there=
fore, the<br>
=C2=A0 =C2=A0defined hash functions are &#39;SHA-1&#39; [1] [18], &#39;SHA-=
224&#39; [1], &#39;SHA-256&#39;<br>
=C2=A0 =C2=A0[1], &#39;SHA-384&#39;[1], &#39;SHA-512&#39; [1], &#39;MD5&#39=
; [4], and &#39;MD2&#39; [3], with<br>
=C2=A0 =C2=A0&#39;SHA-256&#39; preferred.=C2=A0 A new IANA registry of Hash=
 Function Textual<br>
=C2=A0 =C2=A0Names, specified in Section 8, allows for addition of future t=
okens,<br>
=C2=A0 =C2=A0but they may only be added if they are included in RFCs that u=
pdate<br>
=C2=A0 =C2=A0or obsolete RFC 3279 [7].&quot;<br>
<br>
Christer&#39;s suggestion: Keep the text, but update the MD2 and MD5 refere=
nces ([3] and [4]), and add the following new paragraph text:<br>
<br>
&quot;For backward compatibility with implementations compliant with RFC 45=
72, the MD2 and MD5 cipher suite are still listed in the syntax. However, i=
mplementations compliant to this specification MUST NOT use them.&quot;<br>
<br>
<br>
Section 8:<br>
-------------<br>
<br>
&quot;Table 1 contains the initial values of this registry.<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 +--------------------+--------<wbr>------------=
----+-----------+<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 | Hash Function Name |=C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 OID=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0| Reference |<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 +--------------------+--------<wbr>------------=
----+-----------+<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 |=C2=A0 =C2=A0 =C2=A0 =C2=A0&quot;md2&quot;=C2=
=A0 =C2=A0 =C2=A0 =C2=A0 |=C2=A0 =C2=A01.2.840.113549.2.2=C2=A0 =C2=A0|=C2=
=A0 RFC 3279 |<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 |=C2=A0 =C2=A0 =C2=A0 =C2=A0&quot;md5&quot;=C2=
=A0 =C2=A0 =C2=A0 =C2=A0 |=C2=A0 =C2=A01.2.840.113549.2.5=C2=A0 =C2=A0|=C2=
=A0 RFC 3279 |<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 |=C2=A0 =C2=A0 =C2=A0 &quot;sha-1&quot;=C2=A0 =
=C2=A0 =C2=A0 =C2=A0|=C2=A0 =C2=A0 =C2=A01.3.14.3.2.26=C2=A0 =C2=A0 =C2=A0 =
|=C2=A0 RFC 3279 |<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 |=C2=A0 =C2=A0 =C2=A0&quot;sha-224&quot;=C2=A0 =
=C2=A0 =C2=A0 | 2.16.840.1.101.3.4.2.4 |=C2=A0 RFC 4055 |<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 |=C2=A0 =C2=A0 =C2=A0&quot;sha-256&quot;=C2=A0 =
=C2=A0 =C2=A0 | 2.16.840.1.101.3.4.2.1 |=C2=A0 RFC 4055 |<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 |=C2=A0 =C2=A0 =C2=A0&quot;sha-384&quot;=C2=A0 =
=C2=A0 =C2=A0 | 2.16.840.1.101.3.4.2.2 |=C2=A0 RFC 4055 |<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 |=C2=A0 =C2=A0 =C2=A0&quot;sha-512&quot;=C2=A0 =
=C2=A0 =C2=A0 | 2.16.840.1.101.3.4.2.3 |=C2=A0 RFC 4055 |<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 +--------------------+--------<wbr>------------=
----+-----------+&quot;<br>
<br>
Christer&#39;s suggestion: Keep the above text, without changes. This is th=
e initial IANA registration, and we don&#39;t change that.<br>
<br>
<br>
Regards,<br>
<br>
Christer<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
<br>
<br>
-----Original Message-----<br>
From: Martin Thomson [mailto:<a href=3D"mailto:martin.thomson@gmail.com">ma=
rtin.thomson@gmail.<wbr>com</a>]<br>
Sent: 04 January 2017 00:07<br>
To: Roman Shpount &lt;<a href=3D"mailto:roman@telurix.com">roman@telurix.co=
m</a>&gt;<br>
Cc: Cullen Jennings &lt;<a href=3D"mailto:fluffy@iii.ca">fluffy@iii.ca</a>&=
gt;; Jonathan Lennox &lt;<a href=3D"mailto:jonathan@vidyo.com">jonathan@vid=
yo.com</a>&gt;; <a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a>; Chr=
ister Holmberg &lt;<a href=3D"mailto:christer.holmberg@ericsson.com">christ=
er.holmberg@ericsson.<wbr>com</a>&gt;<br>
Subject: Re: [MMUSIC] draft-4572-update: Spec contains references to a numb=
er of obsoleted RFCs<br>
<br>
On 4 January 2017 at 04:44, Roman Shpount &lt;<a href=3D"mailto:roman@telur=
ix.com">roman@telurix.com</a>&gt; wrote:<br>
&gt; I agree, I think MD2 and MD5 should be defined in the grammar but<br>
&gt; specification should state that they MUST NOT be used. This way there<=
br>
&gt; are no potential backwards interop problems.<br>
<br>
To address Christer&#39;s original concern, I would say that you don&#39;t =
need a reference to the algorithms to achieve that.=C2=A0 An informative re=
ference is as far as you might go.<br>
</div></div></blockquote></div><br></div>

--94eb2c06e7945fd2ba054545a3ee--


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From: Bernard Aboba <bernard.aboba@gmail.com>
Date: Wed, 4 Jan 2017 10:59:03 -0800
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To: Eric Rescorla <ekr@rtfm.com>
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Cc: "mmusic \(E-mail\)" <mmusic@ietf.org>, Robin Raymond <robin@opticaltone.com>, draft-ietf-rtcweb-jsep@tools.ietf.org, RTCWeb IETF <rtcweb@ietf.org>, Cullen Jennings <fluffy@iii.ca>, Colin Perkins <csp@csperkins.org>, draft-ietf-mmusic-sdp-bundle-negotiation@ietf.org
Subject: Re: [MMUSIC] [rtcweb] RTP demux in JSEP and BUNDLE
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I would like to emphasize the importance of standardizing "routing rule"
behavior at an appropriate level of detail.

Developers are encountering interoperability problems and are filing bugs
against implementations.

To address the issues we are seeing, it is necessary to get to the level of
detail in the JSEP-17 Section 6 text.

On Fri, Dec 9, 2016 at 10:01 AM, Bernard Aboba <bernard.aboba@gmail.com>
wrote:

> I agree with Eric and Cullen that a more explicit algorithm is needed.
> The proposed text is not specific enough to resolve observed implementati=
on
> differences.
>
> Cullen said:
>
> " I'd also like it be clear that MID takes priority over PT."
>
> [BA] Yes.  This is something that multiple implementations appear to agre=
e
> on.  Also, Peter's suggestion that a MID mismatch result in a packet drop
> should be restored.
>
> "I believe we agreed that if the PT was unique, then it would be used for
> demux as well."
>
> [BA] That is my understanding as well.
>
> However, I'd also like to understand what happens if the PT is not
> unique.  The proposed text does not provide enough detail and existing
> implementations differ in how they treat non-unique PTs. I am familiar wi=
th
> an implementation that declares SDP with non-unique PTs to be in error.
> Another implementation fills a Payload Type table with a single value, wi=
th
> the value selected depending on whether SSRCs are also specified (e.g.
> specification of SSRCs is taken as an indication that only SSRC matching =
is
> desired, and therefore that a Payload Type entry is not needed).  The ORT=
C
> API spec says to latch the SSRC on a Payload Type match and then remove t=
he
> Payload Type table entry, but some implementations have found this leads =
to
> problems so they have foresaken either the latching or the PT removal (or
> both).
>
> Cullen said:
>
> "I prefer the much more explicit algorithm particularly for the RTCP
> handling as implementors have a hard time figuring out wha they need to d=
o
> to process the RTCP."
>
> [BA] I would also prefer that we have a more explicit algorithm here,
> because we have seen very substantial implementation differences.  One
> implementation I'm familiar with sends the RTCP packets to all RtpSender
> and RtpReceiver objects.  While this is inefficient, it does avoid sendin=
g
> RTCP packets to the wrong objects.  Another implementation sorts the RTCP
> reports as indicated in the proposed text - but has found that the requir=
ed
> handling is so message-dependent that it leads to bugs (e.g. FIR and APP
> messages have caused issues in particular).  So this approach is more
> efficient, unless we are willing to get into detail on every RTCP message=
,
> it will fall short.
>
>
> On Fri, Dec 9, 2016 at 9:13 AM, Eric Rescorla <ekr@rtfm.com> wrote:
>
>> I agree with Cullen that the algorithm structure used in current JSEP is
>> the better structure.
>>
>> Magnus, Colin, do you think you could rewrite your text in that structur=
e?
>>
>> -Ekr
>>
>>
>> On Fri, Dec 9, 2016 at 7:10 AM, Cullen Jennings <fluffy@iii.ca> wrote:
>>
>>>
>>> In general, I think it makes sense to put most the details in bundle an=
d
>>> overview in jsep as you have proposed here.
>>>
>>> I believe we agreed that if the PT was unique, then that would be used
>>> for the demux as well. I'd like to see that explicitly spelled out in t=
his
>>> text instead of just left as an option allowed but not really specified=
 by
>>> this text.  I'd also like it be clear that MID takes priority over PT.
>>>
>>> I prefer the much more explicit algorithm particularly for the RTCP
>>> handling as implementors have a hard time figuring out wha they need to=
 do
>>> to process the RTCP.
>>>
>>>
>>>
>>> > On Dec 8, 2016, at 3:54 AM, Colin Perkins <csp@csperkins.org> wrote:
>>> >
>>> > Hi,
>>> >
>>> > [cc=E2=80=99ing RTCWEB and MMUSIC, since this relates to both JSEP an=
d BUNDLE
>>> drafts]
>>> >
>>> > There=E2=80=99s been a lot of discussion on the lists, and at the mee=
ting in
>>> Seoul, around how RTP streams are mapped onto higher-level, application
>>> meaningful, semantic roles. In particular, around how RTP streams map o=
nto
>>> JSEP objects for WebRTC. Magnus Westerlund and I would like to propose =
the
>>> following updates to JSEP and BUNDLE to try to clarify the behaviour.
>>> >
>>> > Comments and feedback very welcome.
>>> >
>>> > Cheers,
>>> > Colin
>>> >
>>> >
>>> >
>>> > # Replacement for Section 6 of draft-ietf-rtcweb-jsep-17
>>> >
>>> > <section title=3D"Processing RTP/RTCP" anchor=3D"sec.rtp.demux">
>>> >    <t>As described in <xref target=3D"RFC3550"/>, RTP packets are
>>> >    associated with RTP streams <xref target=3D"RFC7656"/>. Metadata
>>> >    about those streams, including source description information
>>> >    and reception quality feedback is conveyed in RTCP packets.
>>> >    Each RTP stream is identified by an SSRC, and each RTP packet
>>> >    carries an SSRC value that is used to associate the packet with
>>> >    the correct RTP stream. RTCP packets also use SSRCs to identify
>>> >    the RTP streams that the reports and metadata relate to.  RTCP
>>> >    packets generally carry multiple SSRC values and report on, or
>>> >    deliver source description information relating to, several RTP
>>> >    streams.</t>
>>> >
>>> >    <t>Each incoming RTP stream, identified by its SSRC, is mapped to
>>> >    an m=3D section in the SDP. The SDP m=3D sections then correspond =
to
>>> >    RtpReceiver objects. This allows each RTP stream to be associated
>>> >    with an RtpTransceiver. Further processing of the RTP stream can
>>> >    then be done at the RtpTransceiver level.  This includes using
>>> >    RID <xref target=3D"I-D.ietf-mmusic-rid"/> to distinguish between
>>> >    multiple Encoded Streams, as well as determine which Source RTP
>>> >    stream should be repaired by a given Redundancy RTP stream.</t>
>>> >
>>> >    <t>The process of mapping RTP streams onto m=3D sections depends o=
n
>>> >    whether streams are bundled or not. If the SDP BUNDLE extension
>>> >    is in use, then RTP streams are mapped onto m=3D sections based on
>>> >    the MID values as described in
>>> >    <xref target=3D"I-D.ietf-mmusic-sdp-bundle-negotiation"/>.  If the
>>> >    SDP BUNDLE extension is not in use, each m=3D section corresponds
>>> >    to a transport layer connection and the RTP streams received on
>>> >    that connection correspond to the m=3D section.</t>
>>> >
>>> >    <t>Incoming RTCP packets contain metadata including reception
>>> >    quality feedback, source description information, and other
>>> >    signalling relating to RTP streams. The RTCP packets are parsed,
>>> >    the associated RTP streams are identified based on the included
>>> >    SSRC values, and the metadata relating to those RTP streams is
>>> >    updated (this might include updating the MID information, used
>>> >    to associate RTP streams with m=3D sections, if the SDP BUNDLE
>>> >    extension is in use). This updated metadata is available to the
>>> >    RtpTransceiver objects associated with those RTP streams.
>>> >    </t>
>>> > </section>
>>> >
>>> >
>>> >
>>> > # Replacement text for section 10.2 of draft-ietf-mmusic-sdp-bundle-n
>>> egotiation-36
>>> >
>>> >     <section anchor=3D"sec-rtp-pt"
>>> >              title=3D"Associating RTP Streams With Correct SDP Media
>>> Description"
>>> >              toc=3D"default">
>>> >       <t>As described in <xref format=3D"default" pageno=3D"false"
>>> >       target=3D"RFC3550"/>, RTP packets are associated with RTP strea=
ms
>>> <xref
>>> >       format=3D"default" pageno=3D"false" target=3D"RFC7656"/>. Each =
RTP
>>> stream is
>>> >       identified by an SSRC value, and each RTP packet carries an SSR=
C
>>> value
>>> >       that is used to associate the packet with the correct RTP
>>> stream. RTCP
>>> >       packets also uses SSRCs to identify on which RTP streams any
>>> report or
>>> >       feedback relate to. Thus, an RTCP packet will commonly carry
>>> multiple
>>> >       SSRC values, and might therefore be providing feedback or repor=
t
>>> on
>>> >       multiple RTP streams. </t>
>>> >
>>> >       <t>In order to be able to process received RTP packets correctl=
y
>>> it
>>> >       must be possible to associate an RTP stream with the correct "m=
=3D"
>>> >       line, as the "m=3D" line and SDP attributes associated with the
>>> "m=3D"
>>> >       line contain information needed to process the packets.</t>
>>> >
>>> >       <t>As all RTP streams associated with a BUNDLE group are part o=
f
>>> the
>>> >       same RTP session and using the same address:port combination fo=
r
>>> >       sending and receiving RTP/RTCP packets, the local address:port
>>> >       combination cannot be used to associate an RTP stream with the
>>> correct
>>> >       "m=3D" line. In addition, multiple RTP streams might be associa=
ted
>>> with
>>> >       the same "m=3D" line.</t>
>>> >
>>> >       <t>Also, as described in <xref format=3D"default" pageno=3D"fal=
se"
>>> >       target=3D"sec-rtp-sessions-pt"/>, the same payload type value
>>> might be
>>> >       used by multiple RTP streams, in which case the payload type
>>> value
>>> >       cannot be used to associate an RTP stream with the correct "m=
=3D"
>>> line.
>>> >       However, there are cases where each "m=3D" line has unique payl=
oad
>>> type
>>> >       values, and then the payload type could serve as hint to the
>>> relevant
>>> >                   "m=3D" line the RTP stream is associated with.</t>
>>> >
>>> >       <t>An offerer and answerer can inform each other which SSRC
>>> values
>>> >       they will use for an RTP stream by using the SDP 'ssrc' attribu=
te
>>> >       <xref format=3D"default" pageno=3D"false" target=3D"RFC5576"/>.
>>> However, an
>>> >       offerer will not know which SSRC values the answerer will use
>>> until
>>> >       the offerer has received the answer providing that information.
>>> Due to
>>> >       this, before the offerer has received the answer, the offerer
>>> will not
>>> >       be able to associate an RTP stream with the correct "m=3D" line
>>> using
>>> >       the SSRC value associated with the RTP stream. In addition, the
>>> >       offerer and answerer may start using new SSRC values mid-sessio=
n,
>>> >       without informing each other using the SDP 'ssrc' attribute.</t=
>
>>> >
>>> >       <t>In order for an offerer and answerer to always be able to
>>> associate
>>> >       an RTP stream with the correct "m=3D" line, the offerer and
>>> answerer
>>> >       using the BUNDLE extension MUST support the mechanism defined i=
n
>>> <xref
>>> >       format=3D"default" pageno=3D"false" target=3D"sec-receiver-id"/=
>,
>>> where the
>>> >       offerer and answerer includes the identification-tag (provided
>>> by the
>>> >       remote peer) associated with an "m=3D" line in the RTP Streams =
and
>>> in
>>> >       RTCP SDES packets part of a BUNDLE group.</t>
>>> >
>>> >       <t>The mapping from an SSRC to an identification-tag is carried
>>> in
>>> >       RTCP SDES packets or in RTP header extensions (<xref
>>> format=3D"default"
>>> >       pageno=3D"false" target=3D"sec-receiver-id"/>). Since a compoun=
d RTCP
>>> >       packet can contain multiple RTCP SDES packets, and each RTCP SD=
ES
>>> >       packet can contain multiple chunks, an RTCP packet can contain
>>> several
>>> >       SSRC to identification-tag mappings. The offerer and answerer
>>> maintain
>>> >       tables mapping RTP streams identified by SSRC to "m=3D" lines
>>> identified
>>> >       by the identification-tag.
>>> >       When receiving an RTP packet carrying a MID header extension
>>> >       with the identification-tag, or an RTCP packet carrying one or
>>> >       more SDES MID items, the offerer or answerer creates a mapping
>>> >       table entry between the SSRC value and the identification-tag,
>>> >       in order to associate the RTP stream associated with that SSRC
>>> >       value with the "m=3D" line corresponding to the
>>> identification-tag.</t>
>>> >
>>> >       <t>The mapping between the SSRC an identification-tag might
>>> change
>>> >       mid-session if, for a given SSRC value, a different
>>> identification-tag
>>> >       is provided in an RTP or RTCP packet. In that case these tables
>>> are
>>> >       updated each time an RTP/RTCP packet containing a new mappings
>>> from
>>> >       SSRC to identification-tag is received. Some considerations for
>>> >       avoiding update flaps are provided in Section 4.2.6 of <xref
>>> >       target=3D"RFC7941"/> which should be followed. </t>
>>> >
>>> >       <t>If an offerer and answerer is not able to associate an RTP
>>> stream
>>> >       with an "m=3D" line (using the mechanisms described in this
>>> section, or
>>> >       using other appropriate mechanism, e.g., based on the payload
>>> type
>>> >       value if it is unique to a single "m=3D" line), it MUST either
>>> drop the
>>> >       RTP packets associated with the RTP stream, or process them in =
an
>>> >       application specific manner, once non-stream specific processin=
g
>>> >       (e.g., related to congestion control) of the RTP packets have
>>> >       occurred.</t>
>>> >
>>> >       <t>When compound RTCP packets are received, they are split
>>> >       into their component RTCP packets and those component RTCP
>>> >       packets are processed based on their RTCP packet type, in
>>> >       the order in which they were placed into the compound RTCP
>>> >       packet. Non-compound RTCP packets are processed based on
>>> >       their RTCP packet type, in the order they are received.
>>> Information
>>> >       in each RTCP packet can relate to one or more RTP streams.
>>> >       For example, RTCP Sender Report (SR) and Receiver Report (RR)
>>> >       packets include an SSRC of sender field that indicates the
>>> >       identity of the participant that sent the RTCP packet, along
>>> >       with a list of Report Blocks. Each report contains data on the
>>> >       reception quality of a single RTP stream, identified by SSRC,
>>> >       as received by the SSRC that sent the RTCP packet. Other RTCP
>>> >       packet types similarly contain references to the SSRC of the
>>> >       sender of the RTCP packet, and the RTP streams to which it
>>> >       refers.</t>
>>> >
>>> >       <t>It should always be possible to process RTCP packets, and
>>> >       store the received information in a data structure associated
>>> >       with an RTP stream, identified by SSRC, for later access and
>>> >       use. It is possible that RTCP packets relating to an SSRC can
>>> >       be received before RTP packets relating to that SSRC, so the
>>> >       data structures relating to an SSRC might need to be created
>>> >       before the corresponding RTP stream is received.</t>
>>> >
>>> >       <t>Similarly, information relating to an RTP stream might be
>>> >       received before the data needed to map it onto an m=3D line is
>>> >       received. Information carried in RTCP packets relating to such
>>> >       an RTP stream that is application and/or "m=3D" line dependent
>>> >       MAY be dropped until the SSRCs is associated with a particular
>>> >       "m=3D" line. However, information to generate RTCP report block=
s
>>> >       and other basic transport level feedback or reporting needs to
>>> >       be retained, so RTCP reports relating to the stream can be
>>> >       generated.</t>
>>> >
>>> >     </section>
>>> >
>>> >
>>> >
>>> >
>>> > --
>>> > Colin Perkins
>>> > https://csperkins.org/
>>> >
>>> >
>>> >
>>> >
>>>
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>
>>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>>
>>
>

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Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">I would like to emphasize the importance of standardizing =
&quot;routing rule&quot; behavior at an appropriate level of detail.=C2=A0<=
div><br></div><div>Developers are encountering interoperability problems an=
d are filing bugs against implementations.</div><div><br></div><div>To addr=
ess the issues we are seeing, it is necessary to get to the level of detail=
 in the JSEP-17 Section 6 text.=C2=A0</div></div><div class=3D"gmail_extra"=
><br><div class=3D"gmail_quote">On Fri, Dec 9, 2016 at 10:01 AM, Bernard Ab=
oba <span dir=3D"ltr">&lt;<a href=3D"mailto:bernard.aboba@gmail.com" target=
=3D"_blank">bernard.aboba@gmail.com</a>&gt;</span> wrote:<br><blockquote cl=
ass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;p=
adding-left:1ex"><div dir=3D"ltr">I agree with Eric and Cullen that a more =
explicit algorithm is needed.=C2=A0 The proposed text is not specific enoug=
h to resolve observed implementation differences.=C2=A0<span class=3D""><di=
v><br></div><div>Cullen said:=C2=A0</div><div><br></div><div>&quot;<span st=
yle=3D"font-size:12.8px">=C2=A0</span><span style=3D"font-size:12.8px">I&#3=
9;d also like it be clear that MID takes priority over PT.&quot;</span></di=
v><div><span style=3D"font-size:12.8px"><br></span></div></span><div><span =
style=3D"font-size:12.8px">[BA] Yes.=C2=A0 This is something that multiple =
implementations appear to agree on.=C2=A0 Also, Peter&#39;s suggestion that=
 a MID mismatch result in a packet drop should be restored.</span></div><di=
v><span style=3D"font-size:12.8px"><br></span></div><div>&quot;I believe we=
 agreed that if the PT was unique, then it would be used for demux as well.=
&quot;</div><div><br></div><div>[BA] That is my understanding as well. =C2=
=A0</div><div><br></div><div>However, I&#39;d also like to understand what =
happens if the PT is not unique.=C2=A0 The proposed text does not provide e=
nough detail and existing implementations differ in how they treat non-uniq=
ue PTs. I am familiar with an implementation that declares SDP with non-uni=
que PTs to be in error.=C2=A0 Another implementation fills a Payload Type t=
able with a single value, with the value selected depending on whether SSRC=
s are also specified (e.g. specification of SSRCs is taken as an indication=
 that only SSRC matching is desired, and therefore that a Payload Type entr=
y is not needed).=C2=A0 The ORTC API spec says to latch the SSRC on a Paylo=
ad Type match and then remove the Payload Type table entry, but some implem=
entations have found this leads to problems so they have foresaken either t=
he latching or the PT removal (or both). =C2=A0</div><span class=3D""><div =
style=3D"font-size:12.8px"><div class=3D"m_-6603285502549914630gmail-adm"><=
div id=3D"m_-6603285502549914630gmail-q_158e4a627b100c1c_1" class=3D"m_-660=
3285502549914630gmail-ajR m_-6603285502549914630gmail-h4"></div></div></div=
><div><br></div><div>Cullen said:=C2=A0</div><div><br></div><div>&quot;<spa=
n style=3D"font-size:12.8px">I prefer the much more explicit algorithm part=
icularly for the RTCP handling as implementors have a hard time figuring ou=
t wha they need to do to process the RTCP.</span>&quot;</div><div><br></div=
></span><div>[BA] I would also prefer that we have a more explicit algorith=
m here, because we have seen very substantial implementation differences.=
=C2=A0 One implementation I&#39;m familiar with sends the RTCP packets to a=
ll RtpSender and RtpReceiver objects.=C2=A0 While this is inefficient, it d=
oes avoid sending RTCP packets to the wrong objects.=C2=A0 Another implemen=
tation sorts the RTCP reports as indicated in the proposed text - but has f=
ound that the required handling is so message-dependent that it leads to bu=
gs (e.g. FIR and APP messages have caused issues in particular).=C2=A0 So t=
his approach is more efficient, unless we are willing to get into detail on=
 every RTCP message, it will fall short.=C2=A0</div><div><br></div></div><d=
iv class=3D"gmail_extra"><br><div class=3D"gmail_quote"><div><div class=3D"=
h5">On Fri, Dec 9, 2016 at 9:13 AM, Eric Rescorla <span dir=3D"ltr">&lt;<a =
href=3D"mailto:ekr@rtfm.com" target=3D"_blank">ekr@rtfm.com</a>&gt;</span> =
wrote:<br></div></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0=
 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class=3D"h5"=
><div dir=3D"ltr">I agree with Cullen that the algorithm structure used in =
current JSEP is the better structure.<div><br></div><div>Magnus, Colin, do =
you think you could rewrite your text in that structure?</div><div><br></di=
v><div>-Ekr</div><div><br></div></div><div class=3D"m_-6603285502549914630H=
OEnZb"><div class=3D"m_-6603285502549914630h5"><div class=3D"gmail_extra"><=
br><div class=3D"gmail_quote">On Fri, Dec 9, 2016 at 7:10 AM, Cullen Jennin=
gs <span dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@iii.ca" target=3D"_blank"=
>fluffy@iii.ca</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" s=
tyle=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><br>
In general, I think it makes sense to put most the details in bundle and ov=
erview in jsep as you have proposed here.<br>
<br>
I believe we agreed that if the PT was unique, then that would be used for =
the demux as well. I&#39;d like to see that explicitly spelled out in this =
text instead of just left as an option allowed but not really specified by =
this text.=C2=A0 I&#39;d also like it be clear that MID takes priority over=
 PT.<br>
<br>
I prefer the much more explicit algorithm particularly for the RTCP handlin=
g as implementors have a hard time figuring out wha they need to do to proc=
ess the RTCP.<br>
<div><div class=3D"m_-6603285502549914630m_8079163472208971269h5"><br>
<br>
<br>
&gt; On Dec 8, 2016, at 3:54 AM, Colin Perkins &lt;<a href=3D"mailto:csp@cs=
perkins.org" target=3D"_blank">csp@csperkins.org</a>&gt; wrote:<br>
&gt;<br>
&gt; Hi,<br>
&gt;<br>
&gt; [cc=E2=80=99ing RTCWEB and MMUSIC, since this relates to both JSEP and=
 BUNDLE drafts]<br>
&gt;<br>
&gt; There=E2=80=99s been a lot of discussion on the lists, and at the meet=
ing in Seoul, around how RTP streams are mapped onto higher-level, applicat=
ion meaningful, semantic roles. In particular, around how RTP streams map o=
nto JSEP objects for WebRTC. Magnus Westerlund and I would like to propose =
the following updates to JSEP and BUNDLE to try to clarify the behaviour.<b=
r>
&gt;<br>
&gt; Comments and feedback very welcome.<br>
&gt;<br>
&gt; Cheers,<br>
&gt; Colin<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; # Replacement for Section 6 of draft-ietf-rtcweb-jsep-17<br>
&gt;<br>
&gt; &lt;section title=3D&quot;Processing RTP/RTCP&quot; anchor=3D&quot;sec=
.rtp.demux&quot;&gt;<br>
&gt;=C2=A0 =C2=A0 &lt;t&gt;As described in &lt;xref target=3D&quot;RFC3550&=
quot;/&gt;, RTP packets are<br>
&gt;=C2=A0 =C2=A0 associated with RTP streams &lt;xref target=3D&quot;RFC76=
56&quot;/&gt;. Metadata<br>
&gt;=C2=A0 =C2=A0 about those streams, including source description informa=
tion<br>
&gt;=C2=A0 =C2=A0 and reception quality feedback is conveyed in RTCP packet=
s.<br>
&gt;=C2=A0 =C2=A0 Each RTP stream is identified by an SSRC, and each RTP pa=
cket<br>
&gt;=C2=A0 =C2=A0 carries an SSRC value that is used to associate the packe=
t with<br>
&gt;=C2=A0 =C2=A0 the correct RTP stream. RTCP packets also use SSRCs to id=
entify<br>
&gt;=C2=A0 =C2=A0 the RTP streams that the reports and metadata relate to.=
=C2=A0 RTCP<br>
&gt;=C2=A0 =C2=A0 packets generally carry multiple SSRC values and report o=
n, or<br>
&gt;=C2=A0 =C2=A0 deliver source description information relating to, sever=
al RTP<br>
&gt;=C2=A0 =C2=A0 streams.&lt;/t&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 &lt;t&gt;Each incoming RTP stream, identified by its SSRC=
, is mapped to<br>
&gt;=C2=A0 =C2=A0 an m=3D section in the SDP. The SDP m=3D sections then co=
rrespond to<br>
&gt;=C2=A0 =C2=A0 RtpReceiver objects. This allows each RTP stream to be as=
sociated<br>
&gt;=C2=A0 =C2=A0 with an RtpTransceiver. Further processing of the RTP str=
eam can<br>
&gt;=C2=A0 =C2=A0 then be done at the RtpTransceiver level.=C2=A0 This incl=
udes using<br>
&gt;=C2=A0 =C2=A0 RID &lt;xref target=3D&quot;I-D.ietf-mmusic-rid&quot;/&gt=
; to distinguish between<br>
&gt;=C2=A0 =C2=A0 multiple Encoded Streams, as well as determine which Sour=
ce RTP<br>
&gt;=C2=A0 =C2=A0 stream should be repaired by a given Redundancy RTP strea=
m.&lt;/t&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 &lt;t&gt;The process of mapping RTP streams onto m=3D sec=
tions depends on<br>
&gt;=C2=A0 =C2=A0 whether streams are bundled or not. If the SDP BUNDLE ext=
ension<br>
&gt;=C2=A0 =C2=A0 is in use, then RTP streams are mapped onto m=3D sections=
 based on<br>
&gt;=C2=A0 =C2=A0 the MID values as described in<br>
&gt;=C2=A0 =C2=A0 &lt;xref target=3D&quot;I-D.ietf-mmusic-sdp-bu<wbr>ndle-n=
egotiation&quot;/&gt;.=C2=A0 If the<br>
&gt;=C2=A0 =C2=A0 SDP BUNDLE extension is not in use, each m=3D section cor=
responds<br>
&gt;=C2=A0 =C2=A0 to a transport layer connection and the RTP streams recei=
ved on<br>
&gt;=C2=A0 =C2=A0 that connection correspond to the m=3D section.&lt;/t&gt;=
<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 &lt;t&gt;Incoming RTCP packets contain metadata including=
 reception<br>
&gt;=C2=A0 =C2=A0 quality feedback, source description information, and oth=
er<br>
&gt;=C2=A0 =C2=A0 signalling relating to RTP streams. The RTCP packets are =
parsed,<br>
&gt;=C2=A0 =C2=A0 the associated RTP streams are identified based on the in=
cluded<br>
&gt;=C2=A0 =C2=A0 SSRC values, and the metadata relating to those RTP strea=
ms is<br>
&gt;=C2=A0 =C2=A0 updated (this might include updating the MID information,=
 used<br>
&gt;=C2=A0 =C2=A0 to associate RTP streams with m=3D sections, if the SDP B=
UNDLE<br>
&gt;=C2=A0 =C2=A0 extension is in use). This updated metadata is available =
to the<br>
&gt;=C2=A0 =C2=A0 RtpTransceiver objects associated with those RTP streams.=
<br>
&gt;=C2=A0 =C2=A0 &lt;/t&gt;<br>
&gt; &lt;/section&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; # Replacement text for section 10.2 of draft-ietf-mmusic-sdp-bundle-n<=
wbr>egotiation-36<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0&lt;section anchor=3D&quot;sec-rtp-pt&quot;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 title=3D&quot;Associat=
ing RTP Streams With Correct SDP Media Description&quot;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 toc=3D&quot;default&qu=
ot;&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;As described in &lt;xref format=3D&=
quot;default&quot; pageno=3D&quot;false&quot;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0target=3D&quot;RFC3550&quot;/&gt;, RTP packe=
ts are associated with RTP streams &lt;xref<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0format=3D&quot;default&quot; pageno=3D&quot;=
false&quot; target=3D&quot;RFC7656&quot;/&gt;. Each RTP stream is<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0identified by an SSRC value, and each RTP pa=
cket carries an SSRC value<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0that is used to associate the packet with th=
e correct RTP stream. RTCP<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0packets also uses SSRCs to identify on which=
 RTP streams any report or<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0feedback relate to. Thus, an RTCP packet wil=
l commonly carry multiple<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0SSRC values, and might therefore be providin=
g feedback or report on<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0multiple RTP streams. &lt;/t&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;In order to be able to process rece=
ived RTP packets correctly it<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0must be possible to associate an RTP stream =
with the correct &quot;m=3D&quot;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0line, as the &quot;m=3D&quot; line and SDP a=
ttributes associated with the &quot;m=3D&quot;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0line contain information needed to process t=
he packets.&lt;/t&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;As all RTP streams associated with =
a BUNDLE group are part of the<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0same RTP session and using the same address:=
port combination for<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0sending and receiving RTP/RTCP packets, the =
local address:port<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0combination cannot be used to associate an R=
TP stream with the correct<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0&quot;m=3D&quot; line. In addition, multiple=
 RTP streams might be associated with<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0the same &quot;m=3D&quot; line.&lt;/t&gt;<br=
>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;Also, as described in &lt;xref form=
at=3D&quot;default&quot; pageno=3D&quot;false&quot;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0target=3D&quot;sec-rtp-sessions-pt&quot;/<wb=
r>&gt;, the same payload type value might be<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0used by multiple RTP streams, in which case =
the payload type value<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0cannot be used to associate an RTP stream wi=
th the correct &quot;m=3D&quot; line.<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0However, there are cases where each &quot;m=
=3D&quot; line has unique payload type<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0values, and then the payload type could serv=
e as hint to the relevant<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0&q=
uot;m=3D&quot; line the RTP stream is associated with.&lt;/t&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;An offerer and answerer can inform =
each other which SSRC values<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0they will use for an RTP stream by using the=
 SDP &#39;ssrc&#39; attribute<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;xref format=3D&quot;default&quot; pageno=
=3D&quot;false&quot; target=3D&quot;RFC5576&quot;/&gt;. However, an<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0offerer will not know which SSRC values the =
answerer will use until<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0the offerer has received the answer providin=
g that information. Due to<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0this, before the offerer has received the an=
swer, the offerer will not<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0be able to associate an RTP stream with the =
correct &quot;m=3D&quot; line using<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0the SSRC value associated with the RTP strea=
m. In addition, the<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0offerer and answerer may start using new SSR=
C values mid-session,<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0without informing each other using the SDP &=
#39;ssrc&#39; attribute.&lt;/t&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;In order for an offerer and answere=
r to always be able to associate<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0an RTP stream with the correct &quot;m=3D&qu=
ot; line, the offerer and answerer<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0using the BUNDLE extension MUST support the =
mechanism defined in &lt;xref<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0format=3D&quot;default&quot; pageno=3D&quot;=
false&quot; target=3D&quot;sec-receiver-id&quot;/&gt;, where the<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0offerer and answerer includes the identifica=
tion-tag (provided by the<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0remote peer) associated with an &quot;m=3D&q=
uot; line in the RTP Streams and in<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0RTCP SDES packets part of a BUNDLE group.&lt=
;/t&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;The mapping from an SSRC to an iden=
tification-tag is carried in<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0RTCP SDES packets or in RTP header extension=
s (&lt;xref format=3D&quot;default&quot;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0pageno=3D&quot;false&quot; target=3D&quot;se=
c-receiver-id&quot;/&gt;). Since a compound RTCP<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0packet can contain multiple RTCP SDES packet=
s, and each RTCP SDES<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0packet can contain multiple chunks, an RTCP =
packet can contain several<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0SSRC to identification-tag mappings. The off=
erer and answerer maintain<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0tables mapping RTP streams identified by SSR=
C to &quot;m=3D&quot; lines identified<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0by the identification-tag.<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0When receiving an RTP packet carrying a MID =
header extension<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0with the identification-tag, or an RTCP pack=
et carrying one or<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0more SDES MID items, the offerer or answerer=
 creates a mapping<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0table entry between the SSRC value and the i=
dentification-tag,<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0in order to associate the RTP stream associa=
ted with that SSRC<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0value with the &quot;m=3D&quot; line corresp=
onding to the identification-tag.&lt;/t&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;The mapping between the SSRC an ide=
ntification-tag might change<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0mid-session if, for a given SSRC value, a di=
fferent identification-tag<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0is provided in an RTP or RTCP packet. In tha=
t case these tables are<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0updated each time an RTP/RTCP packet contain=
ing a new mappings from<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0SSRC to identification-tag is received. Some=
 considerations for<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0avoiding update flaps are provided in Sectio=
n 4.2.6 of &lt;xref<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0target=3D&quot;RFC7941&quot;/&gt; which shou=
ld be followed. &lt;/t&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;If an offerer and answerer is not a=
ble to associate an RTP stream<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0with an &quot;m=3D&quot; line (using the mec=
hanisms described in this section, or<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0using other appropriate mechanism, e.g., bas=
ed on the payload type<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0value if it is unique to a single &quot;m=3D=
&quot; line), it MUST either drop the<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0RTP packets associated with the RTP stream, =
or process them in an<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0application specific manner, once non-stream=
 specific processing<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0(e.g., related to congestion control) of the=
 RTP packets have<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0occurred.&lt;/t&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;When compound RTCP packets are rece=
ived, they are split<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0into their component RTCP packets and those =
component RTCP<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0packets are processed based on their RTCP pa=
cket type, in<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0the order in which they were placed into the=
 compound RTCP<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0packet. Non-compound RTCP packets are proces=
sed based on<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0their RTCP packet type, in the order they ar=
e received. Information<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0in each RTCP packet can relate to one or mor=
e RTP streams.<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0For example, RTCP Sender Report (SR) and Rec=
eiver Report (RR)<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0packets include an SSRC of sender field that=
 indicates the<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0identity of the participant that sent the RT=
CP packet, along<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0with a list of Report Blocks. Each report co=
ntains data on the<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0reception quality of a single RTP stream, id=
entified by SSRC,<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0as received by the SSRC that sent the RTCP p=
acket. Other RTCP<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0packet types similarly contain references to=
 the SSRC of the<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0sender of the RTCP packet, and the RTP strea=
ms to which it<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0refers.&lt;/t&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;It should always be possible to pro=
cess RTCP packets, and<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0store the received information in a data str=
ucture associated<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0with an RTP stream, identified by SSRC, for =
later access and<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0use. It is possible that RTCP packets relati=
ng to an SSRC can<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0be received before RTP packets relating to t=
hat SSRC, so the<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0data structures relating to an SSRC might ne=
ed to be created<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0before the corresponding RTP stream is recei=
ved.&lt;/t&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;Similarly, information relating to =
an RTP stream might be<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0received before the data needed to map it on=
to an m=3D line is<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0received. Information carried in RTCP packet=
s relating to such<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0an RTP stream that is application and/or &qu=
ot;m=3D&quot; line dependent<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0MAY be dropped until the SSRCs is associated=
 with a particular<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0&quot;m=3D&quot; line. However, information =
to generate RTCP report blocks<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0and other basic transport level feedback or =
reporting needs to<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0be retained, so RTCP reports relating to the=
 stream can be<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0generated.&lt;/t&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0&lt;/section&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; --<br>
&gt; Colin Perkins<br>
&gt; <a href=3D"https://csperkins.org/" rel=3D"noreferrer" target=3D"_blank=
">https://csperkins.org/</a><br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
<br>
</div></div>______________________________<wbr>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/rtcweb</a><br=
>
</blockquote></div><br></div>
</div></div><br></div></div><span class=3D"">______________________________=
<wbr>_________________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.org</a><br=
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 target=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/mmusic</a><br=
>
<br></span></blockquote></div><br></div>
</blockquote></div><br></div>

--001a113dc310112fd105454962ed--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Roman Shpount <roman@telurix.com>
Thread-Topic: [MMUSIC] draft-4572-update: Spec contains references to a number of obsoleted RFCs
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Subject: Re: [MMUSIC] draft-4572-update: Spec contains references to a number of obsoleted RFCs
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From nobody Thu Jan  5 06:08:02 2017
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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 of the IETF.

        Title           : Connection-Oriented Media Transport over TLS in SDP
        Authors         : Jonathan Lennox
                          Christer Holmberg
	Filename        : draft-ietf-mmusic-4572-update-10.txt
	Pages           : 17
	Date            : 2017-01-05

Abstract:
   This document specifies how to establish secure connection-oriented
   media transport sessions over the Transport Layer Security (TLS)
   protocol using the Session Description Protocol (SDP).  It defines a
   new SDP protocol identifier, 'TCP/TLS'.  It also defines the syntax
   and semantics for an SDP 'fingerprint' attribute that identifies the
   certificate that will be presented for the TLS session.  This
   mechanism allows media transport over TLS connections to be
   established securely, so long as the integrity of session
   descriptions is assured.

   This document obsoletes RFC 4572, by clarifying the usage of multiple
   fingerprints.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-4572-update/

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-mmusic-4572-update-10

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-4572-update-10


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: Eric Rescorla <ekr@rtfm.com>
Date: Thu, 5 Jan 2017 10:56:33 -0800
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To: Bernard Aboba <bernard.aboba@gmail.com>
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Cc: "mmusic \(E-mail\)" <mmusic@ietf.org>, Robin Raymond <robin@opticaltone.com>, draft-ietf-rtcweb-jsep@tools.ietf.org, RTCWeb IETF <rtcweb@ietf.org>, Cullen Jennings <fluffy@iii.ca>, Colin Perkins <csp@csperkins.org>, draft-ietf-mmusic-sdp-bundle-negotiation@ietf.org
Subject: Re: [MMUSIC] [rtcweb] RTP demux in JSEP and BUNDLE
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This discussion seems to have stalled. I think it's clear that:

- There's a strong (though not necessarily unanimous) feeling on the JSEP
side that we need to specify an actual algorithm in the way that JSEP did.
- Magnus and Colin feel like the algorithm in JSEP isn't adequate.

Given that, I think the best way to proceed is to schedule a virtual
interim to jointly produce text in algorithmic form that addresses Colin
and Magnus's concerns. This approach worked well the last time we had some
contentious wordsmithing (the rules around IP address handling). Chairs, do
you think you could arrange this sometime soon so it doesn't hold up JSEP.

-Ekr




On Wed, Jan 4, 2017 at 10:59 AM, Bernard Aboba <bernard.aboba@gmail.com>
wrote:

> I would like to emphasize the importance of standardizing "routing rule"
> behavior at an appropriate level of detail.
>
> Developers are encountering interoperability problems and are filing bugs
> against implementations.
>
> To address the issues we are seeing, it is necessary to get to the level
> of detail in the JSEP-17 Section 6 text.
>
> On Fri, Dec 9, 2016 at 10:01 AM, Bernard Aboba <bernard.aboba@gmail.com>
> wrote:
>
>> I agree with Eric and Cullen that a more explicit algorithm is needed.
>> The proposed text is not specific enough to resolve observed implementat=
ion
>> differences.
>>
>> Cullen said:
>>
>> " I'd also like it be clear that MID takes priority over PT."
>>
>> [BA] Yes.  This is something that multiple implementations appear to
>> agree on.  Also, Peter's suggestion that a MID mismatch result in a pack=
et
>> drop should be restored.
>>
>> "I believe we agreed that if the PT was unique, then it would be used fo=
r
>> demux as well."
>>
>> [BA] That is my understanding as well.
>>
>> However, I'd also like to understand what happens if the PT is not
>> unique.  The proposed text does not provide enough detail and existing
>> implementations differ in how they treat non-unique PTs. I am familiar w=
ith
>> an implementation that declares SDP with non-unique PTs to be in error.
>> Another implementation fills a Payload Type table with a single value, w=
ith
>> the value selected depending on whether SSRCs are also specified (e.g.
>> specification of SSRCs is taken as an indication that only SSRC matching=
 is
>> desired, and therefore that a Payload Type entry is not needed).  The OR=
TC
>> API spec says to latch the SSRC on a Payload Type match and then remove =
the
>> Payload Type table entry, but some implementations have found this leads=
 to
>> problems so they have foresaken either the latching or the PT removal (o=
r
>> both).
>>
>> Cullen said:
>>
>> "I prefer the much more explicit algorithm particularly for the RTCP
>> handling as implementors have a hard time figuring out wha they need to =
do
>> to process the RTCP."
>>
>> [BA] I would also prefer that we have a more explicit algorithm here,
>> because we have seen very substantial implementation differences.  One
>> implementation I'm familiar with sends the RTCP packets to all RtpSender
>> and RtpReceiver objects.  While this is inefficient, it does avoid sendi=
ng
>> RTCP packets to the wrong objects.  Another implementation sorts the RTC=
P
>> reports as indicated in the proposed text - but has found that the requi=
red
>> handling is so message-dependent that it leads to bugs (e.g. FIR and APP
>> messages have caused issues in particular).  So this approach is more
>> efficient, unless we are willing to get into detail on every RTCP messag=
e,
>> it will fall short.
>>
>>
>> On Fri, Dec 9, 2016 at 9:13 AM, Eric Rescorla <ekr@rtfm.com> wrote:
>>
>>> I agree with Cullen that the algorithm structure used in current JSEP i=
s
>>> the better structure.
>>>
>>> Magnus, Colin, do you think you could rewrite your text in that
>>> structure?
>>>
>>> -Ekr
>>>
>>>
>>> On Fri, Dec 9, 2016 at 7:10 AM, Cullen Jennings <fluffy@iii.ca> wrote:
>>>
>>>>
>>>> In general, I think it makes sense to put most the details in bundle
>>>> and overview in jsep as you have proposed here.
>>>>
>>>> I believe we agreed that if the PT was unique, then that would be used
>>>> for the demux as well. I'd like to see that explicitly spelled out in =
this
>>>> text instead of just left as an option allowed but not really specifie=
d by
>>>> this text.  I'd also like it be clear that MID takes priority over PT.
>>>>
>>>> I prefer the much more explicit algorithm particularly for the RTCP
>>>> handling as implementors have a hard time figuring out wha they need t=
o do
>>>> to process the RTCP.
>>>>
>>>>
>>>>
>>>> > On Dec 8, 2016, at 3:54 AM, Colin Perkins <csp@csperkins.org> wrote:
>>>> >
>>>> > Hi,
>>>> >
>>>> > [cc=E2=80=99ing RTCWEB and MMUSIC, since this relates to both JSEP a=
nd BUNDLE
>>>> drafts]
>>>> >
>>>> > There=E2=80=99s been a lot of discussion on the lists, and at the me=
eting in
>>>> Seoul, around how RTP streams are mapped onto higher-level, applicatio=
n
>>>> meaningful, semantic roles. In particular, around how RTP streams map =
onto
>>>> JSEP objects for WebRTC. Magnus Westerlund and I would like to propose=
 the
>>>> following updates to JSEP and BUNDLE to try to clarify the behaviour.
>>>> >
>>>> > Comments and feedback very welcome.
>>>> >
>>>> > Cheers,
>>>> > Colin
>>>> >
>>>> >
>>>> >
>>>> > # Replacement for Section 6 of draft-ietf-rtcweb-jsep-17
>>>> >
>>>> > <section title=3D"Processing RTP/RTCP" anchor=3D"sec.rtp.demux">
>>>> >    <t>As described in <xref target=3D"RFC3550"/>, RTP packets are
>>>> >    associated with RTP streams <xref target=3D"RFC7656"/>. Metadata
>>>> >    about those streams, including source description information
>>>> >    and reception quality feedback is conveyed in RTCP packets.
>>>> >    Each RTP stream is identified by an SSRC, and each RTP packet
>>>> >    carries an SSRC value that is used to associate the packet with
>>>> >    the correct RTP stream. RTCP packets also use SSRCs to identify
>>>> >    the RTP streams that the reports and metadata relate to.  RTCP
>>>> >    packets generally carry multiple SSRC values and report on, or
>>>> >    deliver source description information relating to, several RTP
>>>> >    streams.</t>
>>>> >
>>>> >    <t>Each incoming RTP stream, identified by its SSRC, is mapped to
>>>> >    an m=3D section in the SDP. The SDP m=3D sections then correspond=
 to
>>>> >    RtpReceiver objects. This allows each RTP stream to be associated
>>>> >    with an RtpTransceiver. Further processing of the RTP stream can
>>>> >    then be done at the RtpTransceiver level.  This includes using
>>>> >    RID <xref target=3D"I-D.ietf-mmusic-rid"/> to distinguish between
>>>> >    multiple Encoded Streams, as well as determine which Source RTP
>>>> >    stream should be repaired by a given Redundancy RTP stream.</t>
>>>> >
>>>> >    <t>The process of mapping RTP streams onto m=3D sections depends =
on
>>>> >    whether streams are bundled or not. If the SDP BUNDLE extension
>>>> >    is in use, then RTP streams are mapped onto m=3D sections based o=
n
>>>> >    the MID values as described in
>>>> >    <xref target=3D"I-D.ietf-mmusic-sdp-bundle-negotiation"/>.  If th=
e
>>>> >    SDP BUNDLE extension is not in use, each m=3D section corresponds
>>>> >    to a transport layer connection and the RTP streams received on
>>>> >    that connection correspond to the m=3D section.</t>
>>>> >
>>>> >    <t>Incoming RTCP packets contain metadata including reception
>>>> >    quality feedback, source description information, and other
>>>> >    signalling relating to RTP streams. The RTCP packets are parsed,
>>>> >    the associated RTP streams are identified based on the included
>>>> >    SSRC values, and the metadata relating to those RTP streams is
>>>> >    updated (this might include updating the MID information, used
>>>> >    to associate RTP streams with m=3D sections, if the SDP BUNDLE
>>>> >    extension is in use). This updated metadata is available to the
>>>> >    RtpTransceiver objects associated with those RTP streams.
>>>> >    </t>
>>>> > </section>
>>>> >
>>>> >
>>>> >
>>>> > # Replacement text for section 10.2 of draft-ietf-mmusic-sdp-bundle-=
n
>>>> egotiation-36
>>>> >
>>>> >     <section anchor=3D"sec-rtp-pt"
>>>> >              title=3D"Associating RTP Streams With Correct SDP Media
>>>> Description"
>>>> >              toc=3D"default">
>>>> >       <t>As described in <xref format=3D"default" pageno=3D"false"
>>>> >       target=3D"RFC3550"/>, RTP packets are associated with RTP stre=
ams
>>>> <xref
>>>> >       format=3D"default" pageno=3D"false" target=3D"RFC7656"/>. Each=
 RTP
>>>> stream is
>>>> >       identified by an SSRC value, and each RTP packet carries an
>>>> SSRC value
>>>> >       that is used to associate the packet with the correct RTP
>>>> stream. RTCP
>>>> >       packets also uses SSRCs to identify on which RTP streams any
>>>> report or
>>>> >       feedback relate to. Thus, an RTCP packet will commonly carry
>>>> multiple
>>>> >       SSRC values, and might therefore be providing feedback or
>>>> report on
>>>> >       multiple RTP streams. </t>
>>>> >
>>>> >       <t>In order to be able to process received RTP packets
>>>> correctly it
>>>> >       must be possible to associate an RTP stream with the correct
>>>> "m=3D"
>>>> >       line, as the "m=3D" line and SDP attributes associated with th=
e
>>>> "m=3D"
>>>> >       line contain information needed to process the packets.</t>
>>>> >
>>>> >       <t>As all RTP streams associated with a BUNDLE group are part
>>>> of the
>>>> >       same RTP session and using the same address:port combination f=
or
>>>> >       sending and receiving RTP/RTCP packets, the local address:port
>>>> >       combination cannot be used to associate an RTP stream with the
>>>> correct
>>>> >       "m=3D" line. In addition, multiple RTP streams might be
>>>> associated with
>>>> >       the same "m=3D" line.</t>
>>>> >
>>>> >       <t>Also, as described in <xref format=3D"default" pageno=3D"fa=
lse"
>>>> >       target=3D"sec-rtp-sessions-pt"/>, the same payload type value
>>>> might be
>>>> >       used by multiple RTP streams, in which case the payload type
>>>> value
>>>> >       cannot be used to associate an RTP stream with the correct "m=
=3D"
>>>> line.
>>>> >       However, there are cases where each "m=3D" line has unique
>>>> payload type
>>>> >       values, and then the payload type could serve as hint to the
>>>> relevant
>>>> >                   "m=3D" line the RTP stream is associated with.</t>
>>>> >
>>>> >       <t>An offerer and answerer can inform each other which SSRC
>>>> values
>>>> >       they will use for an RTP stream by using the SDP 'ssrc'
>>>> attribute
>>>> >       <xref format=3D"default" pageno=3D"false" target=3D"RFC5576"/>=
.
>>>> However, an
>>>> >       offerer will not know which SSRC values the answerer will use
>>>> until
>>>> >       the offerer has received the answer providing that information=
.
>>>> Due to
>>>> >       this, before the offerer has received the answer, the offerer
>>>> will not
>>>> >       be able to associate an RTP stream with the correct "m=3D" lin=
e
>>>> using
>>>> >       the SSRC value associated with the RTP stream. In addition, th=
e
>>>> >       offerer and answerer may start using new SSRC values
>>>> mid-session,
>>>> >       without informing each other using the SDP 'ssrc' attribute.</=
t>
>>>> >
>>>> >       <t>In order for an offerer and answerer to always be able to
>>>> associate
>>>> >       an RTP stream with the correct "m=3D" line, the offerer and
>>>> answerer
>>>> >       using the BUNDLE extension MUST support the mechanism defined
>>>> in <xref
>>>> >       format=3D"default" pageno=3D"false" target=3D"sec-receiver-id"=
/>,
>>>> where the
>>>> >       offerer and answerer includes the identification-tag (provided
>>>> by the
>>>> >       remote peer) associated with an "m=3D" line in the RTP Streams
>>>> and in
>>>> >       RTCP SDES packets part of a BUNDLE group.</t>
>>>> >
>>>> >       <t>The mapping from an SSRC to an identification-tag is carrie=
d
>>>> in
>>>> >       RTCP SDES packets or in RTP header extensions (<xref
>>>> format=3D"default"
>>>> >       pageno=3D"false" target=3D"sec-receiver-id"/>). Since a compou=
nd
>>>> RTCP
>>>> >       packet can contain multiple RTCP SDES packets, and each RTCP
>>>> SDES
>>>> >       packet can contain multiple chunks, an RTCP packet can contain
>>>> several
>>>> >       SSRC to identification-tag mappings. The offerer and answerer
>>>> maintain
>>>> >       tables mapping RTP streams identified by SSRC to "m=3D" lines
>>>> identified
>>>> >       by the identification-tag.
>>>> >       When receiving an RTP packet carrying a MID header extension
>>>> >       with the identification-tag, or an RTCP packet carrying one or
>>>> >       more SDES MID items, the offerer or answerer creates a mapping
>>>> >       table entry between the SSRC value and the identification-tag,
>>>> >       in order to associate the RTP stream associated with that SSRC
>>>> >       value with the "m=3D" line corresponding to the
>>>> identification-tag.</t>
>>>> >
>>>> >       <t>The mapping between the SSRC an identification-tag might
>>>> change
>>>> >       mid-session if, for a given SSRC value, a different
>>>> identification-tag
>>>> >       is provided in an RTP or RTCP packet. In that case these table=
s
>>>> are
>>>> >       updated each time an RTP/RTCP packet containing a new mappings
>>>> from
>>>> >       SSRC to identification-tag is received. Some considerations fo=
r
>>>> >       avoiding update flaps are provided in Section 4.2.6 of <xref
>>>> >       target=3D"RFC7941"/> which should be followed. </t>
>>>> >
>>>> >       <t>If an offerer and answerer is not able to associate an RTP
>>>> stream
>>>> >       with an "m=3D" line (using the mechanisms described in this
>>>> section, or
>>>> >       using other appropriate mechanism, e.g., based on the payload
>>>> type
>>>> >       value if it is unique to a single "m=3D" line), it MUST either
>>>> drop the
>>>> >       RTP packets associated with the RTP stream, or process them in
>>>> an
>>>> >       application specific manner, once non-stream specific processi=
ng
>>>> >       (e.g., related to congestion control) of the RTP packets have
>>>> >       occurred.</t>
>>>> >
>>>> >       <t>When compound RTCP packets are received, they are split
>>>> >       into their component RTCP packets and those component RTCP
>>>> >       packets are processed based on their RTCP packet type, in
>>>> >       the order in which they were placed into the compound RTCP
>>>> >       packet. Non-compound RTCP packets are processed based on
>>>> >       their RTCP packet type, in the order they are received.
>>>> Information
>>>> >       in each RTCP packet can relate to one or more RTP streams.
>>>> >       For example, RTCP Sender Report (SR) and Receiver Report (RR)
>>>> >       packets include an SSRC of sender field that indicates the
>>>> >       identity of the participant that sent the RTCP packet, along
>>>> >       with a list of Report Blocks. Each report contains data on the
>>>> >       reception quality of a single RTP stream, identified by SSRC,
>>>> >       as received by the SSRC that sent the RTCP packet. Other RTCP
>>>> >       packet types similarly contain references to the SSRC of the
>>>> >       sender of the RTCP packet, and the RTP streams to which it
>>>> >       refers.</t>
>>>> >
>>>> >       <t>It should always be possible to process RTCP packets, and
>>>> >       store the received information in a data structure associated
>>>> >       with an RTP stream, identified by SSRC, for later access and
>>>> >       use. It is possible that RTCP packets relating to an SSRC can
>>>> >       be received before RTP packets relating to that SSRC, so the
>>>> >       data structures relating to an SSRC might need to be created
>>>> >       before the corresponding RTP stream is received.</t>
>>>> >
>>>> >       <t>Similarly, information relating to an RTP stream might be
>>>> >       received before the data needed to map it onto an m=3D line is
>>>> >       received. Information carried in RTCP packets relating to such
>>>> >       an RTP stream that is application and/or "m=3D" line dependent
>>>> >       MAY be dropped until the SSRCs is associated with a particular
>>>> >       "m=3D" line. However, information to generate RTCP report bloc=
ks
>>>> >       and other basic transport level feedback or reporting needs to
>>>> >       be retained, so RTCP reports relating to the stream can be
>>>> >       generated.</t>
>>>> >
>>>> >     </section>
>>>> >
>>>> >
>>>> >
>>>> >
>>>> > --
>>>> > Colin Perkins
>>>> > https://csperkins.org/
>>>> >
>>>> >
>>>> >
>>>> >
>>>>
>>>> _______________________________________________
>>>> rtcweb mailing list
>>>> rtcweb@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>
>>>
>>>
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>
>>>
>>
>

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<div dir=3D"ltr">This discussion seems to have stalled. I think it&#39;s cl=
ear that:<div><br></div><div>- There&#39;s a strong (though not necessarily=
 unanimous) feeling on the JSEP side that we need to specify an actual algo=
rithm in the way that JSEP did.</div><div>- Magnus and Colin feel like the =
algorithm in JSEP isn&#39;t adequate.</div><div><br></div><div>Given that, =
I think the best way to proceed is to schedule a virtual interim to jointly=
 produce text in algorithmic form that addresses Colin and Magnus&#39;s con=
cerns. This approach worked well the last time we had some contentious word=
smithing (the rules around IP address handling). Chairs, do you think you c=
ould arrange this sometime soon so it doesn&#39;t hold up JSEP.</div><div><=
br></div><div>-Ekr</div><div><br></div><div><div><br></div><div><br></div><=
/div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Wed=
, Jan 4, 2017 at 10:59 AM, Bernard Aboba <span dir=3D"ltr">&lt;<a href=3D"m=
ailto:bernard.aboba@gmail.com" target=3D"_blank">bernard.aboba@gmail.com</a=
>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 =
0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr">I wo=
uld like to emphasize the importance of standardizing &quot;routing rule&qu=
ot; behavior at an appropriate level of detail.=C2=A0<div><br></div><div>De=
velopers are encountering interoperability problems and are filing bugs aga=
inst implementations.</div><div><br></div><div>To address the issues we are=
 seeing, it is necessary to get to the level of detail in the JSEP-17 Secti=
on 6 text.=C2=A0</div></div><div class=3D"HOEnZb"><div class=3D"h5"><div cl=
ass=3D"gmail_extra"><br><div class=3D"gmail_quote">On Fri, Dec 9, 2016 at 1=
0:01 AM, Bernard Aboba <span dir=3D"ltr">&lt;<a href=3D"mailto:bernard.abob=
a@gmail.com" target=3D"_blank">bernard.aboba@gmail.com</a>&gt;</span> wrote=
:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-le=
ft:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr">I agree with Eric and =
Cullen that a more explicit algorithm is needed.=C2=A0 The proposed text is=
 not specific enough to resolve observed implementation differences.=C2=A0<=
span><div><br></div><div>Cullen said:=C2=A0</div><div><br></div><div>&quot;=
<span style=3D"font-size:12.8px">=C2=A0</span><span style=3D"font-size:12.8=
px">I&#39;d also like it be clear that MID takes priority over PT.&quot;</s=
pan></div><div><span style=3D"font-size:12.8px"><br></span></div></span><di=
v><span style=3D"font-size:12.8px">[BA] Yes.=C2=A0 This is something that m=
ultiple implementations appear to agree on.=C2=A0 Also, Peter&#39;s suggest=
ion that a MID mismatch result in a packet drop should be restored.</span><=
/div><div><span style=3D"font-size:12.8px"><br></span></div><div>&quot;I be=
lieve we agreed that if the PT was unique, then it would be used for demux =
as well.&quot;</div><div><br></div><div>[BA] That is my understanding as we=
ll. =C2=A0</div><div><br></div><div>However, I&#39;d also like to understan=
d what happens if the PT is not unique.=C2=A0 The proposed text does not pr=
ovide enough detail and existing implementations differ in how they treat n=
on-unique PTs. I am familiar with an implementation that declares SDP with =
non-unique PTs to be in error.=C2=A0 Another implementation fills a Payload=
 Type table with a single value, with the value selected depending on wheth=
er SSRCs are also specified (e.g. specification of SSRCs is taken as an ind=
ication that only SSRC matching is desired, and therefore that a Payload Ty=
pe entry is not needed).=C2=A0 The ORTC API spec says to latch the SSRC on =
a Payload Type match and then remove the Payload Type table entry, but some=
 implementations have found this leads to problems so they have foresaken e=
ither the latching or the PT removal (or both). =C2=A0</div><span><div styl=
e=3D"font-size:12.8px"><div class=3D"m_7291791861056250763m_-66032855025499=
14630gmail-adm"><div id=3D"m_7291791861056250763m_-6603285502549914630gmail=
-q_158e4a627b100c1c_1" class=3D"m_7291791861056250763m_-6603285502549914630=
gmail-ajR m_7291791861056250763m_-6603285502549914630gmail-h4"></div></div>=
</div><div><br></div><div>Cullen said:=C2=A0</div><div><br></div><div>&quot=
;<span style=3D"font-size:12.8px">I prefer the much more explicit algorithm=
 particularly for the RTCP handling as implementors have a hard time figuri=
ng out wha they need to do to process the RTCP.</span>&quot;</div><div><br>=
</div></span><div>[BA] I would also prefer that we have a more explicit alg=
orithm here, because we have seen very substantial implementation differenc=
es.=C2=A0 One implementation I&#39;m familiar with sends the RTCP packets t=
o all RtpSender and RtpReceiver objects.=C2=A0 While this is inefficient, i=
t does avoid sending RTCP packets to the wrong objects.=C2=A0 Another imple=
mentation sorts the RTCP reports as indicated in the proposed text - but ha=
s found that the required handling is so message-dependent that it leads to=
 bugs (e.g. FIR and APP messages have caused issues in particular).=C2=A0 S=
o this approach is more efficient, unless we are willing to get into detail=
 on every RTCP message, it will fall short.=C2=A0</div><div><br></div></div=
><div class=3D"gmail_extra"><br><div class=3D"gmail_quote"><div><div class=
=3D"m_7291791861056250763h5">On Fri, Dec 9, 2016 at 9:13 AM, Eric Rescorla =
<span dir=3D"ltr">&lt;<a href=3D"mailto:ekr@rtfm.com" target=3D"_blank">ekr=
@rtfm.com</a>&gt;</span> wrote:<br></div></div><blockquote class=3D"gmail_q=
uote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1e=
x"><div><div class=3D"m_7291791861056250763h5"><div dir=3D"ltr">I agree wit=
h Cullen that the algorithm structure used in current JSEP is the better st=
ructure.<div><br></div><div>Magnus, Colin, do you think you could rewrite y=
our text in that structure?</div><div><br></div><div>-Ekr</div><div><br></d=
iv></div><div class=3D"m_7291791861056250763m_-6603285502549914630HOEnZb"><=
div class=3D"m_7291791861056250763m_-6603285502549914630h5"><div class=3D"g=
mail_extra"><br><div class=3D"gmail_quote">On Fri, Dec 9, 2016 at 7:10 AM, =
Cullen Jennings <span dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@iii.ca" targ=
et=3D"_blank">fluffy@iii.ca</a>&gt;</span> wrote:<br><blockquote class=3D"g=
mail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-l=
eft:1ex"><br>
In general, I think it makes sense to put most the details in bundle and ov=
erview in jsep as you have proposed here.<br>
<br>
I believe we agreed that if the PT was unique, then that would be used for =
the demux as well. I&#39;d like to see that explicitly spelled out in this =
text instead of just left as an option allowed but not really specified by =
this text.=C2=A0 I&#39;d also like it be clear that MID takes priority over=
 PT.<br>
<br>
I prefer the much more explicit algorithm particularly for the RTCP handlin=
g as implementors have a hard time figuring out wha they need to do to proc=
ess the RTCP.<br>
<div><div class=3D"m_7291791861056250763m_-6603285502549914630m_80791634722=
08971269h5"><br>
<br>
<br>
&gt; On Dec 8, 2016, at 3:54 AM, Colin Perkins &lt;<a href=3D"mailto:csp@cs=
perkins.org" target=3D"_blank">csp@csperkins.org</a>&gt; wrote:<br>
&gt;<br>
&gt; Hi,<br>
&gt;<br>
&gt; [cc=E2=80=99ing RTCWEB and MMUSIC, since this relates to both JSEP and=
 BUNDLE drafts]<br>
&gt;<br>
&gt; There=E2=80=99s been a lot of discussion on the lists, and at the meet=
ing in Seoul, around how RTP streams are mapped onto higher-level, applicat=
ion meaningful, semantic roles. In particular, around how RTP streams map o=
nto JSEP objects for WebRTC. Magnus Westerlund and I would like to propose =
the following updates to JSEP and BUNDLE to try to clarify the behaviour.<b=
r>
&gt;<br>
&gt; Comments and feedback very welcome.<br>
&gt;<br>
&gt; Cheers,<br>
&gt; Colin<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; # Replacement for Section 6 of draft-ietf-rtcweb-jsep-17<br>
&gt;<br>
&gt; &lt;section title=3D&quot;Processing RTP/RTCP&quot; anchor=3D&quot;sec=
.rtp.demux&quot;&gt;<br>
&gt;=C2=A0 =C2=A0 &lt;t&gt;As described in &lt;xref target=3D&quot;RFC3550&=
quot;/&gt;, RTP packets are<br>
&gt;=C2=A0 =C2=A0 associated with RTP streams &lt;xref target=3D&quot;RFC76=
56&quot;/&gt;. Metadata<br>
&gt;=C2=A0 =C2=A0 about those streams, including source description informa=
tion<br>
&gt;=C2=A0 =C2=A0 and reception quality feedback is conveyed in RTCP packet=
s.<br>
&gt;=C2=A0 =C2=A0 Each RTP stream is identified by an SSRC, and each RTP pa=
cket<br>
&gt;=C2=A0 =C2=A0 carries an SSRC value that is used to associate the packe=
t with<br>
&gt;=C2=A0 =C2=A0 the correct RTP stream. RTCP packets also use SSRCs to id=
entify<br>
&gt;=C2=A0 =C2=A0 the RTP streams that the reports and metadata relate to.=
=C2=A0 RTCP<br>
&gt;=C2=A0 =C2=A0 packets generally carry multiple SSRC values and report o=
n, or<br>
&gt;=C2=A0 =C2=A0 deliver source description information relating to, sever=
al RTP<br>
&gt;=C2=A0 =C2=A0 streams.&lt;/t&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 &lt;t&gt;Each incoming RTP stream, identified by its SSRC=
, is mapped to<br>
&gt;=C2=A0 =C2=A0 an m=3D section in the SDP. The SDP m=3D sections then co=
rrespond to<br>
&gt;=C2=A0 =C2=A0 RtpReceiver objects. This allows each RTP stream to be as=
sociated<br>
&gt;=C2=A0 =C2=A0 with an RtpTransceiver. Further processing of the RTP str=
eam can<br>
&gt;=C2=A0 =C2=A0 then be done at the RtpTransceiver level.=C2=A0 This incl=
udes using<br>
&gt;=C2=A0 =C2=A0 RID &lt;xref target=3D&quot;I-D.ietf-mmusic-rid&quot;/&gt=
; to distinguish between<br>
&gt;=C2=A0 =C2=A0 multiple Encoded Streams, as well as determine which Sour=
ce RTP<br>
&gt;=C2=A0 =C2=A0 stream should be repaired by a given Redundancy RTP strea=
m.&lt;/t&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 &lt;t&gt;The process of mapping RTP streams onto m=3D sec=
tions depends on<br>
&gt;=C2=A0 =C2=A0 whether streams are bundled or not. If the SDP BUNDLE ext=
ension<br>
&gt;=C2=A0 =C2=A0 is in use, then RTP streams are mapped onto m=3D sections=
 based on<br>
&gt;=C2=A0 =C2=A0 the MID values as described in<br>
&gt;=C2=A0 =C2=A0 &lt;xref target=3D&quot;I-D.ietf-mmusic-sdp-bu<wbr>ndle-n=
egotiation&quot;/&gt;.=C2=A0 If the<br>
&gt;=C2=A0 =C2=A0 SDP BUNDLE extension is not in use, each m=3D section cor=
responds<br>
&gt;=C2=A0 =C2=A0 to a transport layer connection and the RTP streams recei=
ved on<br>
&gt;=C2=A0 =C2=A0 that connection correspond to the m=3D section.&lt;/t&gt;=
<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 &lt;t&gt;Incoming RTCP packets contain metadata including=
 reception<br>
&gt;=C2=A0 =C2=A0 quality feedback, source description information, and oth=
er<br>
&gt;=C2=A0 =C2=A0 signalling relating to RTP streams. The RTCP packets are =
parsed,<br>
&gt;=C2=A0 =C2=A0 the associated RTP streams are identified based on the in=
cluded<br>
&gt;=C2=A0 =C2=A0 SSRC values, and the metadata relating to those RTP strea=
ms is<br>
&gt;=C2=A0 =C2=A0 updated (this might include updating the MID information,=
 used<br>
&gt;=C2=A0 =C2=A0 to associate RTP streams with m=3D sections, if the SDP B=
UNDLE<br>
&gt;=C2=A0 =C2=A0 extension is in use). This updated metadata is available =
to the<br>
&gt;=C2=A0 =C2=A0 RtpTransceiver objects associated with those RTP streams.=
<br>
&gt;=C2=A0 =C2=A0 &lt;/t&gt;<br>
&gt; &lt;/section&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; # Replacement text for section 10.2 of draft-ietf-mmusic-sdp-bundle-n<=
wbr>egotiation-36<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0&lt;section anchor=3D&quot;sec-rtp-pt&quot;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 title=3D&quot;Associat=
ing RTP Streams With Correct SDP Media Description&quot;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 toc=3D&quot;default&qu=
ot;&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;As described in &lt;xref format=3D&=
quot;default&quot; pageno=3D&quot;false&quot;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0target=3D&quot;RFC3550&quot;/&gt;, RTP packe=
ts are associated with RTP streams &lt;xref<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0format=3D&quot;default&quot; pageno=3D&quot;=
false&quot; target=3D&quot;RFC7656&quot;/&gt;. Each RTP stream is<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0identified by an SSRC value, and each RTP pa=
cket carries an SSRC value<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0that is used to associate the packet with th=
e correct RTP stream. RTCP<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0packets also uses SSRCs to identify on which=
 RTP streams any report or<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0feedback relate to. Thus, an RTCP packet wil=
l commonly carry multiple<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0SSRC values, and might therefore be providin=
g feedback or report on<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0multiple RTP streams. &lt;/t&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;In order to be able to process rece=
ived RTP packets correctly it<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0must be possible to associate an RTP stream =
with the correct &quot;m=3D&quot;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0line, as the &quot;m=3D&quot; line and SDP a=
ttributes associated with the &quot;m=3D&quot;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0line contain information needed to process t=
he packets.&lt;/t&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;As all RTP streams associated with =
a BUNDLE group are part of the<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0same RTP session and using the same address:=
port combination for<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0sending and receiving RTP/RTCP packets, the =
local address:port<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0combination cannot be used to associate an R=
TP stream with the correct<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0&quot;m=3D&quot; line. In addition, multiple=
 RTP streams might be associated with<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0the same &quot;m=3D&quot; line.&lt;/t&gt;<br=
>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;Also, as described in &lt;xref form=
at=3D&quot;default&quot; pageno=3D&quot;false&quot;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0target=3D&quot;sec-rtp-sessions-pt&quot;/<wb=
r>&gt;, the same payload type value might be<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0used by multiple RTP streams, in which case =
the payload type value<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0cannot be used to associate an RTP stream wi=
th the correct &quot;m=3D&quot; line.<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0However, there are cases where each &quot;m=
=3D&quot; line has unique payload type<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0values, and then the payload type could serv=
e as hint to the relevant<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0&q=
uot;m=3D&quot; line the RTP stream is associated with.&lt;/t&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;An offerer and answerer can inform =
each other which SSRC values<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0they will use for an RTP stream by using the=
 SDP &#39;ssrc&#39; attribute<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;xref format=3D&quot;default&quot; pageno=
=3D&quot;false&quot; target=3D&quot;RFC5576&quot;/&gt;. However, an<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0offerer will not know which SSRC values the =
answerer will use until<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0the offerer has received the answer providin=
g that information. Due to<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0this, before the offerer has received the an=
swer, the offerer will not<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0be able to associate an RTP stream with the =
correct &quot;m=3D&quot; line using<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0the SSRC value associated with the RTP strea=
m. In addition, the<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0offerer and answerer may start using new SSR=
C values mid-session,<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0without informing each other using the SDP &=
#39;ssrc&#39; attribute.&lt;/t&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;In order for an offerer and answere=
r to always be able to associate<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0an RTP stream with the correct &quot;m=3D&qu=
ot; line, the offerer and answerer<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0using the BUNDLE extension MUST support the =
mechanism defined in &lt;xref<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0format=3D&quot;default&quot; pageno=3D&quot;=
false&quot; target=3D&quot;sec-receiver-id&quot;/&gt;, where the<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0offerer and answerer includes the identifica=
tion-tag (provided by the<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0remote peer) associated with an &quot;m=3D&q=
uot; line in the RTP Streams and in<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0RTCP SDES packets part of a BUNDLE group.&lt=
;/t&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;The mapping from an SSRC to an iden=
tification-tag is carried in<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0RTCP SDES packets or in RTP header extension=
s (&lt;xref format=3D&quot;default&quot;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0pageno=3D&quot;false&quot; target=3D&quot;se=
c-receiver-id&quot;/&gt;). Since a compound RTCP<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0packet can contain multiple RTCP SDES packet=
s, and each RTCP SDES<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0packet can contain multiple chunks, an RTCP =
packet can contain several<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0SSRC to identification-tag mappings. The off=
erer and answerer maintain<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0tables mapping RTP streams identified by SSR=
C to &quot;m=3D&quot; lines identified<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0by the identification-tag.<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0When receiving an RTP packet carrying a MID =
header extension<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0with the identification-tag, or an RTCP pack=
et carrying one or<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0more SDES MID items, the offerer or answerer=
 creates a mapping<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0table entry between the SSRC value and the i=
dentification-tag,<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0in order to associate the RTP stream associa=
ted with that SSRC<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0value with the &quot;m=3D&quot; line corresp=
onding to the identification-tag.&lt;/t&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;The mapping between the SSRC an ide=
ntification-tag might change<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0mid-session if, for a given SSRC value, a di=
fferent identification-tag<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0is provided in an RTP or RTCP packet. In tha=
t case these tables are<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0updated each time an RTP/RTCP packet contain=
ing a new mappings from<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0SSRC to identification-tag is received. Some=
 considerations for<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0avoiding update flaps are provided in Sectio=
n 4.2.6 of &lt;xref<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0target=3D&quot;RFC7941&quot;/&gt; which shou=
ld be followed. &lt;/t&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;If an offerer and answerer is not a=
ble to associate an RTP stream<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0with an &quot;m=3D&quot; line (using the mec=
hanisms described in this section, or<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0using other appropriate mechanism, e.g., bas=
ed on the payload type<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0value if it is unique to a single &quot;m=3D=
&quot; line), it MUST either drop the<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0RTP packets associated with the RTP stream, =
or process them in an<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0application specific manner, once non-stream=
 specific processing<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0(e.g., related to congestion control) of the=
 RTP packets have<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0occurred.&lt;/t&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;When compound RTCP packets are rece=
ived, they are split<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0into their component RTCP packets and those =
component RTCP<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0packets are processed based on their RTCP pa=
cket type, in<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0the order in which they were placed into the=
 compound RTCP<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0packet. Non-compound RTCP packets are proces=
sed based on<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0their RTCP packet type, in the order they ar=
e received. Information<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0in each RTCP packet can relate to one or mor=
e RTP streams.<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0For example, RTCP Sender Report (SR) and Rec=
eiver Report (RR)<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0packets include an SSRC of sender field that=
 indicates the<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0identity of the participant that sent the RT=
CP packet, along<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0with a list of Report Blocks. Each report co=
ntains data on the<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0reception quality of a single RTP stream, id=
entified by SSRC,<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0as received by the SSRC that sent the RTCP p=
acket. Other RTCP<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0packet types similarly contain references to=
 the SSRC of the<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0sender of the RTCP packet, and the RTP strea=
ms to which it<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0refers.&lt;/t&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;It should always be possible to pro=
cess RTCP packets, and<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0store the received information in a data str=
ucture associated<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0with an RTP stream, identified by SSRC, for =
later access and<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0use. It is possible that RTCP packets relati=
ng to an SSRC can<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0be received before RTP packets relating to t=
hat SSRC, so the<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0data structures relating to an SSRC might ne=
ed to be created<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0before the corresponding RTP stream is recei=
ved.&lt;/t&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;Similarly, information relating to =
an RTP stream might be<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0received before the data needed to map it on=
to an m=3D line is<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0received. Information carried in RTCP packet=
s relating to such<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0an RTP stream that is application and/or &qu=
ot;m=3D&quot; line dependent<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0MAY be dropped until the SSRCs is associated=
 with a particular<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0&quot;m=3D&quot; line. However, information =
to generate RTCP report blocks<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0and other basic transport level feedback or =
reporting needs to<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0be retained, so RTCP reports relating to the=
 stream can be<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0generated.&lt;/t&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0&lt;/section&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; --<br>
&gt; Colin Perkins<br>
&gt; <a href=3D"https://csperkins.org/" rel=3D"noreferrer" target=3D"_blank=
">https://csperkins.org/</a><br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
<br>
</div></div>______________________________<wbr>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/rtcweb</a><br=
>
</blockquote></div><br></div>
</div></div><br></div></div><span>______________________________<wbr>______=
___________<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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/mmusic</a><br=
>
<br></span></blockquote></div><br></div>
</blockquote></div><br></div>
</div></div></blockquote></div><br></div>

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From: Peter Thatcher <pthatcher@google.com>
Date: Fri, 06 Jan 2017 19:23:50 +0000
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To: Colin Perkins <csp@csperkins.org>, "mmusic (E-mail)" <mmusic@ietf.org>, rtcweb@ietf.org
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Cc: Magnus Westerlund <magnus.westerlund@ericsson.com>, draft-ietf-rtcweb-jsep@tools.ietf.org, draft-ietf-mmusic-sdp-bundle-negotiation@ietf.org
Subject: Re: [MMUSIC] RTP demux in JSEP and BUNDLE
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When I read your text, I get the impression that you want to simplify the
algorithm to only supporting MIDs for demux with SSRC latching.  No PT
demux.   No SSRC demux (except for ones latched by MID).   In other words,
if you want to use BUNDLE, you must use MID and only MID.

But it's not clear if that's the case.  Perhaps I missed something.  I
agree with others that a more explicit algorithm would make your intention
clear.

To help, here is my interpretation of your text as an algorithm.  Please
let me know if it's a correct interpretation:

    <section title=3D"Appendix B" anchor=3D"sec.appendix-b">
      <t>To prepare for demultiplexing RTP packets to the correct "m=3D"
      line, the following steps MUST be followed for each BUNDLE
      group.</t>

      <t>
        <list>
          <t>Construct a table mapping MID to "m=3D" line for each "m=3D"
          line in this BUNDLE group.</t>

          <t>Construct an empty table mapping incoming SSRC to "m=3D" line.
</t>
        </list>
      </t>

      <t>As "m=3D" lines are added or removed from the BUNDLE groups, or
      their configurations are changed, the tables above must also be
      updated.</t>

      <t>For each RTP packet received, the following steps MUST be
      followed to route the packet to the correct "m=3D" section within
      a BUNDLE group:</t>

      <t>
        <list>
          <t>If the packet has a MID and that MID is not in the table
          mapping MID to "m=3D" line, drop the packet and stop.</t>

          <t>If the packet has a MID and that MID is in the table
          mapping MID to "m=3D" line, update the incoming SSRC mapping
          table to include an entry that maps the packet's SSRC to the
          "m=3D" line for that MID.</t>

          <t>If the packet's SSRC is in the incoming SSRC mapping
          table, route the packet to the associated "m=3D" line and
          stop.</t>

          <t>Otherwise, drop the packet.</t>
        </list>
      </t>

... similar for RTCP ...
    </section>





On Thu, Dec 8, 2016, 2:55 AM Colin Perkins <csp@csperkins.org> wrote:

Hi,

[cc=E2=80=99ing RTCWEB and MMUSIC, since this relates to both JSEP and BUND=
LE
drafts]

There=E2=80=99s been a lot of discussion on the lists, and at the meeting i=
n Seoul,
around how RTP streams are mapped onto higher-level, application
meaningful, semantic roles. In particular, around how RTP streams map onto
JSEP objects for WebRTC. Magnus Westerlund and I would like to propose the
following updates to JSEP and BUNDLE to try to clarify the behaviour.

Comments and feedback very welcome.

Cheers,
Colin



# Replacement for Section 6 of draft-ietf-rtcweb-jsep-17

 <section title=3D"Processing RTP/RTCP" anchor=3D"sec.rtp.demux">
    <t>As described in <xref target=3D"RFC3550"/>, RTP packets are
    associated with RTP streams <xref target=3D"RFC7656"/>. Metadata
    about those streams, including source description information
    and reception quality feedback is conveyed in RTCP packets.
    Each RTP stream is identified by an SSRC, and each RTP packet
    carries an SSRC value that is used to associate the packet with
    the correct RTP stream. RTCP packets also use SSRCs to identify
    the RTP streams that the reports and metadata relate to.  RTCP
    packets generally carry multiple SSRC values and report on, or
    deliver source description information relating to, several RTP
    streams.</t>

    <t>Each incoming RTP stream, identified by its SSRC, is mapped to
    an m=3D section in the SDP. The SDP m=3D sections then correspond to
    RtpReceiver objects. This allows each RTP stream to be associated
    with an RtpTransceiver. Further processing of the RTP stream can
    then be done at the RtpTransceiver level.  This includes using
    RID <xref target=3D"I-D.ietf-mmusic-rid"/> to distinguish between
    multiple Encoded Streams, as well as determine which Source RTP
    stream should be repaired by a given Redundancy RTP stream.</t>

    <t>The process of mapping RTP streams onto m=3D sections depends on
    whether streams are bundled or not. If the SDP BUNDLE extension
    is in use, then RTP streams are mapped onto m=3D sections based on
    the MID values as described in
    <xref target=3D"I-D.ietf-mmusic-sdp-bundle-negotiation"/>.  If the
    SDP BUNDLE extension is not in use, each m=3D section corresponds
    to a transport layer connection and the RTP streams received on
    that connection correspond to the m=3D section.</t>

    <t>Incoming RTCP packets contain metadata including reception
    quality feedback, source description information, and other
    signalling relating to RTP streams. The RTCP packets are parsed,
    the associated RTP streams are identified based on the included
    SSRC values, and the metadata relating to those RTP streams is
    updated (this might include updating the MID information, used
    to associate RTP streams with m=3D sections, if the SDP BUNDLE
    extension is in use). This updated metadata is available to the
    RtpTransceiver objects associated with those RTP streams.
    </t>
 </section>



# Replacement text for section 10.2 of
draft-ietf-mmusic-sdp-bundle-negotiation-36

     <section anchor=3D"sec-rtp-pt"
              title=3D"Associating RTP Streams With Correct SDP Media
Description"
              toc=3D"default">
       <t>As described in <xref format=3D"default" pageno=3D"false"
       target=3D"RFC3550"/>, RTP packets are associated with RTP streams <x=
ref
       format=3D"default" pageno=3D"false" target=3D"RFC7656"/>. Each RTP s=
tream
is
       identified by an SSRC value, and each RTP packet carries an SSRC
value
       that is used to associate the packet with the correct RTP stream.
RTCP
       packets also uses SSRCs to identify on which RTP streams any report
or
       feedback relate to. Thus, an RTCP packet will commonly carry multipl=
e
       SSRC values, and might therefore be providing feedback or report on
       multiple RTP streams. </t>

       <t>In order to be able to process received RTP packets correctly it
       must be possible to associate an RTP stream with the correct "m=3D"
       line, as the "m=3D" line and SDP attributes associated with the "m=
=3D"
       line contain information needed to process the packets.</t>

       <t>As all RTP streams associated with a BUNDLE group are part of the
       same RTP session and using the same address:port combination for
       sending and receiving RTP/RTCP packets, the local address:port
       combination cannot be used to associate an RTP stream with the
correct
       "m=3D" line. In addition, multiple RTP streams might be associated w=
ith
       the same "m=3D" line.</t>

       <t>Also, as described in <xref format=3D"default" pageno=3D"false"
       target=3D"sec-rtp-sessions-pt"/>, the same payload type value might =
be
       used by multiple RTP streams, in which case the payload type value
       cannot be used to associate an RTP stream with the correct "m=3D" li=
ne.
       However, there are cases where each "m=3D" line has unique payload t=
ype
       values, and then the payload type could serve as hint to the relevan=
t
                    "m=3D" line the RTP stream is associated with.</t>

       <t>An offerer and answerer can inform each other which SSRC values
       they will use for an RTP stream by using the SDP 'ssrc' attribute
       <xref format=3D"default" pageno=3D"false" target=3D"RFC5576"/>. Howe=
ver, an
       offerer will not know which SSRC values the answerer will use until
       the offerer has received the answer providing that information. Due
to
       this, before the offerer has received the answer, the offerer will
not
       be able to associate an RTP stream with the correct "m=3D" line usin=
g
       the SSRC value associated with the RTP stream. In addition, the
       offerer and answerer may start using new SSRC values mid-session,
       without informing each other using the SDP 'ssrc' attribute.</t>

       <t>In order for an offerer and answerer to always be able to
associate
       an RTP stream with the correct "m=3D" line, the offerer and answerer
       using the BUNDLE extension MUST support the mechanism defined in
<xref
       format=3D"default" pageno=3D"false" target=3D"sec-receiver-id"/>, wh=
ere the
       offerer and answerer includes the identification-tag (provided by th=
e
       remote peer) associated with an "m=3D" line in the RTP Streams and i=
n
       RTCP SDES packets part of a BUNDLE group.</t>

       <t>The mapping from an SSRC to an identification-tag is carried in
       RTCP SDES packets or in RTP header extensions (<xref format=3D"defau=
lt"
       pageno=3D"false" target=3D"sec-receiver-id"/>). Since a compound RTC=
P
       packet can contain multiple RTCP SDES packets, and each RTCP SDES
       packet can contain multiple chunks, an RTCP packet can contain
several
       SSRC to identification-tag mappings. The offerer and answerer
maintain
       tables mapping RTP streams identified by SSRC to "m=3D" lines
identified
       by the identification-tag.
       When receiving an RTP packet carrying a MID header extension
       with the identification-tag, or an RTCP packet carrying one or
       more SDES MID items, the offerer or answerer creates a mapping
       table entry between the SSRC value and the identification-tag,
       in order to associate the RTP stream associated with that SSRC
       value with the "m=3D" line corresponding to the identification-tag.<=
/t>

       <t>The mapping between the SSRC an identification-tag might change
       mid-session if, for a given SSRC value, a different
identification-tag
       is provided in an RTP or RTCP packet. In that case these tables are
       updated each time an RTP/RTCP packet containing a new mappings from
       SSRC to identification-tag is received. Some considerations for
       avoiding update flaps are provided in Section 4.2.6 of <xref
       target=3D"RFC7941"/> which should be followed. </t>

       <t>If an offerer and answerer is not able to associate an RTP stream
       with an "m=3D" line (using the mechanisms described in this section,=
 or
       using other appropriate mechanism, e.g., based on the payload type
       value if it is unique to a single "m=3D" line), it MUST either drop =
the
       RTP packets associated with the RTP stream, or process them in an
       application specific manner, once non-stream specific processing
       (e.g., related to congestion control) of the RTP packets have
       occurred.</t>

       <t>When compound RTCP packets are received, they are split
       into their component RTCP packets and those component RTCP
       packets are processed based on their RTCP packet type, in
       the order in which they were placed into the compound RTCP
       packet. Non-compound RTCP packets are processed based on
       their RTCP packet type, in the order they are received. Information
       in each RTCP packet can relate to one or more RTP streams.
       For example, RTCP Sender Report (SR) and Receiver Report (RR)
       packets include an SSRC of sender field that indicates the
       identity of the participant that sent the RTCP packet, along
       with a list of Report Blocks. Each report contains data on the
       reception quality of a single RTP stream, identified by SSRC,
       as received by the SSRC that sent the RTCP packet. Other RTCP
       packet types similarly contain references to the SSRC of the
       sender of the RTCP packet, and the RTP streams to which it
       refers.</t>

       <t>It should always be possible to process RTCP packets, and
       store the received information in a data structure associated
       with an RTP stream, identified by SSRC, for later access and
       use. It is possible that RTCP packets relating to an SSRC can
       be received before RTP packets relating to that SSRC, so the
       data structures relating to an SSRC might need to be created
       before the corresponding RTP stream is received.</t>

       <t>Similarly, information relating to an RTP stream might be
       received before the data needed to map it onto an m=3D line is
       received. Information carried in RTCP packets relating to such
       an RTP stream that is application and/or "m=3D" line dependent
       MAY be dropped until the SSRCs is associated with a particular
       "m=3D" line. However, information to generate RTCP report blocks
       and other basic transport level feedback or reporting needs to
       be retained, so RTCP reports relating to the stream can be
       generated.</t>

     </section>




--
Colin Perkins
https://csperkins.org/




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

--94eb2c08da88d6256d054571f51e
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">When I read your text, I get the impression that you want =
to simplify the algorithm to only supporting MIDs for demux with SSRC latch=
ing.=C2=A0 No PT demux. =C2=A0 No SSRC demux (except for ones latched by MI=
D). =C2=A0 In other words, if you want to use BUNDLE, you must use MID and =
only MID.<div><br></div><div>But it&#39;s not clear if that&#39;s the case.=
=C2=A0 Perhaps I missed something.=C2=A0 I agree with others that a more ex=
plicit algorithm would make your intention clear.<div><br></div><div>To hel=
p, here is my interpretation of your text as an algorithm.=C2=A0 Please let=
 me know if it&#39;s a correct interpretation:</div><div><br></div><div><di=
v>=C2=A0 =C2=A0 &lt;section title=3D&quot;Appendix B&quot; anchor=3D&quot;s=
ec.appendix-b&quot;&gt;</div><div>=C2=A0 =C2=A0 =C2=A0 &lt;t&gt;To prepare =
for demultiplexing RTP packets to the correct &quot;m=3D&quot;</div><div>=
=C2=A0 =C2=A0 =C2=A0 line, the following steps MUST be followed for each BU=
NDLE</div><div>=C2=A0 =C2=A0 =C2=A0 group.&lt;/t&gt;</div><div><br></div><d=
iv>=C2=A0 =C2=A0 =C2=A0 &lt;t&gt;</div><div>=C2=A0 =C2=A0 =C2=A0 =C2=A0 &lt=
;list&gt;</div><div>=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 &lt;t&gt;Construct a=
 table mapping MID to &quot;m=3D&quot; line for each &quot;m=3D&quot;</div>=
<div>=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 line in this BUNDLE group.&lt;/t&gt=
;</div><div><br></div><div>=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 &lt;t&gt;Cons=
truct an empty table mapping incoming SSRC to &quot;m=3D&quot; line. &lt;/t=
&gt;</div><div>=C2=A0 =C2=A0 =C2=A0 =C2=A0 &lt;/list&gt;</div><div>=C2=A0 =
=C2=A0 =C2=A0 &lt;/t&gt;</div><div><br></div><div>=C2=A0 =C2=A0 =C2=A0 &lt;=
t&gt;As &quot;m=3D&quot; lines are added or removed from the BUNDLE groups,=
 or</div><div>=C2=A0 =C2=A0 =C2=A0 their configurations are changed, the ta=
bles above must also be</div><div>=C2=A0 =C2=A0 =C2=A0 updated.&lt;/t&gt;</=
div><div><br></div><div>=C2=A0 =C2=A0 =C2=A0 &lt;t&gt;For each RTP packet r=
eceived, the following steps MUST be</div><div>=C2=A0 =C2=A0 =C2=A0 followe=
d to route the packet to the correct &quot;m=3D&quot; section within</div><=
div>=C2=A0 =C2=A0 =C2=A0 a BUNDLE group:&lt;/t&gt;</div><div><br></div><div=
>=C2=A0 =C2=A0 =C2=A0 &lt;t&gt;</div><div>=C2=A0 =C2=A0 =C2=A0 =C2=A0 &lt;l=
ist&gt;</div><div>=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 &lt;t&gt;If the packet=
 has a MID and that MID is not in the table</div><div>=C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 mapping MID to &quot;m=3D&quot; line, drop the packet and sto=
p.&lt;/t&gt;</div><div><br></div><div>=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 &l=
t;t&gt;If the packet has a MID and that MID is in the table</div><div>=C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 mapping MID to &quot;m=3D&quot; line, updat=
e the incoming SSRC mapping</div><div>=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 ta=
ble to include an entry that maps the packet&#39;s SSRC to the</div><div>=
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 &quot;m=3D&quot; line for that MID.&lt;/=
t&gt;</div><div><br></div><div>=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 &lt;t&gt;=
If the packet&#39;s SSRC is in the incoming SSRC mapping</div><div>=C2=A0 =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 table, route the packet to the associated &quot=
;m=3D&quot; line and</div><div>=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 stop.&lt;=
/t&gt;</div><div><br></div><div>=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 &lt;t&gt=
;Otherwise, drop the packet.&lt;/t&gt;</div><div>=C2=A0 =C2=A0 =C2=A0 =C2=
=A0 &lt;/list&gt;</div><div>=C2=A0 =C2=A0 =C2=A0 &lt;/t&gt;</div><div><br><=
/div><div>... similar for RTCP ...</div><div>=C2=A0 =C2=A0 &lt;/section&gt;=
</div></div><div><div><div class=3D"gmail_msg"><br class=3D"gmail_msg"><div=
 dir=3D"auto" class=3D"gmail_msg"><br class=3D"gmail_msg"></div><div dir=3D=
"auto" class=3D"gmail_msg"><br class=3D"gmail_msg"><div dir=3D"auto" class=
=3D"gmail_msg"><br class=3D"gmail_msg"><br class=3D"gmail_msg"><div class=
=3D"gmail_quote gmail_msg"><div dir=3D"ltr" class=3D"gmail_msg">On Thu, Dec=
 8, 2016, 2:55 AM Colin Perkins &lt;<a href=3D"mailto:csp@csperkins.org" cl=
ass=3D"gmail_msg" target=3D"_blank">csp@csperkins.org</a>&gt; wrote:<br cla=
ss=3D"gmail_msg"></div><blockquote class=3D"gmail_quote gmail_msg" style=3D=
"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi,<br clas=
s=3D"gmail_msg">
<br class=3D"gmail_msg">
[cc=E2=80=99ing RTCWEB and MMUSIC, since this relates to both JSEP and BUND=
LE drafts]<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
There=E2=80=99s been a lot of discussion on the lists, and at the meeting i=
n Seoul, around how RTP streams are mapped onto higher-level, application m=
eaningful, semantic roles. In particular, around how RTP streams map onto J=
SEP objects for WebRTC. Magnus Westerlund and I would like to propose the f=
ollowing updates to JSEP and BUNDLE to try to clarify the behaviour.<br cla=
ss=3D"gmail_msg">
<br class=3D"gmail_msg">
Comments and feedback very welcome.<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
Cheers,<br class=3D"gmail_msg">
Colin<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
# Replacement for Section 6 of draft-ietf-rtcweb-jsep-17<br class=3D"gmail_=
msg">
<br class=3D"gmail_msg">
=C2=A0&lt;section title=3D&quot;Processing RTP/RTCP&quot; anchor=3D&quot;se=
c.rtp.demux&quot;&gt;<br class=3D"gmail_msg">
=C2=A0 =C2=A0 &lt;t&gt;As described in &lt;xref target=3D&quot;RFC3550&quot=
;/&gt;, RTP packets are<br class=3D"gmail_msg">
=C2=A0 =C2=A0 associated with RTP streams &lt;xref target=3D&quot;RFC7656&q=
uot;/&gt;. Metadata<br class=3D"gmail_msg">
=C2=A0 =C2=A0 about those streams, including source description information=
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 and reception quality feedback is conveyed in RTCP packets.<b=
r class=3D"gmail_msg">
=C2=A0 =C2=A0 Each RTP stream is identified by an SSRC, and each RTP packet=
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 carries an SSRC value that is used to associate the packet wi=
th<br class=3D"gmail_msg">
=C2=A0 =C2=A0 the correct RTP stream. RTCP packets also use SSRCs to identi=
fy<br class=3D"gmail_msg">
=C2=A0 =C2=A0 the RTP streams that the reports and metadata relate to.=C2=
=A0 RTCP<br class=3D"gmail_msg">
=C2=A0 =C2=A0 packets generally carry multiple SSRC values and report on, o=
r<br class=3D"gmail_msg">
=C2=A0 =C2=A0 deliver source description information relating to, several R=
TP<br class=3D"gmail_msg">
=C2=A0 =C2=A0 streams.&lt;/t&gt;<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 &lt;t&gt;Each incoming RTP stream, identified by its SSRC, is=
 mapped to<br class=3D"gmail_msg">
=C2=A0 =C2=A0 an m=3D section in the SDP. The SDP m=3D sections then corres=
pond to<br class=3D"gmail_msg">
=C2=A0 =C2=A0 RtpReceiver objects. This allows each RTP stream to be associ=
ated<br class=3D"gmail_msg">
=C2=A0 =C2=A0 with an RtpTransceiver. Further processing of the RTP stream =
can<br class=3D"gmail_msg">
=C2=A0 =C2=A0 then be done at the RtpTransceiver level.=C2=A0 This includes=
 using<br class=3D"gmail_msg">
=C2=A0 =C2=A0 RID &lt;xref target=3D&quot;I-D.ietf-mmusic-rid&quot;/&gt; to=
 distinguish between<br class=3D"gmail_msg">
=C2=A0 =C2=A0 multiple Encoded Streams, as well as determine which Source R=
TP<br class=3D"gmail_msg">
=C2=A0 =C2=A0 stream should be repaired by a given Redundancy RTP stream.&l=
t;/t&gt;<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 &lt;t&gt;The process of mapping RTP streams onto m=3D section=
s depends on<br class=3D"gmail_msg">
=C2=A0 =C2=A0 whether streams are bundled or not. If the SDP BUNDLE extensi=
on<br class=3D"gmail_msg">
=C2=A0 =C2=A0 is in use, then RTP streams are mapped onto m=3D sections bas=
ed on<br class=3D"gmail_msg">
=C2=A0 =C2=A0 the MID values as described in<br class=3D"gmail_msg">
=C2=A0 =C2=A0 &lt;xref target=3D&quot;I-D.ietf-mmusic-sdp-bundle-negotiatio=
n&quot;/&gt;.=C2=A0 If the<br class=3D"gmail_msg">
=C2=A0 =C2=A0 SDP BUNDLE extension is not in use, each m=3D section corresp=
onds<br class=3D"gmail_msg">
=C2=A0 =C2=A0 to a transport layer connection and the RTP streams received =
on<br class=3D"gmail_msg">
=C2=A0 =C2=A0 that connection correspond to the m=3D section.&lt;/t&gt;<br =
class=3D"gmail_msg">
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 &lt;t&gt;Incoming RTCP packets contain metadata including rec=
eption<br class=3D"gmail_msg">
=C2=A0 =C2=A0 quality feedback, source description information, and other<b=
r class=3D"gmail_msg">
=C2=A0 =C2=A0 signalling relating to RTP streams. The RTCP packets are pars=
ed,<br class=3D"gmail_msg">
=C2=A0 =C2=A0 the associated RTP streams are identified based on the includ=
ed<br class=3D"gmail_msg">
=C2=A0 =C2=A0 SSRC values, and the metadata relating to those RTP streams i=
s<br class=3D"gmail_msg">
=C2=A0 =C2=A0 updated (this might include updating the MID information, use=
d<br class=3D"gmail_msg">
=C2=A0 =C2=A0 to associate RTP streams with m=3D sections, if the SDP BUNDL=
E<br class=3D"gmail_msg">
=C2=A0 =C2=A0 extension is in use). This updated metadata is available to t=
he<br class=3D"gmail_msg">
=C2=A0 =C2=A0 RtpTransceiver objects associated with those RTP streams.<br =
class=3D"gmail_msg">
=C2=A0 =C2=A0 &lt;/t&gt;<br class=3D"gmail_msg">
=C2=A0&lt;/section&gt;<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
# Replacement text for section 10.2 of draft-ietf-mmusic-sdp-bundle-negotia=
tion-36<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0&lt;section anchor=3D&quot;sec-rtp-pt&quot;<br class=3D=
"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 title=3D&quot;Associating =
RTP Streams With Correct SDP Media Description&quot;<br class=3D"gmail_msg"=
>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 toc=3D&quot;default&quot;&=
gt;<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;As described in &lt;xref format=3D&quot=
;default&quot; pageno=3D&quot;false&quot;<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0target=3D&quot;RFC3550&quot;/&gt;, RTP packets a=
re associated with RTP streams &lt;xref<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0format=3D&quot;default&quot; pageno=3D&quot;fals=
e&quot; target=3D&quot;RFC7656&quot;/&gt;. Each RTP stream is<br class=3D"g=
mail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0identified by an SSRC value, and each RTP packet=
 carries an SSRC value<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0that is used to associate the packet with the co=
rrect RTP stream. RTCP<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0packets also uses SSRCs to identify on which RTP=
 streams any report or<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0feedback relate to. Thus, an RTCP packet will co=
mmonly carry multiple<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0SSRC values, and might therefore be providing fe=
edback or report on<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0multiple RTP streams. &lt;/t&gt;<br class=3D"gma=
il_msg">
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;In order to be able to process received=
 RTP packets correctly it<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0must be possible to associate an RTP stream with=
 the correct &quot;m=3D&quot;<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0line, as the &quot;m=3D&quot; line and SDP attri=
butes associated with the &quot;m=3D&quot;<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0line contain information needed to process the p=
ackets.&lt;/t&gt;<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;As all RTP streams associated with a BU=
NDLE group are part of the<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0same RTP session and using the same address:port=
 combination for<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0sending and receiving RTP/RTCP packets, the loca=
l address:port<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0combination cannot be used to associate an RTP s=
tream with the correct<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&quot;m=3D&quot; line. In addition, multiple RTP=
 streams might be associated with<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0the same &quot;m=3D&quot; line.&lt;/t&gt;<br cla=
ss=3D"gmail_msg">
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;Also, as described in &lt;xref format=
=3D&quot;default&quot; pageno=3D&quot;false&quot;<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0target=3D&quot;sec-rtp-sessions-pt&quot;/&gt;, t=
he same payload type value might be<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0used by multiple RTP streams, in which case the =
payload type value<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0cannot be used to associate an RTP stream with t=
he correct &quot;m=3D&quot; line.<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0However, there are cases where each &quot;m=3D&q=
uot; line has unique payload type<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0values, and then the payload type could serve as=
 hint to the relevant<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 &quot=
;m=3D&quot; line the RTP stream is associated with.&lt;/t&gt;<br class=3D"g=
mail_msg">
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;An offerer and answerer can inform each=
 other which SSRC values<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0they will use for an RTP stream by using the SDP=
 &#39;ssrc&#39; attribute<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;xref format=3D&quot;default&quot; pageno=3D&=
quot;false&quot; target=3D&quot;RFC5576&quot;/&gt;. However, an<br class=3D=
"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0offerer will not know which SSRC values the answ=
erer will use until<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0the offerer has received the answer providing th=
at information. Due to<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0this, before the offerer has received the answer=
, the offerer will not<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0be able to associate an RTP stream with the corr=
ect &quot;m=3D&quot; line using<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0the SSRC value associated with the RTP stream. I=
n addition, the<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0offerer and answerer may start using new SSRC va=
lues mid-session,<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0without informing each other using the SDP &#39;=
ssrc&#39; attribute.&lt;/t&gt;<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;In order for an offerer and answerer to=
 always be able to associate<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0an RTP stream with the correct &quot;m=3D&quot; =
line, the offerer and answerer<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0using the BUNDLE extension MUST support the mech=
anism defined in &lt;xref<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0format=3D&quot;default&quot; pageno=3D&quot;fals=
e&quot; target=3D&quot;sec-receiver-id&quot;/&gt;, where the<br class=3D"gm=
ail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0offerer and answerer includes the identification=
-tag (provided by the<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0remote peer) associated with an &quot;m=3D&quot;=
 line in the RTP Streams and in<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0RTCP SDES packets part of a BUNDLE group.&lt;/t&=
gt;<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;The mapping from an SSRC to an identifi=
cation-tag is carried in<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0RTCP SDES packets or in RTP header extensions (&=
lt;xref format=3D&quot;default&quot;<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0pageno=3D&quot;false&quot; target=3D&quot;sec-re=
ceiver-id&quot;/&gt;). Since a compound RTCP<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0packet can contain multiple RTCP SDES packets, a=
nd each RTCP SDES<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0packet can contain multiple chunks, an RTCP pack=
et can contain several<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0SSRC to identification-tag mappings. The offerer=
 and answerer maintain<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0tables mapping RTP streams identified by SSRC to=
 &quot;m=3D&quot; lines identified<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0by the identification-tag.<br class=3D"gmail_msg=
">
=C2=A0 =C2=A0 =C2=A0 =C2=A0When receiving an RTP packet carrying a MID head=
er extension<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0with the identification-tag, or an RTCP packet c=
arrying one or<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0more SDES MID items, the offerer or answerer cre=
ates a mapping<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0table entry between the SSRC value and the ident=
ification-tag,<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0in order to associate the RTP stream associated =
with that SSRC<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0value with the &quot;m=3D&quot; line correspondi=
ng to the identification-tag.&lt;/t&gt;<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;The mapping between the SSRC an identif=
ication-tag might change<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0mid-session if, for a given SSRC value, a differ=
ent identification-tag<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0is provided in an RTP or RTCP packet. In that ca=
se these tables are<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0updated each time an RTP/RTCP packet containing =
a new mappings from<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0SSRC to identification-tag is received. Some con=
siderations for<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0avoiding update flaps are provided in Section 4.=
2.6 of &lt;xref<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0target=3D&quot;RFC7941&quot;/&gt; which should b=
e followed. &lt;/t&gt;<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;If an offerer and answerer is not able =
to associate an RTP stream<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0with an &quot;m=3D&quot; line (using the mechani=
sms described in this section, or<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0using other appropriate mechanism, e.g., based o=
n the payload type<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0value if it is unique to a single &quot;m=3D&quo=
t; line), it MUST either drop the<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0RTP packets associated with the RTP stream, or p=
rocess them in an<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0application specific manner, once non-stream spe=
cific processing<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0(e.g., related to congestion control) of the RTP=
 packets have<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0occurred.&lt;/t&gt;<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;When compound RTCP packets are received=
, they are split<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0into their component RTCP packets and those comp=
onent RTCP<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0packets are processed based on their RTCP packet=
 type, in<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0the order in which they were placed into the com=
pound RTCP<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0packet. Non-compound RTCP packets are processed =
based on<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0their RTCP packet type, in the order they are re=
ceived. Information<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0in each RTCP packet can relate to one or more RT=
P streams.<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0For example, RTCP Sender Report (SR) and Receive=
r Report (RR)<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0packets include an SSRC of sender field that ind=
icates the<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0identity of the participant that sent the RTCP p=
acket, along<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0with a list of Report Blocks. Each report contai=
ns data on the<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0reception quality of a single RTP stream, identi=
fied by SSRC,<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0as received by the SSRC that sent the RTCP packe=
t. Other RTCP<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0packet types similarly contain references to the=
 SSRC of the<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0sender of the RTCP packet, and the RTP streams t=
o which it<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0refers.&lt;/t&gt;<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;It should always be possible to process=
 RTCP packets, and<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0store the received information in a data structu=
re associated<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0with an RTP stream, identified by SSRC, for late=
r access and<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0use. It is possible that RTCP packets relating t=
o an SSRC can<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0be received before RTP packets relating to that =
SSRC, so the<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0data structures relating to an SSRC might need t=
o be created<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0before the corresponding RTP stream is received.=
&lt;/t&gt;<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;Similarly, information relating to an R=
TP stream might be<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0received before the data needed to map it onto a=
n m=3D line is<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0received. Information carried in RTCP packets re=
lating to such<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0an RTP stream that is application and/or &quot;m=
=3D&quot; line dependent<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0MAY be dropped until the SSRCs is associated wit=
h a particular<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&quot;m=3D&quot; line. However, information to g=
enerate RTCP report blocks<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0and other basic transport level feedback or repo=
rting needs to<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0be retained, so RTCP reports relating to the str=
eam can be<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0generated.&lt;/t&gt;<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0&lt;/section&gt;<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
--<br class=3D"gmail_msg">
Colin Perkins<br class=3D"gmail_msg">
<a href=3D"https://csperkins.org/" rel=3D"noreferrer" class=3D"gmail_msg" t=
arget=3D"_blank">https://csperkins.org/</a><br class=3D"gmail_msg">
<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
_______________________________________________<br class=3D"gmail_msg">
mmusic mailing list<br class=3D"gmail_msg">
<a href=3D"mailto:mmusic@ietf.org" class=3D"gmail_msg" target=3D"_blank">mm=
usic@ietf.org</a><br class=3D"gmail_msg">
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" rel=3D"noreferrer"=
 class=3D"gmail_msg" target=3D"_blank">https://www.ietf.org/mailman/listinf=
o/mmusic</a><br class=3D"gmail_msg">
</blockquote></div></div></div></div></div></div></div></div>

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From: Peter Thatcher <pthatcher@google.com>
Date: Fri, 06 Jan 2017 22:49:14 +0000
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Cc: Magnus Westerlund <magnus.westerlund@ericsson.com>, draft-ietf-rtcweb-jsep@tools.ietf.org, draft-ietf-mmusic-sdp-bundle-negotiation@ietf.org
Subject: Re: [MMUSIC] RTP demux in JSEP and BUNDLE
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I made a PR for JSEP that incorporates your text *and* adds a more specific
algorithm which only uses MID and SSRC latching (my interpretation of your
text, which I'm still not sure is correct):

https://github.com/rtcweb-wg/jsep/pull/489


In practice, I think this is mostly word smithing and doesn't require a
virtual interim.  The big questions, which perhaps to require a virtual
interim, are:

1.  Do we want to include PT demux in the algorithm (when neither SSRC nor
MID match)?
2.  Do we want to include signaled SSRCs in the demux algorithm (the SSRC
table is loaded with signaled SSRCs)?

The question to both of those is "yes" in PR 411 and "no" in PR 489.

On Fri, Jan 6, 2017 at 11:23 AM Peter Thatcher <pthatcher@google.com> wrote=
:

> When I read your text, I get the impression that you want to simplify the
> algorithm to only supporting MIDs for demux with SSRC latching.  No PT
> demux.   No SSRC demux (except for ones latched by MID).   In other words=
,
> if you want to use BUNDLE, you must use MID and only MID.
>
> But it's not clear if that's the case.  Perhaps I missed something.  I
> agree with others that a more explicit algorithm would make your intentio=
n
> clear.
>
> To help, here is my interpretation of your text as an algorithm.  Please
> let me know if it's a correct interpretation:
>
>     <section title=3D"Appendix B" anchor=3D"sec.appendix-b">
>       <t>To prepare for demultiplexing RTP packets to the correct "m=3D"
>       line, the following steps MUST be followed for each BUNDLE
>       group.</t>
>
>       <t>
>         <list>
>           <t>Construct a table mapping MID to "m=3D" line for each "m=3D"
>           line in this BUNDLE group.</t>
>
>           <t>Construct an empty table mapping incoming SSRC to "m=3D" lin=
e.
> </t>
>         </list>
>       </t>
>
>       <t>As "m=3D" lines are added or removed from the BUNDLE groups, or
>       their configurations are changed, the tables above must also be
>       updated.</t>
>
>       <t>For each RTP packet received, the following steps MUST be
>       followed to route the packet to the correct "m=3D" section within
>       a BUNDLE group:</t>
>
>       <t>
>         <list>
>           <t>If the packet has a MID and that MID is not in the table
>           mapping MID to "m=3D" line, drop the packet and stop.</t>
>
>           <t>If the packet has a MID and that MID is in the table
>           mapping MID to "m=3D" line, update the incoming SSRC mapping
>           table to include an entry that maps the packet's SSRC to the
>           "m=3D" line for that MID.</t>
>
>           <t>If the packet's SSRC is in the incoming SSRC mapping
>           table, route the packet to the associated "m=3D" line and
>           stop.</t>
>
>           <t>Otherwise, drop the packet.</t>
>         </list>
>       </t>
>
> ... similar for RTCP ...
>     </section>
>
>
>
>
>
> On Thu, Dec 8, 2016, 2:55 AM Colin Perkins <csp@csperkins.org> wrote:
>
> Hi,
>
> [cc=E2=80=99ing RTCWEB and MMUSIC, since this relates to both JSEP and BU=
NDLE
> drafts]
>
> There=E2=80=99s been a lot of discussion on the lists, and at the meeting=
 in
> Seoul, around how RTP streams are mapped onto higher-level, application
> meaningful, semantic roles. In particular, around how RTP streams map ont=
o
> JSEP objects for WebRTC. Magnus Westerlund and I would like to propose th=
e
> following updates to JSEP and BUNDLE to try to clarify the behaviour.
>
> Comments and feedback very welcome.
>
> Cheers,
> Colin
>
>
>
> # Replacement for Section 6 of draft-ietf-rtcweb-jsep-17
>
>  <section title=3D"Processing RTP/RTCP" anchor=3D"sec.rtp.demux">
>     <t>As described in <xref target=3D"RFC3550"/>, RTP packets are
>     associated with RTP streams <xref target=3D"RFC7656"/>. Metadata
>     about those streams, including source description information
>     and reception quality feedback is conveyed in RTCP packets.
>     Each RTP stream is identified by an SSRC, and each RTP packet
>     carries an SSRC value that is used to associate the packet with
>     the correct RTP stream. RTCP packets also use SSRCs to identify
>     the RTP streams that the reports and metadata relate to.  RTCP
>     packets generally carry multiple SSRC values and report on, or
>     deliver source description information relating to, several RTP
>     streams.</t>
>
>     <t>Each incoming RTP stream, identified by its SSRC, is mapped to
>     an m=3D section in the SDP. The SDP m=3D sections then correspond to
>     RtpReceiver objects. This allows each RTP stream to be associated
>     with an RtpTransceiver. Further processing of the RTP stream can
>     then be done at the RtpTransceiver level.  This includes using
>     RID <xref target=3D"I-D.ietf-mmusic-rid"/> to distinguish between
>     multiple Encoded Streams, as well as determine which Source RTP
>     stream should be repaired by a given Redundancy RTP stream.</t>
>
>     <t>The process of mapping RTP streams onto m=3D sections depends on
>     whether streams are bundled or not. If the SDP BUNDLE extension
>     is in use, then RTP streams are mapped onto m=3D sections based on
>     the MID values as described in
>     <xref target=3D"I-D.ietf-mmusic-sdp-bundle-negotiation"/>.  If the
>     SDP BUNDLE extension is not in use, each m=3D section corresponds
>     to a transport layer connection and the RTP streams received on
>     that connection correspond to the m=3D section.</t>
>
>     <t>Incoming RTCP packets contain metadata including reception
>     quality feedback, source description information, and other
>     signalling relating to RTP streams. The RTCP packets are parsed,
>     the associated RTP streams are identified based on the included
>     SSRC values, and the metadata relating to those RTP streams is
>     updated (this might include updating the MID information, used
>     to associate RTP streams with m=3D sections, if the SDP BUNDLE
>     extension is in use). This updated metadata is available to the
>     RtpTransceiver objects associated with those RTP streams.
>     </t>
>  </section>
>
>
>
> # Replacement text for section 10.2 of
> draft-ietf-mmusic-sdp-bundle-negotiation-36
>
>      <section anchor=3D"sec-rtp-pt"
>               title=3D"Associating RTP Streams With Correct SDP Media
> Description"
>               toc=3D"default">
>        <t>As described in <xref format=3D"default" pageno=3D"false"
>        target=3D"RFC3550"/>, RTP packets are associated with RTP streams
> <xref
>        format=3D"default" pageno=3D"false" target=3D"RFC7656"/>. Each RTP=
 stream
> is
>        identified by an SSRC value, and each RTP packet carries an SSRC
> value
>        that is used to associate the packet with the correct RTP stream.
> RTCP
>        packets also uses SSRCs to identify on which RTP streams any repor=
t
> or
>        feedback relate to. Thus, an RTCP packet will commonly carry
> multiple
>        SSRC values, and might therefore be providing feedback or report o=
n
>        multiple RTP streams. </t>
>
>        <t>In order to be able to process received RTP packets correctly i=
t
>        must be possible to associate an RTP stream with the correct "m=3D=
"
>        line, as the "m=3D" line and SDP attributes associated with the "m=
=3D"
>        line contain information needed to process the packets.</t>
>
>        <t>As all RTP streams associated with a BUNDLE group are part of t=
he
>        same RTP session and using the same address:port combination for
>        sending and receiving RTP/RTCP packets, the local address:port
>        combination cannot be used to associate an RTP stream with the
> correct
>        "m=3D" line. In addition, multiple RTP streams might be associated
> with
>        the same "m=3D" line.</t>
>
>        <t>Also, as described in <xref format=3D"default" pageno=3D"false"
>        target=3D"sec-rtp-sessions-pt"/>, the same payload type value migh=
t be
>        used by multiple RTP streams, in which case the payload type value
>        cannot be used to associate an RTP stream with the correct "m=3D"
> line.
>        However, there are cases where each "m=3D" line has unique payload
> type
>        values, and then the payload type could serve as hint to the
> relevant
>                     "m=3D" line the RTP stream is associated with.</t>
>
>        <t>An offerer and answerer can inform each other which SSRC values
>        they will use for an RTP stream by using the SDP 'ssrc' attribute
>        <xref format=3D"default" pageno=3D"false" target=3D"RFC5576"/>. Ho=
wever,
> an
>        offerer will not know which SSRC values the answerer will use unti=
l
>        the offerer has received the answer providing that information. Du=
e
> to
>        this, before the offerer has received the answer, the offerer will
> not
>        be able to associate an RTP stream with the correct "m=3D" line us=
ing
>        the SSRC value associated with the RTP stream. In addition, the
>        offerer and answerer may start using new SSRC values mid-session,
>        without informing each other using the SDP 'ssrc' attribute.</t>
>
>        <t>In order for an offerer and answerer to always be able to
> associate
>        an RTP stream with the correct "m=3D" line, the offerer and answer=
er
>        using the BUNDLE extension MUST support the mechanism defined in
> <xref
>        format=3D"default" pageno=3D"false" target=3D"sec-receiver-id"/>, =
where
> the
>        offerer and answerer includes the identification-tag (provided by
> the
>        remote peer) associated with an "m=3D" line in the RTP Streams and=
 in
>        RTCP SDES packets part of a BUNDLE group.</t>
>
>        <t>The mapping from an SSRC to an identification-tag is carried in
>        RTCP SDES packets or in RTP header extensions (<xref
> format=3D"default"
>        pageno=3D"false" target=3D"sec-receiver-id"/>). Since a compound R=
TCP
>        packet can contain multiple RTCP SDES packets, and each RTCP SDES
>        packet can contain multiple chunks, an RTCP packet can contain
> several
>        SSRC to identification-tag mappings. The offerer and answerer
> maintain
>        tables mapping RTP streams identified by SSRC to "m=3D" lines
> identified
>        by the identification-tag.
>        When receiving an RTP packet carrying a MID header extension
>        with the identification-tag, or an RTCP packet carrying one or
>        more SDES MID items, the offerer or answerer creates a mapping
>        table entry between the SSRC value and the identification-tag,
>        in order to associate the RTP stream associated with that SSRC
>        value with the "m=3D" line corresponding to the
> identification-tag.</t>
>
>        <t>The mapping between the SSRC an identification-tag might change
>        mid-session if, for a given SSRC value, a different
> identification-tag
>        is provided in an RTP or RTCP packet. In that case these tables ar=
e
>        updated each time an RTP/RTCP packet containing a new mappings fro=
m
>        SSRC to identification-tag is received. Some considerations for
>        avoiding update flaps are provided in Section 4.2.6 of <xref
>        target=3D"RFC7941"/> which should be followed. </t>
>
>        <t>If an offerer and answerer is not able to associate an RTP stre=
am
>        with an "m=3D" line (using the mechanisms described in this sectio=
n,
> or
>        using other appropriate mechanism, e.g., based on the payload type
>        value if it is unique to a single "m=3D" line), it MUST either dro=
p
> the
>        RTP packets associated with the RTP stream, or process them in an
>        application specific manner, once non-stream specific processing
>        (e.g., related to congestion control) of the RTP packets have
>        occurred.</t>
>
>        <t>When compound RTCP packets are received, they are split
>        into their component RTCP packets and those component RTCP
>        packets are processed based on their RTCP packet type, in
>        the order in which they were placed into the compound RTCP
>        packet. Non-compound RTCP packets are processed based on
>        their RTCP packet type, in the order they are received. Informatio=
n
>        in each RTCP packet can relate to one or more RTP streams.
>        For example, RTCP Sender Report (SR) and Receiver Report (RR)
>        packets include an SSRC of sender field that indicates the
>        identity of the participant that sent the RTCP packet, along
>        with a list of Report Blocks. Each report contains data on the
>        reception quality of a single RTP stream, identified by SSRC,
>        as received by the SSRC that sent the RTCP packet. Other RTCP
>        packet types similarly contain references to the SSRC of the
>        sender of the RTCP packet, and the RTP streams to which it
>        refers.</t>
>
>        <t>It should always be possible to process RTCP packets, and
>        store the received information in a data structure associated
>        with an RTP stream, identified by SSRC, for later access and
>        use. It is possible that RTCP packets relating to an SSRC can
>        be received before RTP packets relating to that SSRC, so the
>        data structures relating to an SSRC might need to be created
>        before the corresponding RTP stream is received.</t>
>
>        <t>Similarly, information relating to an RTP stream might be
>        received before the data needed to map it onto an m=3D line is
>        received. Information carried in RTCP packets relating to such
>        an RTP stream that is application and/or "m=3D" line dependent
>        MAY be dropped until the SSRCs is associated with a particular
>        "m=3D" line. However, information to generate RTCP report blocks
>        and other basic transport level feedback or reporting needs to
>        be retained, so RTCP reports relating to the stream can be
>        generated.</t>
>
>      </section>
>
>
>
>
> --
> Colin Perkins
> https://csperkins.org/
>
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>
>

--001a113f26fc621bbc054574d43b
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">I made a PR for JSEP that incorporates your text *and* add=
s a more specific algorithm which only uses MID and SSRC latching (my inter=
pretation of your text, which I&#39;m still not sure is correct):<div><br><=
/div><div><a href=3D"https://github.com/rtcweb-wg/jsep/pull/489">https://gi=
thub.com/rtcweb-wg/jsep/pull/489</a><br></div><div><br></div><div><br></div=
><div>In practice, I think this is mostly word smithing and doesn&#39;t req=
uire a virtual interim.=C2=A0 The big questions, which perhaps to require a=
 virtual interim, are:</div><div><br></div><div>1.=C2=A0 Do we want to incl=
ude PT demux in the algorithm (when neither SSRC nor MID match)? =C2=A0</di=
v><div>2.=C2=A0 Do we want to include signaled SSRCs in the demux algorithm=
 (the SSRC table is loaded with signaled SSRCs)?</div><div><br></div><div>T=
he question to both of those is &quot;yes&quot; in PR 411 and &quot;no&quot=
; in PR 489.</div></div><br><div class=3D"gmail_quote"><div dir=3D"ltr">On =
Fri, Jan 6, 2017 at 11:23 AM Peter Thatcher &lt;<a href=3D"mailto:pthatcher=
@google.com">pthatcher@google.com</a>&gt; wrote:<br></div><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex"><div dir=3D"ltr">When I read your text, I get the impression =
that you want to simplify the algorithm to only supporting MIDs for demux w=
ith SSRC latching.=C2=A0 No PT demux. =C2=A0 No SSRC demux (except for ones=
 latched by MID). =C2=A0 In other words, if you want to use BUNDLE, you mus=
t use MID and only MID.<div><br></div><div>But it&#39;s not clear if that&#=
39;s the case.=C2=A0 Perhaps I missed something.=C2=A0 I agree with others =
that a more explicit algorithm would make your intention clear.<div><br></d=
iv><div>To help, here is my interpretation of your text as an algorithm.=C2=
=A0 Please let me know if it&#39;s a correct interpretation:</div><div><br>=
</div><div><div>=C2=A0 =C2=A0 &lt;section title=3D&quot;Appendix B&quot; an=
chor=3D&quot;sec.appendix-b&quot;&gt;</div><div>=C2=A0 =C2=A0 =C2=A0 &lt;t&=
gt;To prepare for demultiplexing RTP packets to the correct &quot;m=3D&quot=
;</div><div>=C2=A0 =C2=A0 =C2=A0 line, the following steps MUST be followed=
 for each BUNDLE</div><div>=C2=A0 =C2=A0 =C2=A0 group.&lt;/t&gt;</div><div>=
<br></div><div>=C2=A0 =C2=A0 =C2=A0 &lt;t&gt;</div><div>=C2=A0 =C2=A0 =C2=
=A0 =C2=A0 &lt;list&gt;</div><div>=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 &lt;t&=
gt;Construct a table mapping MID to &quot;m=3D&quot; line for each &quot;m=
=3D&quot;</div><div>=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 line in this BUNDLE =
group.&lt;/t&gt;</div><div><br></div><div>=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 &lt;t&gt;Construct an empty table mapping incoming SSRC to &quot;m=3D&q=
uot; line. &lt;/t&gt;</div><div>=C2=A0 =C2=A0 =C2=A0 =C2=A0 &lt;/list&gt;</=
div><div>=C2=A0 =C2=A0 =C2=A0 &lt;/t&gt;</div><div><br></div><div>=C2=A0 =
=C2=A0 =C2=A0 &lt;t&gt;As &quot;m=3D&quot; lines are added or removed from =
the BUNDLE groups, or</div><div>=C2=A0 =C2=A0 =C2=A0 their configurations a=
re changed, the tables above must also be</div><div>=C2=A0 =C2=A0 =C2=A0 up=
dated.&lt;/t&gt;</div><div><br></div><div>=C2=A0 =C2=A0 =C2=A0 &lt;t&gt;For=
 each RTP packet received, the following steps MUST be</div><div>=C2=A0 =C2=
=A0 =C2=A0 followed to route the packet to the correct &quot;m=3D&quot; sec=
tion within</div><div>=C2=A0 =C2=A0 =C2=A0 a BUNDLE group:&lt;/t&gt;</div><=
div><br></div><div>=C2=A0 =C2=A0 =C2=A0 &lt;t&gt;</div><div>=C2=A0 =C2=A0 =
=C2=A0 =C2=A0 &lt;list&gt;</div><div>=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 &lt=
;t&gt;If the packet has a MID and that MID is not in the table</div><div>=
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 mapping MID to &quot;m=3D&quot; line, dr=
op the packet and stop.&lt;/t&gt;</div><div><br></div><div>=C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 &lt;t&gt;If the packet has a MID and that MID is in th=
e table</div><div>=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 mapping MID to &quot;m=
=3D&quot; line, update the incoming SSRC mapping</div><div>=C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 table to include an entry that maps the packet&#39;s S=
SRC to the</div><div>=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 &quot;m=3D&quot; li=
ne for that MID.&lt;/t&gt;</div><div><br></div><div>=C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 &lt;t&gt;If the packet&#39;s SSRC is in the incoming SSRC map=
ping</div><div>=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 table, route the packet t=
o the associated &quot;m=3D&quot; line and</div><div>=C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 stop.&lt;/t&gt;</div><div><br></div><div>=C2=A0 =C2=A0 =C2=A0=
 =C2=A0 =C2=A0 &lt;t&gt;Otherwise, drop the packet.&lt;/t&gt;</div><div>=C2=
=A0 =C2=A0 =C2=A0 =C2=A0 &lt;/list&gt;</div><div>=C2=A0 =C2=A0 =C2=A0 &lt;/=
t&gt;</div><div><br></div><div>... similar for RTCP ...</div><div>=C2=A0 =
=C2=A0 &lt;/section&gt;</div></div><div><div><div class=3D"gmail_msg"><br c=
lass=3D"gmail_msg"><div dir=3D"auto" class=3D"gmail_msg"><br class=3D"gmail=
_msg"></div><div dir=3D"auto" class=3D"gmail_msg"><br class=3D"gmail_msg"><=
div dir=3D"auto" class=3D"gmail_msg"><br class=3D"gmail_msg"><br class=3D"g=
mail_msg"><div class=3D"gmail_quote gmail_msg"><div dir=3D"ltr" class=3D"gm=
ail_msg">On Thu, Dec 8, 2016, 2:55 AM Colin Perkins &lt;<a href=3D"mailto:c=
sp@csperkins.org" class=3D"gmail_msg" target=3D"_blank">csp@csperkins.org</=
a>&gt; wrote:<br class=3D"gmail_msg"></div><blockquote class=3D"gmail_quote=
 gmail_msg" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-l=
eft:1ex">Hi,<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
[cc=E2=80=99ing RTCWEB and MMUSIC, since this relates to both JSEP and BUND=
LE drafts]<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
There=E2=80=99s been a lot of discussion on the lists, and at the meeting i=
n Seoul, around how RTP streams are mapped onto higher-level, application m=
eaningful, semantic roles. In particular, around how RTP streams map onto J=
SEP objects for WebRTC. Magnus Westerlund and I would like to propose the f=
ollowing updates to JSEP and BUNDLE to try to clarify the behaviour.<br cla=
ss=3D"gmail_msg">
<br class=3D"gmail_msg">
Comments and feedback very welcome.<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
Cheers,<br class=3D"gmail_msg">
Colin<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
# Replacement for Section 6 of draft-ietf-rtcweb-jsep-17<br class=3D"gmail_=
msg">
<br class=3D"gmail_msg">
=C2=A0&lt;section title=3D&quot;Processing RTP/RTCP&quot; anchor=3D&quot;se=
c.rtp.demux&quot;&gt;<br class=3D"gmail_msg">
=C2=A0 =C2=A0 &lt;t&gt;As described in &lt;xref target=3D&quot;RFC3550&quot=
;/&gt;, RTP packets are<br class=3D"gmail_msg">
=C2=A0 =C2=A0 associated with RTP streams &lt;xref target=3D&quot;RFC7656&q=
uot;/&gt;. Metadata<br class=3D"gmail_msg">
=C2=A0 =C2=A0 about those streams, including source description information=
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 and reception quality feedback is conveyed in RTCP packets.<b=
r class=3D"gmail_msg">
=C2=A0 =C2=A0 Each RTP stream is identified by an SSRC, and each RTP packet=
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 carries an SSRC value that is used to associate the packet wi=
th<br class=3D"gmail_msg">
=C2=A0 =C2=A0 the correct RTP stream. RTCP packets also use SSRCs to identi=
fy<br class=3D"gmail_msg">
=C2=A0 =C2=A0 the RTP streams that the reports and metadata relate to.=C2=
=A0 RTCP<br class=3D"gmail_msg">
=C2=A0 =C2=A0 packets generally carry multiple SSRC values and report on, o=
r<br class=3D"gmail_msg">
=C2=A0 =C2=A0 deliver source description information relating to, several R=
TP<br class=3D"gmail_msg">
=C2=A0 =C2=A0 streams.&lt;/t&gt;<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 &lt;t&gt;Each incoming RTP stream, identified by its SSRC, is=
 mapped to<br class=3D"gmail_msg">
=C2=A0 =C2=A0 an m=3D section in the SDP. The SDP m=3D sections then corres=
pond to<br class=3D"gmail_msg">
=C2=A0 =C2=A0 RtpReceiver objects. This allows each RTP stream to be associ=
ated<br class=3D"gmail_msg">
=C2=A0 =C2=A0 with an RtpTransceiver. Further processing of the RTP stream =
can<br class=3D"gmail_msg">
=C2=A0 =C2=A0 then be done at the RtpTransceiver level.=C2=A0 This includes=
 using<br class=3D"gmail_msg">
=C2=A0 =C2=A0 RID &lt;xref target=3D&quot;I-D.ietf-mmusic-rid&quot;/&gt; to=
 distinguish between<br class=3D"gmail_msg">
=C2=A0 =C2=A0 multiple Encoded Streams, as well as determine which Source R=
TP<br class=3D"gmail_msg">
=C2=A0 =C2=A0 stream should be repaired by a given Redundancy RTP stream.&l=
t;/t&gt;<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 &lt;t&gt;The process of mapping RTP streams onto m=3D section=
s depends on<br class=3D"gmail_msg">
=C2=A0 =C2=A0 whether streams are bundled or not. If the SDP BUNDLE extensi=
on<br class=3D"gmail_msg">
=C2=A0 =C2=A0 is in use, then RTP streams are mapped onto m=3D sections bas=
ed on<br class=3D"gmail_msg">
=C2=A0 =C2=A0 the MID values as described in<br class=3D"gmail_msg">
=C2=A0 =C2=A0 &lt;xref target=3D&quot;I-D.ietf-mmusic-sdp-bundle-negotiatio=
n&quot;/&gt;.=C2=A0 If the<br class=3D"gmail_msg">
=C2=A0 =C2=A0 SDP BUNDLE extension is not in use, each m=3D section corresp=
onds<br class=3D"gmail_msg">
=C2=A0 =C2=A0 to a transport layer connection and the RTP streams received =
on<br class=3D"gmail_msg">
=C2=A0 =C2=A0 that connection correspond to the m=3D section.&lt;/t&gt;<br =
class=3D"gmail_msg">
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 &lt;t&gt;Incoming RTCP packets contain metadata including rec=
eption<br class=3D"gmail_msg">
=C2=A0 =C2=A0 quality feedback, source description information, and other<b=
r class=3D"gmail_msg">
=C2=A0 =C2=A0 signalling relating to RTP streams. The RTCP packets are pars=
ed,<br class=3D"gmail_msg">
=C2=A0 =C2=A0 the associated RTP streams are identified based on the includ=
ed<br class=3D"gmail_msg">
=C2=A0 =C2=A0 SSRC values, and the metadata relating to those RTP streams i=
s<br class=3D"gmail_msg">
=C2=A0 =C2=A0 updated (this might include updating the MID information, use=
d<br class=3D"gmail_msg">
=C2=A0 =C2=A0 to associate RTP streams with m=3D sections, if the SDP BUNDL=
E<br class=3D"gmail_msg">
=C2=A0 =C2=A0 extension is in use). This updated metadata is available to t=
he<br class=3D"gmail_msg">
=C2=A0 =C2=A0 RtpTransceiver objects associated with those RTP streams.<br =
class=3D"gmail_msg">
=C2=A0 =C2=A0 &lt;/t&gt;<br class=3D"gmail_msg">
=C2=A0&lt;/section&gt;<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
# Replacement text for section 10.2 of draft-ietf-mmusic-sdp-bundle-negotia=
tion-36<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0&lt;section anchor=3D&quot;sec-rtp-pt&quot;<br class=3D=
"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 title=3D&quot;Associating =
RTP Streams With Correct SDP Media Description&quot;<br class=3D"gmail_msg"=
>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 toc=3D&quot;default&quot;&=
gt;<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;As described in &lt;xref format=3D&quot=
;default&quot; pageno=3D&quot;false&quot;<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0target=3D&quot;RFC3550&quot;/&gt;, RTP packets a=
re associated with RTP streams &lt;xref<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0format=3D&quot;default&quot; pageno=3D&quot;fals=
e&quot; target=3D&quot;RFC7656&quot;/&gt;. Each RTP stream is<br class=3D"g=
mail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0identified by an SSRC value, and each RTP packet=
 carries an SSRC value<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0that is used to associate the packet with the co=
rrect RTP stream. RTCP<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0packets also uses SSRCs to identify on which RTP=
 streams any report or<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0feedback relate to. Thus, an RTCP packet will co=
mmonly carry multiple<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0SSRC values, and might therefore be providing fe=
edback or report on<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0multiple RTP streams. &lt;/t&gt;<br class=3D"gma=
il_msg">
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;In order to be able to process received=
 RTP packets correctly it<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0must be possible to associate an RTP stream with=
 the correct &quot;m=3D&quot;<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0line, as the &quot;m=3D&quot; line and SDP attri=
butes associated with the &quot;m=3D&quot;<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0line contain information needed to process the p=
ackets.&lt;/t&gt;<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;As all RTP streams associated with a BU=
NDLE group are part of the<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0same RTP session and using the same address:port=
 combination for<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0sending and receiving RTP/RTCP packets, the loca=
l address:port<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0combination cannot be used to associate an RTP s=
tream with the correct<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&quot;m=3D&quot; line. In addition, multiple RTP=
 streams might be associated with<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0the same &quot;m=3D&quot; line.&lt;/t&gt;<br cla=
ss=3D"gmail_msg">
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;Also, as described in &lt;xref format=
=3D&quot;default&quot; pageno=3D&quot;false&quot;<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0target=3D&quot;sec-rtp-sessions-pt&quot;/&gt;, t=
he same payload type value might be<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0used by multiple RTP streams, in which case the =
payload type value<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0cannot be used to associate an RTP stream with t=
he correct &quot;m=3D&quot; line.<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0However, there are cases where each &quot;m=3D&q=
uot; line has unique payload type<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0values, and then the payload type could serve as=
 hint to the relevant<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 &quot=
;m=3D&quot; line the RTP stream is associated with.&lt;/t&gt;<br class=3D"g=
mail_msg">
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;An offerer and answerer can inform each=
 other which SSRC values<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0they will use for an RTP stream by using the SDP=
 &#39;ssrc&#39; attribute<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;xref format=3D&quot;default&quot; pageno=3D&=
quot;false&quot; target=3D&quot;RFC5576&quot;/&gt;. However, an<br class=3D=
"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0offerer will not know which SSRC values the answ=
erer will use until<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0the offerer has received the answer providing th=
at information. Due to<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0this, before the offerer has received the answer=
, the offerer will not<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0be able to associate an RTP stream with the corr=
ect &quot;m=3D&quot; line using<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0the SSRC value associated with the RTP stream. I=
n addition, the<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0offerer and answerer may start using new SSRC va=
lues mid-session,<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0without informing each other using the SDP &#39;=
ssrc&#39; attribute.&lt;/t&gt;<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;In order for an offerer and answerer to=
 always be able to associate<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0an RTP stream with the correct &quot;m=3D&quot; =
line, the offerer and answerer<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0using the BUNDLE extension MUST support the mech=
anism defined in &lt;xref<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0format=3D&quot;default&quot; pageno=3D&quot;fals=
e&quot; target=3D&quot;sec-receiver-id&quot;/&gt;, where the<br class=3D"gm=
ail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0offerer and answerer includes the identification=
-tag (provided by the<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0remote peer) associated with an &quot;m=3D&quot;=
 line in the RTP Streams and in<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0RTCP SDES packets part of a BUNDLE group.&lt;/t&=
gt;<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;The mapping from an SSRC to an identifi=
cation-tag is carried in<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0RTCP SDES packets or in RTP header extensions (&=
lt;xref format=3D&quot;default&quot;<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0pageno=3D&quot;false&quot; target=3D&quot;sec-re=
ceiver-id&quot;/&gt;). Since a compound RTCP<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0packet can contain multiple RTCP SDES packets, a=
nd each RTCP SDES<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0packet can contain multiple chunks, an RTCP pack=
et can contain several<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0SSRC to identification-tag mappings. The offerer=
 and answerer maintain<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0tables mapping RTP streams identified by SSRC to=
 &quot;m=3D&quot; lines identified<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0by the identification-tag.<br class=3D"gmail_msg=
">
=C2=A0 =C2=A0 =C2=A0 =C2=A0When receiving an RTP packet carrying a MID head=
er extension<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0with the identification-tag, or an RTCP packet c=
arrying one or<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0more SDES MID items, the offerer or answerer cre=
ates a mapping<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0table entry between the SSRC value and the ident=
ification-tag,<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0in order to associate the RTP stream associated =
with that SSRC<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0value with the &quot;m=3D&quot; line correspondi=
ng to the identification-tag.&lt;/t&gt;<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;The mapping between the SSRC an identif=
ication-tag might change<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0mid-session if, for a given SSRC value, a differ=
ent identification-tag<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0is provided in an RTP or RTCP packet. In that ca=
se these tables are<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0updated each time an RTP/RTCP packet containing =
a new mappings from<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0SSRC to identification-tag is received. Some con=
siderations for<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0avoiding update flaps are provided in Section 4.=
2.6 of &lt;xref<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0target=3D&quot;RFC7941&quot;/&gt; which should b=
e followed. &lt;/t&gt;<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;If an offerer and answerer is not able =
to associate an RTP stream<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0with an &quot;m=3D&quot; line (using the mechani=
sms described in this section, or<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0using other appropriate mechanism, e.g., based o=
n the payload type<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0value if it is unique to a single &quot;m=3D&quo=
t; line), it MUST either drop the<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0RTP packets associated with the RTP stream, or p=
rocess them in an<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0application specific manner, once non-stream spe=
cific processing<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0(e.g., related to congestion control) of the RTP=
 packets have<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0occurred.&lt;/t&gt;<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;When compound RTCP packets are received=
, they are split<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0into their component RTCP packets and those comp=
onent RTCP<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0packets are processed based on their RTCP packet=
 type, in<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0the order in which they were placed into the com=
pound RTCP<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0packet. Non-compound RTCP packets are processed =
based on<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0their RTCP packet type, in the order they are re=
ceived. Information<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0in each RTCP packet can relate to one or more RT=
P streams.<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0For example, RTCP Sender Report (SR) and Receive=
r Report (RR)<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0packets include an SSRC of sender field that ind=
icates the<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0identity of the participant that sent the RTCP p=
acket, along<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0with a list of Report Blocks. Each report contai=
ns data on the<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0reception quality of a single RTP stream, identi=
fied by SSRC,<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0as received by the SSRC that sent the RTCP packe=
t. Other RTCP<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0packet types similarly contain references to the=
 SSRC of the<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0sender of the RTCP packet, and the RTP streams t=
o which it<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0refers.&lt;/t&gt;<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;It should always be possible to process=
 RTCP packets, and<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0store the received information in a data structu=
re associated<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0with an RTP stream, identified by SSRC, for late=
r access and<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0use. It is possible that RTCP packets relating t=
o an SSRC can<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0be received before RTP packets relating to that =
SSRC, so the<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0data structures relating to an SSRC might need t=
o be created<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0before the corresponding RTP stream is received.=
&lt;/t&gt;<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;Similarly, information relating to an R=
TP stream might be<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0received before the data needed to map it onto a=
n m=3D line is<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0received. Information carried in RTCP packets re=
lating to such<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0an RTP stream that is application and/or &quot;m=
=3D&quot; line dependent<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0MAY be dropped until the SSRCs is associated wit=
h a particular<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&quot;m=3D&quot; line. However, information to g=
enerate RTCP report blocks<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0and other basic transport level feedback or repo=
rting needs to<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0be retained, so RTCP reports relating to the str=
eam can be<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0generated.&lt;/t&gt;<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
=C2=A0 =C2=A0 =C2=A0&lt;/section&gt;<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
--<br class=3D"gmail_msg">
Colin Perkins<br class=3D"gmail_msg">
<a href=3D"https://csperkins.org/" rel=3D"noreferrer" class=3D"gmail_msg" t=
arget=3D"_blank">https://csperkins.org/</a><br class=3D"gmail_msg">
<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
<br class=3D"gmail_msg">
_______________________________________________<br class=3D"gmail_msg">
mmusic mailing list<br class=3D"gmail_msg">
<a href=3D"mailto:mmusic@ietf.org" class=3D"gmail_msg" target=3D"_blank">mm=
usic@ietf.org</a><br class=3D"gmail_msg">
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" rel=3D"noreferrer"=
 class=3D"gmail_msg" target=3D"_blank">https://www.ietf.org/mailman/listinf=
o/mmusic</a><br class=3D"gmail_msg">
</blockquote></div></div></div></div></div></div></div></div></blockquote><=
/div>

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Date: Fri, 6 Jan 2017 16:48:02 -0800
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To: Peter Thatcher <pthatcher@google.com>
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Cc: Magnus Westerlund <magnus.westerlund@ericsson.com>, "mmusic \(E-mail\)" <mmusic@ietf.org>, draft-ietf-rtcweb-jsep@tools.ietf.org, "rtcweb@ietf.org" <rtcweb@ietf.org>, Colin Perkins <csp@csperkins.org>, draft-ietf-mmusic-sdp-bundle-negotiation@ietf.org
Subject: Re: [MMUSIC] RTP demux in JSEP and BUNDLE
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Peter said:

"
In practice, I think this is mostly word smithing and doesn't require a
virtual interim.  The big questions, which perhaps to require a virtual
interim, are:

1.  Do we want to include PT demux in the algorithm (when neither SSRC nor
MID match)?
2.  Do we want to include signaled SSRCs in the demux algorithm (the SSRC
table is loaded with signaled SSRCs)?
"

[BA]  I believe that we need to cover both #1 and #2.

1. There have been bugs filed against implementations that relate to PT
demux (e.g. SSRC change scenario).  So IMHO covering PT demux is
necessary.  JSEP-17 was pretty close, but tere are a few issues that still
need to be discussed:
     a.  SSRC latching on a PT match.  If the PT can change without SSRC
changing (e.g. in a video scenario where all codecs have the same
clockrate), it seems that latching could result in mis-routing.
     b. Whether the PT_table is filled if SSRCs are signaled in codecs
and/or RTX/FEC.  Implementations are doing different things here, and that
is going to cause problems.

2. Including signaled SSRCs in the demux algorithm is a practical
requirement because SSRC signaling is widely implemented and MID isn't
yet.  Also, when discussing RTCP segment routing, doesn't the SSRC_table
come into play (e.g. in deciding which receiver to route a Sender Report
to)?

On Fri, Jan 6, 2017 at 2:49 PM, Peter Thatcher <pthatcher@google.com> wrote=
:

> I made a PR for JSEP that incorporates your text *and* adds a more
> specific algorithm which only uses MID and SSRC latching (my interpretati=
on
> of your text, which I'm still not sure is correct):
>
> https://github.com/rtcweb-wg/jsep/pull/489
>
>
> In practice, I think this is mostly word smithing and doesn't require a
> virtual interim.  The big questions, which perhaps to require a virtual
> interim, are:
>
> 1.  Do we want to include PT demux in the algorithm (when neither SSRC no=
r
> MID match)?
> 2.  Do we want to include signaled SSRCs in the demux algorithm (the SSRC
> table is loaded with signaled SSRCs)?
>
> The question to both of those is "yes" in PR 411 and "no" in PR 489.
>
> On Fri, Jan 6, 2017 at 11:23 AM Peter Thatcher <pthatcher@google.com>
> wrote:
>
>> When I read your text, I get the impression that you want to simplify th=
e
>> algorithm to only supporting MIDs for demux with SSRC latching.  No PT
>> demux.   No SSRC demux (except for ones latched by MID).   In other word=
s,
>> if you want to use BUNDLE, you must use MID and only MID.
>>
>> But it's not clear if that's the case.  Perhaps I missed something.  I
>> agree with others that a more explicit algorithm would make your intenti=
on
>> clear.
>>
>> To help, here is my interpretation of your text as an algorithm.  Please
>> let me know if it's a correct interpretation:
>>
>>     <section title=3D"Appendix B" anchor=3D"sec.appendix-b">
>>       <t>To prepare for demultiplexing RTP packets to the correct "m=3D"
>>       line, the following steps MUST be followed for each BUNDLE
>>       group.</t>
>>
>>       <t>
>>         <list>
>>           <t>Construct a table mapping MID to "m=3D" line for each "m=3D=
"
>>           line in this BUNDLE group.</t>
>>
>>           <t>Construct an empty table mapping incoming SSRC to "m=3D" li=
ne.
>> </t>
>>         </list>
>>       </t>
>>
>>       <t>As "m=3D" lines are added or removed from the BUNDLE groups, or
>>       their configurations are changed, the tables above must also be
>>       updated.</t>
>>
>>       <t>For each RTP packet received, the following steps MUST be
>>       followed to route the packet to the correct "m=3D" section within
>>       a BUNDLE group:</t>
>>
>>       <t>
>>         <list>
>>           <t>If the packet has a MID and that MID is not in the table
>>           mapping MID to "m=3D" line, drop the packet and stop.</t>
>>
>>           <t>If the packet has a MID and that MID is in the table
>>           mapping MID to "m=3D" line, update the incoming SSRC mapping
>>           table to include an entry that maps the packet's SSRC to the
>>           "m=3D" line for that MID.</t>
>>
>>           <t>If the packet's SSRC is in the incoming SSRC mapping
>>           table, route the packet to the associated "m=3D" line and
>>           stop.</t>
>>
>>           <t>Otherwise, drop the packet.</t>
>>         </list>
>>       </t>
>>
>> ... similar for RTCP ...
>>     </section>
>>
>>
>>
>>
>>
>> On Thu, Dec 8, 2016, 2:55 AM Colin Perkins <csp@csperkins.org> wrote:
>>
>> Hi,
>>
>> [cc=E2=80=99ing RTCWEB and MMUSIC, since this relates to both JSEP and B=
UNDLE
>> drafts]
>>
>> There=E2=80=99s been a lot of discussion on the lists, and at the meetin=
g in
>> Seoul, around how RTP streams are mapped onto higher-level, application
>> meaningful, semantic roles. In particular, around how RTP streams map on=
to
>> JSEP objects for WebRTC. Magnus Westerlund and I would like to propose t=
he
>> following updates to JSEP and BUNDLE to try to clarify the behaviour.
>>
>> Comments and feedback very welcome.
>>
>> Cheers,
>> Colin
>>
>>
>>
>> # Replacement for Section 6 of draft-ietf-rtcweb-jsep-17
>>
>>  <section title=3D"Processing RTP/RTCP" anchor=3D"sec.rtp.demux">
>>     <t>As described in <xref target=3D"RFC3550"/>, RTP packets are
>>     associated with RTP streams <xref target=3D"RFC7656"/>. Metadata
>>     about those streams, including source description information
>>     and reception quality feedback is conveyed in RTCP packets.
>>     Each RTP stream is identified by an SSRC, and each RTP packet
>>     carries an SSRC value that is used to associate the packet with
>>     the correct RTP stream. RTCP packets also use SSRCs to identify
>>     the RTP streams that the reports and metadata relate to.  RTCP
>>     packets generally carry multiple SSRC values and report on, or
>>     deliver source description information relating to, several RTP
>>     streams.</t>
>>
>>     <t>Each incoming RTP stream, identified by its SSRC, is mapped to
>>     an m=3D section in the SDP. The SDP m=3D sections then correspond to
>>     RtpReceiver objects. This allows each RTP stream to be associated
>>     with an RtpTransceiver. Further processing of the RTP stream can
>>     then be done at the RtpTransceiver level.  This includes using
>>     RID <xref target=3D"I-D.ietf-mmusic-rid"/> to distinguish between
>>     multiple Encoded Streams, as well as determine which Source RTP
>>     stream should be repaired by a given Redundancy RTP stream.</t>
>>
>>     <t>The process of mapping RTP streams onto m=3D sections depends on
>>     whether streams are bundled or not. If the SDP BUNDLE extension
>>     is in use, then RTP streams are mapped onto m=3D sections based on
>>     the MID values as described in
>>     <xref target=3D"I-D.ietf-mmusic-sdp-bundle-negotiation"/>.  If the
>>     SDP BUNDLE extension is not in use, each m=3D section corresponds
>>     to a transport layer connection and the RTP streams received on
>>     that connection correspond to the m=3D section.</t>
>>
>>     <t>Incoming RTCP packets contain metadata including reception
>>     quality feedback, source description information, and other
>>     signalling relating to RTP streams. The RTCP packets are parsed,
>>     the associated RTP streams are identified based on the included
>>     SSRC values, and the metadata relating to those RTP streams is
>>     updated (this might include updating the MID information, used
>>     to associate RTP streams with m=3D sections, if the SDP BUNDLE
>>     extension is in use). This updated metadata is available to the
>>     RtpTransceiver objects associated with those RTP streams.
>>     </t>
>>  </section>
>>
>>
>>
>> # Replacement text for section 10.2 of draft-ietf-mmusic-sdp-bundle-
>> negotiation-36
>>
>>      <section anchor=3D"sec-rtp-pt"
>>               title=3D"Associating RTP Streams With Correct SDP Media
>> Description"
>>               toc=3D"default">
>>        <t>As described in <xref format=3D"default" pageno=3D"false"
>>        target=3D"RFC3550"/>, RTP packets are associated with RTP streams
>> <xref
>>        format=3D"default" pageno=3D"false" target=3D"RFC7656"/>. Each RT=
P
>> stream is
>>        identified by an SSRC value, and each RTP packet carries an SSRC
>> value
>>        that is used to associate the packet with the correct RTP stream.
>> RTCP
>>        packets also uses SSRCs to identify on which RTP streams any
>> report or
>>        feedback relate to. Thus, an RTCP packet will commonly carry
>> multiple
>>        SSRC values, and might therefore be providing feedback or report =
on
>>        multiple RTP streams. </t>
>>
>>        <t>In order to be able to process received RTP packets correctly =
it
>>        must be possible to associate an RTP stream with the correct "m=
=3D"
>>        line, as the "m=3D" line and SDP attributes associated with the "=
m=3D"
>>        line contain information needed to process the packets.</t>
>>
>>        <t>As all RTP streams associated with a BUNDLE group are part of
>> the
>>        same RTP session and using the same address:port combination for
>>        sending and receiving RTP/RTCP packets, the local address:port
>>        combination cannot be used to associate an RTP stream with the
>> correct
>>        "m=3D" line. In addition, multiple RTP streams might be associate=
d
>> with
>>        the same "m=3D" line.</t>
>>
>>        <t>Also, as described in <xref format=3D"default" pageno=3D"false=
"
>>        target=3D"sec-rtp-sessions-pt"/>, the same payload type value mig=
ht
>> be
>>        used by multiple RTP streams, in which case the payload type valu=
e
>>        cannot be used to associate an RTP stream with the correct "m=3D"
>> line.
>>        However, there are cases where each "m=3D" line has unique payloa=
d
>> type
>>        values, and then the payload type could serve as hint to the
>> relevant
>>                     "m=3D" line the RTP stream is associated with.</t>
>>
>>        <t>An offerer and answerer can inform each other which SSRC value=
s
>>        they will use for an RTP stream by using the SDP 'ssrc' attribute
>>        <xref format=3D"default" pageno=3D"false" target=3D"RFC5576"/>. H=
owever,
>> an
>>        offerer will not know which SSRC values the answerer will use unt=
il
>>        the offerer has received the answer providing that information.
>> Due to
>>        this, before the offerer has received the answer, the offerer wil=
l
>> not
>>        be able to associate an RTP stream with the correct "m=3D" line u=
sing
>>        the SSRC value associated with the RTP stream. In addition, the
>>        offerer and answerer may start using new SSRC values mid-session,
>>        without informing each other using the SDP 'ssrc' attribute.</t>
>>
>>        <t>In order for an offerer and answerer to always be able to
>> associate
>>        an RTP stream with the correct "m=3D" line, the offerer and answe=
rer
>>        using the BUNDLE extension MUST support the mechanism defined in
>> <xref
>>        format=3D"default" pageno=3D"false" target=3D"sec-receiver-id"/>,=
 where
>> the
>>        offerer and answerer includes the identification-tag (provided by
>> the
>>        remote peer) associated with an "m=3D" line in the RTP Streams an=
d in
>>        RTCP SDES packets part of a BUNDLE group.</t>
>>
>>        <t>The mapping from an SSRC to an identification-tag is carried i=
n
>>        RTCP SDES packets or in RTP header extensions (<xref
>> format=3D"default"
>>        pageno=3D"false" target=3D"sec-receiver-id"/>). Since a compound =
RTCP
>>        packet can contain multiple RTCP SDES packets, and each RTCP SDES
>>        packet can contain multiple chunks, an RTCP packet can contain
>> several
>>        SSRC to identification-tag mappings. The offerer and answerer
>> maintain
>>        tables mapping RTP streams identified by SSRC to "m=3D" lines
>> identified
>>        by the identification-tag.
>>        When receiving an RTP packet carrying a MID header extension
>>        with the identification-tag, or an RTCP packet carrying one or
>>        more SDES MID items, the offerer or answerer creates a mapping
>>        table entry between the SSRC value and the identification-tag,
>>        in order to associate the RTP stream associated with that SSRC
>>        value with the "m=3D" line corresponding to the
>> identification-tag.</t>
>>
>>        <t>The mapping between the SSRC an identification-tag might chang=
e
>>        mid-session if, for a given SSRC value, a different
>> identification-tag
>>        is provided in an RTP or RTCP packet. In that case these tables a=
re
>>        updated each time an RTP/RTCP packet containing a new mappings fr=
om
>>        SSRC to identification-tag is received. Some considerations for
>>        avoiding update flaps are provided in Section 4.2.6 of <xref
>>        target=3D"RFC7941"/> which should be followed. </t>
>>
>>        <t>If an offerer and answerer is not able to associate an RTP
>> stream
>>        with an "m=3D" line (using the mechanisms described in this secti=
on,
>> or
>>        using other appropriate mechanism, e.g., based on the payload typ=
e
>>        value if it is unique to a single "m=3D" line), it MUST either dr=
op
>> the
>>        RTP packets associated with the RTP stream, or process them in an
>>        application specific manner, once non-stream specific processing
>>        (e.g., related to congestion control) of the RTP packets have
>>        occurred.</t>
>>
>>        <t>When compound RTCP packets are received, they are split
>>        into their component RTCP packets and those component RTCP
>>        packets are processed based on their RTCP packet type, in
>>        the order in which they were placed into the compound RTCP
>>        packet. Non-compound RTCP packets are processed based on
>>        their RTCP packet type, in the order they are received. Informati=
on
>>        in each RTCP packet can relate to one or more RTP streams.
>>        For example, RTCP Sender Report (SR) and Receiver Report (RR)
>>        packets include an SSRC of sender field that indicates the
>>        identity of the participant that sent the RTCP packet, along
>>        with a list of Report Blocks. Each report contains data on the
>>        reception quality of a single RTP stream, identified by SSRC,
>>        as received by the SSRC that sent the RTCP packet. Other RTCP
>>        packet types similarly contain references to the SSRC of the
>>        sender of the RTCP packet, and the RTP streams to which it
>>        refers.</t>
>>
>>        <t>It should always be possible to process RTCP packets, and
>>        store the received information in a data structure associated
>>        with an RTP stream, identified by SSRC, for later access and
>>        use. It is possible that RTCP packets relating to an SSRC can
>>        be received before RTP packets relating to that SSRC, so the
>>        data structures relating to an SSRC might need to be created
>>        before the corresponding RTP stream is received.</t>
>>
>>        <t>Similarly, information relating to an RTP stream might be
>>        received before the data needed to map it onto an m=3D line is
>>        received. Information carried in RTCP packets relating to such
>>        an RTP stream that is application and/or "m=3D" line dependent
>>        MAY be dropped until the SSRCs is associated with a particular
>>        "m=3D" line. However, information to generate RTCP report blocks
>>        and other basic transport level feedback or reporting needs to
>>        be retained, so RTCP reports relating to the stream can be
>>        generated.</t>
>>
>>      </section>
>>
>>
>>
>>
>> --
>> Colin Perkins
>> https://csperkins.org/
>>
>>
>>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>>
>>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>
>

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

<div dir=3D"ltr">Peter said:=C2=A0<div><br></div><div>&quot;</div><div styl=
e=3D"font-size:12.8px">In practice, I think this is mostly word smithing an=
d doesn&#39;t require a virtual interim.=C2=A0 The big questions, which per=
haps to require a virtual interim, are:</div><div style=3D"font-size:12.8px=
"><br></div><div style=3D"font-size:12.8px">1.=C2=A0 Do we want to include =
PT demux in the algorithm (when neither SSRC nor MID match)? =C2=A0</div><d=
iv style=3D"font-size:12.8px">2.=C2=A0 Do we want to include signaled SSRCs=
 in the demux algorithm (the SSRC table is loaded with signaled SSRCs)?</di=
v><div>&quot;</div><div><br></div><div>[BA] =C2=A0I believe that we need to=
 cover both #1 and #2.=C2=A0</div><div><br></div><div>1. There have been bu=
gs filed against implementations that relate to PT demux (e.g. SSRC change =
scenario).=C2=A0 So IMHO covering PT demux is necessary.=C2=A0 JSEP-17 was =
pretty close, but tere are a few issues that still need to be discussed:</d=
iv><div>=C2=A0 =C2=A0 =C2=A0a.=C2=A0 SSRC latching on a PT match.=C2=A0 If =
the PT can change without SSRC changing (e.g. in a video scenario where all=
 codecs have the same clockrate), it seems that latching could result in mi=
s-routing.=C2=A0</div><div>=C2=A0 =C2=A0 =C2=A0b. Whether the PT_table is f=
illed if SSRCs are signaled in codecs and/or RTX/FEC.=C2=A0 Implementations=
 are doing different things here, and that is going to cause problems. =C2=
=A0</div><div><br></div><div>2. Including signaled SSRCs in the demux algor=
ithm is a practical requirement because SSRC signaling is widely implemente=
d and MID isn&#39;t yet.=C2=A0 Also, when discussing RTCP segment routing, =
doesn&#39;t the SSRC_table come into play (e.g. in deciding which receiver =
to route a Sender Report to)?=C2=A0</div></div><div class=3D"gmail_extra"><=
br><div class=3D"gmail_quote">On Fri, Jan 6, 2017 at 2:49 PM, Peter Thatche=
r <span dir=3D"ltr">&lt;<a href=3D"mailto:pthatcher@google.com" target=3D"_=
blank">pthatcher@google.com</a>&gt;</span> wrote:<br><blockquote class=3D"g=
mail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-l=
eft:1ex"><div dir=3D"ltr">I made a PR for JSEP that incorporates your text =
*and* adds a more specific algorithm which only uses MID and SSRC latching =
(my interpretation of your text, which I&#39;m still not sure is correct):<=
div><br></div><div><a href=3D"https://github.com/rtcweb-wg/jsep/pull/489" t=
arget=3D"_blank">https://github.com/rtcweb-wg/<wbr>jsep/pull/489</a><br></d=
iv><div><br></div><div><br></div><div>In practice, I think this is mostly w=
ord smithing and doesn&#39;t require a virtual interim.=C2=A0 The big quest=
ions, which perhaps to require a virtual interim, are:</div><div><br></div>=
<div>1.=C2=A0 Do we want to include PT demux in the algorithm (when neither=
 SSRC nor MID match)? =C2=A0</div><div>2.=C2=A0 Do we want to include signa=
led SSRCs in the demux algorithm (the SSRC table is loaded with signaled SS=
RCs)?</div><div><br></div><div>The question to both of those is &quot;yes&q=
uot; in PR 411 and &quot;no&quot; in PR 489.</div></div><div class=3D"HOEnZ=
b"><div class=3D"h5"><br><div class=3D"gmail_quote"><div dir=3D"ltr">On Fri=
, Jan 6, 2017 at 11:23 AM Peter Thatcher &lt;<a href=3D"mailto:pthatcher@go=
ogle.com" target=3D"_blank">pthatcher@google.com</a>&gt; wrote:<br></div><b=
lockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px =
#ccc solid;padding-left:1ex"><div dir=3D"ltr">When I read your text, I get =
the impression that you want to simplify the algorithm to only supporting M=
IDs for demux with SSRC latching.=C2=A0 No PT demux. =C2=A0 No SSRC demux (=
except for ones latched by MID). =C2=A0 In other words, if you want to use =
BUNDLE, you must use MID and only MID.<div><br></div><div>But it&#39;s not =
clear if that&#39;s the case.=C2=A0 Perhaps I missed something.=C2=A0 I agr=
ee with others that a more explicit algorithm would make your intention cle=
ar.<div><br></div><div>To help, here is my interpretation of your text as a=
n algorithm.=C2=A0 Please let me know if it&#39;s a correct interpretation:=
</div><div><br></div><div><div>=C2=A0 =C2=A0 &lt;section title=3D&quot;Appe=
ndix B&quot; anchor=3D&quot;sec.appendix-b&quot;&gt;</div><div>=C2=A0 =C2=
=A0 =C2=A0 &lt;t&gt;To prepare for demultiplexing RTP packets to the correc=
t &quot;m=3D&quot;</div><div>=C2=A0 =C2=A0 =C2=A0 line, the following steps=
 MUST be followed for each BUNDLE</div><div>=C2=A0 =C2=A0 =C2=A0 group.&lt;=
/t&gt;</div><div><br></div><div>=C2=A0 =C2=A0 =C2=A0 &lt;t&gt;</div><div>=
=C2=A0 =C2=A0 =C2=A0 =C2=A0 &lt;list&gt;</div><div>=C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 &lt;t&gt;Construct a table mapping MID to &quot;m=3D&quot; line =
for each &quot;m=3D&quot;</div><div>=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 line=
 in this BUNDLE group.&lt;/t&gt;</div><div><br></div><div>=C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 &lt;t&gt;Construct an empty table mapping incoming SSRC t=
o &quot;m=3D&quot; line. &lt;/t&gt;</div><div>=C2=A0 =C2=A0 =C2=A0 =C2=A0 &=
lt;/list&gt;</div><div>=C2=A0 =C2=A0 =C2=A0 &lt;/t&gt;</div><div><br></div>=
<div>=C2=A0 =C2=A0 =C2=A0 &lt;t&gt;As &quot;m=3D&quot; lines are added or r=
emoved from the BUNDLE groups, or</div><div>=C2=A0 =C2=A0 =C2=A0 their conf=
igurations are changed, the tables above must also be</div><div>=C2=A0 =C2=
=A0 =C2=A0 updated.&lt;/t&gt;</div><div><br></div><div>=C2=A0 =C2=A0 =C2=A0=
 &lt;t&gt;For each RTP packet received, the following steps MUST be</div><d=
iv>=C2=A0 =C2=A0 =C2=A0 followed to route the packet to the correct &quot;m=
=3D&quot; section within</div><div>=C2=A0 =C2=A0 =C2=A0 a BUNDLE group:&lt;=
/t&gt;</div><div><br></div><div>=C2=A0 =C2=A0 =C2=A0 &lt;t&gt;</div><div>=
=C2=A0 =C2=A0 =C2=A0 =C2=A0 &lt;list&gt;</div><div>=C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 &lt;t&gt;If the packet has a MID and that MID is not in the tabl=
e</div><div>=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 mapping MID to &quot;m=3D&qu=
ot; line, drop the packet and stop.&lt;/t&gt;</div><div><br></div><div>=C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 &lt;t&gt;If the packet has a MID and that M=
ID is in the table</div><div>=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 mapping MID=
 to &quot;m=3D&quot; line, update the incoming SSRC mapping</div><div>=C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 table to include an entry that maps the pac=
ket&#39;s SSRC to the</div><div>=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 &quot;m=
=3D&quot; line for that MID.&lt;/t&gt;</div><div><br></div><div>=C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 &lt;t&gt;If the packet&#39;s SSRC is in the incomi=
ng SSRC mapping</div><div>=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 table, route t=
he packet to the associated &quot;m=3D&quot; line and</div><div>=C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 stop.&lt;/t&gt;</div><div><br></div><div>=C2=A0 =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 &lt;t&gt;Otherwise, drop the packet.&lt;/t&gt;<=
/div><div>=C2=A0 =C2=A0 =C2=A0 =C2=A0 &lt;/list&gt;</div><div>=C2=A0 =C2=A0=
 =C2=A0 &lt;/t&gt;</div><div><br></div><div>... similar for RTCP ...</div><=
div>=C2=A0 =C2=A0 &lt;/section&gt;</div></div><div><div><div class=3D"m_-85=
52972565696268422gmail_msg"><br class=3D"m_-8552972565696268422gmail_msg"><=
div dir=3D"auto" class=3D"m_-8552972565696268422gmail_msg"><br class=3D"m_-=
8552972565696268422gmail_msg"></div><div dir=3D"auto" class=3D"m_-855297256=
5696268422gmail_msg"><br class=3D"m_-8552972565696268422gmail_msg"><div dir=
=3D"auto" class=3D"m_-8552972565696268422gmail_msg"><br class=3D"m_-8552972=
565696268422gmail_msg"><br class=3D"m_-8552972565696268422gmail_msg"><div c=
lass=3D"gmail_quote m_-8552972565696268422gmail_msg"><div dir=3D"ltr" class=
=3D"m_-8552972565696268422gmail_msg">On Thu, Dec 8, 2016, 2:55 AM Colin Per=
kins &lt;<a href=3D"mailto:csp@csperkins.org" class=3D"m_-85529725656962684=
22gmail_msg" target=3D"_blank">csp@csperkins.org</a>&gt; wrote:<br class=3D=
"m_-8552972565696268422gmail_msg"></div><blockquote class=3D"gmail_quote m_=
-8552972565696268422gmail_msg" style=3D"margin:0 0 0 .8ex;border-left:1px #=
ccc solid;padding-left:1ex">Hi,<br class=3D"m_-8552972565696268422gmail_msg=
">
<br class=3D"m_-8552972565696268422gmail_msg">
[cc=E2=80=99ing RTCWEB and MMUSIC, since this relates to both JSEP and BUND=
LE drafts]<br class=3D"m_-8552972565696268422gmail_msg">
<br class=3D"m_-8552972565696268422gmail_msg">
There=E2=80=99s been a lot of discussion on the lists, and at the meeting i=
n Seoul, around how RTP streams are mapped onto higher-level, application m=
eaningful, semantic roles. In particular, around how RTP streams map onto J=
SEP objects for WebRTC. Magnus Westerlund and I would like to propose the f=
ollowing updates to JSEP and BUNDLE to try to clarify the behaviour.<br cla=
ss=3D"m_-8552972565696268422gmail_msg">
<br class=3D"m_-8552972565696268422gmail_msg">
Comments and feedback very welcome.<br class=3D"m_-8552972565696268422gmail=
_msg">
<br class=3D"m_-8552972565696268422gmail_msg">
Cheers,<br class=3D"m_-8552972565696268422gmail_msg">
Colin<br class=3D"m_-8552972565696268422gmail_msg">
<br class=3D"m_-8552972565696268422gmail_msg">
<br class=3D"m_-8552972565696268422gmail_msg">
<br class=3D"m_-8552972565696268422gmail_msg">
# Replacement for Section 6 of draft-ietf-rtcweb-jsep-17<br class=3D"m_-855=
2972565696268422gmail_msg">
<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0&lt;section title=3D&quot;Processing RTP/RTCP&quot; anchor=3D&quot;se=
c.rtp.demux&quot;&gt;<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 &lt;t&gt;As described in &lt;xref target=3D&quot;RFC3550&quot=
;/&gt;, RTP packets are<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 associated with RTP streams &lt;xref target=3D&quot;RFC7656&q=
uot;/&gt;. Metadata<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 about those streams, including source description information=
<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 and reception quality feedback is conveyed in RTCP packets.<b=
r class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 Each RTP stream is identified by an SSRC, and each RTP packet=
<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 carries an SSRC value that is used to associate the packet wi=
th<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 the correct RTP stream. RTCP packets also use SSRCs to identi=
fy<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 the RTP streams that the reports and metadata relate to.=C2=
=A0 RTCP<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 packets generally carry multiple SSRC values and report on, o=
r<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 deliver source description information relating to, several R=
TP<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 streams.&lt;/t&gt;<br class=3D"m_-8552972565696268422gmail_ms=
g">
<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 &lt;t&gt;Each incoming RTP stream, identified by its SSRC, is=
 mapped to<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 an m=3D section in the SDP. The SDP m=3D sections then corres=
pond to<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 RtpReceiver objects. This allows each RTP stream to be associ=
ated<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 with an RtpTransceiver. Further processing of the RTP stream =
can<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 then be done at the RtpTransceiver level.=C2=A0 This includes=
 using<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 RID &lt;xref target=3D&quot;I-D.ietf-mmusic-rid&quot;/&gt; to=
 distinguish between<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 multiple Encoded Streams, as well as determine which Source R=
TP<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 stream should be repaired by a given Redundancy RTP stream.&l=
t;/t&gt;<br class=3D"m_-8552972565696268422gmail_msg">
<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 &lt;t&gt;The process of mapping RTP streams onto m=3D section=
s depends on<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 whether streams are bundled or not. If the SDP BUNDLE extensi=
on<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 is in use, then RTP streams are mapped onto m=3D sections bas=
ed on<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 the MID values as described in<br class=3D"m_-855297256569626=
8422gmail_msg">
=C2=A0 =C2=A0 &lt;xref target=3D&quot;I-D.ietf-mmusic-sdp-<wbr>bundle-negot=
iation&quot;/&gt;.=C2=A0 If the<br class=3D"m_-8552972565696268422gmail_msg=
">
=C2=A0 =C2=A0 SDP BUNDLE extension is not in use, each m=3D section corresp=
onds<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 to a transport layer connection and the RTP streams received =
on<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 that connection correspond to the m=3D section.&lt;/t&gt;<br =
class=3D"m_-8552972565696268422gmail_msg">
<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 &lt;t&gt;Incoming RTCP packets contain metadata including rec=
eption<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 quality feedback, source description information, and other<b=
r class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 signalling relating to RTP streams. The RTCP packets are pars=
ed,<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 the associated RTP streams are identified based on the includ=
ed<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 SSRC values, and the metadata relating to those RTP streams i=
s<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 updated (this might include updating the MID information, use=
d<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 to associate RTP streams with m=3D sections, if the SDP BUNDL=
E<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 extension is in use). This updated metadata is available to t=
he<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 RtpTransceiver objects associated with those RTP streams.<br =
class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 &lt;/t&gt;<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0&lt;/section&gt;<br class=3D"m_-8552972565696268422gmail_msg">
<br class=3D"m_-8552972565696268422gmail_msg">
<br class=3D"m_-8552972565696268422gmail_msg">
<br class=3D"m_-8552972565696268422gmail_msg">
# Replacement text for section 10.2 of draft-ietf-mmusic-sdp-bundle-<wbr>ne=
gotiation-36<br class=3D"m_-8552972565696268422gmail_msg">
<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0&lt;section anchor=3D&quot;sec-rtp-pt&quot;<br class=3D=
"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 title=3D&quot;Associating =
RTP Streams With Correct SDP Media Description&quot;<br class=3D"m_-8552972=
565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 toc=3D&quot;default&quot;&=
gt;<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;As described in &lt;xref format=3D&quot=
;default&quot; pageno=3D&quot;false&quot;<br class=3D"m_-855297256569626842=
2gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0target=3D&quot;RFC3550&quot;/&gt;, RTP packets a=
re associated with RTP streams &lt;xref<br class=3D"m_-8552972565696268422g=
mail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0format=3D&quot;default&quot; pageno=3D&quot;fals=
e&quot; target=3D&quot;RFC7656&quot;/&gt;. Each RTP stream is<br class=3D"m=
_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0identified by an SSRC value, and each RTP packet=
 carries an SSRC value<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0that is used to associate the packet with the co=
rrect RTP stream. RTCP<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0packets also uses SSRCs to identify on which RTP=
 streams any report or<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0feedback relate to. Thus, an RTCP packet will co=
mmonly carry multiple<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0SSRC values, and might therefore be providing fe=
edback or report on<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0multiple RTP streams. &lt;/t&gt;<br class=3D"m_-=
8552972565696268422gmail_msg">
<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;In order to be able to process received=
 RTP packets correctly it<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0must be possible to associate an RTP stream with=
 the correct &quot;m=3D&quot;<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0line, as the &quot;m=3D&quot; line and SDP attri=
butes associated with the &quot;m=3D&quot;<br class=3D"m_-85529725656962684=
22gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0line contain information needed to process the p=
ackets.&lt;/t&gt;<br class=3D"m_-8552972565696268422gmail_msg">
<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;As all RTP streams associated with a BU=
NDLE group are part of the<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0same RTP session and using the same address:port=
 combination for<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0sending and receiving RTP/RTCP packets, the loca=
l address:port<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0combination cannot be used to associate an RTP s=
tream with the correct<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&quot;m=3D&quot; line. In addition, multiple RTP=
 streams might be associated with<br class=3D"m_-8552972565696268422gmail_m=
sg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0the same &quot;m=3D&quot; line.&lt;/t&gt;<br cla=
ss=3D"m_-8552972565696268422gmail_msg">
<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;Also, as described in &lt;xref format=
=3D&quot;default&quot; pageno=3D&quot;false&quot;<br class=3D"m_-8552972565=
696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0target=3D&quot;sec-rtp-sessions-pt&quot;/<wbr>&g=
t;, the same payload type value might be<br class=3D"m_-8552972565696268422=
gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0used by multiple RTP streams, in which case the =
payload type value<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0cannot be used to associate an RTP stream with t=
he correct &quot;m=3D&quot; line.<br class=3D"m_-8552972565696268422gmail_m=
sg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0However, there are cases where each &quot;m=3D&q=
uot; line has unique payload type<br class=3D"m_-8552972565696268422gmail_m=
sg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0values, and then the payload type could serve as=
 hint to the relevant<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 &quot=
;m=3D&quot; line the RTP stream is associated with.&lt;/t&gt;<br class=3D"m=
_-8552972565696268422gmail_msg">
<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;An offerer and answerer can inform each=
 other which SSRC values<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0they will use for an RTP stream by using the SDP=
 &#39;ssrc&#39; attribute<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;xref format=3D&quot;default&quot; pageno=3D&=
quot;false&quot; target=3D&quot;RFC5576&quot;/&gt;. However, an<br class=3D=
"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0offerer will not know which SSRC values the answ=
erer will use until<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0the offerer has received the answer providing th=
at information. Due to<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0this, before the offerer has received the answer=
, the offerer will not<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0be able to associate an RTP stream with the corr=
ect &quot;m=3D&quot; line using<br class=3D"m_-8552972565696268422gmail_msg=
">
=C2=A0 =C2=A0 =C2=A0 =C2=A0the SSRC value associated with the RTP stream. I=
n addition, the<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0offerer and answerer may start using new SSRC va=
lues mid-session,<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0without informing each other using the SDP &#39;=
ssrc&#39; attribute.&lt;/t&gt;<br class=3D"m_-8552972565696268422gmail_msg"=
>
<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;In order for an offerer and answerer to=
 always be able to associate<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0an RTP stream with the correct &quot;m=3D&quot; =
line, the offerer and answerer<br class=3D"m_-8552972565696268422gmail_msg"=
>
=C2=A0 =C2=A0 =C2=A0 =C2=A0using the BUNDLE extension MUST support the mech=
anism defined in &lt;xref<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0format=3D&quot;default&quot; pageno=3D&quot;fals=
e&quot; target=3D&quot;sec-receiver-id&quot;/&gt;, where the<br class=3D"m_=
-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0offerer and answerer includes the identification=
-tag (provided by the<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0remote peer) associated with an &quot;m=3D&quot;=
 line in the RTP Streams and in<br class=3D"m_-8552972565696268422gmail_msg=
">
=C2=A0 =C2=A0 =C2=A0 =C2=A0RTCP SDES packets part of a BUNDLE group.&lt;/t&=
gt;<br class=3D"m_-8552972565696268422gmail_msg">
<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;The mapping from an SSRC to an identifi=
cation-tag is carried in<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0RTCP SDES packets or in RTP header extensions (&=
lt;xref format=3D&quot;default&quot;<br class=3D"m_-8552972565696268422gmai=
l_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0pageno=3D&quot;false&quot; target=3D&quot;sec-re=
ceiver-id&quot;/&gt;). Since a compound RTCP<br class=3D"m_-855297256569626=
8422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0packet can contain multiple RTCP SDES packets, a=
nd each RTCP SDES<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0packet can contain multiple chunks, an RTCP pack=
et can contain several<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0SSRC to identification-tag mappings. The offerer=
 and answerer maintain<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0tables mapping RTP streams identified by SSRC to=
 &quot;m=3D&quot; lines identified<br class=3D"m_-8552972565696268422gmail_=
msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0by the identification-tag.<br class=3D"m_-855297=
2565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0When receiving an RTP packet carrying a MID head=
er extension<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0with the identification-tag, or an RTCP packet c=
arrying one or<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0more SDES MID items, the offerer or answerer cre=
ates a mapping<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0table entry between the SSRC value and the ident=
ification-tag,<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0in order to associate the RTP stream associated =
with that SSRC<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0value with the &quot;m=3D&quot; line correspondi=
ng to the identification-tag.&lt;/t&gt;<br class=3D"m_-8552972565696268422g=
mail_msg">
<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;The mapping between the SSRC an identif=
ication-tag might change<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0mid-session if, for a given SSRC value, a differ=
ent identification-tag<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0is provided in an RTP or RTCP packet. In that ca=
se these tables are<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0updated each time an RTP/RTCP packet containing =
a new mappings from<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0SSRC to identification-tag is received. Some con=
siderations for<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0avoiding update flaps are provided in Section 4.=
2.6 of &lt;xref<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0target=3D&quot;RFC7941&quot;/&gt; which should b=
e followed. &lt;/t&gt;<br class=3D"m_-8552972565696268422gmail_msg">
<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;If an offerer and answerer is not able =
to associate an RTP stream<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0with an &quot;m=3D&quot; line (using the mechani=
sms described in this section, or<br class=3D"m_-8552972565696268422gmail_m=
sg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0using other appropriate mechanism, e.g., based o=
n the payload type<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0value if it is unique to a single &quot;m=3D&quo=
t; line), it MUST either drop the<br class=3D"m_-8552972565696268422gmail_m=
sg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0RTP packets associated with the RTP stream, or p=
rocess them in an<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0application specific manner, once non-stream spe=
cific processing<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0(e.g., related to congestion control) of the RTP=
 packets have<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0occurred.&lt;/t&gt;<br class=3D"m_-8552972565696=
268422gmail_msg">
<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;When compound RTCP packets are received=
, they are split<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0into their component RTCP packets and those comp=
onent RTCP<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0packets are processed based on their RTCP packet=
 type, in<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0the order in which they were placed into the com=
pound RTCP<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0packet. Non-compound RTCP packets are processed =
based on<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0their RTCP packet type, in the order they are re=
ceived. Information<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0in each RTCP packet can relate to one or more RT=
P streams.<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0For example, RTCP Sender Report (SR) and Receive=
r Report (RR)<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0packets include an SSRC of sender field that ind=
icates the<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0identity of the participant that sent the RTCP p=
acket, along<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0with a list of Report Blocks. Each report contai=
ns data on the<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0reception quality of a single RTP stream, identi=
fied by SSRC,<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0as received by the SSRC that sent the RTCP packe=
t. Other RTCP<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0packet types similarly contain references to the=
 SSRC of the<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0sender of the RTCP packet, and the RTP streams t=
o which it<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0refers.&lt;/t&gt;<br class=3D"m_-855297256569626=
8422gmail_msg">
<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;It should always be possible to process=
 RTCP packets, and<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0store the received information in a data structu=
re associated<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0with an RTP stream, identified by SSRC, for late=
r access and<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0use. It is possible that RTCP packets relating t=
o an SSRC can<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0be received before RTP packets relating to that =
SSRC, so the<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0data structures relating to an SSRC might need t=
o be created<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0before the corresponding RTP stream is received.=
&lt;/t&gt;<br class=3D"m_-8552972565696268422gmail_msg">
<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;t&gt;Similarly, information relating to an R=
TP stream might be<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0received before the data needed to map it onto a=
n m=3D line is<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0received. Information carried in RTCP packets re=
lating to such<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0an RTP stream that is application and/or &quot;m=
=3D&quot; line dependent<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0MAY be dropped until the SSRCs is associated wit=
h a particular<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0&quot;m=3D&quot; line. However, information to g=
enerate RTCP report blocks<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0and other basic transport level feedback or repo=
rting needs to<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0be retained, so RTCP reports relating to the str=
eam can be<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0 =C2=A0generated.&lt;/t&gt;<br class=3D"m_-855297256569=
6268422gmail_msg">
<br class=3D"m_-8552972565696268422gmail_msg">
=C2=A0 =C2=A0 =C2=A0&lt;/section&gt;<br class=3D"m_-8552972565696268422gmai=
l_msg">
<br class=3D"m_-8552972565696268422gmail_msg">
<br class=3D"m_-8552972565696268422gmail_msg">
<br class=3D"m_-8552972565696268422gmail_msg">
<br class=3D"m_-8552972565696268422gmail_msg">
--<br class=3D"m_-8552972565696268422gmail_msg">
Colin Perkins<br class=3D"m_-8552972565696268422gmail_msg">
<a href=3D"https://csperkins.org/" rel=3D"noreferrer" class=3D"m_-855297256=
5696268422gmail_msg" target=3D"_blank">https://csperkins.org/</a><br class=
=3D"m_-8552972565696268422gmail_msg">
<br class=3D"m_-8552972565696268422gmail_msg">
<br class=3D"m_-8552972565696268422gmail_msg">
<br class=3D"m_-8552972565696268422gmail_msg">
<br class=3D"m_-8552972565696268422gmail_msg">
______________________________<wbr>_________________<br class=3D"m_-8552972=
565696268422gmail_msg">
mmusic mailing list<br class=3D"m_-8552972565696268422gmail_msg">
<a href=3D"mailto:mmusic@ietf.org" class=3D"m_-8552972565696268422gmail_msg=
" target=3D"_blank">mmusic@ietf.org</a><br class=3D"m_-8552972565696268422g=
mail_msg">
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" rel=3D"noreferrer"=
 class=3D"m_-8552972565696268422gmail_msg" target=3D"_blank">https://www.ie=
tf.org/mailman/<wbr>listinfo/mmusic</a><br class=3D"m_-8552972565696268422g=
mail_msg">
</blockquote></div></div></div></div></div></div></div></div></blockquote><=
/div>
</div></div><br>______________________________<wbr>_________________<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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/mmusic</a><br=
>
<br></blockquote></div><br></div>

--f403045ddf7acf8ed60545767d23--


From nobody Mon Jan  9 14:09:26 2017
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-21.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 of the IETF.

        Title           : Session Description Protocol (SDP) Offer/Answer Procedures For Stream Control Transmission Protocol (SCTP) over Datagram Transport Layer Security (DTLS) Transport.
        Authors         : Christer Holmberg
                          Roman Shpount
                          Salvatore Loreto
                          Gonzalo Camarillo
	Filename        : draft-ietf-mmusic-sctp-sdp-21.txt
	Pages           : 25
	Date            : 2017-01-09

Abstract:
   The Stream Control Transmission Protocol (SCTP) is a transport
   protocol used to establish associations between two endpoints.
   draft-ietf-tsvwg-sctp-dtls-encaps-09 specifies how SCTP can be used
   on top of the Datagram Transport Layer Security (DTLS) protocol,
   referred to as SCTP-over-DTLS.

   This specification defines the following new Session Description
   Protocol (SDP) protocol identifiers (proto values):'UDP/DTLS/SCTP'
   and 'TCP/DTLS/SCTP'.  This specification also specifies how to use
   the new proto values with the SDP Offer/Answer mechanism for
   negotiating SCTP-over-DTLS associations.


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:
https://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-21

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https://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-sctp-sdp-21


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From nobody Tue Jan 10 05:18:12 2017
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Adam Roach <adam@nostrum.com>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] BUNDLE: meaning of "unspecified" when describing "bundle-only"
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Subject: Re: [MMUSIC] BUNDLE: meaning of "unspecified" when describing "bundle-only"
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Hi Adam,

My suggestion would be alternative #2. Because, if an endpoint supports
the attribute it doesn=B9t really matter what the port value is, so there i=
s
no need to reject the m- line.

Regards,

Christer


On 29/12/16 23:15, "mmusic on behalf of Adam Roach"
<mmusic-bounces@ietf.org on behalf of adam@nostrum.com> wrote:

>We've recently come across an issue with the way the "bundle-only"
>attribute is described in the current document. The current language
>regarding port handling reads:
>
>    The usage of the 'bundle-only' attribute is only defined for a
>    bundled "m=3D" line with a zero port value, within an offer. Other
>    usage is unspecified.
>
>Usually, when we have this kind of language, we still ensure that
>behavior is well defined, to help avoid unnecessary interop failures. I
>see a couple of different options here:
>
> 1. Remove the final sentence and add language saying that creators of
>    SDP MUST NOT include a "bundle-only" attribute in an m-section that
>    has a non-zero port, and that recipients of such SDP {SHOULD,MUST}
>    reject it; or
>
> 2. Retain language saying that including a "bundle-only" attribute in a
>    non-zero m-section is unspecified, but add normative language along
>    the lines of: "implementations that receive an m-section with a
>    non-zero port that also contains a 'bundle-only' attribute MUST
>    ignore the {attribute,port}."
>
>I don't have a preference between these choices, but I think we do need
>clarity. To be absolutely clear, this feedback is based on actual
>implementation interop failures in the field. This problem is not
>theoretical.
>
>/a
>
>_______________________________________________
>mmusic mailing list
>mmusic@ietf.org
>https://www.ietf.org/mailman/listinfo/mmusic


From nobody Tue Jan 10 06:52:13 2017
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-dtls-sdp-16.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 of the IETF.

        Title           : Using the SDP Offer/Answer Mechanism for DTLS
        Authors         : Christer Holmberg
                          Roman Shpount
	Filename        : draft-ietf-mmusic-dtls-sdp-16.txt
	Pages           : 25
	Date            : 2017-01-10

Abstract:
   This document defines the SDP offer/answer procedures for negotiating
   and establishing a DTLS association.  The document also defines the
   criteria for when a new DTLS association must be established.  The
   document updates RFC 5763 and RFC 7345, by replacing common SDP
   offer/answer procedures with a reference to this specification.

   This document defines a new SDP media-level attribute, 'dtls-id'.


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

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

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-dtls-sdp-16


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From nobody Tue Jan 10 07:03:42 2017
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: Draft new version: draft-dtls-sdp-16
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Subject: [MMUSIC] Draft new version: draft-dtls-sdp-16
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--_000_D49AC25F15627christerholmbergericssoncom_
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Hi,

Nobody objected to making the dtls-id attribute value globally unique, so I=
 have submitted a new version (-16) of draft-ietf-mmusic-dtls-sdp in order =
to reflex that.

Now a DTLS association is only identified by the dtls-id attribute values o=
f the offerer and answerer. The fingerprint is no longer used, as the previ=
ous text was confusing to many (no matter how we tried to fix it).

Regards,

Christer

--_000_D49AC25F15627christerholmbergericssoncom_
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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; color: rgb(0, 0, 0); font-size: 14px; font-fami=
ly: Calibri, sans-serif;">
<div>Hi,</div>
<div><br>
</div>
<div>Nobody objected to making the dtls-id attribute value globally unique,=
 so I have submitted a new version (-16) of&nbsp;<span style=3D"orphans: 2;=
 white-space: pre-wrap; widows: 2;">draft-ietf-mmusic-dtls-sdp in order to =
reflex that.</span></div>
<div><span style=3D"orphans: 2; white-space: pre-wrap; widows: 2;"><br>
</span></div>
<div><span style=3D"orphans: 2; white-space: pre-wrap; widows: 2;">Now a DT=
LS association is only identified by the dtls-id attribute values of the of=
ferer and answerer. The fingerprint is no longer used, as the previous text=
 was confusing to many (no matter
 how we tried to fix it).</span></div>
<div><span style=3D"orphans: 2; white-space: pre-wrap; widows: 2;"><br>
</span></div>
<div><span style=3D"orphans: 2; white-space: pre-wrap; widows: 2;">Regards,=
</span></div>
<div><span style=3D"orphans: 2; white-space: pre-wrap; widows: 2;"><br>
</span></div>
<div><span style=3D"orphans: 2; white-space: pre-wrap; widows: 2;">Christer=
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--_000_D49AC25F15627christerholmbergericssoncom_--


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Subject: Re: [MMUSIC] JSEP Issue #394: What appears in m= lines.
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Den 2016-12-22 kl. 18:30, skrev Eric Rescorla:
>
>
> On Thu, Dec 22, 2016 at 7:15 AM, Magnus Westerlund
>
>     Okay, I agree that having any rules for what you should offer here
>     based on what the actual outcome is not the best idea here. I think
>     the rules should be based on capability and intent. So if you only
>     are going offer TCP candidates, then you clearly should use TCP/â€¦,
>
>
> I'm going to push on this. If we've already agreed that mismatches are
> normal, why should we do that? Wouldn't it be better to just effectively
> deprecate this field?
>

So, my main argument for including any rules for how to set it, would be 
that the this part of the PROTO field still has meaning outside of 
WebRTC context. There one may actually have it to match what is goind to 
be used. Thus, it can provide some guidance to gateways on what to 
expect. It might be that we should actually set it to UDP for those not 
supporting ICE/TCP, and to TCP for the ones that supports both UDP and 
TCP. That would still provide information to the gateway.

I think simple rules based on implementation capabilities for how to set 
it are better than no rules. Especially as it would carry some 
information, not zero, which is the result of no rules.

Cheers

Magnus Westerlund

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Ericsson AB                 | Phone  +46 10 7148287
FÃ¤rÃ¶gatan 6                 | Mobile +46 73 0949079
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Subject: Re: [MMUSIC] JSEP Issue #394: What appears in m= lines.
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Den 2017-01-02 kl. 14:42, skrev Harald Alvestrand:
> Den 22. des. 2016 18:30, skrev Eric Rescorla:
>>      __
>>
>>     Okay, I agree that having any rules for what you should offer here
>>     based on what the actual outcome is not the best idea here. I think
>>     the rules should be based on capability and intent. So if you only
>>     are going offer TCP candidates, then you clearly should use TCP/…,
>>
>>
>> I'm going to push on this. If we've already agreed that mismatches are
>> normal, why should we do that? Wouldn't it be better to just effectively
>> deprecate this field?
>>
>
> Concur with EKR.
>
> This field (the TCP/UDP part of the protocol field) was defined in a way
> that makes its original purpose (to negotiate profiles) not only
> impossible while using ICE, but actively harmful, because you end up
> telling lies in very many cases, so people will have to accept things
> that don't match reality.
>
> Make the world as simple for implementors as possible: Send a fixed
> string, and accept all strings.
>

Okay, can we please be precise with what is suggested here. Because what 
you write above, and what EKR proposed is not really the same thing. The 
issue proposed that it was fine to set either of the strings. You saying 
a fixed string. I want to ask you which UDP/... or TCP/..., or even 
ICE/DTLS/RTP/SAVPF?

 From a strict clarity issue using ICE/... would be my preferred. But, 
that is probably going to throw some implementations, especially 
gateways. It will also require us to actually go write the registration 
to explain what it is.

Using UDP is likely the most compatible and will only fail in gateway 
cases where there are no UDP candidates and the gateway fail to pick 
that up.

Using TCP when there are no TCP candidates are equally misleading to 
gateway cases.

The above is why I proposed that implementation capabilities and any 
configuration to limit what functions being used would be an appropriate 
choice.

I hope at least we can agree that an JSEP Answer MUST use the PROTO 
string of the OFFER?

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
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Den 10. jan. 2017 16:44, skrev Magnus Westerlund:
> Den 2017-01-02 kl. 14:42, skrev Harald Alvestrand:
>> Den 22. des. 2016 18:30, skrev Eric Rescorla:
>>>      __
>>>
>>>     Okay, I agree that having any rules for what you should offer here
>>>     based on what the actual outcome is not the best idea here. I think
>>>     the rules should be based on capability and intent. So if you only
>>>     are going offer TCP candidates, then you clearly should use TCP/…,
>>>
>>>
>>> I'm going to push on this. If we've already agreed that mismatches are
>>> normal, why should we do that? Wouldn't it be better to just effectively
>>> deprecate this field?
>>>
>>
>> Concur with EKR.
>>
>> This field (the TCP/UDP part of the protocol field) was defined in a way
>> that makes its original purpose (to negotiate profiles) not only
>> impossible while using ICE, but actively harmful, because you end up
>> telling lies in very many cases, so people will have to accept things
>> that don't match reality.
>>
>> Make the world as simple for implementors as possible: Send a fixed
>> string, and accept all strings.
>>
> 
> Okay, can we please be precise with what is suggested here. Because what
> you write above, and what EKR proposed is not really the same thing. The
> issue proposed that it was fine to set either of the strings. You saying
> a fixed string. I want to ask you which UDP/... or TCP/..., or even
> ICE/DTLS/RTP/SAVPF?

The current JSEP language is:

5.1.3.  Profile Names and Interoperability

   For media m= sections, JSEP endpoints MUST support both the "UDP/TLS/
   RTP/SAVPF" and "TCP/DTLS/RTP/SAVPF" profiles and MUST indicate one of
   these two profiles for each media m= line they produce in an offer.
   For data m= sections, JSEP endpoints must support both the "UDP/DTLS/
   SCTP" and "TCP/DTLS/SCTP" profiles and MUST indicate one of these two
   profiles for each data m= line they produce in an offer.  Because ICE
   can select either TCP or UDP transport depending on network
   conditions, both advertisements are consistent with ICE eventually
   selecting either either UDP or TCP.

The current proposal (as I interpret EKR) is to replace

        ....and MUST indicate one of these two
   profiles for each data m= line they produce in an offer.  Because ICE
   can select either TCP or UDP transport depending on network
   conditions, both advertisements are consistent with ICE eventually
   selecting either either UDP or TCP.

with

            ....and MUST indicate UDP/TLS/RTP/SAVPF for each data m= line
   they produce in an offer.  Because ICE
   can select either TCP or UDP transport depending on network
   conditions, this consistent with ICE eventually
   selecting either either UDP or TCP.

> 
> From a strict clarity issue using ICE/... would be my preferred. But,
> that is probably going to throw some implementations, especially
> gateways. It will also require us to actually go write the registration
> to explain what it is.

We lost clarity when ICE was introduced without redesigning the
"protocol" field. Trying to get it back is not helping anyone.

> 
> Using UDP is likely the most compatible and will only fail in gateway
> cases where there are no UDP candidates and the gateway fail to pick
> that up.
> 
> Using TCP when there are no TCP candidates are equally misleading to
> gateway cases.
> 
> The above is why I proposed that implementation capabilities and any
> configuration to limit what functions being used would be an appropriate
> choice.
> 
> I hope at least we can agree that an JSEP Answer MUST use the PROTO
> string of the OFFER?

We agreed on that long ago.

Section 5.3.1 of JSEP:

   o  The <proto> field MUST be set to exactly match the <proto> field
      for the corresponding m= line in the offer.



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Subject: Re: [MMUSIC] JSEP Issue #394: What appears in m= lines.
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On 1/10/17 10:44 AM, Magnus Westerlund wrote:

> From a strict clarity issue using ICE/... would be my preferred. But,

+1

> that is probably going to throw some implementations, especially
> gateways. It will also require us to actually go write the registration
> to explain what it is.

Having to write a registration to explain what it is would be a good thing!

	Thanks,
	Paul


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From: Jonathan Lennox <jonathan@vidyo.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: [rtcweb] [MMUSIC] JSEP Issue #394: What appears in m= lines.
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From: Roman Shpount <roman@telurix.com>
Date: Tue, 10 Jan 2017 13:08:07 -0500
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Subject: Re: [MMUSIC] JSEP Issue #394: What appears in m= lines.
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We can do the registration for ICE transport tag in ICE-SIP-SDP. ICE tag
will indicate that ICE is required for this offer/answer exchange to
succeed. When ICE transport tag is used, c= and m= SDP line would carry no
address information and will contain "c=IN IP4 0.0.0.0" and port 9 in the
m= line. Also, if ICE/ transport tag is used, we should remove the
requirement for a re-INVITE after ICE nomination process is complete, since
it serves no purpose when no address is specified in c= and m= lines.

If we are not going to define ICE/ transport, then I propose the following:

1. When c= line is set to "IN IP4 0.0.0.0" and m= line is set to port 9, so
that these lines carry no valid address information as it is specified in
JSEP, protocol should be UDP/DTLS/SAVPF for RTP or UDP/DTLS/SCTP for data

2. When c= line and m= line carry valid address information, protocol
should match the candidate specified in c= and m= lines.

3. In cases when c= line and m= line carry valid address information, when
initial offer and answer are generated, they must include UDP candidates
and UDP candidates must be used as defaults. Once ICE nomination process is
complete, only the active candidate pair MUST be included in SDP, and the
transport for the active pair must be used.

I am not sure about the proper procedure for c= and m= lines during the ICE
nomination process, since active candidate can change during the
offer/answer exchange which can cause protocol line mismatch. One option is
to continue re-sending the initial offer and answer.

P.S. In the quoted text someone mentioned "UDP/TLS/RTP/SAVPF", should it be
"UDP/DTLS/RTP/SAVPF"?

Regards,
_____________
Roman Shpount

--001a1149e640cf636b0545c15db7
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<div dir=3D"ltr">We can do the registration for ICE transport tag in ICE-SI=
P-SDP. ICE tag will indicate that ICE is required for this offer/answer exc=
hange to succeed. When ICE transport tag is used, c=3D and m=3D SDP line wo=
uld carry no address information and will contain &quot;c=3DIN IP4 0.0.0.0&=
quot; and port 9 in the m=3D line. Also, if ICE/ transport tag is used, we =
should remove the requirement for a re-INVITE after ICE nomination process =
is complete, since it serves no purpose when no address is specified in c=
=3D and m=3D lines.<div><br></div><div>If we are not going to define ICE/ t=
ransport, then I propose the following:</div><div><br></div><div>1. When c=
=3D line is set to &quot;IN IP4 0.0.0.0&quot; and m=3D line is set to port =
9, so that these lines carry no valid address information as it is specifie=
d in JSEP, protocol should be=C2=A0<span style=3D"color:rgb(0,0,0);font-siz=
e:12.8px">UDP/DTLS/</span><span style=3D"color:rgb(0,0,0);font-size:12.8px"=
>SAVPF for RTP or=C2=A0</span><span style=3D"color:rgb(0,0,0);font-size:12.=
8px">UDP/DTLS/</span><span style=3D"color:rgb(0,0,0);font-size:12.8px">SCTP=
 for data</span></div><div><span style=3D"color:rgb(0,0,0);font-size:12.8px=
"><br></span></div><div><span style=3D"color:rgb(0,0,0);font-size:12.8px">2=
. When=C2=A0</span>c=3D line and m=3D line carry valid address information,=
 protocol should match the candidate specified in c=3D and m=3D lines.=C2=
=A0</div><div><br></div><div>3. In cases<span style=3D"color:rgb(0,0,0);fon=
t-size:12.8px">=C2=A0when=C2=A0</span>c=3D line and m=3D line carry valid a=
ddress information,=C2=A0when initial offer and answer are generated, they =
must include UDP candidates and UDP candidates must be used as defaults. On=
ce ICE nomination process is complete, only the active candidate pair MUST =
be included in SDP, and the transport for the active pair must be used.</di=
v><div><br></div><div>I am not sure about the proper procedure for c=3D and=
 m=3D lines during the ICE nomination process, since active candidate can c=
hange during the offer/answer exchange which can cause protocol line mismat=
ch. One option is to continue re-sending the initial offer and answer.</div=
><div><br></div><div>P.S. In the quoted text someone mentioned &quot;<span =
style=3D"color:rgb(0,0,0);font-size:12.8px">UDP/TLS/</span><span style=3D"c=
olor:rgb(0,0,0);font-size:12.8px">RTP/SAVPF&quot;, should it be=C2=A0</span=
>&quot;<span style=3D"color:rgb(0,0,0);font-size:12.8px">UDP/DTLS/</span><s=
pan style=3D"color:rgb(0,0,0);font-size:12.8px">RTP/SAVPF&quot;?</span><br>=
</div><div><span style=3D"color:rgb(0,0,0);font-size:12.8px"><br></span></d=
iv><div><div>Regards, =C2=A0</div></div><div class=3D"gmail_extra"><div><di=
v class=3D"gmail_signature">_____________<br>Roman Shpount</div></div></div=
></div>

--001a1149e640cf636b0545c15db7--


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To: Roman Shpount <roman@telurix.com>
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From: Paul Kyzivat <paul.kyzivat@comcast.net>
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Subject: Re: [MMUSIC] JSEP Issue #394: What appears in m= lines.
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On 1/10/17 1:08 PM, Roman Shpount wrote:
> We can do the registration for ICE transport tag in ICE-SIP-SDP. ICE tag
> will indicate that ICE is required for this offer/answer exchange to
> succeed. When ICE transport tag is used, c= and m= SDP line would carry
> no address information and will contain "c=IN IP4 0.0.0.0" and port 9 in
> the m= line. Also, if ICE/ transport tag is used, we should remove the
> requirement for a re-INVITE after ICE nomination process is complete,
> since it serves no purpose when no address is specified in c= and m= lines.
>
> If we are not going to define ICE/ transport, then I propose the following:
>
> 1. When c= line is set to "IN IP4 0.0.0.0" and m= line is set to port 9,
> so that these lines carry no valid address information as it is
> specified in JSEP, protocol should be UDP/DTLS/SAVPF for RTP
> or UDP/DTLS/SCTP for data

To do this cleanly, IMO there should be no singling out of IP4. The fact 
is that in this case the c= line is not needed. (And this isn't the only 
case where it isn't. It also isn't needed for MSRP and for websockets - 
things that use URLs in attributes instead of c=.) So perhaps we should 
revise 4566bis to make c= optional for m= lines where it isn't needed. 
Or define a new addrtype for use in c= (e.g. simply "IP" or "IP4/6") and 
define a placeholder address value for this case. For example:

   c=IN IP -

> 2. When c= line and m= line carry valid address information, protocol
> should match the candidate specified in c= and m= lines.
>
> 3. In cases when c= line and m= line carry valid address
> information, when initial offer and answer are generated, they must
> include UDP candidates and UDP candidates must be used as defaults. Once
> ICE nomination process is complete, only the active candidate pair MUST
> be included in SDP, and the transport for the active pair must be used.

I think this could be handled by having different <proto> values:

ICE/DTLS      DTLS over either UDP or TCP
ICE/UDP/DTLS  DTLS over only UDP
ICE/TCP/DTLS  DTLS over only TCP

ICE/DTLS/RTP/SAVPF      Like above, but for RTP/SAVPF
ICE/UDP/DTLS/RTP/SAVPF  "
ICE/TCP/DTLS/RTP/SAVPF  "

So this allows restricting the type of candidates that are allowed, or 
not. And it avoids "lying" about the type that is to be used.

	Thanks,
	Paul


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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: Roman Shpount <roman@telurix.com>, Alan Ford <alan.ford@gmail.com>, "Tom Kristensen (tomkrist)" <tomkrist@cisco.com>, Tom Kristensen <2mkristensen@gmail.com>
Thread-Topic: [bfcpbis] [MMUSIC] m= line protocol in case of ICE
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Date: Tue, 10 Jan 2017 20:17:03 +0000
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Cc: "ice@ietf.org" <ice@ietf.org>, "bfcpbis@ietf.org" <bfcpbis@ietf.org>, "mmusic@ietf.org" <mmusic@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] [bfcpbis]  m= line protocol in case of ICE
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From: Roman Shpount <roman@telurix.com>
Date: Tue, 10 Jan 2017 15:35:12 -0500
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Subject: Re: [MMUSIC] JSEP Issue #394: What appears in m= lines.
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On Tue, Jan 10, 2017 at 2:27 PM, Paul Kyzivat <paul.kyzivat@comcast.net>
wrote:

> On 1/10/17 1:08 PM, Roman Shpount wrote:
>
>> 1. When c= line is set to "IN IP4 0.0.0.0" and m= line is set to port 9,
>> so that these lines carry no valid address information as it is
>> specified in JSEP, protocol should be UDP/DTLS/SAVPF for RTP
>> or UDP/DTLS/SCTP for data
>>
>
> To do this cleanly, IMO there should be no singling out of IP4. The fact
> is that in this case the c= line is not needed. (And this isn't the only
> case where it isn't. It also isn't needed for MSRP and for websockets -
> things that use URLs in attributes instead of c=.) So perhaps we should
> revise 4566bis to make c= optional for m= lines where it isn't needed. Or
> define a new addrtype for use in c= (e.g. simply "IP" or "IP4/6") and
> define a placeholder address value for this case. For example:
>
>   c=IN IP -


It is cleaner, but this needs to be defined somewhere. Current JSEP uses
"IN IP4 0.0.0.0" and port 9.

2. When c= line and m= line carry valid address information, protocol
>> should match the candidate specified in c= and m= lines.
>>
>> 3. In cases when c= line and m= line carry valid address
>> information, when initial offer and answer are generated, they must
>> include UDP candidates and UDP candidates must be used as defaults. Once
>> ICE nomination process is complete, only the active candidate pair MUST
>> be included in SDP, and the transport for the active pair must be used.
>>
>
> I think this could be handled by having different <proto> values:
>
> ICE/DTLS      DTLS over either UDP or TCP
> ICE/UDP/DTLS  DTLS over only UDP
> ICE/TCP/DTLS  DTLS over only TCP
>
> ICE/DTLS/RTP/SAVPF      Like above, but for RTP/SAVPF
> ICE/UDP/DTLS/RTP/SAVPF  "
> ICE/TCP/DTLS/RTP/SAVPF  "
>
> So this allows restricting the type of candidates that are allowed, or
> not. And it avoids "lying" about the type that is to be used.
>

We can define it this way, but I am not sure defining anything beyond
ICE/DTLS/RTP/SAVPF and ICE/DTLS/SCTP is actually needed. Once we specify
that ICE is used, actual candidates can be examined to see which candidate
types are supported.

Regards,
_____________
Roman Shpount

--94eb2c06e794cb4a530545c36bee
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">On Tue, Jan 10, 2017 at 2:27 PM, Paul Kyzivat <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:paul.kyzivat@comcast.net" target=3D"_blank">paul.kyzivat@co=
mcast.net</a>&gt;</span> wrote:<br></div></div><div class=3D"gmail_quote"><=
blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-l=
eft:1px solid rgb(204,204,204);padding-left:1ex"><span class=3D"gmail-">On =
1/10/17 1:08 PM, Roman Shpount wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left:1px solid rgb(204,204,204);padding-left:1ex">1. When c=3D line is set =
to &quot;IN IP4 0.0.0.0&quot; and m=3D line is set to port 9,<br>
so that these lines carry no valid address information as it is<br>
specified in JSEP, protocol should be UDP/DTLS/SAVPF for RTP<br>
or UDP/DTLS/SCTP for data<br>
</blockquote>
<br></span>
To do this cleanly, IMO there should be no singling out of IP4. The fact is=
 that in this case the c=3D line is not needed. (And this isn&#39;t the onl=
y case where it isn&#39;t. It also isn&#39;t needed for MSRP and for websoc=
kets - things that use URLs in attributes instead of c=3D.) So perhaps we s=
hould revise 4566bis to make c=3D optional for m=3D lines where it isn&#39;=
t needed. Or define a new addrtype for use in c=3D (e.g. simply &quot;IP&qu=
ot; or &quot;IP4/6&quot;) and define a placeholder address value for this c=
ase. For example:<br>
<br>
=C2=A0 c=3DIN IP -</blockquote><div><br></div><div>It is cleaner, but this =
needs to be defined somewhere. Current JSEP uses &quot;IN IP4 0.0.0.0&quot;=
 and port 9.</div><div><br></div><blockquote class=3D"gmail_quote" style=3D=
"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-le=
ft:1ex"><span class=3D"gmail-"><blockquote class=3D"gmail_quote" style=3D"m=
argin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left=
:1ex">2. When c=3D line and m=3D line carry valid address information, prot=
ocol<br>
should match the candidate specified in c=3D and m=3D lines.<br>
<br>
3. In cases when c=3D line and m=3D line carry valid address<br>
information, when initial offer and answer are generated, they must<br>
include UDP candidates and UDP candidates must be used as defaults. Once<br=
>
ICE nomination process is complete, only the active candidate pair MUST<br>
be included in SDP, and the transport for the active pair must be used.<br>
</blockquote>
<br></span>
I think this could be handled by having different &lt;proto&gt; values:<br>
<br>
ICE/DTLS=C2=A0 =C2=A0 =C2=A0 DTLS over either UDP or TCP<br>
ICE/UDP/DTLS=C2=A0 DTLS over only UDP<br>
ICE/TCP/DTLS=C2=A0 DTLS over only TCP<br>
<br>
ICE/DTLS/RTP/SAVPF=C2=A0 =C2=A0 =C2=A0 Like above, but for RTP/SAVPF<br>
ICE/UDP/DTLS/RTP/SAVPF=C2=A0 &quot;<br>
ICE/TCP/DTLS/RTP/SAVPF=C2=A0 &quot;<br>
<br>
So this allows restricting the type of candidates that are allowed, or not.=
 And it avoids &quot;lying&quot; about the type that is to be used.<br></bl=
ockquote><div><br></div><div>We can define it this way, but I am not sure d=
efining anything beyond ICE/DTLS/RTP/SAVPF and ICE/DTLS/SCTP is actually ne=
eded. Once we specify that ICE is used, actual candidates can be examined t=
o see which candidate types are supported.</div><div><br></div><div>Regards=
,</div><div><div class=3D"gmail_signature">_____________<br>Roman Shpount</=
div></div><div>=C2=A0</div></div></div></div>

--94eb2c06e794cb4a530545c36bee--


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To: Roman Shpount <roman@telurix.com>
References: <52E4A8FC978E0241AE652516E24CAF001E483F95@ESESSMB309.ericsson.se> <CABcZeBPznLKNHek-SGE5Ly6QTOBL-j65sZBb5MbwQVkmBkpyFw@mail.gmail.com> <9110d772-9269-7fed-3ed4-5269d49acb84@alvestrand.no> <282955c7-d077-105b-6a99-a0f5ede87d91@ericsson.com> <9ff375eb-0efc-cb7e-6f26-c48f17c55275@comcast.net> <CAD5OKxvtRxajLddLQjTMMiLYqE6-Z=msX29NM2O532ydmZbSAQ@mail.gmail.com> <715b5012-0186-ea3b-08fb-954eae652c1d@comcast.net> <CAD5OKxs5NK8cshdeJRrPD2fj30AwBwLeNpqE_EAPaAZ-rzv6iw@mail.gmail.com>
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Date: Tue, 10 Jan 2017 17:17:58 -0500
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Subject: Re: [MMUSIC] JSEP Issue #394: What appears in m= lines.
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On 1/10/17 3:35 PM, Roman Shpount wrote:
> On Tue, Jan 10, 2017 at 2:27 PM, Paul Kyzivat <paul.kyzivat@comcast.net
> <mailto:paul.kyzivat@comcast.net>> wrote:
>
>     On 1/10/17 1:08 PM, Roman Shpount wrote:
>
>         1. When c= line is set to "IN IP4 0.0.0.0" and m= line is set to
>         port 9,
>         so that these lines carry no valid address information as it is
>         specified in JSEP, protocol should be UDP/DTLS/SAVPF for RTP
>         or UDP/DTLS/SCTP for data
>
>
>     To do this cleanly, IMO there should be no singling out of IP4. The
>     fact is that in this case the c= line is not needed. (And this isn't
>     the only case where it isn't. It also isn't needed for MSRP and for
>     websockets - things that use URLs in attributes instead of c=.) So
>     perhaps we should revise 4566bis to make c= optional for m= lines
>     where it isn't needed. Or define a new addrtype for use in c= (e.g.
>     simply "IP" or "IP4/6") and define a placeholder address value for
>     this case. For example:
>
>       c=IN IP -
>
>
> It is cleaner, but this needs to be defined somewhere. Current JSEP uses
> "IN IP4 0.0.0.0" and port 9.

4566bis is still in progress, so it could be done there.

>         2. When c= line and m= line carry valid address information,
>         protocol
>         should match the candidate specified in c= and m= lines.
>
>         3. In cases when c= line and m= line carry valid address
>         information, when initial offer and answer are generated, they must
>         include UDP candidates and UDP candidates must be used as
>         defaults. Once
>         ICE nomination process is complete, only the active candidate
>         pair MUST
>         be included in SDP, and the transport for the active pair must
>         be used.
>
>
>     I think this could be handled by having different <proto> values:
>
>     ICE/DTLS      DTLS over either UDP or TCP
>     ICE/UDP/DTLS  DTLS over only UDP
>     ICE/TCP/DTLS  DTLS over only TCP
>
>     ICE/DTLS/RTP/SAVPF      Like above, but for RTP/SAVPF
>     ICE/UDP/DTLS/RTP/SAVPF  "
>     ICE/TCP/DTLS/RTP/SAVPF  "
>
>     So this allows restricting the type of candidates that are allowed,
>     or not. And it avoids "lying" about the type that is to be used.
>
>
> We can define it this way, but I am not sure defining anything beyond
> ICE/DTLS/RTP/SAVPF and ICE/DTLS/SCTP is actually needed. Once we specify
> that ICE is used, actual candidates can be examined to see which
> candidate types are supported.

Perhaps choice of UDP/TCP/both can be left to the candidates. I guess if 
one side wants to restrict to one or the other it can offer only 
candidates of the type it wants.

	Thanks.
	Paul


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To: Roman Shpount <roman@telurix.com>, Paul Kyzivat <paul.kyzivat@comcast.net>
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Subject: Re: [MMUSIC] JSEP Issue #394: What appears in m= lines.
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Den 2017-01-10 kl. 19:08, skrev Roman Shpount:

> P.S. In the quoted text someone mentioned "UDP/TLS/RTP/SAVPF", should it
> be "UDP/DTLS/RTP/SAVPF"?

No, actually UDP/TLS/RTP/SAVPF is what is registered for DTLS-SRTP over 
UDP with RTP SAVPF profile. See	[RFC5764]. The "UDP/DTLS/RTP/SAVPF" is 
not registered.

So the alternatives for the dependency of default candidate protocol 
type in your proposal would be:

UDP: "UDP/TLS/RTP/SAVPF"
TCP: "TCP/DTLS/RTP/SAVPF"

Cheers

Magnus Westerlund

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Färögatan 6                 | Mobile +46 73 0949079
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From: Roman Shpount <roman@telurix.com>
Date: Wed, 11 Jan 2017 16:01:29 -0500
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To: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Cc: Paul Kyzivat <paul.kyzivat@comcast.net>, "mmusic@ietf.org" <mmusic@ietf.org>
Subject: [MMUSIC] Transport tags
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What should we use for the new protocols that we are defining?

Should it be UDP/TLS/SCTP or UDP/DTLS/SCTP?
Should it be UDP/TLS/BFCP or UDP/DTLS/BFCP?

I would like to see some consistency here.

Regards,
_____________
Roman Shpount

On Wed, Jan 11, 2017 at 3:30 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> Den 2017-01-10 kl. 19:08, skrev Roman Shpount:
>
> P.S. In the quoted text someone mentioned "UDP/TLS/RTP/SAVPF", should it
>> be "UDP/DTLS/RTP/SAVPF"?
>>
>
> No, actually UDP/TLS/RTP/SAVPF is what is registered for DTLS-SRTP over
> UDP with RTP SAVPF profile. See [RFC5764]. The "UDP/DTLS/RTP/SAVPF" is no=
t
> registered.
>
> So the alternatives for the dependency of default candidate protocol type
> in your proposal would be:
>
> UDP: "UDP/TLS/RTP/SAVPF"
> TCP: "TCP/DTLS/RTP/SAVPF"
>
>
> 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 should we use for the new protocols that we are defin=
ing?<div><br></div><div>Should it be UDP/TLS/SCTP or UDP/DTLS/SCTP?</div><d=
iv>Should it be UDP/TLS/BFCP or UDP/DTLS/BFCP?</div><div><br></div><div>I w=
ould like to see some consistency here.</div><div><br></div><div>Regards,<d=
iv class=3D"gmail_extra"><div><div class=3D"gmail_signature">_____________<=
br>Roman Shpount</div></div>
<br><div class=3D"gmail_quote">On Wed, Jan 11, 2017 at 3:30 AM, Magnus West=
erlund <span dir=3D"ltr">&lt;<a href=3D"mailto:magnus.westerlund@ericsson.c=
om" target=3D"_blank">magnus.westerlund@ericsson.com</a>&gt;</span> wrote:<=
br><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;bord=
er-left:1px solid rgb(204,204,204);padding-left:1ex"><span class=3D"gmail-"=
>Den 2017-01-10 kl. 19:08, skrev Roman Shpount:<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left:1px solid rgb(204,204,204);padding-left:1ex">
P.S. In the quoted text someone mentioned &quot;UDP/TLS/RTP/SAVPF&quot;, sh=
ould it<br>
be &quot;UDP/DTLS/RTP/SAVPF&quot;?<br>
</blockquote>
<br></span>
No, actually UDP/TLS/RTP/SAVPF is what is registered for DTLS-SRTP over UDP=
 with RTP SAVPF profile. See [RFC5764]. The &quot;UDP/DTLS/RTP/SAVPF&quot; =
is not registered.<br>
<br>
So the alternatives for the dependency of default candidate protocol type i=
n your proposal would be:<br>
<br>
UDP: &quot;UDP/TLS/RTP/SAVPF&quot;<br>
TCP: &quot;TCP/DTLS/RTP/SAVPF&quot;<div class=3D"gmail-HOEnZb"><div class=
=3D"gmail-h5"><br>
<br>
Cheers<br>
<br>
Magnus Westerlund<br>
<br>
------------------------------<wbr>------------------------------<wbr>-----=
-----<br>
Services, Media and Network features, Ericsson Research EAB/TXM<br>
------------------------------<wbr>------------------------------<wbr>-----=
-----<br>
Ericsson AB=C2=A0 =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" tar=
get=3D"_blank">+46 10 7148287</a><br>
F=C3=A4r=C3=B6gatan 6=C2=A0 =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"+467309490=
79" target=3D"_blank">+46 73 0949079</a><br>
SE-164 80 Stockholm, Sweden | mailto: <a href=3D"mailto:magnus.westerlund@e=
ricsson.com" target=3D"_blank">magnus.westerlund@ericsson.com</a><br>
------------------------------<wbr>------------------------------<wbr>-----=
-----<br>
<br>
</div></div></blockquote></div><br></div></div></div>

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From: Roman Shpount <roman@telurix.com>
Date: Wed, 11 Jan 2017 19:33:13 -0500
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Subject: Re: [MMUSIC] Transport tags
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--001a114a6f02d4db910545dadc38
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I guess BFCP will need TCP/DTLS/BFCP added to the list.

Should we correct draft-ietf-mmusic-sctp-sdp draft to match and change
UDP/DTLS/SCTP to UDP/TLS/SCTP?

Regards,

_____________
Roman Shpount

On Wed, Jan 11, 2017 at 5:05 PM, Charles Eckel (eckelcu) <eckelcu@cisco.com=
>
wrote:

> Here is what we have for BFCP in https://tools.ietf.org/html/
> draft-ietf-bfcpbis-rfc4583bis-16#section-13:
>
>
>
>                        +--------------+------------+
>
>                        | Value        | Reference  |
>
>                        +--------------+------------+
>
>                        | TCP/BFCP     | [RFC XXXX] |
>
>                        | TCP/TLS/BFCP | [RFC XXXX] |
>
>                        | UDP/BFCP     | [RFC XXXX] |
>
>                        | UDP/TLS/BFCP | [RFC XXXX] |
>
>                        +--------------+------------+
>
>
>
> Cheers,
>
> Charles
>
>
>
> *From: *mmusic <mmusic-bounces@ietf.org> on behalf of Roman Shpount <
> roman@telurix.com>
> *Date: *Wednesday, January 11, 2017 at 1:01 PM
> *To: *Magnus Westerlund <magnus.westerlund@ericsson.com>
> *Cc: *Paul Kyzivat <paul.kyzivat@comcast.net>, "mmusic@ietf.org" <
> mmusic@ietf.org>
> *Subject: *[MMUSIC] Transport tags
>
>
>
> What should we use for the new protocols that we are defining?
>
>
>
> Should it be UDP/TLS/SCTP or UDP/DTLS/SCTP?
>
> Should it be UDP/TLS/BFCP or UDP/DTLS/BFCP?
>
>
>
> I would like to see some consistency here.
>
>
>
> Regards,
>
> _____________
> Roman Shpount
>
>
>
> On Wed, Jan 11, 2017 at 3:30 AM, Magnus Westerlund <
> magnus.westerlund@ericsson.com> wrote:
>
> Den 2017-01-10 kl. 19:08, skrev Roman Shpount:
>
> P.S. In the quoted text someone mentioned "UDP/TLS/RTP/SAVPF", should it
> be "UDP/DTLS/RTP/SAVPF"?
>
>
> No, actually UDP/TLS/RTP/SAVPF is what is registered for DTLS-SRTP over
> UDP with RTP SAVPF profile. See [RFC5764]. The "UDP/DTLS/RTP/SAVPF" is no=
t
> registered.
>
> So the alternatives for the dependency of default candidate protocol type
> in your proposal would be:
>
> UDP: "UDP/TLS/RTP/SAVPF"
> TCP: "TCP/DTLS/RTP/SAVPF"
>
>
>
> 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
> ----------------------------------------------------------------------
>
>
>
>

--001a114a6f02d4db910545dadc38
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">I guess BFCP will need TCP/DTLS/BFCP added to the list.<br=
><div><br></div>Should we correct draft-ietf-mmusic-sctp-sdp draft to match=
 and change UDP/DTLS/SCTP to UDP/TLS/SCTP?<div><br></div><div>Regards,</div=
></div><div class=3D"gmail_extra"><br clear=3D"all"><div><div class=3D"gmai=
l_signature" data-smartmail=3D"gmail_signature">_____________<br>Roman Shpo=
unt</div></div>
<br><div class=3D"gmail_quote">On Wed, Jan 11, 2017 at 5:05 PM, Charles Eck=
el (eckelcu) <span dir=3D"ltr">&lt;<a href=3D"mailto:eckelcu@cisco.com" tar=
get=3D"_blank">eckelcu@cisco.com</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex">







<div bgcolor=3D"white" lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div class=3D"m_3936043015517947387WordSection1">
<p class=3D"MsoNormal">Here is what we have for BFCP in <a href=3D"https://=
tools.ietf.org/html/draft-ietf-bfcpbis-rfc4583bis-16#section-13" target=3D"=
_blank">https://tools.ietf.org/html/<wbr>draft-ietf-bfcpbis-rfc4583bis-<wbr=
>16#section-13</a>:<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
 +--------------+------------+<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
 | Value=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 | Reference=C2=A0 |<u></=
u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
 +--------------+------------+<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
 | TCP/BFCP=C2=A0=C2=A0=C2=A0=C2=A0 | [RFC XXXX] |<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 =C2=
=A0| TCP/TLS/BFCP | [RFC XXXX] |<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
 | UDP/BFCP=C2=A0=C2=A0=C2=A0=C2=A0 | [RFC XXXX] |<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
 | UDP/TLS/BFCP | [RFC XXXX] |<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
 +--------------+------------+<u></u><u></u></span></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">Cheers,<u></u><u></u></p>
<p class=3D"MsoNormal">Charles<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<blockquote style=3D"border:none;border-left:solid #b5c4df 4.5pt;padding:0i=
n 0in 0in 4.0pt;margin-left:3.75pt;margin-right:0in">
<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-family:Calibri;color:black">F=
rom: </span>
</b><span style=3D"font-family:Calibri;color:black">mmusic &lt;<a href=3D"m=
ailto:mmusic-bounces@ietf.org" target=3D"_blank">mmusic-bounces@ietf.org</a=
>&gt; on behalf of Roman Shpount &lt;<a href=3D"mailto:roman@telurix.com" t=
arget=3D"_blank">roman@telurix.com</a>&gt;<br>
<b>Date: </b>Wednesday, January 11, 2017 at 1:01 PM<br>
<b>To: </b>Magnus Westerlund &lt;<a href=3D"mailto:magnus.westerlund@ericss=
on.com" target=3D"_blank">magnus.westerlund@ericsson.<wbr>com</a>&gt;<br>
<b>Cc: </b>Paul Kyzivat &lt;<a href=3D"mailto:paul.kyzivat@comcast.net" tar=
get=3D"_blank">paul.kyzivat@comcast.net</a>&gt;, &quot;<a href=3D"mailto:mm=
usic@ietf.org" target=3D"_blank">mmusic@ietf.org</a>&quot; &lt;<a href=3D"m=
ailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.org</a>&gt;<br>
<b>Subject: </b>[MMUSIC] Transport tags<u></u><u></u></span></p>
</div><div><div class=3D"h5">
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">What should we use for the new protocols that we are=
 defining?
<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Should it be UDP/TLS/SCTP or UDP/DTLS/SCTP?<u></u><u=
></u></p>
</div>
<div>
<p class=3D"MsoNormal">Should it be UDP/TLS/BFCP or UDP/DTLS/BFCP?<u></u><u=
></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">I would like to see some consistency here.<u></u><u>=
</u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Regards, <u></u><u></u></p>
<div>
<div>
<div>
<p class=3D"MsoNormal">_____________<br>
Roman Shpount<u></u><u></u></p>
</div>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">On Wed, Jan 11, 2017 at 3:30 AM, Magnus Westerlund &=
lt;<a href=3D"mailto:magnus.westerlund@ericsson.com" target=3D"_blank">magn=
us.westerlund@ericsson.<wbr>com</a>&gt; wrote:<u></u><u></u></p>
<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">
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span class=3D"m_3936=
043015517947387gmail-">Den 2017-01-10 kl. 19:08, skrev Roman Shpount:<u></u=
><u></u></span></p>
<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">
<p class=3D"MsoNormal">P.S. In the quoted text someone mentioned &quot;UDP/=
TLS/RTP/SAVPF&quot;, should it<br>
be &quot;UDP/DTLS/RTP/SAVPF&quot;?<u></u><u></u></p>
</blockquote>
<p class=3D"MsoNormal"><br>
No, actually UDP/TLS/RTP/SAVPF is what is registered for DTLS-SRTP over UDP=
 with RTP SAVPF profile. See [RFC5764]. The &quot;UDP/DTLS/RTP/SAVPF&quot; =
is not registered.<br>
<br>
So the alternatives for the dependency of default candidate protocol type i=
n your proposal would be:<br>
<br>
UDP: &quot;UDP/TLS/RTP/SAVPF&quot;<br>
TCP: &quot;TCP/DTLS/RTP/SAVPF&quot; <u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
<br>
Cheers<br>
<br>
Magnus Westerlund<br>
<br>
------------------------------<wbr>------------------------------<wbr>-----=
-----<br>
Services, Media and Network features, Ericsson Research EAB/TXM<br>
------------------------------<wbr>------------------------------<wbr>-----=
-----<br>
Ericsson AB=C2=A0 =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" target=3D"_blank">
+46 10 7148287</a><br>
F=C3=A4r=C3=B6gatan 6=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0| Mobile <a href=3D"tel:%2B46%2073%200949079" target=3D"_blank">
+46 73 0949079</a><br>
SE-164 80 Stockholm, Sweden | mailto: <a href=3D"mailto:magnus.westerlund@e=
ricsson.com" target=3D"_blank">
magnus.westerlund@ericsson.com</a><br>
------------------------------<wbr>------------------------------<wbr>-----=
-----<u></u><u></u></p>
</div>
</div>
</blockquote>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
</div>
</div>
</div></div></blockquote>
</div>
</div>

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

--001a114a6f02d4db910545dadc38--


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Subject: [MMUSIC] [Editorial Errata Reported] RFC4566 (4903)
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The following errata report has been submitted for RFC4566,
"SDP: Session Description Protocol".

--------------------------------------
You may review the report below and at:
http://www.rfc-editor.org/errata_search.php?rfc=4566&eid=4903

--------------------------------------
Type: Editorial
Reported by: Jxck <block.rxckin.beats@gmail.com>

Section: 9

Original Text
-------------
   ; SDP Syntax
   session-description = proto-version
                         origin-field
                         session-name-field
                         information-field
                         uri-field
                         email-fields
                         phone-fields
                         connection-field
                         bandwidth-fields
                         time-fields
                         key-field
                         attribute-fields
                         media-descriptions

   connection-field =    [%x63 "=" nettype SP addrtype SP
                         connection-address CRLF]
                         ;a connection field must be present
                         ;in every media description or at the
                         ;session-level

   media-descriptions =  *( media-field
                         information-field
                         *connection-field
                         bandwidth-fields
                         key-field
                         attribute-fields )


Corrected Text
--------------
   ; SDP Syntax
   session-description = proto-version
                         origin-field
                         session-name-field
                         information-field
                         uri-field
                         email-fields
                         phone-fields
                         [connection-field]
                         bandwidth-fields
                         time-fields
                         key-field
                         attribute-fields
                         media-descriptions

   connection-field =    %x63 "=" nettype SP addrtype SP
                         connection-address CRLF
                         ;a connection field must be present
                         ;in every media description or at the
                         ;session-level

   media-descriptions =  *( media-field
                         information-field
                         *connection-field
                         bandwidth-fields
                         key-field
                         attribute-fields )


Notes
-----
in BNF of SDP, 
connection-field itself is optional [0, 1], but media-description has multiple [0,) connection-field.
it seems conflict, because multiple of optional fields doesn't finish because multiple filed can't find when to end.

so change connection-field itself not optional (remove []) and, replace session-descriptions connection-field as optional.

Instructions:
-------------
This erratum is currently posted as "Reported". If necessary, please
use "Reply All" to discuss whether it should be verified or
rejected. When a decision is reached, the verifying party  
can log in to change the status and edit the report, if necessary. 

--------------------------------------
RFC4566 (draft-ietf-mmusic-sdp-new-26)
--------------------------------------
Title               : SDP: Session Description Protocol
Publication Date    : July 2006
Author(s)           : M. Handley, V. Jacobson, C. Perkins
Category            : PROPOSED STANDARD
Source              : Multiparty Multimedia Session Control RAI
Area                : Real-time Applications and Infrastructure
Stream              : IETF
Verifying Party     : IESG


From nobody Wed Jan 11 23:49:15 2017
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To: Roman Shpount <roman@telurix.com>
References: <CAD5OKxvEqvah+ZJdPKJdGmKob84X8WaCqKQ2GEiatVbOSmEjDw@mail.gmail.com>
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [MMUSIC] Transport tags
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Hi,

I think one should actually use the DTLS or TLS depending on what one 
requires. It one requires DTLS, then use DTLS, if is working with plain 
TLS, then use TLS. The UDP/TLS/RTP/SAVPF is part of the first DTLS-SRTP 
registrations that was done. As plain TLS can't work in this particular 
combination due to UDP there is no actual risk for having to seperate 
between TLS and DTLS. However, if one stack is TCP/FRAMING/(D)TLS/FOO 
then there are potentially important differences if this is TLS or DTLS.

Cheers

Magnus

Den 2017-01-11 kl. 22:01, skrev Roman Shpount:
> What should we use for the new protocols that we are defining?
>
> Should it be UDP/TLS/SCTP or UDP/DTLS/SCTP?
> Should it be UDP/TLS/BFCP or UDP/DTLS/BFCP?
>
> I would like to see some consistency here.
>
> Regards,
> _____________
> Roman Shpount
>
> On Wed, Jan 11, 2017 at 3:30 AM, Magnus Westerlund
> <magnus.westerlund@ericsson.com <mailto:magnus.westerlund@ericsson.com>>
> wrote:
>
>     Den 2017-01-10 kl. 19:08, skrev Roman Shpount:
>
>         P.S. In the quoted text someone mentioned "UDP/TLS/RTP/SAVPF",
>         should it
>         be "UDP/DTLS/RTP/SAVPF"?
>
>
>     No, actually UDP/TLS/RTP/SAVPF is what is registered for DTLS-SRTP
>     over UDP with RTP SAVPF profile. See [RFC5764]. The
>     "UDP/DTLS/RTP/SAVPF" is not registered.
>
>     So the alternatives for the dependency of default candidate protocol
>     type in your proposal would be:
>
>     UDP: "UDP/TLS/RTP/SAVPF"
>     TCP: "TCP/DTLS/RTP/SAVPF"
>
>
>     Cheers
>
>     Magnus Westerlund
>
>     ----------------------------------------------------------------------
>     Services, Media and Network features, Ericsson Research EAB/TXM
>     ----------------------------------------------------------------------
>     Ericsson AB                 | Phone  +46 10 7148287
>     <tel:%2B46%2010%207148287>
>     FÃ¤rÃ¶gatan 6                 | Mobile +46 73 0949079
>     <tel:%2B46%2073%200949079>
>     SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>     <mailto:magnus.westerlund@ericsson.com>
>     ----------------------------------------------------------------------
>
>


-- 

Magnus Westerlund

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


From nobody Thu Jan 12 06:33:40 2017
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Subject: Re: [MMUSIC] [Editorial Errata Reported] RFC4566 (4903)
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RFC Errata System <rfc-editor@rfc-editor.org> writes:
> Original Text

>    connection-field =    [%x63 "=" nettype SP addrtype SP
>                          connection-address CRLF]
>                          ;a connection field must be present
>                          ;in every media description or at the
>                          ;session-level

> Corrected Text

>    connection-field =    %x63 "=" nettype SP addrtype SP
>                          connection-address CRLF
>                          ;a connection field must be present
>                          ;in every media description or at the
>                          ;session-level
>

> in BNF of SDP, connection-field itself is optional [0, 1], but
> media-description has multiple [0,) connection-field.  it seems
> conflict, because multiple of optional fields doesn't finish because
> multiple filed can't find when to end.

As a grammar, a set of BNF productions, the original text is correct.
The erratum is correct that parser generators may have a difficult time
processing that grammar, but strictly speaking, that is not a problem
with the RFC.

Personally, I do not like the style of the grammar, which is to present
all of the alternatives as required and then each alternative contains
its own optionality.  That is,

   session-description = proto-version
                         origin-field
                         session-name-field
                         information-field
                         uri-field
                         email-fields
			 [...]

   information-field =   [%x69 "=" text CRLF]

   uri-field =           [%x75 "=" uri CRLF]

   email-fields =        *(%x65 "=" email-address CRLF)

I would prefer

   session-description = proto-version
                         origin-field
                         session-name-field
                         [information-field]
                         [uri-field]
                         *email-field
			 [...]

   information-field =   %x69 "=" text CRLF

   uri-field =           %x75 "=" uri CRLF

   email-field =         %x65 "=" email-address CRLF

But that is a style issue, and it is best to define connection-field
using the same style as every other field.

Dale


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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: Roman Shpount <roman@telurix.com>
Thread-Topic: [MMUSIC] Transport tags
Thread-Index: AQHSbE3p2SC5GZHiXUyd4kk6fM3mOqEzs4yAgACvV4CAAHqSgA==
Date: Thu, 12 Jan 2017 15:51:56 +0000
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References: <CAD5OKxvEqvah+ZJdPKJdGmKob84X8WaCqKQ2GEiatVbOSmEjDw@mail.gmail.com> <6A083F67-4ECB-4703-A881-EB2F074F309E@cisco.com> <CAD5OKxt9iVE8Sgax5rjzwyhNCupeXLCVDjz3iMU+A3eTjAnGoA@mail.gmail.com>
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Archived-At: <https://mailarchive.ietf.org/arch/msg/mmusic/KLGMQ_BwlyiMNa0wYDKxgW7__tI>
Cc: Magnus Westerlund <magnus.westerlund@ericsson.com>, Christer Holmberg <christer.holmberg@ericsson.com>, "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <paul.kyzivat@comcast.net>
Subject: Re: [MMUSIC] Transport tags
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From: Roman Shpount <roman@telurix.com>
Date: Thu, 12 Jan 2017 10:56:44 -0500
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--94eb2c033ce8940ac50545e7c3ae
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Magnus,

Does it mean for the future protocols we should do DTLS? As
in UDP/DTLS/BFCP and UDP/DTLS/SCTP?

And we should consider UDP/TLS/RTP as legacy that does not follow the
preferred rule.

Regards,

_____________
Roman Shpount

On Thu, Jan 12, 2017 at 2:48 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> Hi,
>
> I think one should actually use the DTLS or TLS depending on what one
> requires. It one requires DTLS, then use DTLS, if is working with plain
> TLS, then use TLS. The UDP/TLS/RTP/SAVPF is part of the first DTLS-SRTP
> registrations that was done. As plain TLS can't work in this particular
> combination due to UDP there is no actual risk for having to seperate
> between TLS and DTLS. However, if one stack is TCP/FRAMING/(D)TLS/FOO the=
n
> there are potentially important differences if this is TLS or DTLS.
>
> Cheers
>
> Magnus
>
> Den 2017-01-11 kl. 22:01, skrev Roman Shpount:
>
>> What should we use for the new protocols that we are defining?
>>
>> Should it be UDP/TLS/SCTP or UDP/DTLS/SCTP?
>> Should it be UDP/TLS/BFCP or UDP/DTLS/BFCP?
>>
>> I would like to see some consistency here.
>>
>> Regards,
>> _____________
>> Roman Shpount
>>
>> On Wed, Jan 11, 2017 at 3:30 AM, Magnus Westerlund
>> <magnus.westerlund@ericsson.com <mailto:magnus.westerlund@ericsson.com>>
>> wrote:
>>
>>     Den 2017-01-10 kl. 19:08, skrev Roman Shpount:
>>
>>         P.S. In the quoted text someone mentioned "UDP/TLS/RTP/SAVPF",
>>         should it
>>         be "UDP/DTLS/RTP/SAVPF"?
>>
>>
>>     No, actually UDP/TLS/RTP/SAVPF is what is registered for DTLS-SRTP
>>     over UDP with RTP SAVPF profile. See [RFC5764]. The
>>     "UDP/DTLS/RTP/SAVPF" is not registered.
>>
>>     So the alternatives for the dependency of default candidate protocol
>>     type in your proposal would be:
>>
>>     UDP: "UDP/TLS/RTP/SAVPF"
>>     TCP: "TCP/DTLS/RTP/SAVPF"
>>
>>
>>     Cheers
>>
>>     Magnus Westerlund
>>
>>     ------------------------------------------------------------
>> ----------
>>     Services, Media and Network features, Ericsson Research EAB/TXM
>>     ------------------------------------------------------------
>> ----------
>>     Ericsson AB                 | Phone  +46 10 7148287
>>     <tel:%2B46%2010%207148287>
>>     F=C3=A4r=C3=B6gatan 6                 | Mobile +46 73 0949079
>>     <tel:%2B46%2073%200949079>
>>     SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>>     <mailto:magnus.westerlund@ericsson.com>
>>     ------------------------------------------------------------
>> ----------
>>
>>
>>
>
> --
>
> 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
> ----------------------------------------------------------------------
>
>

--94eb2c033ce8940ac50545e7c3ae
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Magnus,<div><br></div><div>Does it mean for the future pro=
tocols we should do DTLS? As in=C2=A0UDP/DTLS/BFCP and UDP/DTLS/SCTP?</div>=
<div><br></div><div>And we should consider UDP/TLS/RTP as legacy that does =
not follow the preferred rule.</div><div><br></div><div>Regards,</div></div=
><div class=3D"gmail_extra"><br clear=3D"all"><div><div class=3D"gmail_sign=
ature" data-smartmail=3D"gmail_signature">_____________<br>Roman Shpount</d=
iv></div>
<br><div class=3D"gmail_quote">On Thu, Jan 12, 2017 at 2:48 AM, Magnus West=
erlund <span dir=3D"ltr">&lt;<a href=3D"mailto:magnus.westerlund@ericsson.c=
om" target=3D"_blank">magnus.westerlund@ericsson.com</a>&gt;</span> wrote:<=
br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left=
:1px #ccc solid;padding-left:1ex">Hi,<br>
<br>
I think one should actually use the DTLS or TLS depending on what one requi=
res. It one requires DTLS, then use DTLS, if is working with plain TLS, the=
n use TLS. The UDP/TLS/RTP/SAVPF is part of the first DTLS-SRTP registratio=
ns that was done. As plain TLS can&#39;t work in this particular combinatio=
n due to UDP there is no actual risk for having to seperate between TLS and=
 DTLS. However, if one stack is TCP/FRAMING/(D)TLS/FOO then there are poten=
tially important differences if this is TLS or DTLS.<br>
<br>
Cheers<br>
<br>
Magnus<span class=3D""><br>
<br>
Den 2017-01-11 kl. 22:01, skrev Roman Shpount:<br>
</span><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-=
left:1px #ccc solid;padding-left:1ex"><span class=3D"">
What should we use for the new protocols that we are defining?<br>
<br>
Should it be UDP/TLS/SCTP or UDP/DTLS/SCTP?<br>
Should it be UDP/TLS/BFCP or UDP/DTLS/BFCP?<br>
<br>
I would like to see some consistency here.<br>
<br>
Regards,<br>
_____________<br>
Roman Shpount<br>
<br>
On Wed, Jan 11, 2017 at 3:30 AM, Magnus Westerlund<br></span>
&lt;<a href=3D"mailto:magnus.westerlund@ericsson.com" target=3D"_blank">mag=
nus.westerlund@ericsson.co<wbr>m</a> &lt;mailto:<a href=3D"mailto:magnus.we=
sterlund@ericsson.com" target=3D"_blank">magnus.westerlund@eric<wbr>sson.co=
m</a>&gt;&gt;<span class=3D""><br>
wrote:<br>
<br>
=C2=A0 =C2=A0 Den 2017-01-10 kl. 19:08, skrev Roman Shpount:<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 P.S. In the quoted text someone mentioned &quot=
;UDP/TLS/RTP/SAVPF&quot;,<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 should it<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 be &quot;UDP/DTLS/RTP/SAVPF&quot;?<br>
<br>
<br>
=C2=A0 =C2=A0 No, actually UDP/TLS/RTP/SAVPF is what is registered for DTLS=
-SRTP<br>
=C2=A0 =C2=A0 over UDP with RTP SAVPF profile. See [RFC5764]. The<br>
=C2=A0 =C2=A0 &quot;UDP/DTLS/RTP/SAVPF&quot; is not registered.<br>
<br>
=C2=A0 =C2=A0 So the alternatives for the dependency of default candidate p=
rotocol<br>
=C2=A0 =C2=A0 type in your proposal would be:<br>
<br>
=C2=A0 =C2=A0 UDP: &quot;UDP/TLS/RTP/SAVPF&quot;<br>
=C2=A0 =C2=A0 TCP: &quot;TCP/DTLS/RTP/SAVPF&quot;<br>
<br>
<br>
=C2=A0 =C2=A0 Cheers<br>
<br>
=C2=A0 =C2=A0 Magnus Westerlund<br>
<br>
=C2=A0 =C2=A0 ------------------------------<wbr>--------------------------=
----<wbr>----------<br>
=C2=A0 =C2=A0 Services, Media and Network features, Ericsson Research EAB/T=
XM<br>
=C2=A0 =C2=A0 ------------------------------<wbr>--------------------------=
----<wbr>----------<br>
=C2=A0 =C2=A0 Ericsson AB=C2=A0 =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"+4=
6107148287" target=3D"_blank">+46 10 7148287</a><br></span>
=C2=A0 =C2=A0 &lt;tel:%2B46%2010%207148287&gt;<span class=3D""><br>
=C2=A0 =C2=A0 F=C3=A4r=C3=B6gatan 6=C2=A0 =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" target=3D"_blank">+46 73 0949079</a><br></span>
=C2=A0 =C2=A0 &lt;tel:%2B46%2073%200949079&gt;<span class=3D""><br>
=C2=A0 =C2=A0 SE-164 80 Stockholm, Sweden | mailto: <a href=3D"mailto:magnu=
s.westerlund@ericsson.com" target=3D"_blank">magnus.westerlund@ericsson.com=
</a><br></span>
=C2=A0 =C2=A0 &lt;mailto:<a href=3D"mailto:magnus.westerlund@ericsson.com" =
target=3D"_blank">magnus.westerlund@eric<wbr>sson.com</a>&gt;<br>
=C2=A0 =C2=A0 ------------------------------<wbr>--------------------------=
----<wbr>----------<br>
<br>
<br><span class=3D"HOEnZb"><font color=3D"#888888">
</font></span></blockquote><span class=3D"HOEnZb"><font color=3D"#888888">
<br>
<br>
-- <br></font></span><div class=3D"HOEnZb"><div class=3D"h5">
<br>
Magnus Westerlund<br>
<br>
------------------------------<wbr>------------------------------<wbr>-----=
-----<br>
Services, Media and Network features, Ericsson Research EAB/TXM<br>
------------------------------<wbr>------------------------------<wbr>-----=
-----<br>
Ericsson AB=C2=A0 =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" tar=
get=3D"_blank">+46 10 7148287</a><br>
F=C3=A4r=C3=B6gatan 6=C2=A0 =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"+467309490=
79" target=3D"_blank">+46 73 0949079</a><br>
SE-164 80 Stockholm, Sweden | mailto: <a href=3D"mailto:magnus.westerlund@e=
ricsson.com" target=3D"_blank">magnus.westerlund@ericsson.com</a><br>
------------------------------<wbr>------------------------------<wbr>-----=
-----<br>
<br>
</div></div></blockquote></div><br></div>

--94eb2c033ce8940ac50545e7c3ae--


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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [MMUSIC] Transport tags
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Den 2017-01-12 kl. 16:56, skrev Roman Shpount:
> Magnus,
>
> Does it mean for the future protocols we should do DTLS? As
> in UDP/DTLS/BFCP and UDP/DTLS/SCTP?

Yes, I think so given that the protocol is DTLS and not TLS.

>
> And we should consider UDP/TLS/RTP as legacy that does not follow the
> preferred rule.

I would.

Cheers

Magnus Westerlund

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From: Christer Holmberg <christer.holmberg@ericsson.com>
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From: Roman Shpount <roman@telurix.com>
Date: Thu, 12 Jan 2017 15:46:00 -0500
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: Magnus Westerlund <magnus.westerlund@ericsson.com>, "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <paul.kyzivat@comcast.net>
Subject: Re: [MMUSIC] Transport tags
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Hi Christer,

Definition for TCP/DTLS/BFCP needs to be added
in draft-ietf-bfcpbis-rfc4583bis.

This is BFCP over DTLS over TCP with RFC 4571 framing. UDP/DTLS/BFCP
together with TCP/DTLS/BFCP will over ICE based on the mechanisms defined
in draft-ietf-mmusic-dtls-sdp. Only UDP/DTLS/BFCP in combination with
TCP/DTLS/BFCP will support ICE. TCP/BFCP and TLS/BFCP will not support ICE
and will have two run with at least one end point on the public IP.

I hope it makes sense.

Regards,

_____________
Roman Shpount

On Thu, Jan 12, 2017 at 3:34 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
>
>
> Now sure I follow: Where TCP with DTLS for BFCP defined? Doesn=E2=80=99t =
TCP based
> BFCP run over TLS?
>
>
>
> That is also what is used in RFC 4583.
>
>
>
> As Roman has indicated, TCP/BFCP and TLS/BFCP won=E2=80=99t work on ICE. =
But, if
> you want to fix, don=E2=80=99t you then have to do more than just change =
the m-
> proto value? Don=E2=80=99t you have to define usage of the framing mechan=
ism?
>
>
>
> Regards,
>
>
>
> Christer
>
>
>
> *From:* Charles Eckel (eckelcu) [mailto:eckelcu@cisco.com]
> *Sent:* 12 January 2017 17:52
> *To:* Roman Shpount <roman@telurix.com>
> *Cc:* Magnus Westerlund <magnus.westerlund@ericsson.com>; Paul Kyzivat <
> paul.kyzivat@comcast.net>; mmusic@ietf.org; Christer Holmberg <
> christer.holmberg@ericsson.com>
>
> *Subject:* Re: [MMUSIC] Transport tags
>
>
>
> Works for me.
>
>
>
> Cheers,
>
> Charles
>
>
>
> *From: *Roman Shpount <roman@telurix.com>
> *Date: *Wednesday, January 11, 2017 at 4:33 PM
> *To: *Charles Eckel <eckelcu@cisco.com>
> *Cc: *Magnus Westerlund <magnus.westerlund@ericsson.com>, Paul Kyzivat <
> paul.kyzivat@comcast.net>, "mmusic@ietf.org" <mmusic@ietf.org>, Christer
> Holmberg <christer.holmberg@ericsson.com>
> *Subject: *Re: [MMUSIC] Transport tags
>
>
>
> I guess BFCP will need TCP/DTLS/BFCP added to the list.
>
>
>
> Should we correct draft-ietf-mmusic-sctp-sdp draft to match and change
> UDP/DTLS/SCTP to UDP/TLS/SCTP?
>
>
>
> Regards,
>
>
> _____________
> Roman Shpount
>
>
>
> On Wed, Jan 11, 2017 at 5:05 PM, Charles Eckel (eckelcu) <
> eckelcu@cisco.com> wrote:
>
> Here is what we have for BFCP in https://tools.ietf.org/html/
> draft-ietf-bfcpbis-rfc4583bis-16#section-13:
>
>
>
>                        +--------------+------------+
>
>                        | Value        | Reference  |
>
>                        +--------------+------------+
>
>                        | TCP/BFCP     | [RFC XXXX] |
>
>                        | TCP/TLS/BFCP | [RFC XXXX] |
>
>                        | UDP/BFCP     | [RFC XXXX] |
>
>                        | UDP/TLS/BFCP | [RFC XXXX] |
>
>                        +--------------+------------+
>
>
>
> Cheers,
>
> Charles
>
>
>
> *From: *mmusic <mmusic-bounces@ietf.org> on behalf of Roman Shpount <
> roman@telurix.com>
> *Date: *Wednesday, January 11, 2017 at 1:01 PM
> *To: *Magnus Westerlund <magnus.westerlund@ericsson.com>
> *Cc: *Paul Kyzivat <paul.kyzivat@comcast.net>, "mmusic@ietf.org" <
> mmusic@ietf.org>
> *Subject: *[MMUSIC] Transport tags
>
>
>
> What should we use for the new protocols that we are defining?
>
>
>
> Should it be UDP/TLS/SCTP or UDP/DTLS/SCTP?
>
> Should it be UDP/TLS/BFCP or UDP/DTLS/BFCP?
>
>
>
> I would like to see some consistency here.
>
>
>
> Regards,
>
> _____________
> Roman Shpount
>
>
>
> On Wed, Jan 11, 2017 at 3:30 AM, Magnus Westerlund <
> magnus.westerlund@ericsson.com> wrote:
>
> Den 2017-01-10 kl. 19:08, skrev Roman Shpount:
>
> P.S. In the quoted text someone mentioned "UDP/TLS/RTP/SAVPF", should it
> be "UDP/DTLS/RTP/SAVPF"?
>
>
> No, actually UDP/TLS/RTP/SAVPF is what is registered for DTLS-SRTP over
> UDP with RTP SAVPF profile. See [RFC5764]. The "UDP/DTLS/RTP/SAVPF" is no=
t
> registered.
>
> So the alternatives for the dependency of default candidate protocol type
> in your proposal would be:
>
> UDP: "UDP/TLS/RTP/SAVPF"
> TCP: "TCP/DTLS/RTP/SAVPF"
>
>
>
> 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
> ----------------------------------------------------------------------
>
>
>
>
>
>

--94eb2c0830de1e957f0545ebce2a
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Hi Christer,<div><br></div><div>Definition for TCP/DTLS/BF=
CP needs to be added in=C2=A0draft-ietf-bfcpbis-rfc4583bis.</div><div><br><=
/div><div>This is BFCP over DTLS over TCP with RFC 4571 framing. UDP/DTLS/B=
FCP together with TCP/DTLS/BFCP=C2=A0will over ICE based on the mechanisms =
defined in draft-ietf-mmusic-dtls-sdp. Only UDP/DTLS/BFCP in combination wi=
th TCP/DTLS/BFCP will support ICE. TCP/BFCP and TLS/BFCP will not support I=
CE and will have two run with at least one end point on the public IP.</div=
><div><br></div><div>I hope it makes sense.</div><div><br></div><div>Regard=
s,</div></div><div class=3D"gmail_extra"><br clear=3D"all"><div><div class=
=3D"gmail_signature" data-smartmail=3D"gmail_signature">_____________<br>Ro=
man Shpount</div></div>
<br><div class=3D"gmail_quote">On Thu, Jan 12, 2017 at 3:34 PM, Christer Ho=
lmberg <span dir=3D"ltr">&lt;<a href=3D"mailto:christer.holmberg@ericsson.c=
om" target=3D"_blank">christer.holmberg@ericsson.com</a>&gt;</span> wrote:<=
br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left=
:1px #ccc solid;padding-left:1ex">





<div bgcolor=3D"white" lang=3D"EN-GB" link=3D"blue" vlink=3D"purple">
<div class=3D"m_9209037337826514988WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;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;,sans-serif;color:#1f497d"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1f497d">Now sure I follow: Where TCP with DTL=
S for BFCP defined? Doesn=E2=80=99t TCP based BFCP run over TLS?<u></u><u><=
/u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1f497d"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1f497d">That is also what is used in RFC 4583=
.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1f497d"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1f497d">As Roman has indicated, TCP/BFCP and =
TLS/BFCP won=E2=80=99t work on ICE. But, if you want to fix, don=E2=80=99t =
you then have to do more than just
 change the m- proto value? Don=E2=80=99t you have to define usage of the f=
raming mechanism?<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1f497d"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1f497d">Regards,<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1f497d"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1f497d">Christer<u></u><u></u></span></p>
<p class=3D"MsoNormal"><a name=3D"m_9209037337826514988__MailEndCompose"><s=
pan style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;co=
lor:#1f497d"><u></u>=C2=A0<u></u></span></a></p>
<div>
<div style=3D"border:none;border-top:solid #e1e1e1 1.0pt;padding:3.0pt 0cm =
0cm 0cm">
<p class=3D"MsoNormal"><b><span lang=3D"EN-US" style=3D"font-size:11.0pt;fo=
nt-family:&quot;Calibri&quot;,sans-serif">From:</span></b><span lang=3D"EN-=
US" style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif"> =
Charles Eckel (eckelcu) [mailto:<a href=3D"mailto:eckelcu@cisco.com" target=
=3D"_blank">eckelcu@cisco.com</a>]
<br>
<b>Sent:</b> 12 January 2017 17:52<br>
<b>To:</b> Roman Shpount &lt;<a href=3D"mailto:roman@telurix.com" target=3D=
"_blank">roman@telurix.com</a>&gt;<br>
<b>Cc:</b> Magnus Westerlund &lt;<a href=3D"mailto:magnus.westerlund@ericss=
on.com" target=3D"_blank">magnus.westerlund@ericsson.<wbr>com</a>&gt;; Paul=
 Kyzivat &lt;<a href=3D"mailto:paul.kyzivat@comcast.net" target=3D"_blank">=
paul.kyzivat@comcast.net</a>&gt;; <a href=3D"mailto:mmusic@ietf.org" target=
=3D"_blank">mmusic@ietf.org</a>; Christer Holmberg &lt;<a href=3D"mailto:ch=
rister.holmberg@ericsson.com" target=3D"_blank">christer.holmberg@ericsson.=
<wbr>com</a>&gt;</span></p><div><div class=3D"h5"><br>
<b>Subject:</b> Re: [MMUSIC] Transport tags<u></u><u></u></div></div><p></p=
>
</div>
</div><div><div class=3D"h5">
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Works for me.<u></u><u></u></sp=
an></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Cheers,<u></u><u></u></span></p=
>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Charles<u></u><u></u></span></p=
>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
<blockquote style=3D"border:none;border-left:solid #b5c4df 4.5pt;padding:0c=
m 0cm 0cm 4.0pt;margin-left:3.75pt;margin-top:5.0pt;margin-right:0cm;margin=
-bottom:5.0pt">
<div style=3D"border:none;border-top:solid #b5c4df 1.0pt;padding:3.0pt 0cm =
0cm 0cm">
<p class=3D"MsoNormal"><b><span lang=3D"EN-US" style=3D"font-family:&quot;C=
alibri&quot;,sans-serif;color:black">From:
</span></b><span lang=3D"EN-US" style=3D"font-family:&quot;Calibri&quot;,sa=
ns-serif;color:black">Roman Shpount &lt;</span><a href=3D"mailto:roman@telu=
rix.com" target=3D"_blank"><span lang=3D"EN-US" style=3D"font-family:&quot;=
Calibri&quot;,sans-serif">roman@telurix.com</span></a><span lang=3D"EN-US" =
style=3D"font-family:&quot;Calibri&quot;,sans-serif;color:black">&gt;<br>
<b>Date: </b>Wednesday, January 11, 2017 at 4:33 PM<br>
<b>To: </b>Charles Eckel &lt;</span><a href=3D"mailto:eckelcu@cisco.com" ta=
rget=3D"_blank"><span lang=3D"EN-US" style=3D"font-family:&quot;Calibri&quo=
t;,sans-serif">eckelcu@cisco.com</span></a><span lang=3D"EN-US" style=3D"fo=
nt-family:&quot;Calibri&quot;,sans-serif;color:black">&gt;<br>
<b>Cc: </b>Magnus Westerlund &lt;</span><a href=3D"mailto:magnus.westerlund=
@ericsson.com" target=3D"_blank"><span lang=3D"EN-US" style=3D"font-family:=
&quot;Calibri&quot;,sans-serif">magnus.westerlund@ericsson.<wbr>com</span><=
/a><span lang=3D"EN-US" style=3D"font-family:&quot;Calibri&quot;,sans-serif=
;color:black">&gt;,
 Paul Kyzivat &lt;</span><a href=3D"mailto:paul.kyzivat@comcast.net" target=
=3D"_blank"><span lang=3D"EN-US" style=3D"font-family:&quot;Calibri&quot;,s=
ans-serif">paul.kyzivat@comcast.net</span></a><span lang=3D"EN-US" style=3D=
"font-family:&quot;Calibri&quot;,sans-serif;color:black">&gt;, &quot;</span=
><a href=3D"mailto:mmusic@ietf.org" target=3D"_blank"><span lang=3D"EN-US" =
style=3D"font-family:&quot;Calibri&quot;,sans-serif">mmusic@ietf.org</span>=
</a><span lang=3D"EN-US" style=3D"font-family:&quot;Calibri&quot;,sans-seri=
f;color:black">&quot;
 &lt;</span><a href=3D"mailto:mmusic@ietf.org" target=3D"_blank"><span lang=
=3D"EN-US" style=3D"font-family:&quot;Calibri&quot;,sans-serif">mmusic@ietf=
.org</span></a><span lang=3D"EN-US" style=3D"font-family:&quot;Calibri&quot=
;,sans-serif;color:black">&gt;, Christer Holmberg &lt;</span><a href=3D"mai=
lto:christer.holmberg@ericsson.com" target=3D"_blank"><span lang=3D"EN-US" =
style=3D"font-family:&quot;Calibri&quot;,sans-serif">christer.holmberg@eric=
sson.<wbr>com</span></a><span lang=3D"EN-US" style=3D"font-family:&quot;Cal=
ibri&quot;,sans-serif;color:black">&gt;<br>
<b>Subject: </b>Re: [MMUSIC] Transport tags<u></u><u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US">I guess BFCP will need TCP/DTLS=
/BFCP added to the list.<u></u><u></u></span></p>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
</div>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Should we correct draft-ietf-mm=
usic-sctp-sdp draft to match and change UDP/DTLS/SCTP to UDP/TLS/SCTP?
<u></u><u></u></span></p>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Regards,<u></u><u></u></span></=
p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><br clear=3D"all">
<u></u><u></u></span></p>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US">_____________<br>
Roman Shpount<u></u><u></u></span></p>
</div>
</div>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US">On Wed, Jan 11, 2017 at 5:05 PM=
, Charles Eckel (eckelcu) &lt;</span><a href=3D"mailto:eckelcu@cisco.com" t=
arget=3D"_blank"><span lang=3D"EN-US">eckelcu@cisco.com</span></a><span lan=
g=3D"EN-US">&gt; wrote:<u></u><u></u></span></p>
<blockquote style=3D"border:none;border-left:solid #cccccc 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0cm;margin-=
bottom:5.0pt">
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Here is what we have for BFCP i=
n
</span><a href=3D"https://tools.ietf.org/html/draft-ietf-bfcpbis-rfc4583bis=
-16#section-13" target=3D"_blank"><span lang=3D"EN-US">https://tools.ietf.o=
rg/html/<wbr>draft-ietf-bfcpbis-rfc4583bis-<wbr>16#section-13</span></a><sp=
an lang=3D"EN-US">:<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">=C2=A0<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:&quot;Courier New&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0 +--------------+------------+</span><span lang=3D"EN-US"><u=
></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:&quot;Courier New&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0 | Value=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 | Referen=
ce=C2=A0 |</span><span lang=3D"EN-US"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:&quot;Courier New&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0 +--------------+------------+</span><span lang=3D"EN-US"><u=
></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:&quot;Courier New&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0 | TCP/BFCP=C2=A0=C2=A0=C2=A0=C2=A0 | [RFC XXXX] |</span><sp=
an lang=3D"EN-US"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:&quot;Courier New&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0 =C2=A0| TCP/TLS/BFCP | [RFC XXXX] |</span><span lang=3D"EN-US"><u=
></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:&quot;Courier New&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0 | UDP/BFCP=C2=A0=C2=A0=C2=A0=C2=A0 | [RFC XXXX] |</span><sp=
an lang=3D"EN-US"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:&quot;Courier New&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0 | UDP/TLS/BFCP | [RFC XXXX] |</span><span lang=3D"EN-US"><u=
></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:&quot;Courier New&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0 +--------------+------------+</span><span lang=3D"EN-US"><u=
></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">=C2=A0<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Cheers,<u></u><u></u></span></p=
>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Charles<u></u><u></u></span></p=
>
<p class=3D"MsoNormal"><span lang=3D"EN-US">=C2=A0<u></u><u></u></span></p>
<blockquote style=3D"border:none;border-left:solid #b5c4df 4.5pt;padding:0c=
m 0cm 0cm 4.0pt;margin-left:3.75pt;margin-top:5.0pt;margin-right:0cm;margin=
-bottom:5.0pt">
<div style=3D"border:none;border-top:solid #b5c4df 1.0pt;padding:3.0pt 0cm =
0cm 0cm">
<p class=3D"MsoNormal"><b><span lang=3D"EN-US" style=3D"font-family:&quot;C=
alibri&quot;,sans-serif;color:black">From:
</span></b><span lang=3D"EN-US" style=3D"font-family:&quot;Calibri&quot;,sa=
ns-serif;color:black">mmusic &lt;</span><a href=3D"mailto:mmusic-bounces@ie=
tf.org" target=3D"_blank"><span lang=3D"EN-US" style=3D"font-family:&quot;C=
alibri&quot;,sans-serif">mmusic-bounces@ietf.org</span></a><span lang=3D"EN=
-US" style=3D"font-family:&quot;Calibri&quot;,sans-serif;color:black">&gt;
 on behalf of Roman Shpount &lt;</span><a href=3D"mailto:roman@telurix.com"=
 target=3D"_blank"><span lang=3D"EN-US" style=3D"font-family:&quot;Calibri&=
quot;,sans-serif">roman@telurix.com</span></a><span lang=3D"EN-US" style=3D=
"font-family:&quot;Calibri&quot;,sans-serif;color:black">&gt;<br>
<b>Date: </b>Wednesday, January 11, 2017 at 1:01 PM<br>
<b>To: </b>Magnus Westerlund &lt;</span><a href=3D"mailto:magnus.westerlund=
@ericsson.com" target=3D"_blank"><span lang=3D"EN-US" style=3D"font-family:=
&quot;Calibri&quot;,sans-serif">magnus.westerlund@ericsson.<wbr>com</span><=
/a><span lang=3D"EN-US" style=3D"font-family:&quot;Calibri&quot;,sans-serif=
;color:black">&gt;<br>
<b>Cc: </b>Paul Kyzivat &lt;</span><a href=3D"mailto:paul.kyzivat@comcast.n=
et" target=3D"_blank"><span lang=3D"EN-US" style=3D"font-family:&quot;Calib=
ri&quot;,sans-serif">paul.kyzivat@comcast.net</span></a><span lang=3D"EN-US=
" style=3D"font-family:&quot;Calibri&quot;,sans-serif;color:black">&gt;,
 &quot;</span><a href=3D"mailto:mmusic@ietf.org" target=3D"_blank"><span la=
ng=3D"EN-US" style=3D"font-family:&quot;Calibri&quot;,sans-serif">mmusic@ie=
tf.org</span></a><span lang=3D"EN-US" style=3D"font-family:&quot;Calibri&qu=
ot;,sans-serif;color:black">&quot; &lt;</span><a href=3D"mailto:mmusic@ietf=
.org" target=3D"_blank"><span lang=3D"EN-US" style=3D"font-family:&quot;Cal=
ibri&quot;,sans-serif">mmusic@ietf.org</span></a><span lang=3D"EN-US" style=
=3D"font-family:&quot;Calibri&quot;,sans-serif;color:black">&gt;<br>
<b>Subject: </b>[MMUSIC] Transport tags</span><span lang=3D"EN-US"><u></u><=
u></u></span></p>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US">=C2=A0<u></u><u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US">What should we use for the new =
protocols that we are defining?
<u></u><u></u></span></p>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US">=C2=A0<u></u><u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Should it be UDP/TLS/SCTP or UD=
P/DTLS/SCTP?<u></u><u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Should it be UDP/TLS/BFCP or UD=
P/DTLS/BFCP?<u></u><u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US">=C2=A0<u></u><u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US">I would like to see some consis=
tency here.<u></u><u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US">=C2=A0<u></u><u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Regards,
<u></u><u></u></span></p>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US">_____________<br>
Roman Shpount<u></u><u></u></span></p>
</div>
</div>
<p class=3D"MsoNormal"><span lang=3D"EN-US">=C2=A0<u></u><u></u></span></p>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US">On Wed, Jan 11, 2017 at 3:30 AM=
, Magnus Westerlund &lt;</span><a href=3D"mailto:magnus.westerlund@ericsson=
.com" target=3D"_blank"><span lang=3D"EN-US">magnus.westerlund@ericsson.<wb=
r>com</span></a><span lang=3D"EN-US">&gt;
 wrote:<u></u><u></u></span></p>
<blockquote style=3D"border:none;border-left:solid #cccccc 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0cm;margin-=
bottom:5.0pt">
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span class=3D"m_9209=
037337826514988m3936043015517947387gmail-"><span lang=3D"EN-US">Den 2017-01=
-10 kl. 19:08, skrev Roman Shpount:</span></span><span lang=3D"EN-US"><u></=
u><u></u></span></p>
<blockquote style=3D"border:none;border-left:solid #cccccc 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0cm;margin-=
bottom:5.0pt">
<p class=3D"MsoNormal"><span lang=3D"EN-US">P.S. In the quoted text someone=
 mentioned &quot;UDP/TLS/RTP/SAVPF&quot;, should it<br>
be &quot;UDP/DTLS/RTP/SAVPF&quot;?<u></u><u></u></span></p>
</blockquote>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><br>
No, actually UDP/TLS/RTP/SAVPF is what is registered for DTLS-SRTP over UDP=
 with RTP SAVPF profile. See [RFC5764]. The &quot;UDP/DTLS/RTP/SAVPF&quot; =
is not registered.<br>
<br>
So the alternatives for the dependency of default candidate protocol type i=
n your proposal would be:<br>
<br>
UDP: &quot;UDP/TLS/RTP/SAVPF&quot;<br>
TCP: &quot;TCP/DTLS/RTP/SAVPF&quot; <u></u><u></u></span></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=3D"EN-US">=
<br>
<br>
Cheers<br>
<br>
Magnus Westerlund<br>
<br>
------------------------------<wbr>------------------------------<wbr>-----=
-----<br>
Services, Media and Network features, Ericsson Research EAB/TXM<br>
------------------------------<wbr>------------------------------<wbr>-----=
-----<br>
Ericsson AB=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0| =
Phone=C2=A0 </span><a href=3D"tel:%2B46%2010%207148287" target=3D"_blank"><=
span lang=3D"EN-US">+46 10 7148287</span></a><span lang=3D"EN-US"><br>
F=C3=A4r=C3=B6gatan 6=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0| Mobile </span><a href=3D"tel:%2B46%2073%200949079" target=3D"_b=
lank"><span lang=3D"EN-US">+46 73 0949079</span></a><span lang=3D"EN-US"><b=
r>
SE-164 80 Stockholm, Sweden | mailto: </span><a href=3D"mailto:magnus.weste=
rlund@ericsson.com" target=3D"_blank"><span lang=3D"EN-US">magnus.westerlun=
d@ericsson.com</span></a><span lang=3D"EN-US"><br>
------------------------------<wbr>------------------------------<wbr>-----=
-----<u></u><u></u></span></p>
</div>
</div>
</blockquote>
</div>
<p class=3D"MsoNormal"><span lang=3D"EN-US">=C2=A0<u></u><u></u></span></p>
</div>
</div>
</div>
</div>
</div>
</blockquote>
</div>
</div>
</blockquote>
</div>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
</div>
</blockquote>
</div></div></div>
</div>

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

--94eb2c0830de1e957f0545ebce2a--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Roman Shpount <roman@telurix.com>
Thread-Topic: [MMUSIC] Transport tags
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Subject: Re: [MMUSIC] Transport tags
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From: Roman Shpount <roman@telurix.com>
Date: Thu, 12 Jan 2017 16:27:57 -0500
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Cc: Magnus Westerlund <magnus.westerlund@ericsson.com>, "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <paul.kyzivat@comcast.net>
Subject: Re: [MMUSIC] Transport tags
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On Thu, Jan 12, 2017 at 4:08 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
>
>
> >Definition for TCP/DTLS/BFCP needs to be added in draft-ietf-bfcpbis-
> rfc4583bis.
>
> >
>
> >This is BFCP over DTLS over TCP with RFC 4571 framing.
>
>
>
> Correct.
>
>
>
> >UDP/DTLS/BFCP together with TCP/DTLS/BFCP will over ICE based on the
> mechanisms defined in draft-ietf-mmusic-dtls-sdp. Only UDP/DTLS/BFCP >in
> combination with TCP/DTLS/BFCP will support ICE. TCP/BFCP and TLS/BFCP will
> not support ICE and will have two run with at least one end point >on the
> public IP.
>
>
>
> I agree with all of that, but that was not my issue :)
>
>
>
> My issue was that it is not enough to simply change the proto value in
> 4583bis. You also need to define the usage of the RFC 4571 framing
> mechanism.
>
>
>
> And, if that is not backward compatible with RFC 4583 (which does NOT use
> 4571 framing), you need to add some text about that also.
>
>
>
I do not think TCP/DTLS/BFCP is defined anywhere, so it does not have to be
backwards compatible. We are not changing the proto value, we should define
a new one and explain how it works. This leaves TCP/BFCP and TLS/BFCP as
they were previously defined and simply states that these transports cannot
be used with ICE. Only UDP/DTLS/BFCP and TCP/DTLS/BFCP can be used with
ICE, with ICE not supported for all the other BFCP flavors.

Only issue I see here is that an offer of BFCP with ICE will not be
backwards compatible with RFC 4583 implementations, but considering that
RFC4583 did not define BFCP over DTLS at all, we should be fine here.

Regards,
_____________
Roman Shpount

--001a114d7c7a1c04e00545ec64ed
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure"><br></div></div><div class=3D"gmail_quote">On Thu, Jan 12, 2017 at 4:0=
8 PM, Christer Holmberg <span dir=3D"ltr">&lt;<a href=3D"mailto:christer.ho=
lmberg@ericsson.com" target=3D"_blank">christer.holmberg@ericsson.com</a>&g=
t;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0px 0=
px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">





<div lang=3D"EN-GB">
<div class=3D"gmail-m_-2807287925862297963WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:calibri,sa=
ns-serif;color:rgb(31,73,125)">Hi,<u></u><u></u></span></p>
<div><span class=3D"gmail-">
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">&gt;Definition for TCP/DTLS/BFCP needs to be added i=
n=C2=A0draft-ietf-bfcpbis-<wbr>rfc4583bis.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">&gt;<u></u>=C2=A0<u></u></p>
</div>
</span><div><span class=3D"gmail-">
<p class=3D"MsoNormal">&gt;This is BFCP over DTLS over TCP with RFC 4571 fr=
aming. <u></u>
<u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:calibri,sa=
ns-serif"><u></u>=C2=A0<u></u></span></p>
</span><p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:cal=
ibri,sans-serif">Correct.<u></u><u></u></span></p><span class=3D"gmail-">
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:calibri,sa=
ns-serif"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal">&gt;UDP/DTLS/BFCP together with TCP/DTLS/BFCP=C2=A0w=
ill over ICE based on the mechanisms defined in draft-ietf-mmusic-dtls-sdp.=
 Only UDP/DTLS/BFCP &gt;in combination with TCP/DTLS/BFCP will support ICE.=
 TCP/BFCP and TLS/BFCP will not support ICE and
 will have two run with at least one end point &gt;on the public IP.<u></u>=
<u></u></p>
</span></div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:calibri,sa=
ns-serif">I agree with all of that, but that was not my issue :)<u></u><u><=
/u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:calibri,sa=
ns-serif"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:calibri,sa=
ns-serif">My issue was that it is not enough to simply change the proto val=
ue in 4583bis. You also need to define the usage of the RFC 4571 framing me=
chanism.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:calibri,sa=
ns-serif"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:calibri,sa=
ns-serif">And, if that is not backward compatible with RFC 4583 (which does=
 NOT use 4571 framing), you need to add some text about that also.<u></u><u=
></u></span></p>
<p class=3D"MsoNormal"><br></p></div></div></div></div></blockquote><div><b=
r></div><div>I do not think TCP/DTLS/BFCP is defined anywhere, so it does n=
ot have to be backwards compatible. We are not changing the proto value, we=
 should define a new one and explain how it works. This leaves TCP/BFCP and=
 TLS/BFCP as they were previously defined and simply states that these tran=
sports cannot be used with ICE. Only UDP/DTLS/BFCP and TCP/DTLS/BFCP can be=
 used with ICE, with ICE not supported for all the other BFCP flavors.=C2=
=A0</div><div><br></div><div>Only issue I see here is that an offer of BFCP=
 with ICE will not be backwards compatible with RFC 4583 implementations, b=
ut considering that RFC4583 did not define BFCP over DTLS at all, we should=
 be fine here.</div><div><br></div><div>Regards,</div><div><div class=3D"gm=
ail_signature">_____________<br>Roman Shpount</div></div><div>=C2=A0</div><=
/div></div></div>

--001a114d7c7a1c04e00545ec64ed--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Roman Shpount <roman@telurix.com>
Thread-Topic: [MMUSIC] Transport tags
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Subject: Re: [MMUSIC] Transport tags
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From: Roman Shpount <roman@telurix.com>
Date: Thu, 12 Jan 2017 17:15:03 -0500
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [MMUSIC] Transport tags
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--001a1149e64093561d0545ed0c50
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On Thu, Jan 12, 2017 at 4:45 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> >I do not think TCP/DTLS/BFCP is defined anywhere, so it does not have to
> be backwards compatible. We are not changing the proto value, we should
> >define a new one and explain how it works. This leaves TCP/BFCP and
> TLS/BFCP as they were previously defined and simply states that these
> transports >cannot be used with ICE. Only UDP/DTLS/BFCP and TCP/DTLS/BFCP
> can be used with ICE, with ICE not supported for all the other BFCP
> flavors.
>
>
>
> Ok, so TCP/BFCP and TLS/BFCP will still be kept in 4583bis, and should be
> used when ICE is not used?
>
>
>
TCP/BFCP, TLS/BFCP and UDP/BFCP will be kept in 4583bis and can only be
used when ICE is not used. UDP/DTLS/BFCP and TCP/DTLS/BFCP are two
transport that can be used with or without ICE.

Regards,
_____________
Roman Shpount

--001a1149e64093561d0545ed0c50
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">On Thu, Jan 12, 2017 at 4:45 PM, Christer Holmberg <span dir=3D"ltr">&=
lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chri=
ster.holmberg@ericsson.com</a>&gt;</span> wrote:<br></div></div><div class=
=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px =
0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">





<div lang=3D"EN-GB">
<div class=3D"gmail-m_-36831143190770620WordSection1">
<p class=3D"MsoNormal">&gt;I do not think TCP/DTLS/BFCP is defined anywhere=
, so it does not have to be backwards compatible. We are not changing the p=
roto value, we should &gt;define a new one and explain how it works. This l=
eaves TCP/BFCP and TLS/BFCP as they were
 previously defined and simply states that these transports &gt;cannot be u=
sed with ICE. Only UDP/DTLS/BFCP and TCP/DTLS/BFCP can be used with ICE, wi=
th ICE not supported for all the other BFCP flavors.=C2=A0<br></p><div><div=
><div><span class=3D"gmail-"><div><p class=3D"MsoNormal"><u></u></p>
</div>
</span><div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:calibri,sa=
ns-serif">Ok, so TCP/BFCP and TLS/BFCP will still be kept in 4583bis, and s=
hould be used when ICE is not used?<u></u><u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal"><br></p></div></div></div></div></div></div></blockq=
uote><div><br></div><div>TCP/BFCP, TLS/BFCP and UDP/BFCP will be kept in 45=
83bis and can only be used when ICE is not used. UDP/DTLS/BFCP and TCP/DTLS=
/BFCP are two transport that can be used with or without ICE.</div><div><br=
></div><div>Regards,</div><div><div class=3D"gmail_signature">_____________=
<br>Roman Shpount</div></div><div>=C2=A0</div></div></div></div>

--001a1149e64093561d0545ed0c50--


From nobody Thu Jan 12 14:25:26 2017
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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 of the IETF.

        Title           : Connection-Oriented Media Transport over TLS in SDP
        Authors         : Jonathan Lennox
                          Christer Holmberg
	Filename        : draft-ietf-mmusic-4572-update-11.txt
	Pages           : 17
	Date            : 2017-01-12

Abstract:
   This document specifies how to establish secure connection-oriented
   media transport sessions over the Transport Layer Security (TLS)
   protocol using the Session Description Protocol (SDP).  It defines a
   new SDP protocol identifier, 'TCP/TLS'.  It also defines the syntax
   and semantics for an SDP 'fingerprint' attribute that identifies the
   certificate that will be presented for the TLS session.  This
   mechanism allows media transport over TLS connections to be
   established securely, so long as the integrity of session
   descriptions is assured.

   This document obsoletes RFC 4572, by clarifying the usage of multiple
   fingerprints.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-4572-update/

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-mmusic-4572-update-11

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-4572-update-11


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 Jan 12 14:46:50 2017
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Subject: [MMUSIC] Review of draft-ietf-mmusic-ice-sip-sdp-10
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I've done a review of draft-ietf-mmusic-ice-sip-sdp-10, and have a 
handful of comments. I've broken these down into three sections: 
big-picture questions, technical comments, and editorial comments. I 
apologize for the lateness in posting this review; the comments are 
substantial, and converting them into email form took far longer than I 
anticipated.


Big Picture Questions

The text in section 9, read literally, deprecates RFCs 4091 and 4092. If 
that is the actual intention, the abstract and "obsoletes" metadata need 
to be updated to reflect this fact. If the deprecation is qualified 
(e.g., the RFCs are still okay in other contexts, but should not be used 
in conjunction with ICE), then the text in section 9 needs to be much 
clearer.

It's not clear to me what the relationship is between this document and 
RFC 5245 (and, for that matter, 6544). Is this supposed to obsolete RFC 
5245? That appears to be the intention. It's also not clear why we would 
obsolete 5245 without also incorporating 6544 -- it seems that doing so 
leaves the use of ICE TCP with SIP to be kind of in limbo.

Assuming the intention *is* to replace RFC 5245, the structure of this 
document seems to leave non-SIP uses of RFC3264 (e.g., WebRTC) stranded. 
If the intention is *not* to replace RFC 5245, the replication of all of 
the SDP syntax seems to be problematic: the the case of a conflict, 
which specification is intended to govern?

In short, I think the relationship between this document, the 
contemporary work (e.g., ICE-BIS), and the historical work on which it 
is based needs to be made much clearer in the document itself, and we 
need to have a clear picture about how to handle SDP for non-SIP uses of 
ICE (e.g., by splitting this document into an SDP part and a SIP part, 
similar to how we split 3264 from 3261).


Technical Comments

I believe that the RECOMMENDED-strength normative statement in the final 
paragraph of 4.1.1.1 overstates things a bit. While this guidance makes 
set in certain circumstances. For example, for devices that are known to 
typically known to be on a single network with mutual routability, using 
a relayed candidate as a default will lead to calls being unnecessarily 
sent out of network and back. I strongly suggest that this paragraph be 
reworked to be a considered discussion of the different trade-offs 
involved in using each kind of candidate, with a non-normative 
suggestion to make the behavior configurable in user agents. What I want 
to avoid here is guidance which leads to devices behind a single ALG 
necessarily going through two relays by default when talking to each 
other (which is what the current guidance can lead to -- I can sketch 
this out in more detail if it's not obvious how this happens).

Section 4.1.1.2 uses the word "media stream" in a way that's not 
consistent with the way that term is defined by RFC 7656, which could 
lead to confusion. In RFC 7656 terminology, "media stream" is an alias 
for "RTP stream," which is identified by a single SSRC. With 
technologies such as simulcast. A single m-line can contain multiple 
media streams. I think you want the term "source stream," with a 
citation of that term in RFC 7656.

Section 4.1.1.2 discusses ice-options attributes in the terms of 
"support", without any discussion of activation. Although this is 
treated (somewhat lightly) later in the document, this smells like the 
same kind of potential morass we ran into with SIP options tags: the 
patchwork means of indicating feature *support* versus feature 
*activation* made it very difficult to specify and implement things in a 
consistent fashion. I strongly recommend that this document spend a bit 
more text discussing this distinction, and maybe even consider a formal 
syntax for distinguishing between supported and activated features.

Section 4.1.3 should probably point out that, in the case of forking, 
the same local candidates are used for all branches of the fork.

Section 4.2, section paragraph: replace "be rejected" with "fail" -- 
there are more ways to fail than rejection, and the action should be the 
same for rejection as for other failures.

Section 4.2.1.2 should add motivation for the requirement imposed by its 
second paragraph, or the requirement should be removed. The need to do 
this is not obvious, and I suspect it may be unnecessary.

Section 4.2.2.2.3 specifies, during offer processing, that the 
implementation "SHOULD wait for [ICE] checks to complete" before sending 
an answer. Keeping in mind that these updates typically happen with the 
SIP UPDATE method, and given that RFC 3261 specifies "TUs SHOULD respond 
immediately to non-INVITE requests" (cf. RFC3261 section 17.1), I'm 
concerned about the timing implications here. We should minimally point 
out that we're explicitly telling UAs to ignore this normative statement 
in RFC3261, and make sure that we've done the analysis to ensure that 
we're not going to cause problems for the SIP state machine (I'm 
concerned, in particular, about unnecessary SIP retransmissions here).

Section 4.2.4.1.4: the final paragraph seems to reiterate behavior 
described in ICE-BIS. It's probably not ideal to describe this in two 
places, as any subtle difference can result in interop problems.

Given the state of ICE today, I think the definition of "candidate" in 
section 5.1 should also include the syntax for TCP transports.

Section 5.1, <connection-address>: the language in here that states "If 
the DNS query returns more than on IP address, on is chosen," implying 
that one is chosen at random. There are specific procedures for how this 
is done (especially with IPv6); the statement should point to the 
selection mechanisms.

Section 11.2.1, final paragraph: We should have text in here addressing 
the fact that user agents that are not willing to receive non-ICE 
answers need to include "Require: ice" in their offer; and that clients 
that reject non-ICE offers should use the 421 response code to do so, 
and should indicate "Require: ice" in their rejections.



Editorial Comments (of varying severity)

The final sentence of the definition of "Default Destination/Candidate" 
is a bit difficult to follow. Suggest: ' For the RTCP component, the 
address and port are indicated using the "a=rtcp" attribute defined in 
[RFC3605], if present; otherwise, the RTCP component address is the same 
as the address of the RTP component, and its port is one greater than 
the port of the RTP component.'

Section 3, final paragraph: remove "the" before "[ICE-BIS]".

Section 4.1.3, second paragraph: eliminate the space before the first 
comma. Place a comma between "its role" and "and processes"

Section 4.1.3, final paragraph: replace "complaint" with "compliant".

The contents of section 4.1.5 appear to be completely redundant with the 
contents of section 4.1.5.1.1. I would recommend eliminating section 
4.1.5. Also, the section numbering here is really baroque and 
unnecessarily deep. For example, section 4.1.5.1 contains no text and 
only one subsection; this is also true of section 4.1.5.2. The more I 
look at this, the more bizarre it appears. I think the correct thing to 
do here is:
(a) Remove all the text currently in section 4.1.5.
(b) Move the paragraph in section 4.1.5.1.1 and the two paragraphs in 
4.1.5.2.1 into section 4.1.5
(c) Eliminate the (now empty) sections 4.1.5.1, 4.1.5.1.1, 4.1.5.2, and 
4.1.5.2.1.

Section 4.2.1.2.1, second paragraph says "The agent MAY include 
additional candidates it did not offer previously" before we explain how 
this can happen. I recommend adding a "(see section 4.2.4.1.1)" after 
"previously".

Section 4.2.2.3, final paragraph: The second line has an "i" that has 
been separated from the rest of its word (which appears on the following 
line).

Section 4.2.3: add commas: "being ICE-unaware, a subsequent" ... "this 
is done, this could result" ... "ICE-unaware, that answer would not"

Section 4.2.3.1.1 should not rely on the section header exclusively for 
context here. The first sentence should read "If ICE support is 
indicated int he SIP answer and the offer was a restart..." Ditto for 
sections 4.2.3.1.3 (add "and ICE is running") and 4.2.3.1.3 ("and ICE is 
completed").

Section 5.1 indicates that ABNF is used for "candidate," but this is not 
repeated for (e.g.) sections 5.3. and 5.4. I would suggest moving the 
statement from 5.1 (and 5.2) into section 5 so that it applies to all 
syntax definitions.

In all of section 5, the attributes really should include examples.

Also, section 5 should indicate which exernally-defined ABNF constructs 
are being used in the document. This is done in some locations, but not 
others. Running the ABNF through a validator will highlight all 
externally-defined elements; it will also help uncover any other issues 
with the ABNF.

All attributes are defined with ABNF of the form: "attribute-name" ":" 
-- this is unnecessary, and should be shortened to "attribute-name:".

Section 5.1, under <connection-address>: change "using an A" to "using 
an A record."

Section 5.1, under <transport> refers to specification of TCP with ICE 
in the future tense rather than the past. This should be reworked.

Section 5.1, under <foundation>, "...in the Frozen algorithm" should be 
followed by "defined in" and then a citation for where the Frozen 
algorithm is defined.

Section 5.1, <priority> -- this just says it's an integer, without 
describing its meaning. Either explain what it means, or cite something 
that does so.

Section 5.1, <rel-addr> and <rel-port>: replace "<rel-addr> and 
<rel-port> is equal" with "<rel-addr> and <rel-port> are equal".

Section 5.2, in the ABNF for "remote-candidate", indicate that 
component-ID is defined in RFC 4566.

Section 5.4, final paragraph: this indicates that the grammar allows for 
"6 bits of randomness per character" -- this should indicate that it 
allows for "6 bits of information per character".

Section 6: rewrite the first sentence using active voice so as to 
positively identify which entities the requirements apply to.

Section 7.1: "Note that" is an awkward way to start a section; I suggest 
striking it.

Section 7.1, first line: replace "may not" with "might not." I had to 
read this several times to realize that the intended meaning here was 
"might not." In colloquial usage, "may not" is a prohibition (e.g., "you 
may not take more than two cookies.")

Section 8.1, first paragraph refers to "ringing the phone of the called 
party." Old-style telephones are only one use case for SIP; this should 
instead read "alerting the called user agent."

Section 8.1.1 second paragraph is HUGE. Consider how to split it up to 
be less "wall of text"y.

Section 8.1.1, fourth paragraph replace "each candidate" with "every 
candidate."

Section 8.1.1, final paragraph: "it's a localized decision" is extremely 
informal and doesn't match the general style for RFCs. Please rephrase.

Section 8.1.2, first paragraph describes a kind of difficult-to-follow 
situation in which an INVITE contains an offer, and the answer is 
provided in a PRACK response, resulting in an empty "200 OK." Two 
comments: (1) this would really benefit from a sequence diagram; and (2) 
are SBCs and ALGs likely to do the right thing with empty 200s?

Section 8.1.2, second paragraph indicates that placing an offer in an 
INVITE the 2xx can cause substantial PDD and clipping; this is true 
regardless of whether ICE is in use. I think you mean to say that ICE 
can exacerbate PDD and clipping in this case rather than causing it.

Sections 8.4, 9, and 11.1 -- remove the redundant RFC numbers (as in 
"defined in RFC 3312 [RFC3312]").

Section 10 refers to timer Ta without indicating where it is defined.

Section 11.1: Replace "SIPS" with "TLS". As Olle is fond of noting, 
there are circumstances in which TLS is feasible but SIPS is not, and 
specifying "TLS" here is sufficient.

Section 11.2: replace "stun" with "STUN."

Section 11.2.1, final paragraph: replace "if its not used" with "if it's 
not used."

Section 11.2.2 -- expand the acronym "NAT" on first use.

Section 11.2.2, first paragraph: replace "application layer SIP" with 
"application-layer SIP"

Section 12.1: Insert "The" at the very beginning of the first sentence.

Section 12.2: Make "RFC 5245" a reference.

Section 12.2, third paragraph should start "[RFC6679] defines the..." 
(add the word "the")

Section 13: make "RFC 5245" and "RFC 6336" references.


/a


From nobody Thu Jan 12 14:50:05 2017
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Subject: [MMUSIC] Last Call: <draft-ietf-mmusic-4572-update-11.txt> (Connection-Oriented Media Transport over TLS in SDP) to Proposed Standard
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The IESG has received a request from the Multiparty Multimedia Session
Control WG (mmusic) to consider the following document:
- 'Connection-Oriented Media Transport over TLS in SDP'
  <draft-ietf-mmusic-4572-update-11.txt> as Proposed Standard

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action. Please send substantive comments to the
ietf@ietf.org mailing lists by 2017-01-26. Exceptionally, comments may be
sent to iesg@ietf.org instead. In either case, please retain the
beginning of the Subject line to allow automated sorting.

Abstract


   This document specifies how to establish secure connection-oriented
   media transport sessions over the Transport Layer Security (TLS)
   protocol using the Session Description Protocol (SDP).  It defines a
   new SDP protocol identifier, 'TCP/TLS'.  It also defines the syntax
   and semantics for an SDP 'fingerprint' attribute that identifies the
   certificate that will be presented for the TLS session.  This
   mechanism allows media transport over TLS connections to be
   established securely, so long as the integrity of session
   descriptions is assured.

   This document obsoletes RFC 4572, by clarifying the usage of multiple
   fingerprints.




The file can be obtained via
https://datatracker.ietf.org/doc/draft-ietf-mmusic-4572-update/

IESG discussion can be tracked via
https://datatracker.ietf.org/doc/draft-ietf-mmusic-4572-update/ballot/


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





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To: Roman Shpount <roman@telurix.com>, Christer Holmberg <christer.holmberg@ericsson.com>
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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [MMUSIC] Transport tags
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Hi,

A question, is the UDP/DTLS/BFCP only used with ICE, or are they also 
used without ICE support. I wonders how this relates to the discussion 
that the UDP/TCP destinction becomes mote when one actually are doing 
ICE/DTLS/BFCP, i.e. use ICE to agree on a datagram capable transport and 
then puts DTLS/BFCP over it, independent on what transport type the 
arrived candidate pairs are on.

Is there a point to the BFCP usages to actually be explicit that we one 
only want to use ICE negotiated DTLS secured BFCP, and thus should have 
a PROTO value for that, i.e. ICE/DTLS/BFCP?

Note, this is a question, not an actual proposal. Just an exploration so 
that you who use the protocol can consider if it makes sense.

Cheers

Magnus

Den 2017-01-12 kl. 23:15, skrev Roman Shpount:
> On Thu, Jan 12, 2017 at 4:45 PM, Christer Holmberg
> <christer.holmberg@ericsson.com <mailto:christer.holmberg@ericsson.com>>
> wrote:
>
>     >I do not think TCP/DTLS/BFCP is defined anywhere, so it does not
>     have to be backwards compatible. We are not changing the proto
>     value, we should >define a new one and explain how it works. This
>     leaves TCP/BFCP and TLS/BFCP as they were previously defined and
>     simply states that these transports >cannot be used with ICE. Only
>     UDP/DTLS/BFCP and TCP/DTLS/BFCP can be used with ICE, with ICE not
>     supported for all the other BFCP flavors.
>
>     __
>
>     __ __
>
>     Ok, so TCP/BFCP and TLS/BFCP will still be kept in 4583bis, and
>     should be used when ICE is not used?____
>
>
>
> TCP/BFCP, TLS/BFCP and UDP/BFCP will be kept in 4583bis and can only be
> used when ICE is not used. UDP/DTLS/BFCP and TCP/DTLS/BFCP are two
> transport that can be used with or without ICE.
>
> Regards,
> _____________
> Roman Shpount
>


-- 

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
FÃ¤rÃ¶gatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
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Subject: Re: [MMUSIC] Transport tags
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Hi,

I see no reason to use UDP/DTLS/BFCP only with ICE.


The question to define an =B3ICE proto=B2 has came up before. It=B9s certai=
nly
an interesting idea, but I think it should be studied within a wider scope
- not for a specific protocol.

Regards,

Christer


On 13/01/17 10:35, "Magnus Westerlund" <magnus.westerlund@ericsson.com>
wrote:

>Hi,
>
>A question, is the UDP/DTLS/BFCP only used with ICE, or are they also
>used without ICE support. I wonders how this relates to the discussion
>that the UDP/TCP destinction becomes mote when one actually are doing
>ICE/DTLS/BFCP, i.e. use ICE to agree on a datagram capable transport and
>then puts DTLS/BFCP over it, independent on what transport type the
>arrived candidate pairs are on.
>
>Is there a point to the BFCP usages to actually be explicit that we one
>only want to use ICE negotiated DTLS secured BFCP, and thus should have
>a PROTO value for that, i.e. ICE/DTLS/BFCP?
>
>Note, this is a question, not an actual proposal. Just an exploration so
>that you who use the protocol can consider if it makes sense.
>
>Cheers
>
>Magnus
>
>Den 2017-01-12 kl. 23:15, skrev Roman Shpount:
>> On Thu, Jan 12, 2017 at 4:45 PM, Christer Holmberg
>> <christer.holmberg@ericsson.com <mailto:christer.holmberg@ericsson.com>>
>> wrote:
>>
>>     >I do not think TCP/DTLS/BFCP is defined anywhere, so it does not
>>     have to be backwards compatible. We are not changing the proto
>>     value, we should >define a new one and explain how it works. This
>>     leaves TCP/BFCP and TLS/BFCP as they were previously defined and
>>     simply states that these transports >cannot be used with ICE. Only
>>     UDP/DTLS/BFCP and TCP/DTLS/BFCP can be used with ICE, with ICE not
>>     supported for all the other BFCP flavors.
>>
>>     __
>>
>>     __ __
>>
>>     Ok, so TCP/BFCP and TLS/BFCP will still be kept in 4583bis, and
>>     should be used when ICE is not used?____
>>
>>
>>
>> TCP/BFCP, TLS/BFCP and UDP/BFCP will be kept in 4583bis and can only be
>> used when ICE is not used. UDP/DTLS/BFCP and TCP/DTLS/BFCP are two
>> transport that can be used with or without ICE.
>>
>> Regards,
>> _____________
>> Roman Shpount
>>
>
>
>--=20
>
>Magnus Westerlund
>
>----------------------------------------------------------------------
>Services, Media and Network features, Ericsson Research EAB/TXM
>----------------------------------------------------------------------
>Ericsson AB                 | Phone  +46 10 7148287
>F=E4r=F6gatan 6                 | Mobile +46 73 0949079
>SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>----------------------------------------------------------------------
>


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: SIP-SDP: Verify decision made in Seoul regarding non-supported transport in m- line of answer
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Subject: [MMUSIC] SIP-SDP: Verify decision made in Seoul regarding non-supported transport in m- line of answer
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--_000_D49E87A915A0Cchristerholmbergericssoncom_
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Hi,

In Seoul we discussed a number of issues related to ICE-SDP. We made some d=
ecisions, and I got action points to verify one of those on the list.

The associated slides can be found here:

https://www.ietf.org/proceedings/97/slides/slides-97-mmusic-ice-sip-sdp-00.=
pdf


The issue (slides 4-6) was when an answerer receives an offer with a m- lin=
e transport that it doesn=92t support. The current O/A rules say that the t=
ransport in the answer must match the transport in the offer.

However, if ICE is used, there may be transports (offered using ICE candida=
tes) that the answerer DOES support.

Based on the discussions, there was strong consensus to go for Alt#2, which=
 was allowing a transport in the m- line of the answer even if the answerer=
 doesn=92t support it, as ICE candidates will be used to determine the tran=
sport.

Does anyone object to Alt#2?

Regards,

Christer













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e-break: after-white-space; color: rgb(0, 0, 0); font-size: 14px; font-fami=
ly: Calibri, sans-serif;">
<div>Hi,</div>
<div><br>
</div>
<div>In Seoul we discussed a number of issues related to ICE-SDP. We made s=
ome decisions, and I got action points to verify one of those on the list.<=
/div>
<div><br>
</div>
<div>The associated slides can be found here:</div>
<div><br>
</div>
<div><a href=3D"https://www.ietf.org/proceedings/97/slides/slides-97-mmusic=
-ice-sip-sdp-00.pdf">https://www.ietf.org/proceedings/97/slides/slides-97-m=
music-ice-sip-sdp-00.pdf</a></div>
<div><br>
</div>
<div><br>
</div>
<div>The issue (slides 4-6) was when an answerer receives an offer with a m=
- line transport that it doesn=92t support. The current O/A rules say that =
the transport in the answer must match the transport in the offer.</div>
<div><br>
</div>
<div>However, if ICE is used, there may be transports (offered using ICE ca=
ndidates) that the answerer DOES support.</div>
<div><br>
</div>
<div>Based on the discussions, there was strong consensus to go for Alt#2, =
which was allowing a transport in the m- line of the answer even if the ans=
werer doesn=92t support it, as ICE candidates will be used to determine the=
 transport.</div>
<div><br>
</div>
<div>Does anyone object to Alt#2?</div>
<div><br>
</div>
<div>Regards,</div>
<div><br>
</div>
<div>Christer</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div>&nbsp;</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
</body>
</html>

--_000_D49E87A915A0Cchristerholmbergericssoncom_--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: ICE-SDP: Suggested text to clarify that an ICE transport switch does not require a new offer (in order to match the transport of the c/m- line with the transport of the new candidate)
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Subject: [MMUSIC] ICE-SDP: Suggested text to clarify that an ICE transport switch does not require a new offer (in order to match the transport of the c/m- line with the transport of the new candidate)
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Hi,

One of the issues that we were going to discuss in Seoul, but didn=92t have=
 much time for, was whether we should clarify that a ICE transport switch (=
using a previously exchanged and verified ICE candidate pair) does not requ=
ire a new offer, in order to align the m- line with the new transport.

I got an action point in Seoul to suggest text for draft-ice-sep, and below=
 is my initial suggestion:


4.2.  Subsequent Offer/Answer Exchanges

   Either agent MAY generate a subsequent offer at any time allowed by
   [RFC3264].  The rules in Section 4.1.5 will cause the controlling
   agent to send an updated offer at the conclusion of ICE processing
   when ICE has selected different candidate pairs from the default
   pairs.  This section defines rules for construction of subsequent
   offers and answers.

   Should a subsequent offer be rejected, ICE processing continues as if
   the subsequent offer had never been made.

   <new>

   As described in section 4.1.5, when ICE concludes, a subsequent offer mi=
ght be required in order to match the transport in the c/m- line of a media=
 stream with the transport of the the selected candidates. Once this has be=
en done, if an endpoint later starts to send media using another candidate,=
 the endpoint does not need to send a subsequent offer (even if the transpo=
rt in the c/m- line does no longer match the transport of the candidate).

   </new>

Regards,

Christer

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ly: Calibri, sans-serif;">
<div><br>
</div>
<div>Hi,</div>
<div><br>
</div>
<div>One of the issues that we were going to discuss in Seoul, but didn=92t=
 have much time for, was whether we should clarify that a ICE transport swi=
tch (using a previously exchanged and verified ICE candidate pair) does not=
 require a new offer, in order to
 align the m- line with the new transport.</div>
<div><br>
</div>
<div>I got an action point in Seoul to suggest text for draft-ice-sep, and =
below is my initial suggestion:</div>
<div><br>
</div>
<div>
<pre style=3D"font-variant-ligatures: normal; orphans: 2; widows: 2; word-w=
rap: break-word; white-space: pre-wrap;">4.2.  Subsequent Offer/Answer Exch=
anges

   Either agent MAY generate a subsequent offer at any time allowed by
   [RFC3264].  The rules in Section 4.1.5 will cause the controlling
   agent to send an updated offer at the conclusion of ICE processing
   when ICE has selected different candidate pairs from the default
   pairs.  This section defines rules for construction of subsequent
   offers and answers.

   Should a subsequent offer be rejected, ICE processing continues as if
   the subsequent offer had never been made.</pre>
<pre style=3D"font-variant-ligatures: normal; orphans: 2; widows: 2; word-w=
rap: break-word; white-space: pre-wrap;">   &lt;new&gt;</pre>
<pre style=3D"font-variant-ligatures: normal; orphans: 2; widows: 2; word-w=
rap: break-word; white-space: pre-wrap;">   As described in section 4.1.5, =
when ICE concludes, a subsequent offer might be required in order to match =
the transport in the c/m- line of a media stream with the transport of the =
the selected candidates. Once this has been done, if an endpoint later star=
ts to send media using another candidate, the endpoint does not need to sen=
d a subsequent offer (even if the transport in the c/m- line does no longer=
 match the transport of the candidate).</pre>
<pre style=3D"font-variant-ligatures: normal; orphans: 2; widows: 2; word-w=
rap: break-word; white-space: pre-wrap;"><span style=3D"font-family: Calibr=
i, sans-serif;">   &lt;/new&gt;</span></pre>
<pre style=3D"font-variant-ligatures: normal; orphans: 2; widows: 2; word-w=
rap: break-word; white-space: pre-wrap;"><span style=3D"font-family: Calibr=
i, sans-serif;">Regards,</span></pre>
<pre style=3D"font-variant-ligatures: normal; orphans: 2; widows: 2; word-w=
rap: break-word; white-space: pre-wrap;"><span style=3D"font-family: Calibr=
i, sans-serif;">Christer</span></pre>
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] SIP-SDP: Verify decision made in Seoul regarding non-supported transport in m- line of answer
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Subject: Re: [MMUSIC] SIP-SDP: Verify decision made in Seoul regarding non-supported transport in m- line of answer
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The subject shall be ICE-SDP: Verify decision made=85

From: mmusic <mmusic-bounces@ietf.org<mailto:mmusic-bounces@ietf.org>> on b=
ehalf of Christer Holmberg <christer.holmberg@ericsson.com<mailto:christer.=
holmberg@ericsson.com>>
Date: Friday 13 January 2017 at 13:32
To: "mmusic@ietf.org<mailto:mmusic@ietf.org>" <mmusic@ietf.org<mailto:mmusi=
c@ietf.org>>
Subject: [MMUSIC] SIP-SDP: Verify decision made in Seoul regarding non-supp=
orted transport in m- line of answer

Hi,

In Seoul we discussed a number of issues related to ICE-SDP. We made some d=
ecisions, and I got action points to verify one of those on the list.

The associated slides can be found here:

https://www.ietf.org/proceedings/97/slides/slides-97-mmusic-ice-sip-sdp-00.=
pdf


The issue (slides 4-6) was when an answerer receives an offer with a m- lin=
e transport that it doesn=92t support. The current O/A rules say that the t=
ransport in the answer must match the transport in the offer.

However, if ICE is used, there may be transports (offered using ICE candida=
tes) that the answerer DOES support.

Based on the discussions, there was strong consensus to go for Alt#2, which=
 was allowing a transport in the m- line of the answer even if the answerer=
 doesn=92t support it, as ICE candidates will be used to determine the tran=
sport.

Does anyone object to Alt#2?

Regards,

Christer













--_000_D49E8E2E15A34christerholmbergericssoncom_
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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; color: rgb(0, 0, 0); font-size: 14px; font-fami=
ly: Calibri, sans-serif;">
<div>The subject shall be ICE-SDP: Verify decision made=85</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<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:=
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 BORDER-RIGHT: medium none; PADDING-TOP: 3pt">
<span style=3D"font-weight:bold">From: </span>mmusic &lt;<a href=3D"mailto:=
mmusic-bounces@ietf.org">mmusic-bounces@ietf.org</a>&gt; on behalf of Chris=
ter Holmberg &lt;<a href=3D"mailto:christer.holmberg@ericsson.com">christer=
.holmberg@ericsson.com</a>&gt;<br>
<span style=3D"font-weight:bold">Date: </span>Friday 13 January 2017 at 13:=
32<br>
<span style=3D"font-weight:bold">To: </span>&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>[MMUSIC] SIP-SDP: Verify d=
ecision made in Seoul regarding non-supported transport in m- line of answe=
r<br>
</div>
<div><br>
</div>
<div>
<div style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line=
-break: after-white-space; color: rgb(0, 0, 0); font-size: 14px; font-famil=
y: Calibri, sans-serif;">
<div>Hi,</div>
<div><br>
</div>
<div>In Seoul we discussed a number of issues related to ICE-SDP. We made s=
ome decisions, and I got action points to verify one of those on the list.<=
/div>
<div><br>
</div>
<div>The associated slides can be found here:</div>
<div><br>
</div>
<div><a href=3D"https://www.ietf.org/proceedings/97/slides/slides-97-mmusic=
-ice-sip-sdp-00.pdf">https://www.ietf.org/proceedings/97/slides/slides-97-m=
music-ice-sip-sdp-00.pdf</a></div>
<div><br>
</div>
<div><br>
</div>
<div>The issue (slides 4-6) was when an answerer receives an offer with a m=
- line transport that it doesn=92t support. The current O/A rules say that =
the transport in the answer must match the transport in the offer.</div>
<div><br>
</div>
<div>However, if ICE is used, there may be transports (offered using ICE ca=
ndidates) that the answerer DOES support.</div>
<div><br>
</div>
<div>Based on the discussions, there was strong consensus to go for Alt#2, =
which was allowing a transport in the m- line of the answer even if the ans=
werer doesn=92t support it, as ICE candidates will be used to determine the=
 transport.</div>
<div><br>
</div>
<div>Does anyone object to Alt#2?</div>
<div><br>
</div>
<div>Regards,</div>
<div><br>
</div>
<div>Christer</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div>&nbsp;</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
</div>
</div>
</span>
</body>
</html>

--_000_D49E8E2E15A34christerholmbergericssoncom_--


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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: Roman Shpount <roman@telurix.com>
Thread-Topic: [MMUSIC] Transport tags
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Subject: Re: [MMUSIC] Transport tags
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UDP/DTLS/BFCP is already defined in 4583bis, except we specified the tag as=
 UDP/TLS/BFCP. I agree it can be used with and without ICE.

Cheers,
Charles

On Jan 12, 2017, at 2:15 PM, Roman Shpount <roman@telurix.com<mailto:roman@=
telurix.com>> wrote:

On Thu, Jan 12, 2017 at 4:45 PM, Christer Holmberg <christer.holmberg@erics=
son.com<mailto:christer.holmberg@ericsson.com>> wrote:
>I do not think TCP/DTLS/BFCP is defined anywhere, so it does not have to b=
e backwards compatible. We are not changing the proto value, we should >def=
ine a new one and explain how it works. This leaves TCP/BFCP and TLS/BFCP a=
s they were previously defined and simply states that these transports >can=
not be used with ICE. Only UDP/DTLS/BFCP and TCP/DTLS/BFCP can be used with=
 ICE, with ICE not supported for all the other BFCP flavors.

Ok, so TCP/BFCP and TLS/BFCP will still be kept in 4583bis, and should be u=
sed when ICE is not used?


TCP/BFCP, TLS/BFCP and UDP/BFCP will be kept in 4583bis and can only be use=
d when ICE is not used. UDP/DTLS/BFCP and TCP/DTLS/BFCP are two transport t=
hat can be used with or without ICE.

Regards,
_____________
Roman Shpount


--_000_EBECE11584F34E9783E265439A2D65F0ciscocom_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body dir=3D"auto">
<div></div>
<div><br>
</div>
<div>
<div><span style=3D"background-color: rgba(255, 255, 255, 0);">UDP/DTLS/BFC=
P is already defined in 4583bis, except we specified the tag as&nbsp;UDP/TL=
S/BFCP. I agree it can be used with and without ICE.&nbsp;</span></div>
<div><span style=3D"background-color: rgba(255, 255, 255, 0);"><br>
</span></div>
<div><span style=3D"background-color: rgba(255, 255, 255, 0);">Cheers,</spa=
n></div>
<div><span style=3D"background-color: rgba(255, 255, 255, 0);">Charles</spa=
n></div>
<div><br>
</div>
On Jan 12, 2017, at 2:15 PM, Roman Shpount &lt;<a href=3D"mailto:roman@telu=
rix.com">roman@telurix.com</a>&gt; wrote:<br>
<br>
</div>
<blockquote type=3D"cite">
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div>
<div class=3D"gmail_signature">On Thu, Jan 12, 2017 at 4:45 PM, Christer Ho=
lmberg <span dir=3D"ltr">
&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chr=
ister.holmberg@ericsson.com</a>&gt;</span> wrote:<br>
</div>
</div>
<div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left:1px solid rgb(204,204,204);padding-left:1ex">
<div lang=3D"EN-GB">
<div class=3D"gmail-m_-36831143190770620WordSection1">
<p class=3D"MsoNormal">&gt;I do not think TCP/DTLS/BFCP is defined anywhere=
, so it does not have to be backwards compatible. We are not changing the p=
roto value, we should &gt;define a new one and explain how it works. This l=
eaves TCP/BFCP and TLS/BFCP as they were
 previously defined and simply states that these transports &gt;cannot be u=
sed with ICE. Only UDP/DTLS/BFCP and TCP/DTLS/BFCP can be used with ICE, wi=
th ICE not supported for all the other BFCP flavors.&nbsp;<br>
</p>
<div>
<div>
<div><span class=3D"gmail-">
<div>
<p class=3D"MsoNormal"><u></u></p>
</div>
</span>
<div>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:calibri,sa=
ns-serif">Ok, so TCP/BFCP and TLS/BFCP will still be kept in 4583bis, and s=
hould be used when ICE is not used?<u></u><u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal"><br>
</p>
</div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>TCP/BFCP, TLS/BFCP and UDP/BFCP will be kept in 4583bis and can only b=
e used when ICE is not used. UDP/DTLS/BFCP and TCP/DTLS/BFCP are two transp=
ort that can be used with or without ICE.</div>
<div><br>
</div>
<div>Regards,</div>
<div>
<div class=3D"gmail_signature">_____________<br>
Roman Shpount</div>
</div>
<div>&nbsp;</div>
</div>
</div>
</div>
</div>
</blockquote>
</body>
</html>

--_000_EBECE11584F34E9783E265439A2D65F0ciscocom_--


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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: [MMUSIC] Transport tags
Thread-Index: AQHSbE3p2SC5GZHiXUyd4kk6fM3mOqEzs4yAgACvV4CAAHqSgIAA1ReAgAADMACAAAZqgIAABU+AgAAExICAAAhlgIAArViA///+Bo4=
Date: Fri, 13 Jan 2017 14:28:24 +0000
Message-ID: <DA5874F4-F20D-4C2B-A5B9-32688EAAA34C@cisco.com>
References: <CAD5OKxvEqvah+ZJdPKJdGmKob84X8WaCqKQ2GEiatVbOSmEjDw@mail.gmail.com> <6A083F67-4ECB-4703-A881-EB2F074F309E@cisco.com> <CAD5OKxt9iVE8Sgax5rjzwyhNCupeXLCVDjz3iMU+A3eTjAnGoA@mail.gmail.com> <E0242B2A-4F7E-4915-8C73-ED68AC69846A@cisco.com> <7594FB04B1934943A5C02806D1A2204B4BF6F650@ESESSMB209.ericsson.se> <CAD5OKxu4nS4naza=f4Y0=TZJ3T5Cwc_XtsoVmP_WbxuFWiN0iA@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B4BF6F72C@ESESSMB209.ericsson.se> <CAD5OKxuyzJP_pFAWVnMMBDATakZOqPKE0ujyXgSSPXRjhASsuQ@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B4BF6F797@ESESSMB209.ericsson.se> <CAD5OKxsH=AhhwxAWiEcwmtUUe-NqEBwmc23tUPm1VVbSCPBWAQ@mail.gmail.com>, <0e376a55-f20b-b589-104a-6e34df5767e9@ericsson.com>
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Cc: Paul Kyzivat <paul.kyzivat@comcast.net>, "mmusic@ietf.org" <mmusic@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] Transport tags
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It is possible and people already do use UDP/TLS/BFCP with ICE and without =
any TCP candidates.=20

Cheers,
Charles

> On Jan 13, 2017, at 12:35 AM, Magnus Westerlund <magnus.westerlund@ericss=
on.com> wrote:
>=20
> Hi,
>=20
> A question, is the UDP/DTLS/BFCP only used with ICE, or are they also use=
d without ICE support. I wonders how this relates to the discussion that th=
e UDP/TCP destinction becomes mote when one actually are doing ICE/DTLS/BFC=
P, i.e. use ICE to agree on a datagram capable transport and then puts DTLS=
/BFCP over it, independent on what transport type the arrived candidate pai=
rs are on.
>=20
> Is there a point to the BFCP usages to actually be explicit that we one o=
nly want to use ICE negotiated DTLS secured BFCP, and thus should have a PR=
OTO value for that, i.e. ICE/DTLS/BFCP?
>=20
> Note, this is a question, not an actual proposal. Just an exploration so =
that you who use the protocol can consider if it makes sense.
>=20
> Cheers
>=20
> Magnus
>=20
>> Den 2017-01-12 kl. 23:15, skrev Roman Shpount:
>> On Thu, Jan 12, 2017 at 4:45 PM, Christer Holmberg
>> <christer.holmberg@ericsson.com <mailto:christer.holmberg@ericsson.com>>
>> wrote:
>>=20
>>    >I do not think TCP/DTLS/BFCP is defined anywhere, so it does not
>>    have to be backwards compatible. We are not changing the proto
>>    value, we should >define a new one and explain how it works. This
>>    leaves TCP/BFCP and TLS/BFCP as they were previously defined and
>>    simply states that these transports >cannot be used with ICE. Only
>>    UDP/DTLS/BFCP and TCP/DTLS/BFCP can be used with ICE, with ICE not
>>    supported for all the other BFCP flavors.
>>=20
>>    __
>>=20
>>    __ __
>>=20
>>    Ok, so TCP/BFCP and TLS/BFCP will still be kept in 4583bis, and
>>    should be used when ICE is not used?____
>>=20
>>=20
>>=20
>> TCP/BFCP, TLS/BFCP and UDP/BFCP will be kept in 4583bis and can only be
>> used when ICE is not used. UDP/DTLS/BFCP and TCP/DTLS/BFCP are two
>> transport that can be used with or without ICE.
>>=20
>> Regards,
>> _____________
>> Roman Shpount
>>=20
>=20
>=20
> --=20
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20


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From: Roman Shpount <roman@telurix.com>
Date: Fri, 13 Jan 2017 14:08:49 -0500
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Subject: Re: [MMUSIC] SIP-SDP: Verify decision made in Seoul regarding non-supported transport in m- line of answer
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As I have mentioned before, I think ALT#1 is a better option: If we propose
mandatory to implement UDP based protocol for each transport family, only
use the mandatory to implement protocol for the default candidate, and the
transport mismatch problem will never occur.

As a bit of background, this issue appeared because of TCP/BFCP, which was
assumed to be the default protocol for BFCP with ICE. Since then, I think
we have decided that TCP/BFCP is not going to be used with ICE. For all the
other protocols that I know (UDP/DTLS/SCTP, UDP/TLS/RTP/SAVPF,
UDP/DTLS/BFCP, RTP/AVP, etc) there is a clear preference that UDP based
protocol should be used for backwards compatibility. I also do not see a
use case which will require that only tcp based candidates would be
included in the offer or the answer. Because of this, I think mandatory to
implement transport is a better approach.

This also have additional benefits of:

1. Providing m=3D and c=3D line information which will not cause the ICE
mismatch with older implementations
2. Not supplying any on the path signaling devices with bogus address
information required in ALT#2 answer
3. Improves general chances of negotiation succeeding with end points not
supporting ICE by picking the protocol more likely to succeed (chance of
TCP/RTP/AVP succeeding, for instance, are much lower then RTP/AVP).

Regards,

_____________
Roman Shpount

On Fri, Jan 13, 2017 at 7:00 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> The subject shall be ICE-SDP: Verify decision made=E2=80=A6
>
> From: mmusic <mmusic-bounces@ietf.org> on behalf of Christer Holmberg <
> christer.holmberg@ericsson.com>
> Date: Friday 13 January 2017 at 13:32
> To: "mmusic@ietf.org" <mmusic@ietf.org>
> Subject: [MMUSIC] SIP-SDP: Verify decision made in Seoul regarding
> non-supported transport in m- line of answer
>
> Hi,
>
> In Seoul we discussed a number of issues related to ICE-SDP. We made some
> decisions, and I got action points to verify one of those on the list.
>
> The associated slides can be found here:
>
> https://www.ietf.org/proceedings/97/slides/slides-
> 97-mmusic-ice-sip-sdp-00.pdf
>
>
> The issue (slides 4-6) was when an answerer receives an offer with a m-
> line transport that it doesn=E2=80=99t support. The current O/A rules say=
 that the
> transport in the answer must match the transport in the offer.
>
> However, if ICE is used, there may be transports (offered using ICE
> candidates) that the answerer DOES support.
>
> Based on the discussions, there was strong consensus to go for Alt#2,
> which was allowing a transport in the m- line of the answer even if the
> answerer doesn=E2=80=99t support it, as ICE candidates will be used to de=
termine
> the transport.
>
> Does anyone object to Alt#2?
>
> Regards,
>
> Christer
>
>
>
>
>
>
>
>
>
>
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>
>

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<div dir=3D"ltr">As I have mentioned before, I think ALT#1 is a better opti=
on: If we propose mandatory to implement UDP based protocol for each transp=
ort family, only use the mandatory to implement protocol for the default ca=
ndidate, and the transport mismatch problem will never occur.<div><br></div=
><div>As a bit of background, this issue appeared because of TCP/BFCP, whic=
h was assumed to be the default protocol for BFCP with ICE. Since then, I t=
hink we have decided that TCP/BFCP is not going to be used with ICE. For al=
l the other protocols that I know (UDP/DTLS/SCTP, UDP/TLS/RTP/SAVPF, UDP/DT=
LS/BFCP, RTP/AVP, etc) there is a clear preference that UDP based protocol =
should be used for backwards compatibility. I also do not see a use case wh=
ich will require that only tcp based candidates would be included in the of=
fer or the answer. Because of this, I think mandatory to implement transpor=
t is a better approach.</div><div><br></div><div>This also have additional =
benefits of:=C2=A0</div><div><br></div><div>1. Providing m=3D and c=3D line=
 information which will not cause the ICE mismatch with older implementatio=
ns</div><div>2. Not supplying any on the path signaling devices with bogus =
address information required in ALT#2 answer</div><div>3. Improves general =
chances of negotiation succeeding with end points not supporting ICE by pic=
king the protocol more likely to succeed (chance of TCP/RTP/AVP succeeding,=
 for instance, are much lower then RTP/AVP).</div><div><br></div><div>Regar=
ds,</div></div><div class=3D"gmail_extra"><br clear=3D"all"><div><div class=
=3D"gmail_signature" data-smartmail=3D"gmail_signature">_____________<br>Ro=
man Shpount</div></div>
<br><div class=3D"gmail_quote">On Fri, Jan 13, 2017 at 7:00 AM, Christer Ho=
lmberg <span dir=3D"ltr">&lt;<a href=3D"mailto:christer.holmberg@ericsson.c=
om" target=3D"_blank">christer.holmberg@ericsson.com</a>&gt;</span> wrote:<=
br><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:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div>The subject shall be ICE-SDP: Verify decision made=E2=80=A6</div>
<div><br>
</div>
<span id=3D"m_-7922315582249877170OLK_SRC_BODY_SECTION">
<div style=3D"font-family:Calibri;font-size:11pt;text-align:left;color:blac=
k;BORDER-BOTTOM:medium none;BORDER-LEFT:medium none;PADDING-BOTTOM:0in;PADD=
ING-LEFT:0in;PADDING-RIGHT:0in;BORDER-TOP:#b5c4df 1pt solid;BORDER-RIGHT:me=
dium none;PADDING-TOP:3pt">
<span style=3D"font-weight:bold">From: </span>mmusic &lt;<a href=3D"mailto:=
mmusic-bounces@ietf.org" target=3D"_blank">mmusic-bounces@ietf.org</a>&gt; =
on behalf of Christer Holmberg &lt;<a href=3D"mailto:christer.holmberg@eric=
sson.com" target=3D"_blank">christer.holmberg@ericsson.<wbr>com</a>&gt;<br>
<span style=3D"font-weight:bold">Date: </span>Friday 13 January 2017 at 13:=
32<br>
<span style=3D"font-weight:bold">To: </span>&quot;<a href=3D"mailto:mmusic@=
ietf.org" target=3D"_blank">mmusic@ietf.org</a>&quot; &lt;<a href=3D"mailto=
:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.org</a>&gt;<br>
<span style=3D"font-weight:bold">Subject: </span>[MMUSIC] SIP-SDP: Verify d=
ecision made in Seoul regarding non-supported transport in m- line of answe=
r<br>
</div><div><div class=3D"h5">
<div><br>
</div>
<div>
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div>Hi,</div>
<div><br>
</div>
<div>In Seoul we discussed a number of issues related to ICE-SDP. We made s=
ome decisions, and I got action points to verify one of those on the list.<=
/div>
<div><br>
</div>
<div>The associated slides can be found here:</div>
<div><br>
</div>
<div><a href=3D"https://www.ietf.org/proceedings/97/slides/slides-97-mmusic=
-ice-sip-sdp-00.pdf" target=3D"_blank">https://www.ietf.org/<wbr>proceeding=
s/97/slides/slides-<wbr>97-mmusic-ice-sip-sdp-00.pdf</a></div>
<div><br>
</div>
<div><br>
</div>
<div>The issue (slides 4-6) was when an answerer receives an offer with a m=
- line transport that it doesn=E2=80=99t support. The current O/A rules say=
 that the transport in the answer must match the transport in the offer.</d=
iv>
<div><br>
</div>
<div>However, if ICE is used, there may be transports (offered using ICE ca=
ndidates) that the answerer DOES support.</div>
<div><br>
</div>
<div>Based on the discussions, there was strong consensus to go for Alt#2, =
which was allowing a transport in the m- line of the answer even if the ans=
werer doesn=E2=80=99t support it, as ICE candidates will be used to determi=
ne the transport.</div>
<div><br>
</div>
<div>Does anyone object to Alt#2?</div>
<div><br>
</div>
<div>Regards,</div>
<div><br>
</div>
<div>Christer</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div>=C2=A0</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
</div>
</div>
</div></div></span>
</div>

<br>______________________________<wbr>_________________<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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/mmusic</a><br=
>
<br></blockquote></div><br></div>

--001a1149e640627a750545fe9037--


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From: Roman Shpount <roman@telurix.com>
Date: Fri, 13 Jan 2017 14:12:40 -0500
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Subject: Re: [MMUSIC] ICE-SDP: Suggested text to clarify that an ICE transport switch does not require a new offer (in order to match the transport of the c/m- line with the transport of the new candidate)
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--001a114d7daa2481d50545fe9e6b
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This looks good to me.

One related question, what happens if another candidate pair is selected
during the time this updated offer is processed? This can cause offer and
answer contain candidates that do not form the pair and use different
transports.

Regards,

_____________
Roman Shpount

On Fri, Jan 13, 2017 at 6:59 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>
> Hi,
>
> One of the issues that we were going to discuss in Seoul, but didn=E2=80=
=99t have
> much time for, was whether we should clarify that a ICE transport switch
> (using a previously exchanged and verified ICE candidate pair) does not
> require a new offer, in order to align the m- line with the new transport=
.
>
> I got an action point in Seoul to suggest text for draft-ice-sep, and
> below is my initial suggestion:
>
> 4.2.  Subsequent Offer/Answer Exchanges
>
>    Either agent MAY generate a subsequent offer at any time allowed by
>    [RFC3264].  The rules in Section 4.1.5 will cause the controlling
>    agent to send an updated offer at the conclusion of ICE processing
>    when ICE has selected different candidate pairs from the default
>    pairs.  This section defines rules for construction of subsequent
>    offers and answers.
>
>    Should a subsequent offer be rejected, ICE processing continues as if
>    the subsequent offer had never been made.
>
>    <new>
>
>    As described in section 4.1.5, when ICE concludes, a subsequent offer =
might be required in order to match the transport in the c/m- line of a med=
ia stream with the transport of the the selected candidates. Once this has =
been done, if an endpoint later starts to send media using another candidat=
e, the endpoint does not need to send a subsequent offer (even if the trans=
port in the c/m- line does no longer match the transport of the candidate).
>
>    </new>
>
> Regards,
>
> Christer
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>
>

--001a114d7daa2481d50545fe9e6b
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<div dir=3D"ltr">This looks good to me.<div><br></div><div>One related ques=
tion, what happens if another candidate pair is selected during the time th=
is updated offer is processed? This can cause offer and answer contain cand=
idates that do not form the pair and use different transports.</div><div><b=
r></div><div>Regards,</div></div><div class=3D"gmail_extra"><br clear=3D"al=
l"><div><div class=3D"gmail_signature" data-smartmail=3D"gmail_signature">_=
____________<br>Roman Shpount</div></div>
<br><div class=3D"gmail_quote">On Fri, Jan 13, 2017 at 6:59 AM, Christer Ho=
lmberg <span dir=3D"ltr">&lt;<a href=3D"mailto:christer.holmberg@ericsson.c=
om" target=3D"_blank">christer.holmberg@ericsson.com</a>&gt;</span> wrote:<=
br><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:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div><br>
</div>
<div>Hi,</div>
<div><br>
</div>
<div>One of the issues that we were going to discuss in Seoul, but didn=E2=
=80=99t have much time for, was whether we should clarify that a ICE transp=
ort switch (using a previously exchanged and verified ICE candidate pair) d=
oes not require a new offer, in order to
 align the m- line with the new transport.</div>
<div><br>
</div>
<div>I got an action point in Seoul to suggest text for draft-ice-sep, and =
below is my initial suggestion:</div>
<div><br>
</div>
<div>
<pre style=3D"font-variant-ligatures:normal;word-wrap:break-word;white-spac=
e:pre-wrap">4.2.  Subsequent Offer/Answer Exchanges

   Either agent MAY generate a subsequent offer at any time allowed by
   [RFC3264].  The rules in Section 4.1.5 will cause the controlling
   agent to send an updated offer at the conclusion of ICE processing
   when ICE has selected different candidate pairs from the default
   pairs.  This section defines rules for construction of subsequent
   offers and answers.

   Should a subsequent offer be rejected, ICE processing continues as if
   the subsequent offer had never been made.</pre>
<pre style=3D"font-variant-ligatures:normal;word-wrap:break-word;white-spac=
e:pre-wrap">   &lt;new&gt;</pre>
<pre style=3D"font-variant-ligatures:normal;word-wrap:break-word;white-spac=
e:pre-wrap">   As described in section 4.1.5, when ICE concludes, a subsequ=
ent offer might be required in order to match the transport in the c/m- lin=
e of a media stream with the transport of the the selected candidates. Once=
 this has been done, if an endpoint later starts to send media using anothe=
r candidate, the endpoint does not need to send a subsequent offer (even if=
 the transport in the c/m- line does no longer match the transport of the c=
andidate).</pre>
<pre style=3D"font-variant-ligatures:normal;word-wrap:break-word;white-spac=
e:pre-wrap"><span style=3D"font-family:Calibri,sans-serif">   &lt;/new&gt;<=
/span></pre>
<pre style=3D"font-variant-ligatures:normal;word-wrap:break-word;white-spac=
e:pre-wrap"><span style=3D"font-family:Calibri,sans-serif">Regards,</span><=
/pre>
<pre style=3D"font-variant-ligatures:normal;word-wrap:break-word;white-spac=
e:pre-wrap"><span style=3D"font-family:Calibri,sans-serif">Christer</span><=
/pre>
</div>
</div>

<br>______________________________<wbr>_________________<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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/mmusic</a><br=
>
<br></blockquote></div><br></div>

--001a114d7daa2481d50545fe9e6b--


From nobody Sun Jan 15 04:52:07 2017
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-data-channel-sdpneg-11.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 of the IETF.

        Title           : SDP-based Data Channel Negotiation
        Authors         : Keith Drage
                          Maridi R. Makaraju
                          Juergen Stoetzer-Bradler
                          Richard Ejzak
                          Jerome Marcon
	Filename        : draft-ietf-mmusic-data-channel-sdpneg-11.txt
	Pages           : 41
	Date            : 2017-01-15

Abstract:
   The Real-Time Communication in WEB-browsers (RTCWeb) working group is
   charged to provide protocols to support direct interactive rich
   communications using audio, video, and data between two peers' web-
   browsers.  For the support of data communication, the RTCWeb working
   group has in particular defined the concept of bi-directional data
   channels over SCTP (Stream Control Transmission Protocol), where each
   data channel might be used to transport other protocols, called
   subprotocols.  Data channel setup can be done using either the in-
   band Data Channel Establishment Protocol (DCEP) or using some out-of-
   band non-DCEP protocol.  This document specifies how the SDP (Session
   Description Protocol) offer/answer exchange can be used to achieve
   such an out-of-band non-DCEP negotiation.  Even though data channels
   are designed for RTCWeb use initially, they may be used by other
   protocols like, but not limited to, the CLUE protocol (which is
   defined by the IETF "ControLling mUltiple streams for tElepresence"
   working group).  This document is intended to be used wherever data
   channels are used.


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

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-mmusic-data-channel-sdpneg-11

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-data-channel-sdpneg-11


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: Bo Burman <bo.burman@ericsson.com>
To: "mmusic (mmusic@ietf.org)" <mmusic@ietf.org>
Thread-Topic: WG Last Call for draft-ietf-mmusic-data-channel-sdpneg-11
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Cc: "draft-ietf-mmusic-data-channel-sdpneg.all@ietf.org" <draft-ietf-mmusic-data-channel-sdpneg.all@ietf.org>
Subject: [MMUSIC] WG Last Call for draft-ietf-mmusic-data-channel-sdpneg-11
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MMUSIC,

This email starts a one week WG last call on version -11 of this draft that=
 ends on January 23, 2017.
There was a previous WG last call on version -09 that resulted in some mino=
r document updates.

The intended status of this document is standards track (Proposed Standard)=
.

Please review and provide any comments you may have on the document. Commen=
ts should be sent to the document authors and the MMUSIC WG list. If you re=
view the document but do not have any comments, please send a note to that =
effect as well.



Please also forward this WGLC to any other interested parties who may be ab=
le to review the draft, asking them to also direct their comments to the au=
thors and the list as above.


The document can be retrieved here:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-data-channel-sdpneg/


Thank you!



        Bo Burman (MMUSIC co-chair)


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<p class=3D"MsoNormal">MMUSIC,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">This email starts a one week WG last call on version=
 -11 of this draft that ends on January 23, 2017.<o:p></o:p></p>
<p class=3D"MsoNormal">There was a previous WG last call on version -09 tha=
t resulted in some minor document updates.<o:p></o:p></p>
<p class=3D"MsoNormal"><br>
The intended status of this document is standards track (Proposed Standard)=
.<br>
<br>
Please review and provide any comments you may have on the document. Commen=
ts should be sent to the document authors and the MMUSIC WG list. If you re=
view the document but do not have any comments, please send a note to that =
effect as well.<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">Please also forward this WGLC to any other intere=
sted parties who may be able to review the draft, asking them to also direc=
t their comments to the authors and the list as above.<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The document can be retrieved here:<o:p></o:p></p>
<p class=3D"MsoNormal"><a href=3D"https://datatracker.ietf.org/doc/draft-ie=
tf-mmusic-data-channel-sdpneg/">https://datatracker.ietf.org/doc/draft-ietf=
-mmusic-data-channel-sdpneg/</a><o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">Thank you!<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Bo Bur=
man (MMUSIC co-chair)<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</body>
</html>

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From: Eric Rescorla <ekr@rtfm.com>
Date: Mon, 16 Jan 2017 05:17:52 -0800
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To: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [MMUSIC] JSEP Issue #394: What appears in m= lines.
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On Tue, Jan 10, 2017 at 7:44 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> Den 2017-01-02 kl. 14:42, skrev Harald Alvestrand:
>
>> Den 22. des. 2016 18:30, skrev Eric Rescorla:
>>
>>>      __
>>>
>>>     Okay, I agree that having any rules for what you should offer here
>>>     based on what the actual outcome is not the best idea here. I think
>>>     the rules should be based on capability and intent. So if you only
>>>     are going offer TCP candidates, then you clearly should use TCP/=E2=
=80=A6,
>>>
>>>
>>> I'm going to push on this. If we've already agreed that mismatches are
>>> normal, why should we do that? Wouldn't it be better to just effectivel=
y
>>> deprecate this field?
>>>
>>>
>> Concur with EKR.
>>
>> This field (the TCP/UDP part of the protocol field) was defined in a way
>> that makes its original purpose (to negotiate profiles) not only
>> impossible while using ICE, but actively harmful, because you end up
>> telling lies in very many cases, so people will have to accept things
>> that don't match reality.
>>
>> Make the world as simple for implementors as possible: Send a fixed
>> string, and accept all strings.
>>
>>
> Okay, can we please be precise with what is suggested here. Because what
> you write above, and what EKR proposed is not really the same thing. The
> issue proposed that it was fine to set either of the strings. You saying =
a
> fixed string. I want to ask you which UDP/... or TCP/..., or even
> ICE/DTLS/RTP/SAVPF?
>
> From a strict clarity issue using ICE/... would be my preferred. But, tha=
t
> is probably going to throw some implementations, especially gateways. It
> will also require us to actually go write the registration to explain wha=
t
> it is.
>
> Using UDP is likely the most compatible and will only fail in gateway
> cases where there are no UDP candidates and the gateway fail to pick that
> up.
>
> Using TCP when there are no TCP candidates are equally misleading to
> gateway cases.
>
> The above is why I proposed that implementation capabilities and any
> configuration to limit what functions being used would be an appropriate
> choice.
>
> I hope at least we can agree that an JSEP Answer MUST use the PROTO strin=
g
> of the OFFER?


Yes. I think implementations should offer UDP/ (ICE/ will cause problems)
and in the answer echo the offer.

-Ekr


>
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
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> ----------------------------------------------------------------------
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Tue, Jan 10, 2017 at 7:44 AM, Magnus Westerlund <span dir=3D"ltr">&l=
t;<a href=3D"mailto:magnus.westerlund@ericsson.com" target=3D"_blank">magnu=
s.westerlund@ericsson.com</a>&gt;</span> wrote:<br><blockquote class=3D"gma=
il_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-lef=
t:1ex"><span class=3D"">Den 2017-01-02 kl. 14:42, skrev Harald Alvestrand:<=
br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Den 22. des. 2016 18:30, skrev Eric Rescorla:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
=C2=A0 =C2=A0 =C2=A0__<br>
<br>
=C2=A0 =C2=A0 Okay, I agree that having any rules for what you should offer=
 here<br>
=C2=A0 =C2=A0 based on what the actual outcome is not the best idea here. I=
 think<br>
=C2=A0 =C2=A0 the rules should be based on capability and intent. So if you=
 only<br>
=C2=A0 =C2=A0 are going offer TCP candidates, then you clearly should use T=
CP/=E2=80=A6,<br>
<br>
<br>
I&#39;m going to push on this. If we&#39;ve already agreed that mismatches =
are<br>
normal, why should we do that? Wouldn&#39;t it be better to just effectivel=
y<br>
deprecate this field?<br>
<br>
</blockquote>
<br>
Concur with EKR.<br>
<br>
This field (the TCP/UDP part of the protocol field) was defined in a way<br=
>
that makes its original purpose (to negotiate profiles) not only<br>
impossible while using ICE, but actively harmful, because you end up<br>
telling lies in very many cases, so people will have to accept things<br>
that don&#39;t match reality.<br>
<br>
Make the world as simple for implementors as possible: Send a fixed<br>
string, and accept all strings.<br>
<br>
</blockquote>
<br></span>
Okay, can we please be precise with what is suggested here. Because what yo=
u write above, and what EKR proposed is not really the same thing. The issu=
e proposed that it was fine to set either of the strings. You saying a fixe=
d string. I want to ask you which UDP/... or TCP/..., or even ICE/DTLS/RTP/=
SAVPF?<br>
<br>
>From a strict clarity issue using ICE/... would be my preferred. But, that =
is probably going to throw some implementations, especially gateways. It wi=
ll also require us to actually go write the registration to explain what it=
 is.<br>
<br>
Using UDP is likely the most compatible and will only fail in gateway cases=
 where there are no UDP candidates and the gateway fail to pick that up.<br=
>
<br>
Using TCP when there are no TCP candidates are equally misleading to gatewa=
y cases.<br>
<br>
The above is why I proposed that implementation capabilities and any config=
uration to limit what functions being used would be an appropriate choice.<=
br>
<br>
I hope at least we can agree that an JSEP Answer MUST use the PROTO string =
of the OFFER?</blockquote><div><br></div><div>Yes. I think implementations =
should offer UDP/ (ICE/ will cause problems) and in the answer echo the off=
er.</div><div><br></div><div>-Ekr</div><div>=C2=A0</div><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex"><span class=3D"im HOEnZb"><br>
<br>
Cheers<br>
<br>
Magnus Westerlund<br>
<br>
------------------------------<wbr>------------------------------<wbr>-----=
-----<br>
Services, Media and Network features, Ericsson Research EAB/TXM<br>
------------------------------<wbr>------------------------------<wbr>-----=
-----<br>
Ericsson AB=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0| =
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get=3D"_blank">+46 10 7148287</a><br>
F=C3=A4r=C3=B6gatan 6=C2=A0 =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"+467309490=
79" target=3D"_blank">+46 73 0949079</a><br>
SE-164 80 Stockholm, Sweden | mailto: <a href=3D"mailto:magnus.westerlund@e=
ricsson.com" target=3D"_blank">magnus.westerlund@ericsson.com</a><br>
------------------------------<wbr>------------------------------<wbr>-----=
-----<br>
<br></span><div class=3D"HOEnZb"><div class=3D"h5">
______________________________<wbr>_________________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.org</a><br=
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<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/mmusic</a><br=
>
</div></div></blockquote></div><br></div></div>

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Eric Rescorla <ekr@rtfm.com>, Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: [MMUSIC] JSEP Issue #394: What appears in m= lines.
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--_000_D4A2966B15C88christerholmbergericssoncom_
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Hi,

=85

>>I hope at least we can agree that an JSEP Answer MUST use the PROTO strin=
g of the OFFER?
>
>Yes. I think implementations should offer UDP/ (ICE/ will cause problems) =
and in the answer echo the offer.

This is related to the generic issue I raised in Seoul, and sent an e-mail =
about last week: is it ok to echo the transport in the m- line proto value =
of the answer even if the answerer doesn=92t support the transport (alt #1)=
?

OR, should we mandate a transport that everyone must support, and mandate t=
o use that transport as m- line proto value in offers and answers (alt #2)?

People in Seoul preferred alt #1, but I know that at least Roman prefers al=
t #2.

Regards,

Christer



--_000_D4A2966B15C88christerholmbergericssoncom_
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e-break: after-white-space; color: rgb(0, 0, 0); font-size: 14px; font-fami=
ly: Calibri, sans-serif;">
<div>Hi,</div>
<div><br>
</div>
<div>=85</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<div>
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>&gt;&gt;I hope at least we can agree that an JSEP Answer MUST use the =
PROTO string of the OFFER?</div>
<div>&gt;</div>
<div>&gt;Yes. I think implementations should offer UDP/ (ICE/ will cause pr=
oblems) and in the answer echo the offer.</div>
</div>
</div>
</div>
</div>
</div>
</span>
<div><br>
</div>
<div>This is related to the generic issue I raised in Seoul, and sent an e-=
mail about last week: is it ok to echo the transport in the m- line proto v=
alue of the answer even if the answerer doesn=92t support the transport (al=
t #1)?</div>
<div><br>
</div>
<div>OR, should we mandate a transport that everyone must support, and mand=
ate to use that transport as m- line proto value in offers and answers (alt=
 #2)?</div>
<div><br>
</div>
<div>People in Seoul preferred alt #1, but I know that at least Roman prefe=
rs alt #2.</div>
<div><br>
</div>
<div>Regards,</div>
<div><br>
</div>
<div>Christer</div>
<div><br>
</div>
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<div>
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<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
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--_000_D4A2966B15C88christerholmbergericssoncom_--


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From: Eric Rescorla <ekr@rtfm.com>
Date: Mon, 16 Jan 2017 05:45:07 -0800
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On Mon, Jan 16, 2017 at 5:28 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> =E2=80=A6
>
> >>I hope at least we can agree that an JSEP Answer MUST use the PROTO
> string of the OFFER?
> >
> >Yes. I think implementations should offer UDP/ (ICE/ will cause problems=
)
> and in the answer echo the offer.
>
> This is related to the generic issue I raised in Seoul, and sent an e-mai=
l
> about last week: is it ok to echo the transport in the m- line proto valu=
e
> of the answer even if the answerer doesn=E2=80=99t support the transport =
(alt #1)?
>
> OR, should we mandate a transport that everyone must support, and mandate
> to use that transport as m- line proto value in offers and answers (alt #=
2)?
>
> People in Seoul preferred alt #1, but I know that at least Roman prefers
> alt #2.
>

I continue to support #1. We should just stop trying to pretend that this
value has useful semantics.

-Ekr


>
> Regards,
>
> Christer
>
>
>

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Mon, Jan 16, 2017 at 5:28 AM, Christer Holmberg <span dir=3D"ltr">&l=
t;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chris=
ter.holmberg@ericsson.com</a>&gt;</span> wrote:<br><blockquote class=3D"gma=
il_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-lef=
t:1ex">



<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div>Hi,</div>
<div><br>
</div>
<div>=E2=80=A6</div><span class=3D"">
<div><br>
</div>
<span id=3D"m_-711194476441389907OLK_SRC_BODY_SECTION">
<div>
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>&gt;&gt;I hope at least we can agree that an JSEP Answer MUST use the =
PROTO string of the OFFER?</div>
<div>&gt;</div>
<div>&gt;Yes. I think implementations should offer UDP/ (ICE/ will cause pr=
oblems) and in the answer echo the offer.</div>
</div>
</div>
</div>
</div>
</div>
</span>
<div><br>
</div>
</span><div>This is related to the generic issue I raised in Seoul, and sen=
t an e-mail about last week: is it ok to echo the transport in the m- line =
proto value of the answer even if the answerer doesn=E2=80=99t support the =
transport (alt #1)?</div>
<div><br>
</div>
<div>OR, should we mandate a transport that everyone must support, and mand=
ate to use that transport as m- line proto value in offers and answers (alt=
 #2)?</div>
<div><br>
</div>
<div>People in Seoul preferred alt #1, but I know that at least Roman prefe=
rs alt #2.</div></div></blockquote><div><br></div><div>I continue to suppor=
t #1. We should just stop trying to pretend that this value has useful sema=
ntics.</div><div><br></div><div>-Ekr</div><div>=C2=A0</div><blockquote clas=
s=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pad=
ding-left:1ex"><div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-siz=
e:14px;font-family:Calibri,sans-serif">
<div><br>
</div>
<div>Regards,</div>
<div><br>
</div>
<div>Christer</div>
<div><br>
</div>
<span id=3D"m_-711194476441389907OLK_SRC_BODY_SECTION">
<div>
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
</div>
</div>
</div>
</div>
</div>
</span>
</div>

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

--001a114d88eaa1cd38054636662f--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Eric Rescorla <ekr@rtfm.com>
Thread-Topic: [MMUSIC] JSEP Issue #394: What appears in m= lines.
Thread-Index: AQHSXHlOCQ7BQRm6Y0eHkrCi1urD06ElMmaAgAy0twCACUUOAIAAJN0A///iwICAADeE8A==
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From: Paul Kyzivat <paul.kyzivat@comcast.net>
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Subject: Re: [MMUSIC] WG Last Call for draft-ietf-mmusic-data-channel-sdpneg-11
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I've followed this document closely throughout its development.
In preparation for this message I reviewed the document again. I found 
some things that need to be addressed:

1) MAJOR: Section 5.1.1.3 and various other places:

I can find nothing here or elsewhere in the document that says whether 
the the stream id in an offer is the id on which the offerer will 
receive, or the ID on which it will send. Of course this is irrelevant 
if both ends agree to use the same id, but that isn't required.

For this to interact properly with attributes for individual streams (in 
dcsa) I think it will be necessary for this to be consistent with the 
say SDP negotiates media sections - namely that the SDP in an offer 
identifies where the offerer wants to *receive* the media.

It will probably require changes in a variety of places to get this 
sorted out.

2) MINOR: Section 5.1.1 says:

    The intention in exchanging these attributes is to create, on two
    peers, without use of DCEP [I-D.ietf-rtcweb-data-protocol], matched
    pairs of oppositely directed data channels having the same set of
    attributes.  It is assumed that the data channel properties
    (reliable/partially reliable, ordered/unordered) are suitable per the
    subprotocol transport requirements.

In this, "matched pairs of oppositely directed data channels" is 
improper terminology. Each channel *is* a matched pair, of oppositely 
directed SCTP streams.

3) MINOR: Section 5.1.2 says:

    In the SDP, each data channel declaration MAY also be followed by
    other media level SDP attributes, ...

I'm concerned with "followed", because that imposes an ordering 
requirement on attributes. Generally there is no ordering requirement on 
SDP attributes other than that they up front if they are session level 
or follow a particular m= line if they are media-level.

Certainly it will be more readable to group these things in a logical 
way, but I don't think it should be required. So I suggest changing 
"followed" to "accompanied".

4) NIT: IdNits reports 1 error and 8 warnings. These need to be addressed.

	Thanks,
	Paul

On 1/16/17 6:45 AM, Bo Burman wrote:
> MMUSIC,
>
>
>
> This email starts a one week WG last call on version -11 of this draft
> that ends on January 23, 2017.
>
> There was a previous WG last call on version -09 that resulted in some
> minor document updates.
>
>
> The intended status of this document is standards track (Proposed Standard).
>
> Please review and provide any comments you may have on the document.
> Comments should be sent to the document authors and the MMUSIC WG list.
> If you review the document but do not have any comments, please send a
> note to that effect as well.
>
>
>
> Please also forward this WGLC to any other interested parties who may be
> able to review the draft, asking them to also direct their comments to
> the authors and the list as above.
>
>
>
> The document can be retrieved here:
>
> https://datatracker.ietf.org/doc/draft-ietf-mmusic-data-channel-sdpneg/
>
>
>
> Thank you!
>
>
>
>         Bo Burman (MMUSIC co-chair)
>
>
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


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From: Roman Shpount <roman@telurix.com>
Date: Mon, 16 Jan 2017 18:37:54 -0500
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Cc: Magnus Westerlund <magnus.westerlund@ericsson.com>, mmusic WG <mmusic@ietf.org>
Subject: Re: [MMUSIC] JSEP Issue #394: What appears in m= lines.
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For JSEP, is there a reason not to require UDP/TLS/RTP/SAVPF and
UDP/DTLS/SCTP in both the offer and the answer?

Is there a use case to offer only tcp candidates in offer or answer?

Regards,
_____________
Roman Shpount

On Mon, Jan 16, 2017 at 11:09 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> =E2=80=A6
>
> >>>>I hope at least we can agree that an JSEP Answer MUST use the PROTO
> string of the OFFER?
> >>>>
> >>>Yes. I think implementations should offer UDP/ (ICE/ will cause
> problems) and in the answer echo the offer.
> >>
> >> This is related to the generic issue I raised in Seoul, and sent an
> e-mail about last week: is it ok to echo
> >> the transport in the m- line proto value of the answer even if the
> answerer doesn=E2=80=99t support the transport (alt #1)?
> >>
> >> OR, should we mandate a transport that everyone must support, and
> mandate to use that transport as m- line proto value in offers and answer=
s
> (alt #2)?
> >>
> >> People in Seoul preferred alt #1, but I know that at least Roman
> prefers alt #2.
> >
> > I continue to support #1. We should just stop trying to pretend that
> this value has useful semantics.
>
> As #1 was also the outcome of Seoul, I think we should move ahead with it=
.
> Chairs?
>
> Note that #1 does not prevent individual protocols from defining MTI
> transports, default m- line proto values etc,  but it would not be a
> general requirement to do so.
>
> Regards,
>
> Christer
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>

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<div dir=3D"ltr">For JSEP, is there a reason not to require UDP/TLS/RTP/SAV=
PF and UDP/DTLS/SCTP in both the offer and the answer?<div><br></div><div>I=
s there a use case to offer only tcp candidates in offer or answer?</div><d=
iv><br></div><div>Regards,<div class=3D"gmail_extra"><div><div class=3D"gma=
il_signature">_____________<br>Roman Shpount</div></div>
<br><div class=3D"gmail_quote">On Mon, Jan 16, 2017 at 11:09 AM, Christer H=
olmberg <span dir=3D"ltr">&lt;<a href=3D"mailto:christer.holmberg@ericsson.=
com" target=3D"_blank">christer.holmberg@ericsson.com</a>&gt;</span> wrote:=
<br><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;bor=
der-left:1px solid rgb(204,204,204);padding-left:1ex"><span class=3D"gmail-=
">Hi,<br>
<br>
=E2=80=A6<br>
<br>
&gt;&gt;&gt;&gt;I hope at least we can agree that an JSEP Answer MUST use t=
he PROTO string of the OFFER?<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;Yes. I think implementations should offer UDP/ (ICE/ will cause=
 problems) and in the answer echo the offer.<br>
&gt;&gt;<br>
&gt;&gt; This is related to the generic issue I raised in Seoul, and sent a=
n e-mail about last week: is it ok to echo<br>
&gt;&gt; the transport in the m- line proto value of the answer even if the=
 answerer doesn=E2=80=99t support the transport (alt #1)?<br>
&gt;&gt;<br>
&gt;&gt; OR, should we mandate a transport that everyone must support, and =
mandate to use that transport as m- line proto value in offers and answers =
(alt #2)?<br>
&gt;&gt;<br>
&gt;&gt; People in Seoul preferred alt #1, but I know that at least Roman p=
refers alt #2.<br>
&gt;<br>
&gt; I continue to support #1. We should just stop trying to pretend that t=
his value has useful semantics.<br>
<br>
</span>As #1 was also the outcome of Seoul, I think we should move ahead wi=
th it. Chairs?<br>
<br>
Note that #1 does not prevent individual protocols from defining MTI transp=
orts, default m- line proto values etc,=C2=A0 but it would not be a general=
 requirement to do so.<br>
<br>
Regards,<br>
<br>
Christer<br>
<div class=3D"gmail-HOEnZb"><div class=3D"gmail-h5"><br>
<br>
______________________________<wbr>_________________<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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/mmusic</a><br=
>
</div></div></blockquote></div><br></div></div></div>

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From: Eric Rescorla <ekr@rtfm.com>
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Subject: Re: [MMUSIC] JSEP Issue #394: What appears in m= lines.
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On Mon, Jan 16, 2017 at 3:37 PM, Roman Shpount <roman@telurix.com> wrote:

> For JSEP, is there a reason not to require UDP/TLS/RTP/SAVPF and
> UDP/DTLS/SCTP in both the offer and the answer?
>
>
As an answerer, you may get TCP/<blah> from non-JSEP endpoints and the
consensus was that echoing them was better

-Ekr

Is there a use case to offer only tcp candidates in offer or answer?
>
> Regards,
> _____________
> Roman Shpount
>
> On Mon, Jan 16, 2017 at 11:09 AM, Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
>
>> Hi,
>>
>> =E2=80=A6
>>
>> >>>>I hope at least we can agree that an JSEP Answer MUST use the PROTO
>> string of the OFFER?
>> >>>>
>> >>>Yes. I think implementations should offer UDP/ (ICE/ will cause
>> problems) and in the answer echo the offer.
>> >>
>> >> This is related to the generic issue I raised in Seoul, and sent an
>> e-mail about last week: is it ok to echo
>> >> the transport in the m- line proto value of the answer even if the
>> answerer doesn=E2=80=99t support the transport (alt #1)?
>> >>
>> >> OR, should we mandate a transport that everyone must support, and
>> mandate to use that transport as m- line proto value in offers and answe=
rs
>> (alt #2)?
>> >>
>> >> People in Seoul preferred alt #1, but I know that at least Roman
>> prefers alt #2.
>> >
>> > I continue to support #1. We should just stop trying to pretend that
>> this value has useful semantics.
>>
>> As #1 was also the outcome of Seoul, I think we should move ahead with
>> it. Chairs?
>>
>> Note that #1 does not prevent individual protocols from defining MTI
>> transports, default m- line proto values etc,  but it would not be a
>> general requirement to do so.
>>
>> Regards,
>>
>> Christer
>>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>>
>
>

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On M=
on, Jan 16, 2017 at 3:37 PM, Roman Shpount <span dir=3D"ltr">&lt;<a href=3D=
"mailto:roman@telurix.com" target=3D"_blank">roman@telurix.com</a>&gt;</spa=
n> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;b=
order-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr">For JSEP, is t=
here a reason not to require UDP/TLS/RTP/SAVPF and UDP/DTLS/SCTP in both th=
e offer and the answer?<div><br></div></div></blockquote><div><br></div><di=
v>As an answerer, you may get TCP/&lt;blah&gt; from non-JSEP endpoints and =
the consensus was that echoing them was better</div><div><br></div><div>-Ek=
r</div><div><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0=
 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div>=
</div><div>Is there a use case to offer only tcp candidates in offer or ans=
wer?</div><div><br></div><div>Regards,<div class=3D"gmail_extra"><div><div =
class=3D"m_3230130407412229628gmail_signature">_____________<span class=3D"=
HOEnZb"><font color=3D"#888888"><br>Roman Shpount</font></span></div></div>
<br><div class=3D"gmail_quote"><div><div class=3D"h5">On Mon, Jan 16, 2017 =
at 11:09 AM, Christer Holmberg <span dir=3D"ltr">&lt;<a href=3D"mailto:chri=
ster.holmberg@ericsson.com" target=3D"_blank">christer.holmberg@ericsson.<w=
br>com</a>&gt;</span> wrote:<br></div></div><blockquote class=3D"gmail_quot=
e" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204)=
;padding-left:1ex"><div><div class=3D"h5"><span class=3D"m_3230130407412229=
628gmail-">Hi,<br>
<br>
=E2=80=A6<br>
<br>
&gt;&gt;&gt;&gt;I hope at least we can agree that an JSEP Answer MUST use t=
he PROTO string of the OFFER?<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;Yes. I think implementations should offer UDP/ (ICE/ will cause=
 problems) and in the answer echo the offer.<br>
&gt;&gt;<br>
&gt;&gt; This is related to the generic issue I raised in Seoul, and sent a=
n e-mail about last week: is it ok to echo<br>
&gt;&gt; the transport in the m- line proto value of the answer even if the=
 answerer doesn=E2=80=99t support the transport (alt #1)?<br>
&gt;&gt;<br>
&gt;&gt; OR, should we mandate a transport that everyone must support, and =
mandate to use that transport as m- line proto value in offers and answers =
(alt #2)?<br>
&gt;&gt;<br>
&gt;&gt; People in Seoul preferred alt #1, but I know that at least Roman p=
refers alt #2.<br>
&gt;<br>
&gt; I continue to support #1. We should just stop trying to pretend that t=
his value has useful semantics.<br>
<br>
</span>As #1 was also the outcome of Seoul, I think we should move ahead wi=
th it. Chairs?<br>
<br>
Note that #1 does not prevent individual protocols from defining MTI transp=
orts, default m- line proto values etc,=C2=A0 but it would not be a general=
 requirement to do so.<br>
<br>
Regards,<br>
<br>
Christer<br>
</div></div><div class=3D"m_3230130407412229628gmail-HOEnZb"><div class=3D"=
m_3230130407412229628gmail-h5"><br>
<br><span class=3D"">
______________________________<wbr>_________________<br>
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</blockquote></div><br></div></div>

--001a11407e082afd6f05463efbf8--


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From: Roman Shpount <roman@telurix.com>
Date: Mon, 16 Jan 2017 19:11:23 -0500
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To: Eric Rescorla <ekr@rtfm.com>
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Cc: Magnus Westerlund <magnus.westerlund@ericsson.com>, mmusic WG <mmusic@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] JSEP Issue #394: What appears in m= lines.
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On Mon, Jan 16, 2017 at 6:59 PM, Eric Rescorla <ekr@rtfm.com> wrote:

> On Mon, Jan 16, 2017 at 3:37 PM, Roman Shpount <roman@telurix.com> wrote:
>
>> For JSEP, is there a reason not to require UDP/TLS/RTP/SAVPF and
>> UDP/DTLS/SCTP in both the offer and the answer?
>>
>>
> As an answerer, you may get TCP/<blah> from non-JSEP endpoints and the
> consensus was that echoing them was better
>
>
Is there a reason why this would ever happen? You use ICE TCP to go through
UDP firewalls when communicating with a server on a public IP. Why would
you ever do only tcp candidates in offer or answer? This is inefficient and
will work worse then offering both udp and tcp candidates. I am all for
options, but they should have some reason for existence. I see no reason
why TCP/anything would ever be used for JSEP offer/answer, unless ICE is
complete and tcp candidate pair is selected.

BTW, we made UDP required protocol for default candidates for
UDP/DTLS/SCTP. I see no reason not to do the same thing for
UDP/TLS/RTP/SAVPF at least in JSEP. It will only make things interop with
non JSEP endpoints better.

Regards,
_____________
Roman Shpount

--001a114f8426f391cc05463f2372
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">On Mon, Jan 16, 2017 at 6:59 PM, Eric Rescorla <span dir=3D"ltr">&lt;<=
a href=3D"mailto:ekr@rtfm.com" target=3D"_blank">ekr@rtfm.com</a>&gt;</span=
> wrote:<br></div></div><div class=3D"gmail_quote"><blockquote class=3D"gma=
il_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,2=
04,204);padding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><div =
class=3D"gmail_quote"><span class=3D"gmail-">On Mon, Jan 16, 2017 at 3:37 P=
M, Roman Shpount <span dir=3D"ltr">&lt;<a href=3D"mailto:roman@telurix.com"=
 target=3D"_blank">roman@telurix.com</a>&gt;</span> wrote:<br><blockquote c=
lass=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px soli=
d rgb(204,204,204);padding-left:1ex"><div dir=3D"ltr">For JSEP, is there a =
reason not to require UDP/TLS/RTP/SAVPF and UDP/DTLS/SCTP in both the offer=
 and the answer?<div><br></div></div></blockquote><div><br></div></span><di=
v>As an answerer, you may get TCP/&lt;blah&gt; from non-JSEP endpoints and =
the consensus was that echoing them was better</div><div><br></div></div></=
div></div></blockquote><div><br></div><div>Is there a reason why this would=
 ever happen? You use ICE TCP to go through UDP firewalls when communicatin=
g with a server on a public IP. Why would you ever do only tcp candidates i=
n offer or answer? This is inefficient and will work worse then offering bo=
th udp and tcp candidates. I am all for options, but they should have some =
reason for existence. I see no reason why TCP/anything would ever be used f=
or JSEP offer/answer, unless ICE is complete and tcp candidate pair is sele=
cted.</div><div><br></div><div>BTW, we made UDP required protocol for defau=
lt candidates for UDP/DTLS/SCTP. I see no reason not to do the same thing f=
or UDP/TLS/RTP/SAVPF at least in JSEP. It will only make things interop wit=
h non JSEP endpoints better.</div><div><br></div><div>Regards,</div><div><d=
iv class=3D"gmail_signature">_____________<br>Roman Shpount</div></div><div=
>=C2=A0</div></div></div></div>

--001a114f8426f391cc05463f2372--


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References: <52E4A8FC978E0241AE652516E24CAF001E483F95@ESESSMB309.ericsson.se> <CABcZeBPznLKNHek-SGE5Ly6QTOBL-j65sZBb5MbwQVkmBkpyFw@mail.gmail.com> <9110d772-9269-7fed-3ed4-5269d49acb84@alvestrand.no> <282955c7-d077-105b-6a99-a0f5ede87d91@ericsson.com> <CABcZeBPtMMR-xC_=pr1umBWY1CPkAm1J=T=Q_1F1bLNkZwtJkg@mail.gmail.com> <D4A2966B.15C88%christer.holmberg@ericsson.com> <CABcZeBOS+b_bdgaTnQfsNAhdf7g=fspyYON2r5=BoKvPD-32Rw@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B4BF78DE0@ESESSMB209.ericsson.se> <CAD5OKxtN=sHrGoQU9D=WLXWQwNpCqOT5P6ZwhkaS1945VnTT-Q@mail.gmail.com> <CABcZeBN+MGKD_opEq7bKeafb46o3=jKyMEKLDKQ-Mj8a5eezyg@mail.gmail.com> <CAD5OKxuqBeE3VkpRp-Leyyf1nzh2wwPG0giwbtcFOwJ8AecG8w@mail.gmail.com>
From: Eric Rescorla <ekr@rtfm.com>
Date: Mon, 16 Jan 2017 16:20:53 -0800
Message-ID: <CABcZeBOrPzqKh2CWMqHCz8vFLvT20WDqL7_FK=SPnZ_PXn_P_A@mail.gmail.com>
To: Roman Shpount <roman@telurix.com>
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Cc: Magnus Westerlund <magnus.westerlund@ericsson.com>, mmusic WG <mmusic@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] JSEP Issue #394: What appears in m= lines.
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--001a114dac0e5257b705463f481a
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On Mon, Jan 16, 2017 at 4:11 PM, Roman Shpount <roman@telurix.com> wrote:

> On Mon, Jan 16, 2017 at 6:59 PM, Eric Rescorla <ekr@rtfm.com> wrote:
>
>> On Mon, Jan 16, 2017 at 3:37 PM, Roman Shpount <roman@telurix.com> wrote:
>>
>>> For JSEP, is there a reason not to require UDP/TLS/RTP/SAVPF and
>>> UDP/DTLS/SCTP in both the offer and the answer?
>>>
>>>
>> As an answerer, you may get TCP/<blah> from non-JSEP endpoints and the
>> consensus was that echoing them was better
>>
>>
> Is there a reason why this would ever happen?
>

It's permitted behavior.



> You use ICE TCP to go through UDP firewalls when communicating with a
> server on a public IP. Why would you ever do only tcp candidates in offer
> or answer? This is inefficient and will work worse then offering both udp
> and tcp candidates. I am all for options, but they should have some reason
> for existence. I see no reason why TCP/anything would ever be used for JSEP
> offer/answer, unless ICE is complete and tcp candidate pair is selected.
>

Well, as I stated above, this isn't about what JSEP endpoints offer but
what they answer
when other endpoints send TCP/<blah>, however foolish you might think it is.


> BTW, we made UDP required protocol for default candidates for
> UDP/DTLS/SCTP. I see no reason not to do the same thing for
> UDP/TLS/RTP/SAVPF at least in JSEP. It will only make things interop with
> non JSEP endpoints better.
>

What's the evidence that answering TCP/TLS/RTP/SAVP with UDP/TLS/RTP/SAVPF
will improve interop?

-Ekr


> Regards,
> _____________
> Roman Shpount
>
>

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Mon, Jan 16, 2017 at 4:11 PM, Roman Shpount <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:roman@telurix.com" target=3D"_blank">roman@telurix.com</a>&=
gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 =
0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div c=
lass=3D"gmail_extra"><span class=3D""><div><div class=3D"m_-686160898818694=
6013gmail_signature">On Mon, Jan 16, 2017 at 6:59 PM, Eric Rescorla <span d=
ir=3D"ltr">&lt;<a href=3D"mailto:ekr@rtfm.com" target=3D"_blank">ekr@rtfm.c=
om</a>&gt;</span> wrote:<br></div></div></span><div class=3D"gmail_quote"><=
span class=3D""><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0=
px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir=
=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><span class=
=3D"m_-6861608988186946013gmail-">On Mon, Jan 16, 2017 at 3:37 PM, Roman Sh=
pount <span dir=3D"ltr">&lt;<a href=3D"mailto:roman@telurix.com" target=3D"=
_blank">roman@telurix.com</a>&gt;</span> wrote:<br><blockquote class=3D"gma=
il_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,2=
04,204);padding-left:1ex"><div dir=3D"ltr">For JSEP, is there a reason not =
to require UDP/TLS/RTP/SAVPF and UDP/DTLS/SCTP in both the offer and the an=
swer?<div><br></div></div></blockquote><div><br></div></span><div>As an ans=
werer, you may get TCP/&lt;blah&gt; from non-JSEP endpoints and the consens=
us was that echoing them was better</div><div><br></div></div></div></div><=
/blockquote><div><br></div></span><div>Is there a reason why this would eve=
r happen? </div></div></div></div></blockquote><div><br></div><div>It&#39;s=
 permitted behavior.</div><div><br></div><div>=C2=A0</div><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gma=
il_quote"><div>You use ICE TCP to go through UDP firewalls when communicati=
ng with a server on a public IP. Why would you ever do only tcp candidates =
in offer or answer? This is inefficient and will work worse then offering b=
oth udp and tcp candidates. I am all for options, but they should have some=
 reason for existence. I see no reason why TCP/anything would ever be used =
for JSEP offer/answer, unless ICE is complete and tcp candidate pair is sel=
ected.</div></div></div></div></blockquote><div><br></div><div>Well, as I s=
tated above, this isn&#39;t about what JSEP endpoints offer but what they a=
nswer</div><div>when other endpoints send TCP/&lt;blah&gt;, however foolish=
 you might think it is.</div><div><br></div><blockquote class=3D"gmail_quot=
e" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">=
<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><div=
><br></div><div>BTW, we made UDP required protocol for default candidates f=
or UDP/DTLS/SCTP. I see no reason not to do the same thing for UDP/TLS/RTP/=
SAVPF at least in JSEP. It will only make things interop with non JSEP endp=
oints better.</div></div></div></div></blockquote><div><br></div><div>What&=
#39;s the evidence that answering TCP/TLS/RTP/SAVP with UDP/TLS/RTP/SAVPF w=
ill improve interop?</div><div><br></div><div>-Ekr</div><div><br></div><blo=
ckquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #c=
cc solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><div=
 class=3D"gmail_quote"><div><br></div><div>Regards,</div><div><div class=3D=
"m_-6861608988186946013gmail_signature">_____________<span class=3D"HOEnZb"=
><font color=3D"#888888"><br>Roman Shpount</font></span></div></div><div>=
=C2=A0</div></div></div></div>
</blockquote></div><br></div></div>

--001a114dac0e5257b705463f481a--


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From: Roman Shpount <roman@telurix.com>
Date: Mon, 16 Jan 2017 19:53:22 -0500
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Subject: Re: [MMUSIC] JSEP Issue #394: What appears in m= lines.
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--001a113776f221612a05463fbaea
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For clarity, this is what I would suggest for JSEP:

JSEP end point MUST send an offer with UDP/DTLS/SCTP and UDP/TLS/RTP/SAVPF.

If JSEP end point receives an offer with UDP/DTLS/SCTP or
UDP/TLS/RTP/SAVPF, it MUST respond with the same proto and MUST include udp
candidates (and use udp as a default candidate if JSEP endpoint is
providing default candidates).

If JSEP end point receives an offer with TCP/DTLS/SCTP or
TCP/DTLS/RTP/SAVPF, it MUST respond with the same proto and MUST include
tcp candidates (and use tcp as a default candidate if JSEP endpoint is
providing default candidates).

If non JSEP end point responds with a protocol that does not match the
offer, this is considered an error regardless of the ICE candidate supplied
in the answer.

So, what I want is for JSEP not to generate offers with TCP/blah and not to
generate answers were protocol does not match the protocol in the offer, or
does not match ICE candidates provided. I think this will improve interop
and can be satisfied by all existing implementations.

Generic requirements for ICE outside of JSEP can be more complex since we
need to explain what will happen when end point receives an offer with
TCP/blah but does not support TCP candidates. Also, outside of JSEP, things
like ICE mismatch come into play. This, of cause, can be discussed and
decided elsewhere, not in JSEP.

Regards,
_____________
Roman Shpount

--001a113776f221612a05463fbaea
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr">For clarity, this is what I would suggest for JSEP:<div><b=
r></div><div>JSEP end point MUST send an offer with UDP/DTLS/SCTP and UDP/T=
LS/RTP/SAVPF.</div><div><br></div><div>If JSEP end point receives an offer =
with UDP/DTLS/SCTP or UDP/TLS/RTP/SAVPF, it MUST respond with the same prot=
o and MUST include udp candidates (and use udp as a default candidate if JS=
EP endpoint is providing default candidates).=C2=A0<br></div><div><br></div=
><div>If JSEP end point receives an offer with TCP/DTLS/SCTP or TCP/DTLS/RT=
P/SAVPF, it MUST respond with the same proto and MUST include tcp candidate=
s (and use tcp as a default candidate if JSEP endpoint=C2=A0is providing de=
fault candidates).=C2=A0</div><div><br></div><div>If non JSEP end point res=
ponds with a protocol that does not match the offer, this is considered an =
error regardless of the ICE candidate supplied in the answer.<br><div class=
=3D"gmail_extra"><br></div><div class=3D"gmail_extra">So, what I want is fo=
r JSEP not to generate offers with TCP/blah and not to generate answers wer=
e protocol does not match the protocol in the offer, or does not match ICE =
candidates provided. I think this will improve interop and can be satisfied=
 by all existing implementations.</div><div class=3D"gmail_extra"><br></div=
><div class=3D"gmail_extra">Generic requirements for ICE outside of JSEP ca=
n be more complex since we need to explain what will happen when end point =
receives an offer with TCP/blah but does not support TCP candidates. Also, =
outside of JSEP, things like ICE mismatch come into play. This, of cause, c=
an be discussed and decided elsewhere, not in JSEP.</div><div class=3D"gmai=
l_extra"><br></div><div class=3D"gmail_extra">Regards,=C2=A0<br clear=3D"al=
l"><div><div class=3D"gmail_signature">_____________<br>Roman Shpount</div>=
</div></div></div></div>

--001a113776f221612a05463fbaea--


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From: Eric Rescorla <ekr@rtfm.com>
Date: Mon, 16 Jan 2017 17:15:06 -0800
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Subject: Re: [MMUSIC] JSEP Issue #394: What appears in m= lines.
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On Mon, Jan 16, 2017 at 4:53 PM, Roman Shpount <roman@telurix.com> wrote:

> For clarity, this is what I would suggest for JSEP:
>
> JSEP end point MUST send an offer with UDP/DTLS/SCTP and UDP/TLS/RTP/SAVPF.
>

Yes.


If JSEP end point receives an offer with UDP/DTLS/SCTP or
> UDP/TLS/RTP/SAVPF, it MUST respond with the same proto
>

Yes.



> and MUST include udp candidates (and use udp as a default candidate if
> JSEP endpoint is providing default candidates).
>

No, I don't agree with this. It should offer whatever candidates it wants
and/or are determined by policy. For example, if the implementation is set
to relay-only, it will be restricted to whatever the TURN relay will do.


If JSEP end point receives an offer with TCP/DTLS/SCTP or
> TCP/DTLS/RTP/SAVPF, it MUST respond with the same proto
>

Yes.



> and MUST include tcp candidates (and use tcp as a default candidate if
> JSEP endpoint is providing default candidates).
>

No, I don't agree for the reasons above.




If non JSEP end point responds with a protocol that does not match the
> offer, this is considered an error regardless of the ICE candidate supplied
> in the answer.
>
> So, what I want is for JSEP not to generate offers with TCP/blah and not
> to generate answers were protocol does not match the protocol in the offer,
>

Yes.



> or does not match ICE candidates provided. I think this will improve
> interop and can be satisfied by all existing implementations.
>

I don't think trying to match the candidates to the proto line is useful,
especially when you are doing trickle ICE.

-Ekr


Generic requirements for ICE outside of JSEP can be more complex since we
> need to explain what will happen when end point receives an offer with
> TCP/blah but does not support TCP candidates. Also, outside of JSEP, things
> like ICE mismatch come into play. This, of cause, can be discussed and
> decided elsewhere, not in JSEP.
>
> Regards,
> _____________
> Roman Shpount
>

--f403045dc8583490b10546400ad7
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Mon, Jan 16, 2017 at 4:53 PM, Roman Shpount <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:roman@telurix.com" target=3D"_blank">roman@telurix.com</a>&=
gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 =
0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr">For cl=
arity, this is what I would suggest for JSEP:<div><br></div><div>JSEP end p=
oint MUST send an offer with UDP/DTLS/SCTP and UDP/TLS/RTP/SAVPF.</div></di=
v></blockquote><div><br></div><div>Yes.</div><div><br></div><div><br></div>=
<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>If JSEP end point rece=
ives an offer with UDP/DTLS/SCTP or UDP/TLS/RTP/SAVPF, it MUST respond with=
 the same proto </div></div></blockquote><div><br></div><div>Yes.</div><div=
><br></div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"marg=
in:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"=
><div>and MUST include udp candidates (and use udp as a default candidate i=
f JSEP endpoint is providing default candidates).=C2=A0<br></div></div></bl=
ockquote><div><br></div><div>No, I don&#39;t agree with this. It should off=
er whatever candidates it wants and/or are determined by policy. For exampl=
e, if the implementation is set to relay-only, it will be restricted to wha=
tever the TURN relay will do.</div><div><br></div><div><br></div><blockquot=
e class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc sol=
id;padding-left:1ex"><div dir=3D"ltr"><div>If JSEP end point receives an of=
fer with TCP/DTLS/SCTP or TCP/DTLS/RTP/SAVPF, it MUST respond with the same=
 proto</div></div></blockquote><div><br></div><div>Yes.</div><div><br></div=
><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .=
8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div>and =
MUST include tcp candidates (and use tcp as a default candidate if JSEP end=
point=C2=A0is providing default candidates).=C2=A0</div></div></blockquote>=
<div><br></div><div>No, I don&#39;t agree for the reasons above.</div><div>=
<br></div><div>=C2=A0</div><div><br></div><div><br></div><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex"><div dir=3D"ltr"><div>If non JSEP end point responds with a p=
rotocol that does not match the offer, this is considered an error regardle=
ss of the ICE candidate supplied in the answer.<br><div class=3D"gmail_extr=
a"><br></div><div class=3D"gmail_extra">So, what I want is for JSEP not to =
generate offers with TCP/blah and not to generate answers were protocol doe=
s not match the protocol in the offer, </div></div></div></blockquote><div>=
<br></div><div>Yes.</div><div><br></div><div>=C2=A0</div><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex"><div dir=3D"ltr"><div><div class=3D"gmail_extra">or does not =
match ICE candidates provided. I think this will improve interop and can be=
 satisfied by all existing implementations.</div></div></div></blockquote><=
div><br></div><div>I don&#39;t think trying to match the candidates to the =
proto line is useful, especially when you are doing trickle ICE.</div><div>=
<br></div><div>-Ekr</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"><div dir=3D"ltr"><div><div class=3D"gmail_extra">Generic require=
ments for ICE outside of JSEP can be more complex since we need to explain =
what will happen when end point receives an offer with TCP/blah but does no=
t support TCP candidates. Also, outside of JSEP, things like ICE mismatch c=
ome into play. This, of cause, can be discussed and decided elsewhere, not =
in JSEP.</div><div class=3D"gmail_extra"><br></div><div class=3D"gmail_extr=
a">Regards,=C2=A0<br clear=3D"all"><div><div class=3D"m_5055532914729063045=
gmail_signature">_____________<span class=3D"HOEnZb"><font color=3D"#888888=
"><br>Roman Shpount</font></span></div></div></div></div></div>
</blockquote></div><br></div></div>

--f403045dc8583490b10546400ad7--


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From: Roman Shpount <roman@telurix.com>
Date: Mon, 16 Jan 2017 20:35:43 -0500
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Subject: Re: [MMUSIC] JSEP Issue #394: What appears in m= lines.
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--001a11479b4a94836c05464051e5
Content-Type: text/plain; charset=UTF-8

After considering your arguments regarding trickle ICE argument I agree
with you.

To summarize, I can live with the following:

1. JSEP end point MUST send an offer with UDP/DTLS/SCTP or UDP/TLS/RTP/SAVPF
2. If JSEP end point receives the offer, it MUST response with the same
transport even if no ice candidate matching this protocol are present in
the answer
3. If non JSEP end point responds with a protocol that does not match the
offer, this is considered an error regardless of the ICE candidate supplied
in the answer.

I do agree to this only within the limit of JSEP. Outside of JSEP it needs
to be discussed in more detail, since things like ICE mismatch will need to
be considered.

Regards,
_____________
Roman Shpount

--001a11479b4a94836c05464051e5
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">After considering your arguments regarding trickle ICE argument I agre=
e with you.<br></div></div><div class=3D"gmail_quote"><div><br></div><div>T=
o summarize, I can live with the following:</div><div><br></div>1. JSEP end=
 point MUST send an offer with UDP/DTLS/SCTP or UDP/TLS/RTP/SAVPF</div><div=
 class=3D"gmail_quote">2. If JSEP end point receives the offer, it MUST res=
ponse with the same transport even if no ice candidate matching this protoc=
ol are present in the answer</div><div class=3D"gmail_quote">3. If non JSEP=
 end point responds with a protocol that does not match the offer, this is =
considered an error regardless of the ICE candidate supplied in the answer.=
</div><div class=3D"gmail_quote"><br></div><div class=3D"gmail_quote">I do =
agree to this only within the limit of JSEP. Outside of JSEP it needs to be=
 discussed in more detail, since things like ICE mismatch will need to be c=
onsidered.</div><div class=3D"gmail_quote"><br></div><div class=3D"gmail_qu=
ote">Regards,<br><div><div class=3D"gmail_signature">_____________<br>Roman=
 Shpount</div></div><div>=C2=A0</div></div></div></div>

--001a11479b4a94836c05464051e5--


From nobody Wed Jan 18 14:24:14 2017
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From: Flemming Andreasen <fandreas@cisco.com>
To: mmusic <mmusic@ietf.org>
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Cc: draft-ietf-mmusic-dtls-sdp@ietf.org
Subject: [MMUSIC] 1 week WGLC on recent changes to draft-ietf-mmusic-dtls-sdp (-14 to -16)
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Greetings

We previously completed WGLC on draft-ietf-mmusic-dtls-sdp, however 
there has been some updates to the draft subsequently. We are hereby 
issuing a 1 week WGLC on the the changes from -14 to -16 only, as 
indicated by the following URL:

https://www.ietf.org/rfcdiff?url1=draft-ietf-mmusic-dtls-sdp-14&url2=draft-ietf-mmusic-dtls-sdp-16

If you have any comments on the updates, please provide those by 
Wednesday, January 25. Comments should be sent to the document authors 
and the MMUSIC WG list.

Thanks

-- Flemming (as MMUSIC co-chair)





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Subject: [MMUSIC] Modifying an approved document: MSID
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This is a multi-part message in MIME format.
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Ted pointed out to me that I sent this to the wrong group.

Chairs and members, please advise.



-------- Forwarded Message --------
Subject: 	[rtcweb] Modifying an approved document: MSID
Date: 	Wed, 18 Jan 2017 23:22:14 +0100
From: 	Harald Alvestrand <harald@alvestrand.no>
To: 	rtcweb@ietf.org <rtcweb@ietf.org>



When reviewing the implications of the PeerConnection API change to
support AddTrack rather than AddStream, I found an issue.

This issue concerns draft-ietf-rtcweb-msid, which is currently in
REF-WAIT state (I believe).

The issue is that it is possible to add a track without specifying a
stream. Since the track's ID needs to be carried, we have to send an
"a=msid" line, but the track's ID is the *second* field on that line,
with the first being the stream's ID.

This creates a problem.

Suggested fix: Insert two lines in the document:

1) On SDP generation:

"If there is no stream associated with the track, use the reserved ID
value '-'"

2) On SDP parsing

"If the stream ID is the reserved value '-', the track is not associated
with a stream, and no stream is signalled or created."

If this is OK with the community, I'll issue an updated draft with this
change.


-- 
Surveillance is pervasive. Go Dark.

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


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<html>
  <head>

    <meta http-equiv="content-type" content="text/html; charset=windows-1252">
  </head>
  <body text="#000000" bgcolor="#FFFFFF">
    <p>Ted pointed out to me that I sent this to the wrong group.</p>
    <p>Chairs and members, please advise.<br>
    </p>
    <div class="moz-forward-container"><br>
      <br>
      -------- Forwarded Message --------
      <table class="moz-email-headers-table" border="0" cellpadding="0"
        cellspacing="0">
        <tbody>
          <tr>
            <th valign="BASELINE" align="RIGHT" nowrap="nowrap">Subject:
            </th>
            <td>[rtcweb] Modifying an approved document: MSID</td>
          </tr>
          <tr>
            <th valign="BASELINE" align="RIGHT" nowrap="nowrap">Date: </th>
            <td>Wed, 18 Jan 2017 23:22:14 +0100</td>
          </tr>
          <tr>
            <th valign="BASELINE" align="RIGHT" nowrap="nowrap">From: </th>
            <td>Harald Alvestrand <a class="moz-txt-link-rfc2396E" href="mailto:harald@alvestrand.no">&lt;harald@alvestrand.no&gt;</a></td>
          </tr>
          <tr>
            <th valign="BASELINE" align="RIGHT" nowrap="nowrap">To: </th>
            <td><a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a> <a class="moz-txt-link-rfc2396E" href="mailto:rtcweb@ietf.org">&lt;rtcweb@ietf.org&gt;</a></td>
          </tr>
        </tbody>
      </table>
      <br>
      <br>
      <pre>When reviewing the implications of the PeerConnection API change to
support AddTrack rather than AddStream, I found an issue.

This issue concerns draft-ietf-rtcweb-msid, which is currently in
REF-WAIT state (I believe).

The issue is that it is possible to add a track without specifying a
stream. Since the track's ID needs to be carried, we have to send an
"a=msid" line, but the track's ID is the *second* field on that line,
with the first being the stream's ID.

This creates a problem.

Suggested fix: Insert two lines in the document:

1) On SDP generation:

"If there is no stream associated with the track, use the reserved ID
value '-'"

2) On SDP parsing

"If the stream ID is the reserved value '-', the track is not associated
with a stream, and no stream is signalled or created."

If this is OK with the community, I'll issue an updated draft with this
change.


-- 
Surveillance is pervasive. Go Dark.

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

--------------AF50912027E1AA8D98B48086--


From nobody Thu Jan 19 10:57:10 2017
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Subject: [MMUSIC] mmusic - New Meeting Session Request for IETF 98
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A new meeting session request has just been submitted by Flemming Andreasen, a Chair of the mmusic working group.


---------------------------------------------------------
Working Group Name: Multiparty Multimedia Session Control
Area Name: Applications and Real-Time Area
Session Requester: Flemming Andreasen

Number of Sessions: 1
Length of Session(s):  2 Hours
Number of Attendees: 40
Conflicts to Avoid: 
 First Priority: quic perc tls ice httpbis rtcweb avtcore avtext sipcore dispatch payload core dots sipbrandy
 Second Priority: clue straw xrblock rmcat stir tram sfc



Special Requests:
  
---------------------------------------------------------


From nobody Thu Jan 19 22:54:10 2017
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To: Flemming Andreasen <fandreas@cisco.com>, mmusic <mmusic@ietf.org>
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From: Christian Groves <Christian.Groves@nteczone.com>
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Subject: Re: [MMUSIC] 1 week WGLC on recent changes to draft-ietf-mmusic-dtls-sdp (-14 to -16)
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A couple of nits:

Cl. 5.1 Extraneous ",":  "...Endpoints MUST support the cipher suites as 
defined in
    [I-D.ietf-mmusic-4572-update].<,>

Cl. 5.3 Para 2: The first sentence should probably use the same 
formulation as paragraph 3, "... that requires the establishment of..."

Cl. 5.4 Note - Missing "s" : "...an answer that include<s>.."

Cl. 6 Para 1 - Spelling error: "mechansim" -> "mechanism"

Otherwise OK.

Regards, Christian


On 19/01/2017 9:24 AM, Flemming Andreasen wrote:
> Greetings
>
> We previously completed WGLC on draft-ietf-mmusic-dtls-sdp, however 
> there has been some updates to the draft subsequently. We are hereby 
> issuing a 1 week WGLC on the the changes from -14 to -16 only, as 
> indicated by the following URL:
>
> https://www.ietf.org/rfcdiff?url1=draft-ietf-mmusic-dtls-sdp-14&url2=draft-ietf-mmusic-dtls-sdp-16 
>
>
> If you have any comments on the updates, please provide those by 
> Wednesday, January 25. Comments should be sent to the document authors 
> and the MMUSIC WG list.
>
> Thanks
>
> -- Flemming (as MMUSIC co-chair)
>
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


From nobody Thu Jan 19 23:17:00 2017
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Subject: Re: [MMUSIC] WG Last Call for draft-ietf-mmusic-data-channel-sdpneg-11
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Hello Paul,

Please see below.

Regards, Christian

On 17/01/2017 6:56 AM, Paul Kyzivat wrote:
> I've followed this document closely throughout its development.
> In preparation for this message I reviewed the document again. I found 
> some things that need to be addressed:
>
> 1) MAJOR: Section 5.1.1.3 and various other places:
>
> I can find nothing here or elsewhere in the document that says whether 
> the the stream id in an offer is the id on which the offerer will 
> receive, or the ID on which it will send. Of course this is irrelevant 
> if both ends agree to use the same id, but that isn't required.
>
> For this to interact properly with attributes for individual streams 
> (in dcsa) I think it will be necessary for this to be consistent with 
> the say SDP negotiates media sections - namely that the SDP in an 
> offer identifies where the offerer wants to *receive* the media.
>
> It will probably require changes in a variety of places to get this 
> sorted out.

[CNG] There is some guidance in clause 5.2.1 around managing stream 
identifiers. Utilising the same Stream ID is required when supporting 
WebRTC datachannel (see clause 6.4/draft-ietf-rtcweb-data-channel-13). 
Clause 5.1.1 / ietf-mmusic-data-channel-sdpneg vaguely mentions that the 
opposite datachannels have the same set of attributes. SCTP stream 
identifier being one of those.

>
> 2) MINOR: Section 5.1.1 says:
>
>    The intention in exchanging these attributes is to create, on two
>    peers, without use of DCEP [I-D.ietf-rtcweb-data-protocol], matched
>    pairs of oppositely directed data channels having the same set of
>    attributes.  It is assumed that the data channel properties
>    (reliable/partially reliable, ordered/unordered) are suitable per the
>    subprotocol transport requirements.
>
> In this, "matched pairs of oppositely directed data channels" is 
> improper terminology. Each channel *is* a matched pair, of oppositely 
> directed SCTP streams.
[CNG] Yes I agree.
> ..snip..


From nobody Thu Jan 19 23:48:29 2017
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To: Bo Burman <bo.burman@ericsson.com>, "mmusic (mmusic@ietf.org)" <mmusic@ietf.org>
References: <AM5PR0701MB2577808960A0AE88E584E3AB8D7D0@AM5PR0701MB2577.eurprd07.prod.outlook.com>
From: Christian Groves <Christian.Groves@nteczone.com>
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Cc: "draft-ietf-mmusic-data-channel-sdpneg.all@ietf.org" <draft-ietf-mmusic-data-channel-sdpneg.all@ietf.org>
Subject: Re: [MMUSIC] WG Last Call for draft-ietf-mmusic-data-channel-sdpneg-11
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Hello,

I've reviewed it and I think its ready to progress.

A minor NIT on the example in Appendix A. It should contain an 
"a=dtls-id:....." attribute as per other examples in the draft.

Regards, Christian


On 16/01/2017 10:45 PM, Bo Burman wrote:
>
> MMUSIC,
>
> This email starts a one week WG last call on version -11 of this draft 
> that ends on January 23, 2017.
>
> There was a previous WG last call on version -09 that resulted in some 
> minor document updates.
>
>
> The intended status of this document is standards track (Proposed 
> Standard).
>
> Please review and provide any comments you may have on the document. 
> Comments should be sent to the document authors and the MMUSIC WG 
> list. If you review the document but do not have any comments, please 
> send a note to that effect as well.
>
> Please also forward this WGLC to any other interested parties who may 
> be able to review the draft, asking them to also direct their comments 
> to the authors and the list as above.
>
> The document can be retrieved here:
>
> https://datatracker.ietf.org/doc/draft-ietf-mmusic-data-channel-sdpneg/
>
> Thank you!
>
>         Bo Burman (MMUSIC co-chair)
>
>
>
> _______________________________________________
> 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: Christian Groves <Christian.Groves@nteczone.com>, Flemming Andreasen <fandreas@cisco.com>, mmusic <mmusic@ietf.org>
Thread-Topic: [MMUSIC] 1 week WGLC on recent changes to draft-ietf-mmusic-dtls-sdp (-14 to -16)
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Subject: Re: [MMUSIC] 1 week WGLC on recent changes to draft-ietf-mmusic-dtls-sdp (-14 to -16)
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Hi Christian,

I=B9ll fix as suggested. Thanks!

Regards,

Christer


On 20/01/17 08:53, "Christian Groves" <Christian.Groves@nteczone.com>
wrote:

>A couple of nits:
>
>Cl. 5.1 Extraneous ",":  "...Endpoints MUST support the cipher suites as
>defined in
>    [I-D.ietf-mmusic-4572-update].<,>
>
>Cl. 5.3 Para 2: The first sentence should probably use the same
>formulation as paragraph 3, "... that requires the establishment of..."
>
>Cl. 5.4 Note - Missing "s" : "...an answer that include<s>.."
>
>Cl. 6 Para 1 - Spelling error: "mechansim" -> "mechanism"
>
>Otherwise OK.
>
>Regards, Christian
>
>
>On 19/01/2017 9:24 AM, Flemming Andreasen wrote:
>> Greetings
>>
>> We previously completed WGLC on draft-ietf-mmusic-dtls-sdp, however
>> there has been some updates to the draft subsequently. We are hereby
>> issuing a 1 week WGLC on the the changes from -14 to -16 only, as
>> indicated by the following URL:
>>
>>=20
>>https://www.ietf.org/rfcdiff?url1=3Ddraft-ietf-mmusic-dtls-sdp-14&url2=3D=
draf
>>t-ietf-mmusic-dtls-sdp-16
>>
>>
>> If you have any comments on the updates, please provide those by
>> Wednesday, January 25. Comments should be sent to the document authors
>> and the MMUSIC WG list.
>>
>> Thanks
>>
>> -- Flemming (as MMUSIC co-chair)
>>
>>
>>
>>
>> _______________________________________________
>> 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: Christer Holmberg <christer.holmberg@ericsson.com>, Christian Groves <Christian.Groves@nteczone.com>, Flemming Andreasen <fandreas@cisco.com>, mmusic <mmusic@ietf.org>
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Subject: Re: [MMUSIC] 1 week WGLC on recent changes to draft-ietf-mmusic-dtls-sdp (-14 to -16)
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Pull request for the WGLC comments created:

https://github.com/cdh4u/draft-dtls-sdp/pull/22

Regards,

Christer

-----Original Message-----
From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
Sent: 20 January 2017 09:58
To: Christian Groves <Christian.Groves@nteczone.com>; Flemming Andreasen <f=
andreas@cisco.com>; mmusic <mmusic@ietf.org>
Cc: draft-ietf-mmusic-dtls-sdp@ietf.org
Subject: Re: [MMUSIC] 1 week WGLC on recent changes to draft-ietf-mmusic-dt=
ls-sdp (-14 to -16)

Hi Christian,

I=B9ll fix as suggested. Thanks!

Regards,

Christer


On 20/01/17 08:53, "Christian Groves" <Christian.Groves@nteczone.com>
wrote:

>A couple of nits:
>
>Cl. 5.1 Extraneous ",":  "...Endpoints MUST support the cipher suites=20
>as defined in
>    [I-D.ietf-mmusic-4572-update].<,>
>
>Cl. 5.3 Para 2: The first sentence should probably use the same=20
>formulation as paragraph 3, "... that requires the establishment of..."
>
>Cl. 5.4 Note - Missing "s" : "...an answer that include<s>.."
>
>Cl. 6 Para 1 - Spelling error: "mechansim" -> "mechanism"
>
>Otherwise OK.
>
>Regards, Christian
>
>
>On 19/01/2017 9:24 AM, Flemming Andreasen wrote:
>> Greetings
>>
>> We previously completed WGLC on draft-ietf-mmusic-dtls-sdp, however=20
>> there has been some updates to the draft subsequently. We are hereby=20
>> issuing a 1 week WGLC on the the changes from -14 to -16 only, as=20
>> indicated by the following URL:
>>
>>=20
>>https://www.ietf.org/rfcdiff?url1=3Ddraft-ietf-mmusic-dtls-sdp-14&url2=3D=
d
>>raf
>>t-ietf-mmusic-dtls-sdp-16
>>
>>
>> If you have any comments on the updates, please provide those by=20
>> Wednesday, January 25. Comments should be sent to the document=20
>> authors and the MMUSIC WG list.
>>
>> Thanks
>>
>> -- Flemming (as MMUSIC co-chair)
>>
>>
>>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>>
>


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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-ice-sip-sdp-11.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 of the IETF.

        Title           : Using Interactive Connectivity Establishment (ICE) with Session Description Protocol (SDP) offer/answer and Session Initiation Protocol (SIP)
        Authors         : Marc Petit-Huguenin
                          Ari Keranen
                          Suhas Nandakumar
	Filename        : draft-ietf-mmusic-ice-sip-sdp-11.txt
	Pages           : 42
	Date            : 2017-01-20

Abstract:
   This document describes how Interactive Connectivity Establishment
   (ICE) is used with Session Description Protocol (SDP) offer/answer
   and Session Initiation Protocol (SIP).


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

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-mmusic-ice-sip-sdp-11

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-ice-sip-sdp-11


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: Suhas Nandakumar <suhasietf@gmail.com>
Date: Fri, 20 Jan 2017 13:11:44 -0800
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Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-ice-sip-sdp-11.txt
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Hello All

 This is a keep alive version. We will follow with a new version once we
discuss review feedback from Adam.

Thanks
Suhas

On Fri, Jan 20, 2017 at 1:09 PM, <internet-drafts@ietf.org> wrote:

>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> This draft is a work item of the Multiparty Multimedia Session Control of
> the IETF.
>
>         Title           : Using Interactive Connectivity Establishment
> (ICE) with Session Description Protocol (SDP) offer/answer and Session
> Initiation Protocol (SIP)
>         Authors         : Marc Petit-Huguenin
>                           Ari Keranen
>                           Suhas Nandakumar
>         Filename        : draft-ietf-mmusic-ice-sip-sdp-11.txt
>         Pages           : 42
>         Date            : 2017-01-20
>
> Abstract:
>    This document describes how Interactive Connectivity Establishment
>    (ICE) is used with Session Description Protocol (SDP) offer/answer
>    and Session Initiation Protocol (SIP).
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-mmusic-ice-sip-sdp/
>
> There's also a htmlized version available at:
> https://tools.ietf.org/html/draft-ietf-mmusic-ice-sip-sdp-11
>
> A diff from the previous version is available at:
> https://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-ice-sip-sdp-11
>
>
> 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
>

--001a11406072d9497005468d1845
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<div dir=3D"ltr">Hello All<div><br></div><div>=C2=A0This is a keep alive ve=
rsion. We will follow with a new version once we discuss review feedback fr=
om Adam.</div><div><br></div><div>Thanks</div><div>Suhas</div></div><div cl=
ass=3D"gmail_extra"><br><div class=3D"gmail_quote">On Fri, Jan 20, 2017 at =
1:09 PM,  <span dir=3D"ltr">&lt;<a href=3D"mailto:internet-drafts@ietf.org"=
 target=3D"_blank">internet-drafts@ietf.org</a>&gt;</span> wrote:<br><block=
quote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc=
 solid;padding-left:1ex"><br>
A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.<br>
This draft is a work item of the Multiparty Multimedia Session Control of t=
he IETF.<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Title=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 Using Interactive Connectivity Establishment (ICE) with Session Descriptio=
n Protocol (SDP) offer/answer and Session Initiation Protocol (SIP)<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Authors=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0: Marc=
 Petit-Huguenin<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 Ari Keranen<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 Suhas Nandakumar<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Filename=C2=A0 =C2=A0 =C2=A0 =C2=A0 : draft-iet=
f-mmusic-ice-sip-sdp-<wbr>11.txt<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Pages=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 42<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Date=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :=
 2017-01-20<br>
<br>
Abstract:<br>
=C2=A0 =C2=A0This document describes how Interactive Connectivity Establish=
ment<br>
=C2=A0 =C2=A0(ICE) is used with Session Description Protocol (SDP) offer/an=
swer<br>
=C2=A0 =C2=A0and Session Initiation Protocol (SIP).<br>
<br>
<br>
The IETF datatracker status page for this draft is:<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-mmusic-ice-sip-sdp/"=
 rel=3D"noreferrer" target=3D"_blank">https://datatracker.ietf.org/<wbr>doc=
/draft-ietf-mmusic-ice-sip-<wbr>sdp/</a><br>
<br>
There&#39;s also a htmlized version available at:<br>
<a href=3D"https://tools.ietf.org/html/draft-ietf-mmusic-ice-sip-sdp-11" re=
l=3D"noreferrer" target=3D"_blank">https://tools.ietf.org/html/<wbr>draft-i=
etf-mmusic-ice-sip-sdp-<wbr>11</a><br>
<br>
A diff from the previous version is available at:<br>
<a href=3D"https://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-ice-sip-sd=
p-11" rel=3D"noreferrer" target=3D"_blank">https://www.ietf.org/rfcdiff?<wb=
r>url2=3Ddraft-ietf-mmusic-ice-<wbr>sip-sdp-11</a><br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n<br>
until the htmlized version and diff are available at <a href=3D"http://tool=
s.ietf.org" rel=3D"noreferrer" target=3D"_blank">tools.ietf.org</a>.<br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" rel=3D"noreferrer" target=
=3D"_blank">ftp://ftp.ietf.org/internet-<wbr>drafts/</a><br>
<br>
______________________________<wbr>_________________<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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/mmusic</a><br=
>
</blockquote></div><br></div>

--001a11406072d9497005468d1845--


From nobody Fri Jan 20 14:01:57 2017
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From: "Ben Campbell" <ben@nostrum.com>
To: draft-ietf-mmusic-sctp-sdp.all@ietf.org, mmusic <mmusic@ietf.org>
Date: Fri, 20 Jan 2017 16:01:49 -0600
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Subject: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21
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Hi,

This is my AD Evaluation of draft-ietf-mmusic-sctp-sdp-21. The draft is 
mostly in good shape, but I would like to at least discuss the 
substantive comments prior to IETF last call.

Thanks!

Ben.

----------------
Substantive Comments

-1, first paragraph: RFC 5572-update is currently in IETF LC. Is there a 
reason not to reference it instead?

-4.1, last paragraph:
I assume this paragraph refers to fmt values used with UDP/DTLS/SCTP or 
TCP/DTLS/SCTP, not fmt values in general. It would help to be explicit 
about that. (Also, it seem like this paragraph belongs in section 4.3).

-4.5: The example is missing the sctp-port attribute.

-5.3: Why is the mux-category SPECIAL rather than CAUTION? IIUC, SPECIAL 
means you need to refer to the rules in the protocol definition. But the 
text here basically say that the rules are undefined.

-6.1: What is meant by saying an "endpoint MUST assume that larger ... 
will be rejected"? Can that be stated in terms of actual procedure (e.g. 
"endpoint MUST NOT send...larger"?

-6.2, "Purpose" field: Please include the unit here, too.

-9.1, 2nd paragraph: Don't the lower layers need to be established 
before the upper layers? Won't removal of the TCP connection or DTLS 
association break an existing SCTP association?

-18.2: Seems like RFC0793 and RFC6544 should be normative references.

Editorial Comments:

-1, 2nd paragraph: Seems like SCTP is also used to transport data :-)

-1, last paragraph: is "strongly RECOMMENDED" intended to be stronger 
than RECOMMENDED? Assuming you don't mean MUST, please consider dropping 
the adverb.

-6.1, typo: thevmaximum

-9.1, 2nd paragraph: s/mange/manage

-9.3, last paragraph: What does "impact" mean in context? Change state? 
(It seems like these actions would likely cause data to be sent at the 
SCTP layer, which seems like a kind of impact.)

-10.3: Does "identical" mean identical to that from the offer?

-10.3, 8th paragraph: s/"closing establishing"/"closing or establishing"


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From: Paul Kyzivat <paul.kyzivat@comcast.net>
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Date: Fri, 20 Jan 2017 18:35:42 -0500
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Subject: Re: [MMUSIC] WG Last Call for draft-ietf-mmusic-data-channel-sdpneg-11
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On 1/20/17 2:16 AM, Christian Groves wrote:
> Hello Paul,
>
> Please see below.
>
> Regards, Christian
>
> On 17/01/2017 6:56 AM, Paul Kyzivat wrote:
>> I've followed this document closely throughout its development.
>> In preparation for this message I reviewed the document again. I found
>> some things that need to be addressed:
>>
>> 1) MAJOR: Section 5.1.1.3 and various other places:
>>
>> I can find nothing here or elsewhere in the document that says whether
>> the the stream id in an offer is the id on which the offerer will
>> receive, or the ID on which it will send. Of course this is irrelevant
>> if both ends agree to use the same id, but that isn't required.
>>
>> For this to interact properly with attributes for individual streams
>> (in dcsa) I think it will be necessary for this to be consistent with
>> the say SDP negotiates media sections - namely that the SDP in an
>> offer identifies where the offerer wants to *receive* the media.
>>
>> It will probably require changes in a variety of places to get this
>> sorted out.
>
> [CNG] There is some guidance in clause 5.2.1 around managing stream
> identifiers. Utilising the same Stream ID is required when supporting
> WebRTC datachannel (see clause 6.4/draft-ietf-rtcweb-data-channel-13).
> Clause 5.1.1 / ietf-mmusic-data-channel-sdpneg vaguely mentions that the
> opposite datachannels have the same set of attributes. SCTP stream
> identifier being one of those.

I was taking that into account, and yet found it incomplete.

However, upon rereading, I did find some language in section 5.2.3 that 
helps clarify this:

    The peer receiving such an SDP offer performs the following:
    ...
    o  For accepted data channels, it creates peer instances for the data
       channels with the agent using the channel parameters described in
       the SDP offer.  Note that the agent is asked to create data
       channels with SCTP stream identifiers contained in the SDP offer
       if the SDP offer is accepted.

The "Note" is what resolves any ambiguity. But it is written as if it 
was an implication that is simply being noted, rather than being a 
normative statement.

So I propose that it be revised as follows:

    o  For accepted data channels, the agent MUST create peer instances
       for the data channels using the SCTP stream identifiers and
       channel parameters contained in the SDP offer.

	Thanks,
	Paul

>> 2) MINOR: Section 5.1.1 says:
>>
>>    The intention in exchanging these attributes is to create, on two
>>    peers, without use of DCEP [I-D.ietf-rtcweb-data-protocol], matched
>>    pairs of oppositely directed data channels having the same set of
>>    attributes.  It is assumed that the data channel properties
>>    (reliable/partially reliable, ordered/unordered) are suitable per the
>>    subprotocol transport requirements.
>>
>> In this, "matched pairs of oppositely directed data channels" is
>> improper terminology. Each channel *is* a matched pair, of oppositely
>> directed SCTP streams.
> [CNG] Yes I agree.
>> ..snip..
>
> _______________________________________________
> 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: Ben Campbell <ben@nostrum.com>, "draft-ietf-mmusic-sctp-sdp.all@ietf.org" <draft-ietf-mmusic-sctp-sdp.all@ietf.org>, mmusic <mmusic@ietf.org>
Thread-Topic: AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - EDITORIAL comments
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Date: Sat, 21 Jan 2017 14:55:46 +0000
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - EDITORIAL comments
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Hi Ben,

Thanks for your review! Please see inline for my replies to your EDITORIAL =
comments (I'll address the technical ones in a separate e-mail).

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

Editorial Comments:

> -1, 2nd paragraph: Seems like SCTP is also used to transport data :-)

Occasionally, yes :)

I suggest the following new text:

   "The Stream Control Transmission Protocol (SCTP) [RFC4960] is a
   reliable transport protocol used to transport data between two
   endpoints using SCTP associations."

----

> -1, last paragraph: is "strongly RECOMMENDED" intended to be stronger tha=
n RECOMMENDED? Assuming you don't mean MUST, please consider dropping the a=
dverb.

I suggest to use "RECOMMENDED".

----

> -6.1, typo: thevmaximum

Will be fixed.

----

>-9.1, 2nd paragraph: s/mange/manage

Will be fixed.

----

>-9.3, last paragraph: What does "impact" mean in context? Change state?=20
>(It seems like these actions would likely cause data to be sent at the SCT=
P layer, which seems like a kind of impact.)

I am not sure I understand what you mean by "sent at the SCTP layer". If an=
 SCTP association is closed, no data can be sent on the STCP layer.

The purpose of the text is to point out that the underlying DTLS associatio=
n is not impacted/affected when closing/establishing SCTP associations usin=
g the 'sctp-port' attribute. That also means that the state is not changed.

----

>-10.3: Does "identical" mean identical to that from the offer?

Correct. Note that the bullet text refers to the text in the first paragrap=
h, which talks about the m- line in the offer.

But, if you think it is unclear, we could say:

"with an m- line proto value [RFC3264] identical to the value in the offer.=
"

----

>-10.3, 8th paragraph: s/"closing establishing"/"closing or establishing"

I suggest to say:

"for closing the currently used, and establishing a new, DTLS association"

----

Regards,

Christer


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Ben Campbell <ben@nostrum.com>, "draft-ietf-mmusic-sctp-sdp.all@ietf.org" <draft-ietf-mmusic-sctp-sdp.all@ietf.org>, mmusic <mmusic@ietf.org>, "Roman Shpount" <roman@telurix.com>
Thread-Topic: AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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Hi,

Below are my replies to Ben's technical comments.

----------------
Substantive Comments

> -1, first paragraph: RFC 5572-update is currently in IETF LC. Is there a =
reason not to reference it instead?

I assume you mean 4572-update :)

I agree that we should reference that spec instead, and I'll fix that.

----

> -4.1, last paragraph:
> I assume this paragraph refers to fmt values used with UDP/DTLS/SCTP or T=
CP/DTLS/SCTP,=20
> not fmt values in general.

Correct.=20

> It would help to be explicit about that.

The first sentence in section 4.1 does say:

   "This section defines the following new SDP Media Description (m-
   line) protocol identifiers (proto values) for describing an SCTP
   association: 'UDP/DTLS/SCTP' and 'TCP/DTLS/SCTP'."

> (Also, it seem like this paragraph belongs in section 4.3).

I could move and combine it with the 4th paragraph of section 4.3:

   "The m- line fmt value, identifying the application-layer protocol,
   MUST be registered by IANA. Section 15.3 defines the IANA registry=20
   for the media format namespace."

----

> -4.5: The example is missing the sctp-port attribute.

I'll add it.

----

> -5.3: Why is the mux-category SPECIAL rather than CAUTION? IIUC, SPECIAL =
means you need to refer
> to the rules in the protocol definition. But the text here basically say =
that the rules are undefined.

I'm ok using CAUTION. I agree it might be more appropriate than SPECIAL.

----

> -6.1: What is meant by saying an "endpoint MUST assume that larger ...=20
> will be rejected"? Can that be stated in terms of actual procedure (e.g.=
=20
> "endpoint MUST NOT send...larger"?

I'd like Roman's input on this, in case he had some cases in mind where end=
point will have to send larger messages.

I guess we could say SHOULD NO send larger message, and explain that one ca=
nnot assume that larger message (if sent for whatever reason) will be accep=
ted by the peer.

----

> -6.2, "Purpose" field: Please include the unit here, too.

I'll change to "maximum message size (indicated in bytes) that..."
                                =20
----

> -9.1, 2nd paragraph: Don't the lower layers need to be established before=
 the
> upper layers? Won't removal of the TCP connection or DTLS association bre=
ak=20
> an existing SCTP association?

Roman had lots of input on this, so I'll let him reply.

----

> -18.2: Seems like RFC0793 and RFC6544 should be normative references.

I'll fix that.

Regards,

Christer


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From: Andy Hutton <andyhutton.ietf@gmail.com>
Date: Mon, 23 Jan 2017 11:51:14 +0000
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--f403045fb638e902d30546c19de1
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I have just submitted this draft.

The draft is the result of discussion in the SIPBrandy working group
regarding how to proceed with
https://tools.ietf.org/html/draft-ietf-sipbrandy-osrtp-01.  The SIPBrandy
WG was not able to progress the opportunistic SRTP work because it required
an update to RFC 4568 which is not covered by the SIPBrandy charter.

This short draft is therefore submitted to MMUSIC with a view to updating
the relevant RFC's and progressing the OSRTP work in SIPBrandy.

Regards
Andy




---------- Forwarded message ----------
From: <internet-drafts@ietf.org>
Date: Mon, Jan 23, 2017 at 11:08 AM
Subject: I-D Action: draft-hutton-mmusic-opportunistic-negotiation-00.txt
To: i-d-announce@ietf.org



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


        Title           : Negotiating SRTP and RTCP Feedback using the
RTP/AVP Profile
        Authors         : Andrew Hutton
                          Roland Jesske
                          Alan Johnston
                          Gonzalo Salgueiro
                          Bernard Aboba
        Filename        : draft-hutton-mmusic-opportunis
tic-negotiation-00.txt
        Pages           : 6
        Date            : 2017-01-23

Abstract:
   This document describes how the use of the Secure Real-time transport
   protocol (SRTP) [RFC3711]. can be negotiated using the AVP (Audio
   Video Profile) defined in [RFC3551].  Such a mechanism is used to
   provide a means for encrypted media to be used in environments where
   support for encryption is not known in advance, and not required.
   The same mechanism is also applied to negotiation of the Extended RTP
   Profile for Real-time Transport Control Protocol Based Feedback (RTP/
   AVPF) [RFC4585].


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-hutton-mmusic-opportu
nistic-negotiation/

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-hutton-mmusic-opportunistic-negotiation-00


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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--f403045fb638e902d30546c19de1
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<div dir=3D"ltr"><div><br></div><div>I have just submitted this draft.</div=
><div><br></div><div>The draft is the result of discussion in the SIPBrandy=
 working group regarding how to proceed with=C2=A0<a href=3D"https://tools.=
ietf.org/html/draft-ietf-sipbrandy-osrtp-01">https://tools.ietf.org/html/dr=
aft-ietf-sipbrandy-osrtp-01</a>.=C2=A0 The SIPBrandy WG was not able to pro=
gress the opportunistic SRTP work because it required an update to RFC 4568=
 which is not covered by the SIPBrandy charter.</div><div><br></div><div>Th=
is short draft is therefore submitted to MMUSIC with a view to updating the=
 relevant RFC&#39;s and progressing the OSRTP work in SIPBrandy.</div><div>=
<br></div><div>Regards</div><div>Andy</div><div><br></div><div><br></div><d=
iv><br></div><br><div class=3D"gmail_quote">---------- Forwarded message --=
--------<br>From: <b class=3D"gmail_sendername"></b> <span dir=3D"ltr">&lt;=
<a href=3D"mailto:internet-drafts@ietf.org" target=3D"_blank">internet-draf=
ts@ietf.org</a>&gt;</span><br>Date: Mon, Jan 23, 2017 at 11:08 AM<br>Subjec=
t: I-D Action: draft-hutton-mmusic-<wbr>opportunistic-negotiation-00.<wbr>t=
xt<br>To: <a href=3D"mailto:i-d-announce@ietf.org" target=3D"_blank">i-d-an=
nounce@ietf.org</a><br><br><br><br>
A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.<br>
<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Title=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 Negotiating SRTP and RTCP Feedback using the RTP/AVP Profile<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Authors=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0: Andr=
ew Hutton<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 Roland Jesske<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 Alan Johnston<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 Gonzalo Salgueiro<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 Bernard Aboba<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Filename=C2=A0 =C2=A0 =C2=A0 =C2=A0 : draft-hut=
ton-mmusic-opportunis<wbr>tic-negotiation-00.txt<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Pages=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 6<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Date=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :=
 2017-01-23<br>
<br>
Abstract:<br>
=C2=A0 =C2=A0This document describes how the use of the Secure Real-time tr=
ansport<br>
=C2=A0 =C2=A0protocol (SRTP) [RFC3711]. can be negotiated using the AVP (Au=
dio<br>
=C2=A0 =C2=A0Video Profile) defined in [RFC3551].=C2=A0 Such a mechanism is=
 used to<br>
=C2=A0 =C2=A0provide a means for encrypted media to be used in environments=
 where<br>
=C2=A0 =C2=A0support for encryption is not known in advance, and not requir=
ed.<br>
=C2=A0 =C2=A0The same mechanism is also applied to negotiation of the Exten=
ded RTP<br>
=C2=A0 =C2=A0Profile for Real-time Transport Control Protocol Based Feedbac=
k (RTP/<br>
=C2=A0 =C2=A0AVPF) [RFC4585].<br>
<br>
<br>
The IETF datatracker status page for this draft is:<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-hutton-mmusic-opportunist=
ic-negotiation/" rel=3D"noreferrer" target=3D"_blank">https://datatracker.i=
etf.org/d<wbr>oc/draft-hutton-mmusic-opportu<wbr>nistic-negotiation/</a><br=
>
<br>
There&#39;s also a htmlized version available at:<br>
<a href=3D"https://tools.ietf.org/html/draft-hutton-mmusic-opportunistic-ne=
gotiation-00" rel=3D"noreferrer" target=3D"_blank">https://tools.ietf.org/h=
tml/dr<wbr>aft-hutton-mmusic-opportunisti<wbr>c-negotiation-00</a><br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n<br>
until the htmlized version and diff are available at <a href=3D"http://tool=
s.ietf.org" rel=3D"noreferrer" target=3D"_blank">tools.ietf.org</a>.<br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" rel=3D"noreferrer" target=
=3D"_blank">ftp://ftp.ietf.org/internet-dr<wbr>afts/</a><br>
<br>
______________________________<wbr>_________________<br>
I-D-Announce mailing list<br>
<a href=3D"mailto:I-D-Announce@ietf.org" target=3D"_blank">I-D-Announce@iet=
f.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/i-d-announceInternet-Draft=
" rel=3D"noreferrer" target=3D"_blank">https://www.ietf.org/mailman/l<wbr>i=
stinfo/i-d-announce<br>
Internet-Draft</a> directories: <a href=3D"http://www.ietf.org/shadow.html"=
 rel=3D"noreferrer" target=3D"_blank">http://www.ietf.org/shadow.htm<wbr>l<=
/a><br>
or <a href=3D"ftp://ftp.ietf.org/ietf/1shadow-sites.txt" rel=3D"noreferrer"=
 target=3D"_blank">ftp://ftp.ietf.org/ietf/1shado<wbr>w-sites.txt</a><br>
</div><br></div>

--f403045fb638e902d30546c19de1--


From nobody Mon Jan 23 07:18:15 2017
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From: Harald Alvestrand <harald@alvestrand.no>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [MMUSIC] Modifying an approved document: MSID
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Den 19. jan. 2017 13:31, skrev Harald Alvestrand:
> Ted pointed out to me that I sent this to the wrong group.
> 
> Chairs and members, please advise.

My proposed change is here:

https://github.com/alvestrand/rtcweb-msid/pull/16

The filed issue is here:

https://github.com/alvestrand/rtcweb-msid/issues/15

(CCing RTCWEB - please keep discussion, if any, on mmusic)

> 
> 
> 
> -------- Forwarded Message --------
> Subject: 	[rtcweb] Modifying an approved document: MSID
> Date: 	Wed, 18 Jan 2017 23:22:14 +0100
> From: 	Harald Alvestrand <harald@alvestrand.no>
> To: 	rtcweb@ietf.org <rtcweb@ietf.org>
> 
> 
> 
> When reviewing the implications of the PeerConnection API change to
> support AddTrack rather than AddStream, I found an issue.
> 
> This issue concerns draft-ietf-rtcweb-msid, which is currently in
> REF-WAIT state (I believe).
> 
> The issue is that it is possible to add a track without specifying a
> stream. Since the track's ID needs to be carried, we have to send an
> "a=msid" line, but the track's ID is the *second* field on that line,
> with the first being the stream's ID.
> 
> This creates a problem.
> 
> Suggested fix: Insert two lines in the document:
> 
> 1) On SDP generation:
> 
> "If there is no stream associated with the track, use the reserved ID
> value '-'"
> 
> 2) On SDP parsing
> 
> "If the stream ID is the reserved value '-', the track is not associated
> with a stream, and no stream is signalled or created."
> 
> If this is OK with the community, I'll issue an updated draft with this
> change.
> 
> 
> -- 
> Surveillance is pervasive. Go Dark.
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> 
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
> 


From nobody Mon Jan 23 16:13:41 2017
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From: "Ben Campbell" <ben@nostrum.com>
To: "Christer Holmberg" <christer.holmberg@ericsson.com>
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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Thanks for the response. Replies inline.

Ben.

On 21 Jan 2017, at 14:23, Christer Holmberg wrote:

> Hi,
>
> Below are my replies to Ben's technical comments.
>
> ----------------
> Substantive Comments
>
>> -1, first paragraph: RFC 5572-update is currently in IETF LC. Is 
>> there a reason not to reference it instead?
>
> I assume you mean 4572-update :)

Yes :-)

>
> I agree that we should reference that spec instead, and I'll fix that.
>
> ----
>
>> -4.1, last paragraph:
>> I assume this paragraph refers to fmt values used with UDP/DTLS/SCTP 
>> or TCP/DTLS/SCTP,
>> not fmt values in general.
>
> Correct.
>
>> It would help to be explicit about that.
>
> The first sentence in section 4.1 does say:
>
>    "This section defines the following new SDP Media Description (m-
>    line) protocol identifiers (proto values) for describing an SCTP
>    association: 'UDP/DTLS/SCTP' and 'TCP/DTLS/SCTP'."

I'm not sure all readers will see the connection between the first and 
last paragraph. I still think it would help to explicitly mention it. 
(Especially if it moves to 4.3)

An alternative, if it were to stay in 4.1 would be to have the opening 
paragraph in 4.1 explicitly mention that it includes the fmt values to 
be used in the context of these new proto values.

>
>> (Also, it seem like this paragraph belongs in section 4.3).
>
> I could move and combine it with the 4th paragraph of section 4.3:
>
>    "The m- line fmt value, identifying the application-layer protocol,
>    MUST be registered by IANA. Section 15.3 defines the IANA registry
>    for the media format namespace."
>
> ----

[...]


> ----
>
>> -5.3: Why is the mux-category SPECIAL rather than CAUTION? IIUC, 
>> SPECIAL means you need to refer
>> to the rules in the protocol definition. But the text here basically 
>> say that the rules are undefined.
>
> I'm ok using CAUTION. I agree it might be more appropriate than 
> SPECIAL.

I'm glad we agree :-) This is probably a big enough change to make sure 
the workgroup also agrees, though.

>
> ----
>
>> -6.1: What is meant by saying an "endpoint MUST assume that larger 
>> ...
>> will be rejected"? Can that be stated in terms of actual procedure 
>> (e.g.
>> "endpoint MUST NOT send...larger"?
>
> I'd like Roman's input on this, in case he had some cases in mind 
> where endpoint will have to send larger messages.
>
> I guess we could say SHOULD NO send larger message, and explain that 
> one cannot assume that larger message (if sent for whatever reason) 
> will be accepted by the peer.

That would be reasonable if it is the right answer. I just want to avoid 
the vagueness of "MUST assume".
>

[...]

> ----
>
>> -9.1, 2nd paragraph: Don't the lower layers need to be established 
>> before the
>> upper layers? Won't removal of the TCP connection or DTLS association 
>> break
>> an existing SCTP association?
>
> Roman had lots of input on this, so I'll let him reply.

Okay

[...]


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From: Roman Shpount <roman@telurix.com>
Date: Mon, 23 Jan 2017 20:16:53 -0500
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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On Mon, Jan 23, 2017 at 7:13 PM, Ben Campbell <ben@nostrum.com> wrote:

> On 21 Jan 2017, at 14:23, Christer Holmberg wrote:
>
>> -9.1, 2nd paragraph: Don't the lower layers need to be established before
>>> the
>>> upper layers? Won't removal of the TCP connection or DTLS association
>>> break
>>> an existing SCTP association?
>>>
>>
>> Roman had lots of input on this, so I'll let him reply.
>>
>
> Okay
>

Replacing of TCP connection or DTLS association does not break the SCTP
association. SCTP is designed to run over unreliable transports, so short
term loss of connectivity when one TCP connection or one DTLS association
is being replaced by the other would be handled by the SCTP re-transmit
timers. Also, when ICE is used, switching from one ICE candidate to
another, including switching between two ICE tcp candidates or switching
from udp to tcp candidate, does not affect existing SCTP association.
Existing SCTP association continues to run despite the underlying candidate
switch and as far as SCTP association is concerned it is running over the
same unreliable transport. Finally, DTLS association can be re-established
to force re-keying the long running connection. This can be done without
affecting SCTP association running on top of these DTLS associations.

This is why we are saying that the SCTP association, the DTLS association
and the TCP connection are managed independently from each other.  Each can
be established and  closed without impacting others. Each protocol uses its
own means to detect connection closure, failure or timeout.

I am also cautious of immediately closing SCTP association on TCP
connection or DTLS association closure. There are certain situations, when
these connections would be closed due to 3pcc collisions only to be
immediately opened again. I think, if SCTP association should be closed, it
needs to be explicitly signaled at SCTP level.

Regards,
_____________
Roman Shpount

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure"><br></div></div><div class=3D"gmail_quote">On Mon, Jan 23, 2017 at 7:1=
3 PM, Ben Campbell <span dir=3D"ltr">&lt;<a href=3D"mailto:ben@nostrum.com"=
 target=3D"_blank">ben@nostrum.com</a>&gt;</span> wrote:<br><blockquote cla=
ss=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid =
rgb(204,204,204);padding-left:1ex"><span class=3D"gmail-">On 21 Jan 2017, a=
t 14:23, Christer Holmberg wrote:<br></span><span class=3D"gmail-"><blockqu=
ote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px=
 solid rgb(204,204,204);padding-left:1ex"><blockquote class=3D"gmail_quote"=
 style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);p=
adding-left:1ex">-9.1, 2nd paragraph: Don&#39;t the lower layers need to be=
 established before the<br>
upper layers? Won&#39;t removal of the TCP connection or DTLS association b=
reak<br>
an existing SCTP association?<br>
</blockquote>
<br>
Roman had lots of input on this, so I&#39;ll let him reply.<br>
</blockquote>
<br></span>
Okay<br></blockquote><div><br></div><div>Replacing of TCP connection or DTL=
S association does not break the SCTP association. SCTP is designed to run =
over unreliable transports, so short term loss of connectivity when one TCP=
 connection or one DTLS association is being replaced by the other would be=
 handled by the SCTP re-transmit timers. Also, when ICE is used, switching =
from one ICE candidate to another, including switching between two ICE tcp =
candidates or switching from udp to tcp candidate, does not affect existing=
 SCTP association. Existing SCTP association continues to run despite the u=
nderlying candidate switch and as far as SCTP association is concerned it i=
s running over the same unreliable transport. Finally, DTLS association can=
 be re-established to force re-keying the long running connection. This can=
 be done without affecting SCTP association running on top of these DTLS as=
sociations.</div><br>This is why we are saying that the SCTP association, t=
he DTLS association and the TCP connection are=C2=A0managed independently f=
rom each other.=C2=A0 Each can be established and=C2=A0 closed without impa=
cting others. Each protocol uses its own means to detect connection closure=
, failure or timeout.</div><div class=3D"gmail_quote"><br></div><div class=
=3D"gmail_quote">I am also cautious of immediately closing SCTP association=
 on TCP connection or DTLS association closure. There are certain situation=
s, when these connections would be closed due to 3pcc collisions only to be=
 immediately opened again. I think, if SCTP association should be closed, i=
t needs to be explicitly signaled at SCTP level. =C2=A0<br><br>Regards,<br>=
_____________<div><div class=3D"gmail_signature">Roman Shpount</div></div><=
div>=C2=A0</div></div></div></div>

--001a11479b4a159f810546ccdf49--


From nobody Mon Jan 23 18:25:42 2017
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From: "Ben Campbell" <ben@nostrum.com>
To: "Christer Holmberg" <christer.holmberg@ericsson.com>
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - EDITORIAL comments
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Thanks for the response. Replies inline.

Ben.


On 21 Jan 2017, at 8:55, Christer Holmberg wrote:

> Hi Ben,
>
> Thanks for your review! Please see inline for my replies to your 
> EDITORIAL comments (I'll address the technical ones in a separate 
> e-mail).
>
> ----------------
>
> Editorial Comments:
>
>> -1, 2nd paragraph: Seems like SCTP is also used to transport data :-)
>
> Occasionally, yes :)
>
> I suggest the following new text:
>
>    "The Stream Control Transmission Protocol (SCTP) [RFC4960] is a
>    reliable transport protocol used to transport data between two
>    endpoints using SCTP associations."

WFM.

[...]

> ----
>
>> -9.3, last paragraph: What does "impact" mean in context? Change 
>> state?
>> (It seems like these actions would likely cause data to be sent at 
>> the SCTP layer, which seems like a kind of impact.)
>
> I am not sure I understand what you mean by "sent at the SCTP layer". 
> If an SCTP association is closed, no data can be sent on the STCP 
> layer.

Sorry, I meant sent at the DTLS layer.

>
> The purpose of the text is to point out that the underlying DTLS 
> association is not impacted/affected when closing/establishing SCTP 
> associations using the 'sctp-port' attribute. That also means that the 
> state is not changed.

My point is that "impact" can mean many things. I think we are saying 
that the state of the DTLS security association is unchanged, right?

>
> ----
>
>> -10.3: Does "identical" mean identical to that from the offer?
>
> Correct. Note that the bullet text refers to the text in the first 
> paragraph, which talks about the m- line in the offer.
>
> But, if you think it is unclear, we could say:
>
> "with an m- line proto value [RFC3264] identical to the value in the 
> offer."

That would help.

>
> ----
>
>> -10.3, 8th paragraph: s/"closing establishing"/"closing or 
>> establishing"
>
> I suggest to say:
>
> "for closing the currently used, and establishing a new, DTLS 
> association"

WFM.


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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - EDITORIAL comments
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Hi,

...

>>> -9.3, last paragraph: What does "impact" mean in context? Change
>>> state?
>>> (It seems like these actions would likely cause data to be sent at
>>> the SCTP layer, which seems like a kind of impact.)
>>
>> I am not sure I understand what you mean by "sent at the SCTP layer".
>> If an SCTP association is closed, no data can be sent on the STCP
>> layer.
>
>Sorry, I meant sent at the DTLS layer.
>
>>
>> The purpose of the text is to point out that the underlying DTLS
>> association is not impacted/affected when closing/establishing SCTP
>> associations using the 'sctp-port' attribute. That also means that the
>> state is not changed.
>
>My point is that "impact" can mean many things. I think we are saying
>that the state of the DTLS security association is unchanged, right?

Yes.

So, would the following change address your issue?

 "NOTE: Closing and establishing a new SCTP association using the SDP
  'sctp-port' attribute will not affect the state of the underlying DTLS
  association.=B2


The last bullet of section 10.5 contains similar text, and I assume you
would have the same issue with that. If so, I could change it:

 "o  NOTE: This specification does not define a mechanism for
    explicitly closing a DTLS association while maintaining the
    overlying SCTP association.  However, if a DTLS association is
    closed and replaced with a new DTLS association, as a result of
    some other action [I-D.ietf-mmusic-dtls-sdp], the state of the
    SCTP association is not affected."


Personally I think =B3affected=B2 fits better than =B3changed=B2, but if yo=
u want
to use =B3changed=B2 I won=B9t argue against you :)


----



>>> -10.3: Does "identical" mean identical to that from the offer?
>>
>> Correct. Note that the bullet text refers to the text in the first
>> paragraph, which talks about the m- line in the offer.
>>
>> But, if you think it is unclear, we could say:
>>
>> "with an m- line proto value [RFC3264] identical to the value in the
>> offer."
>
>That would help.

I=B9ll change as suggested.


Regards,

Christer


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Ben Campbell <ben@nostrum.com>
Thread-Topic: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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Hi,

...


>>> -4.1, last paragraph:
>>> I assume this paragraph refers to fmt values used with UDP/DTLS/SCTP
>>> or TCP/DTLS/SCTP,
>>> not fmt values in general.
>>
>> Correct.
>>
>>> It would help to be explicit about that.
>>
>> The first sentence in section 4.1 does say:
>>
>>    "This section defines the following new SDP Media Description (m-
>>    line) protocol identifiers (proto values) for describing an SCTP
>>    association: 'UDP/DTLS/SCTP' and 'TCP/DTLS/SCTP'."
>
>I'm not sure all readers will see the connection between the first and
>last paragraph. I still think it would help to explicitly mention it.
>(Especially if it moves to 4.3)
>
>An alternative, if it were to stay in 4.1 would be to have the opening
>paragraph in 4.1 explicitly mention that it includes the fmt values to
>be used in the context of these new proto values.
>
>>
>>> (Also, it seem like this paragraph belongs in section 4.3).
>>
>> I could move and combine it with the 4th paragraph of section 4.3:
>>
>>    "The m- line fmt value, identifying the application-layer protocol,
>>    MUST be registered by IANA. Section 15.3 defines the IANA registry
>>    for the media format namespace."


Ok, so my suggestion is to:

1) Move the paragraph to section 4.3, as described above
2) Modify the first paragraph in section 4.1:

"This section defines the following new SDP Media Description (m-
   line) protocol identifiers (proto values) for describing an SCTP
   association: 'UDP/DTLS/SCTP' and 'TCP/DTLS/SCTP'.  The section also
   describes how an m- line, associated with the proto values, is
created, and how the the media format (=8Cfmt=B9) values are used."



----


>>> -5.3: Why is the mux-category SPECIAL rather than CAUTION? IIUC,
>>> SPECIAL means you need to refer
>>> to the rules in the protocol definition. But the text here basically
>>> say that the rules are undefined.
>>
>> I'm ok using CAUTION. I agree it might be more appropriate than
>> SPECIAL.
>
>I'm glad we agree :-)

You only mentioned section 5.3, but I think the same applies also to
section 6.3 (MUX category for the max-message-size-attribute), so I
suggest to change both to CAUTION.

>This is probably a big enough change to make sure the workgroup also
>agrees, though.


*Sounds of crickets*

Yes, it does agree :)


 ----


>>> -6.1: What is meant by saying an "endpoint MUST assume that larger
>>> ...
>>> will be rejected"? Can that be stated in terms of actual procedure
>>> (e.g.
>>> "endpoint MUST NOT send...larger"?
>>
>> I'd like Roman's input on this, in case he had some cases in mind
>> where endpoint will have to send larger messages.
>>
>> I guess we could say SHOULD NO send larger message, and explain that
>> one cannot assume that larger message (if sent for whatever reason)
>> will be accepted by the peer.
>
>That would be reasonable if it is the right answer. I just want to avoid
>the vagueness of "MUST assume".

I suggest the following modified text:

"An SCTP endpoint SHOULD NOT send a SCTP user message with a message size
that is larger than the maximum size indicated by the peer. If the SCTP
endpoint needs (for whatever reason) to send a larger message, it cannot
be assumed that the message will be accepted by the peer.=B2


----


>>>-9.1, 2nd paragraph: Don't the lower layers need to be established
>>> before the
>>> upper layers? Won't removal of the TCP connection or DTLS association
>>> break
>>> an existing SCTP association?
>>
>> Roman had lots of input on this, so I'll let him reply.
>
>Okay

See Roman=B9s reply.

Regards,

Christer


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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - EDITORIAL comments
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On 24 Jan 2017, at 7:03, Christer Holmberg wrote:

> Hi,
>
> ...
>
>>>> -9.3, last paragraph: What does "impact" mean in context? Change
>>>> state?
>>>> (It seems like these actions would likely cause data to be sent at
>>>> the SCTP layer, which seems like a kind of impact.)
>>>
>>> I am not sure I understand what you mean by "sent at the SCTP 
>>> layer".
>>> If an SCTP association is closed, no data can be sent on the STCP
>>> layer.
>>
>> Sorry, I meant sent at the DTLS layer.
>>
>>>
>>> The purpose of the text is to point out that the underlying DTLS
>>> association is not impacted/affected when closing/establishing SCTP
>>> associations using the 'sctp-port' attribute. That also means that 
>>> the
>>> state is not changed.
>>
>> My point is that "impact" can mean many things. I think we are saying
>> that the state of the DTLS security association is unchanged, right?
>
> Yes.
>
> So, would the following change address your issue?
>
>  "NOTE: Closing and establishing a new SCTP association using the SDP
>   'sctp-port' attribute will not affect the state of the underlying 
> DTLS
>   association.Â²
>
>
> The last bullet of section 10.5 contains similar text, and I assume 
> you
> would have the same issue with that. If so, I could change it:
>
>  "o  NOTE: This specification does not define a mechanism for
>     explicitly closing a DTLS association while maintaining the
>     overlying SCTP association.  However, if a DTLS association is
>     closed and replaced with a new DTLS association, as a result of
>     some other action [I-D.ietf-mmusic-dtls-sdp], the state of the
>     SCTP association is not affected."
>

Both work for me.

>
> Personally I think Â³affectedÂ² fits better than Â³changedÂ², but if 
> you want
> to use Â³changedÂ² I wonÂ¹t argue against you :)

"Affected" is fine. It's the "...state of..." part that fixes it in my 
mind. :-)

[...]

Thanks!

Ben.


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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: Andy Hutton <andyhutton.ietf@gmail.com>, "mmusic-chairs@ietf.org" <mmusic-chairs@ietf.org>, "sipbrandy-chairs@ietf.org" <sipbrandy-chairs@ietf.org>, Ben Campbell <ben@nostrum.com>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Fwd: I-D Action: draft-hutton-mmusic-opportunistic-negotiation-00.txt
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Date: Tue, 24 Jan 2017 15:47:56 +0000
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Subject: Re: [MMUSIC] Fwd: I-D Action: draft-hutton-mmusic-opportunistic-negotiation-00.txt
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--_000_6AD021E52C844BED88D8DFD5773C768Bciscocom_--


From nobody Tue Jan 24 07:50:43 2017
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From: "Ben Campbell" <ben@nostrum.com>
To: "Christer Holmberg" <christer.holmberg@ericsson.com>
Date: Tue, 24 Jan 2017 09:50:33 -0600
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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On 24 Jan 2017, at 7:20, Christer Holmberg wrote:

> Hi,
>
> ...
>
>
>>>> -4.1, last paragraph:
>>>> I assume this paragraph refers to fmt values used with 
>>>> UDP/DTLS/SCTP
>>>> or TCP/DTLS/SCTP,
>>>> not fmt values in general.
>>>
>>> Correct.
>>>
>>>> It would help to be explicit about that.
>>>
>>> The first sentence in section 4.1 does say:
>>>
>>>    "This section defines the following new SDP Media Description (m-
>>>    line) protocol identifiers (proto values) for describing an SCTP
>>>    association: 'UDP/DTLS/SCTP' and 'TCP/DTLS/SCTP'."
>>
>> I'm not sure all readers will see the connection between the first 
>> and
>> last paragraph. I still think it would help to explicitly mention it.
>> (Especially if it moves to 4.3)
>>
>> An alternative, if it were to stay in 4.1 would be to have the 
>> opening
>> paragraph in 4.1 explicitly mention that it includes the fmt values 
>> to
>> be used in the context of these new proto values.
>>
>>>
>>>> (Also, it seem like this paragraph belongs in section 4.3).
>>>
>>> I could move and combine it with the 4th paragraph of section 4.3:
>>>
>>>    "The m- line fmt value, identifying the application-layer 
>>> protocol,
>>>    MUST be registered by IANA. Section 15.3 defines the IANA 
>>> registry
>>>    for the media format namespace."
>
>
> Ok, so my suggestion is to:
>
> 1) Move the paragraph to section 4.3, as described above
> 2) Modify the first paragraph in section 4.1:
>
> "This section defines the following new SDP Media Description (m-
>    line) protocol identifiers (proto values) for describing an SCTP
>    association: 'UDP/DTLS/SCTP' and 'TCP/DTLS/SCTP'.  The section also
>    describes how an m- line, associated with the proto values, is
> created, and how the the media format (Å’fmtÂ¹) values are used."

The change helps, but does it still make sense if the last paragraph of 
4.1 moves to 4.3?

>
>
>
> ----
>
>
>>>> -5.3: Why is the mux-category SPECIAL rather than CAUTION? IIUC,
>>>> SPECIAL means you need to refer
>>>> to the rules in the protocol definition. But the text here 
>>>> basically
>>>> say that the rules are undefined.
>>>
>>> I'm ok using CAUTION. I agree it might be more appropriate than
>>> SPECIAL.
>>
>> I'm glad we agree :-)
>
> You only mentioned section 5.3, but I think the same applies also to
> section 6.3 (MUX category for the max-message-size-attribute), so I
> suggest to change both to CAUTION.
>
>> This is probably a big enough change to make sure the workgroup also
>> agrees, though.
>
>
> *Sounds of crickets*
>
> Yes, it does agree :)

Ok, if no one objects before the IETF LC completes, we will assume 
agreement :-)

>
>
>  ----
>
>
>>>> -6.1: What is meant by saying an "endpoint MUST assume that larger
>>>> ...
>>>> will be rejected"? Can that be stated in terms of actual procedure
>>>> (e.g.
>>>> "endpoint MUST NOT send...larger"?
>>>
>>> I'd like Roman's input on this, in case he had some cases in mind
>>> where endpoint will have to send larger messages.
>>>
>>> I guess we could say SHOULD NO send larger message, and explain that
>>> one cannot assume that larger message (if sent for whatever reason)
>>> will be accepted by the peer.
>>
>> That would be reasonable if it is the right answer. I just want to 
>> avoid
>> the vagueness of "MUST assume".
>
> I suggest the following modified text:
>
> "An SCTP endpoint SHOULD NOT send a SCTP user message with a message 
> size
> that is larger than the maximum size indicated by the peer. If the 
> SCTP
> endpoint needs (for whatever reason) to send a larger message, it 
> cannot
> be assumed that the message will be accepted by the peer.Â²

I'm okay with that, but don't be surprised if we get questions about 
whether it ever would ever make sense to send a message that you can't 
assume the peer will accept. (I wonder if it's more a matter of 
"probably not accept", given the peer has already stated its preference.

>
>
> ----
>
>
>>>> -9.1, 2nd paragraph: Don't the lower layers need to be established
>>>> before the
>>>> upper layers? Won't removal of the TCP connection or DTLS 
>>>> association
>>>> break
>>>> an existing SCTP association?
>>>
>>> Roman had lots of input on this, so I'll let him reply.
>>
>> Okay
>
> See RomanÂ¹s reply.

Thanks, I will reply there.

Ben.


From nobody Tue Jan 24 08:38:45 2017
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From: Paul Kyzivat <paul.kyzivat@comcast.net>
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Date: Tue, 24 Jan 2017 11:38:39 -0500
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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Roman,

On 1/23/17 8:16 PM, Roman Shpount wrote:
>
> On Mon, Jan 23, 2017 at 7:13 PM, Ben Campbell <ben@nostrum.com
> <mailto:ben@nostrum.com>> wrote:
>
>     On 21 Jan 2017, at 14:23, Christer Holmberg wrote:
>
>             -9.1, 2nd paragraph: Don't the lower layers need to be
>             established before the
>             upper layers? Won't removal of the TCP connection or DTLS
>             association break
>             an existing SCTP association?
>
>
>         Roman had lots of input on this, so I'll let him reply.
>
>
>     Okay
>
>
> Replacing of TCP connection or DTLS association does not break the SCTP
> association. SCTP is designed to run over unreliable transports, so
> short term loss of connectivity when one TCP connection or one DTLS
> association is being replaced by the other would be handled by the SCTP
> re-transmit timers. Also, when ICE is used, switching from one ICE
> candidate to another, including switching between two ICE tcp candidates
> or switching from udp to tcp candidate, does not affect existing SCTP
> association. Existing SCTP association continues to run despite the
> underlying candidate switch and as far as SCTP association is concerned
> it is running over the same unreliable transport. Finally, DTLS
> association can be re-established to force re-keying the long running
> connection. This can be done without affecting SCTP association running
> on top of these DTLS associations.
>
> This is why we are saying that the SCTP association, the DTLS
> association and the TCP connection are managed independently from each
> other.  Each can be established and  closed without impacting others.
> Each protocol uses its own means to detect connection closure, failure
> or timeout.

I can see that this can be a good thing when it is what you want/need.

OTOH, if one of the two parties involved changes, and there is no state 
sharing, you need to establish a new SCTP association. This may turn out 
to be the case even when the offerer doesn't know that the other party 
has changed.

So how are these two cases distinguished in the signaling? AFAICT the 
only thing available is sctp-port. If the offerer doesn't realize a 
change is coming, then it won't change its port. And in that case the 
answerer won't know what port its predecessor used, so it might choose 
the same value.

Of course, this is largely a 3pcc scenario, and if it is initiated in 
the middle then the middle can probably know to change the port somewhere.

But I'm not sure if that is the only case.

	Thanks,
	Paul


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From: Paul Kyzivat <paul.kyzivat@comcast.net>
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments: CAUTION
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On 1/24/17 8:20 AM, Christer Holmberg wrote:

> You only mentioned section 5.3, but I think the same applies also to
> section 6.3 (MUX category for the max-message-size-attribute), so I
> suggest to change both to CAUTION.
>
>> This is probably a big enough change to make sure the workgroup also
>> agrees, though.
>
>
> *Sounds of crickets*
>
> Yes, it does agree :)

This change seems good to me.

	Thanks,
	Paul


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Ben Campbell <ben@nostrum.com>
Thread-Topic: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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From: Roman Shpount <roman@telurix.com>
Date: Tue, 24 Jan 2017 14:18:42 -0500
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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On Tue, Jan 24, 2017 at 11:38 AM, Paul Kyzivat <paul.kyzivat@comcast.net>
wrote:

> Roman,
>
> On 1/23/17 8:16 PM, Roman Shpount wrote:
>
>>
>> On Mon, Jan 23, 2017 at 7:13 PM, Ben Campbell <ben@nostrum.com
>> <mailto:ben@nostrum.com>> wrote:
>>
>>     On 21 Jan 2017, at 14:23, Christer Holmberg wrote:
>>
>>             -9.1, 2nd paragraph: Don't the lower layers need to be
>>             established before the
>>             upper layers? Won't removal of the TCP connection or DTLS
>>             association break
>>             an existing SCTP association?
>>
>>
>>         Roman had lots of input on this, so I'll let him reply.
>>
>>
>>     Okay
>>
>>
>> Replacing of TCP connection or DTLS association does not break the SCTP
>> association. SCTP is designed to run over unreliable transports, so
>> short term loss of connectivity when one TCP connection or one DTLS
>> association is being replaced by the other would be handled by the SCTP
>> re-transmit timers. Also, when ICE is used, switching from one ICE
>> candidate to another, including switching between two ICE tcp candidates
>> or switching from udp to tcp candidate, does not affect existing SCTP
>> association. Existing SCTP association continues to run despite the
>> underlying candidate switch and as far as SCTP association is concerned
>> it is running over the same unreliable transport. Finally, DTLS
>> association can be re-established to force re-keying the long running
>> connection. This can be done without affecting SCTP association running
>> on top of these DTLS associations.
>>
>> This is why we are saying that the SCTP association, the DTLS
>> association and the TCP connection are managed independently from each
>> other.  Each can be established and  closed without impacting others.
>> Each protocol uses its own means to detect connection closure, failure
>> or timeout.
>>
>
> I can see that this can be a good thing when it is what you want/need.
>
> OTOH, if one of the two parties involved changes, and there is no state
> sharing, you need to establish a new SCTP association. This may turn out to
> be the case even when the offerer doesn't know that the other party has
> changed.
>
> So how are these two cases distinguished in the signaling? AFAICT the only
> thing available is sctp-port. If the offerer doesn't realize a change is
> coming, then it won't change its port. And in that case the answerer won't
> know what port its predecessor used, so it might choose the same value.
>
> Of course, this is largely a 3pcc scenario, and if it is initiated in the
> middle then the middle can probably know to change the port somewhere.
>
> But I'm not sure if that is the only case.
>

We have considered that issue and assumed that selecting a random sctp-port
is sufficient to determine if this is the same vs new connection. There is
some possibility of collisions when answering party accidentally picks the
same port as the one which was previously used for SCTP communications, but
this is similar to ICE ufrag collisions, dtls-id collisions or to-tag
collisions. I do agree that sctp-port has less random bits then ufrag,
dtls-id, or to-tag, so it is more likely to collide. Please let us know if
this is a sufficient point of concern to warrant adding more randomness by
changing sctp-port definition (making it longer) or adding another
attribute. Another solution would be to require SCTP association restart on
DTLS association restart and use random bits in dtls-id minimize the chance
of collision.

In general, I am not strongly attached to not restarting SCTP association
on underlying transport change. I am even less attached to restarting SCTP
on DTLS restart.

What I do feel strongly about is:

1. Keeping SCTP association when ICE candidates change
2. Keeping SCTP association on ICE restart

Keeping SCTP association on ICE restart has exactly the same properties as
keeping SCTP association on transport change (it is unclear if ICE restart
was caused by the same endpoint or by endpoint change due to 3pcc). Since
ICE restart and transport change are essentially the same thing, I proposed
that SCTP association is preserved in both cases. If we tie SCTP
association restart to DTLS association restart we will get the desired
result for ICE only at the cost of loosing re-key ability for the existing
SCTP association.

So, my questions are:
1. Is possibility of sctp-port collision large enough concern to change
when SCTP association is restarted?
2. Would it be acceptable to lose ability to re-key DTLS association while
running the same SCTP assignations and force SCTP association restart on
DTLS association restart?

Regards,
_____________
Roman Shpount

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

<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"m_6132186892=
640943501gmail-m_-6991839479850713400gmail_signature">On Tue, Jan 24, 2017 =
at 11:38 AM, Paul Kyzivat <span dir=3D"ltr">&lt;<a href=3D"mailto:paul.kyzi=
vat@comcast.net" target=3D"_blank">paul.kyzivat@comcast.net</a>&gt;</span> =
wrote:<br></div></div><div class=3D"gmail_quote"><blockquote class=3D"gmail=
_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204=
,204);padding-left:1ex">Roman,<span class=3D"m_6132186892640943501gmail-m_-=
6991839479850713400gmail-"><br>
<br>
On 1/23/17 8:16 PM, Roman Shpount wrote:<br>
</span><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;=
border-left:1px solid rgb(204,204,204);padding-left:1ex"><span class=3D"m_6=
132186892640943501gmail-m_-6991839479850713400gmail-">
<br>
On Mon, Jan 23, 2017 at 7:13 PM, Ben Campbell &lt;<a href=3D"mailto:ben@nos=
trum.com" target=3D"_blank">ben@nostrum.com</a><br></span><div><div class=
=3D"m_6132186892640943501gmail-m_-6991839479850713400gmail-h5">
&lt;mailto:<a href=3D"mailto:ben@nostrum.com" target=3D"_blank">ben@nostrum=
.com</a>&gt;&gt; wrote:<br>
<br>
=C2=A0 =C2=A0 On 21 Jan 2017, at 14:23, Christer Holmberg wrote:<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 -9.1, 2nd paragraph: Don&#39;t th=
e lower layers need to be<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 established before the<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 upper layers? Won&#39;t removal o=
f the TCP connection or DTLS<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 association break<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 an existing SCTP association?<br>
<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Roman had lots of input on this, so I&#39;ll le=
t him reply.<br>
<br>
<br>
=C2=A0 =C2=A0 Okay<br>
<br>
<br>
Replacing of TCP connection or DTLS association does not break the SCTP<br>
association. SCTP is designed to run over unreliable transports, so<br>
short term loss of connectivity when one TCP connection or one DTLS<br>
association is being replaced by the other would be handled by the SCTP<br>
re-transmit timers. Also, when ICE is used, switching from one ICE<br>
candidate to another, including switching between two ICE tcp candidates<br=
>
or switching from udp to tcp candidate, does not affect existing SCTP<br>
association. Existing SCTP association continues to run despite the<br>
underlying candidate switch and as far as SCTP association is concerned<br>
it is running over the same unreliable transport. Finally, DTLS<br>
association can be re-established to force re-keying the long running<br>
connection. This can be done without affecting SCTP association running<br>
on top of these DTLS associations.<br>
<br>
This is why we are saying that the SCTP association, the DTLS<br>
association and the TCP connection are managed independently from each<br>
other.=C2=A0 Each can be established and=C2=A0 closed without impacting oth=
ers.<br>
Each protocol uses its own means to detect connection closure, failure<br>
or timeout.<br>
</div></div></blockquote>
<br>
I can see that this can be a good thing when it is what you want/need.<br>
<br>
OTOH, if one of the two parties involved changes, and there is no state sha=
ring, you need to establish a new SCTP association. This may turn out to be=
 the case even when the offerer doesn&#39;t know that the other party has c=
hanged.<br>
<br>
So how are these two cases distinguished in the signaling? AFAICT the only =
thing available is sctp-port. If the offerer doesn&#39;t realize a change i=
s coming, then it won&#39;t change its port. And in that case the answerer =
won&#39;t know what port its predecessor used, so it might choose the same =
value.<br>
<br>
Of course, this is largely a 3pcc scenario, and if it is initiated in the m=
iddle then the middle can probably know to change the port somewhere.<br>
<br>
But I&#39;m not sure if that is the only case.<br></blockquote><div><br></d=
iv><div>We have considered that issue and assumed that selecting a random s=
ctp-port is sufficient to determine if this is the same vs new connection. =
There is some possibility of collisions when answering party accidentally p=
icks the same port as the one which was previously used for SCTP communicat=
ions, but this is similar to ICE ufrag collisions,=C2=A0dtls-id collisions =
or to-tag collisions. I do agree that sctp-port has less random bits then u=
frag, dtls-id, or to-tag, so it is more likely to collide. Please let us kn=
ow if this is a sufficient point of concern to warrant adding more randomne=
ss by changing sctp-port definition (making it longer) or adding another at=
tribute. Another solution would be to require SCTP association restart on D=
TLS association restart and use random bits in dtls-id minimize the chance =
of collision.</div><div><br></div><div>In general, I am not strongly attach=
ed to not restarting SCTP association on underlying transport change. I am =
even less attached to restarting SCTP on DTLS restart.</div><div><br></div>=
<div>What I do feel strongly about is:</div><div><br></div><div>1. Keeping =
SCTP association when ICE candidates change</div><div>2. Keeping SCTP assoc=
iation on ICE restart</div><div><br></div><div>Keeping SCTP association on =
ICE restart has exactly the same properties as keeping SCTP association on =
transport change (it is unclear if ICE restart was caused by the same endpo=
int or by endpoint change due to 3pcc). Since ICE restart and transport cha=
nge are essentially the same thing, I proposed that SCTP association is pre=
served in both cases. If we tie SCTP association restart to DTLS associatio=
n restart we will get the desired result for ICE only at the cost of loosin=
g re-key ability for the existing SCTP association.</div><div><br></div><di=
v>So, my questions are:</div><div>1. Is possibility of sctp-port collision =
large enough concern to change when SCTP association is restarted?</div><di=
v>2. Would it be acceptable to lose ability to re-key DTLS association whil=
e running the same SCTP assignations and force SCTP association restart on =
DTLS association restart?=C2=A0</div><div><br></div><div>Regards,</div><div=
><div class=3D"m_6132186892640943501gmail-m_-6991839479850713400gmail_signa=
ture">_____________<br>Roman Shpount</div></div><div>=C2=A0</div></div></di=
v></div>

--001a11479b4a009e0b0546dbfc9f--


From nobody Tue Jan 24 11:22:21 2017
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To: "Christer Holmberg" <christer.holmberg@ericsson.com>
Date: Tue, 24 Jan 2017 13:22:11 -0600
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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On 24 Jan 2017, at 13:16, Christer Holmberg wrote:

> Hi,
>
>>>>>> -4.1, last paragraph:
>>>>>> I assume this paragraph refers to fmt values used with
>>>>>> UDP/DTLS/SCTP or TCP/DTLS/SCTP, not fmt values in general.
>>>>>
>>>>> Correct.
>>>>>
>>>>>> It would help to be explicit about that.
>>>>>
>>>>> The first sentence in section 4.1 does say:
>>>>>
>>>>>    "This section defines the following new SDP Media Description 
>>>>> (m-
>>>>>    line) protocol identifiers (proto values) for describing an 
>>>>> SCTP
>>>>>    association: 'UDP/DTLS/SCTP' and 'TCP/DTLS/SCTP'."
>>>>
>>>> I'm not sure all readers will see the connection between the first
>>>> and last paragraph. I still think it would help to explicitly 
>>>> mention
>>>> it.
>>>> (Especially if it moves to 4.3)
>>>>
>>>> An alternative, if it were to stay in 4.1 would be to have the
>>>> opening paragraph in 4.1 explicitly mention that it includes the 
>>>> fmt
>>>> values to be used in the context of these new proto values.
>>>>
>>>>>
>>>>>> (Also, it seem like this paragraph belongs in section 4.3).
>>>>>
>>>>> I could move and combine it with the 4th paragraph of section 4.3:
>>>>>
>>>>>    "The m- line fmt value, identifying the application-layer
>>>>>    protocol, MUST be registered by IANA. Section 15.3 defines the 
>>>>> IANA
>>>>>    registry for the media format namespace."
>>>>>
>>>>>
>>> Ok, so my suggestion is to:
>>>
>>> 1) Move the paragraph to section 4.3, as described above
>>> 2) Modify the first paragraph in section 4.1:
>>>
>>> "This section defines the following new SDP Media Description (m-
>>>    line) protocol identifiers (proto values) for describing an SCTP
>>>    association: 'UDP/DTLS/SCTP' and 'TCP/DTLS/SCTP'.  The section 
>>> also
>>>    describes how an m- line, associated with the proto values, is
>>>    created, and how the the media format (Å’fmtÂ¹) values are used."
>>
>> The change helps, but does it still make sense if the last paragraph 
>> of
>> 4.1 moves to 4.3?
>
> So, you suggest to not move the last paragraph to 4.3?

No, my suggestion is to either (put the additional sentence in the first 
paragraph of 4.1 and keep last paragraph in 4.1) _OR_ (move the 
paragraph to 4.3 and put the clarification in _that_ paragraph.)

(Personally I like the latter, but either are okay.)


>
> ----
>
>
>>>>>> -6.1: What is meant by saying an "endpoint MUST assume that 
>>>>>> larger
>>>>>> ...
>>>>>> will be rejected"? Can that be stated in terms of actual 
>>>>>> procedure
>>>>>> (e.g.
>>>>>> "endpoint MUST NOT send...larger"?
>>>>>
>>>>> I'd like Roman's input on this, in case he had some cases in mind
>>>>> where endpoint will have to send larger messages.
>>>>>
>>>>> I guess we could say SHOULD NO send larger message, and explain 
>>>>> that
>>>>> one cannot assume that larger message (if sent for whatever 
>>>>> reason)
>>>>> will be accepted by the peer.
>>>>
>>>> That would be reasonable if it is the right answer. I just want to
>>>> avoid the vagueness of "MUST assume".
>>>
>>> I suggest the following modified text:
>>>
>>> "An SCTP endpoint SHOULD NOT send a SCTP user message with a message
>>> size that is larger than the maximum size indicated by the peer. If
>>> the SCTP endpoint needs (for whatever reason) to send a larger
>>> message, it cannot be assumed that the message will be accepted by 
>>> the
>>> peer."
>>
>> I'm okay with that, but don't be surprised if we get questions about 
>> whether it ever would ever make
>> sense to send a message that you can't assume the peer will accept. 
>> (I wonder if it's more a matter
>> of "probably not accept", given the peer has already stated its 
>> preference.
>
> If nobody objects, I am ok saying "MUST NOT send".
>
> Because, if there is a case where it would be needed, the peers simply 
> need to support and negotiate a larger value.

Okay with me. If someone comes up with a plausible reason one might need 
to send larger messages, please speak up before the IETF LC completes. 
:-)

Ben.


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Ben Campbell <ben@nostrum.com>
Thread-Topic: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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From nobody Tue Jan 24 12:10:07 2017
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From: "Ben Campbell" <ben@nostrum.com>
To: "Christer Holmberg" <christer.holmberg@ericsson.com>
Date: Tue, 24 Jan 2017 14:10:00 -0600
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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All looks good to me. Please submit a revision when you are ready, and I 
will request the IETF Last call.

Thanks!

Ben.

On 24 Jan 2017, at 14:04, Christer Holmberg wrote:

> Hi,
>
>>>>>>>> -4.1, last paragraph:
>>>>>>>> I assume this paragraph refers to fmt values used with
>>>>>>>> UDP/DTLS/SCTP or TCP/DTLS/SCTP, not fmt values in general.
>>>>>>>
>>>>>>> Correct.
>>>>>>>
>>>>>>>> It would help to be explicit about that.
>>>>>>>
>>>>>>> The first sentence in section 4.1 does say:
>>>>>>>
>>>>>>>    "This section defines the following new SDP Media Description
>>>>>>>      (m- line) protocol identifiers (proto values) for 
>>>>>>> describing an
>>>>>>>      SCTP association: 'UDP/DTLS/SCTP' and 'TCP/DTLS/SCTP'."
>>>>>>
>>>>>> I'm not sure all readers will see the connection between the 
>>>>>> first
>>>>>> and last paragraph. I still think it would help to explicitly
>>>>>> mention it.
>>>>>> (Especially if it moves to 4.3)
>>>>>>
>>>>>> An alternative, if it were to stay in 4.1 would be to have the
>>>>>> opening paragraph in 4.1 explicitly mention that it includes the
>>>>>> fmt values to be used in the context of these new proto values.
>>>>>>>
>>>>>>>
>>>>>>>> (Also, it seem like this paragraph belongs in section 4.3).
>>>>>>>
>>>>>>> I could move and combine it with the 4th paragraph of section 
>>>>>>> 4.3:
>>>>>>>
>>>>>>>    "The m- line fmt value, identifying the application-layer
>>>>>>>    protocol, MUST be registered by IANA. Section 15.3 defines 
>>>>>>> the
>>>>>>>    IANA registry for the media format namespace."
>>>>>>>
>>>>>>>
>>>>> Ok, so my suggestion is to:
>>>>>
>>>>> 1) Move the paragraph to section 4.3, as described above
>>>>> 2) Modify the first paragraph in section 4.1:
>>>>>
>>>>> "This section defines the following new SDP Media Description (m-
>>>>>    line) protocol identifiers (proto values) for describing an 
>>>>> SCTP
>>>>>    association: 'UDP/DTLS/SCTP' and 'TCP/DTLS/SCTP'.  The section
>>>>>    also describes how an m- line, associated with the proto 
>>>>> values, is
>>>>>    created, and how the the media format (Å’fmtÂ¹) values are 
>>>>> used."
>>>>
>>>> The change helps, but does it still make sense if the last 
>>>> paragraph
>>>> of 4.1 moves to 4.3?
>>>
>>> So, you suggest to not move the last paragraph to 4.3?
>>
>> No, my suggestion is to either (put the additional sentence in the 
>> first paragraph of 4.1 and keep last
>> paragraph in 4.1) _OR_ (move the paragraph to 4.3 and put the 
>> clarification in _that_ paragraph.)
>
> Ok, so:
>
> 1) Don't modify the 1st paragraph in 4.1
> 2) Move the last paragraph to 4.3
> 3) Modify the paragraph in 4.3:
>
>       "When the 'UDP/DTLS/SCTP' and 'TCP/DTLS/SCTP' proto values,
>         the m- line fmt value, identifying the application-layer
>         rotocol, MUST be registered by IANA. Section 15.3 defines the
>         IANA registry for the media format namespace."
>
> ----
>
>>>>>>>> -6.1: What is meant by saying an "endpoint MUST assume that
>>>>>>>> larger ...
>>>>>>>> will be rejected"? Can that be stated in terms of actual
>>>>>>>> procedure (e.g.
>>>>>>>> "endpoint MUST NOT send...larger"?
>>>>>>>
>>>>>>> I'd like Roman's input on this, in case he had some cases in 
>>>>>>> mind
>>>>>>> where endpoint will have to send larger messages.
>>>>>>>
>>>>>>> I guess we could say SHOULD NO send larger message, and explain
>>>>>>> that one cannot assume that larger message (if sent for whatever
>>>>>>> reason)
>>>>>>> will be accepted by the peer.
>>>>>>
>>>>>> That would be reasonable if it is the right answer. I just want 
>>>>>> to
>>>>>> avoid the vagueness of "MUST assume".
>>>>>
>>>>> I suggest the following modified text:
>>>>>
>>>>> "An SCTP endpoint SHOULD NOT send a SCTP user message with a 
>>>>> message
>>>>> size that is larger than the maximum size indicated by the peer. 
>>>>> If
>>>>> the SCTP endpoint needs (for whatever reason) to send a larger
>>>>> message, it cannot be assumed that the message will be accepted by
>>>>> the peer."
>>>>
>>>> I'm okay with that, but don't be surprised if we get questions 
>>>> about
>>>> whether it ever would ever make sense to send a message that you
>>>> can't assume the peer will accept.
>>>> (I wonder if it's more a matter
>>>> of "probably not accept", given the peer has already stated its
>>>> preference.
>>>
>>> If nobody objects, I am ok saying "MUST NOT send".
>>>
>>> Because, if there is a case where it would be needed, the peers 
>>> simply
>>> need to support and negotiate a larger value.
>>
>> Okay with me. If someone comes up with a plausible reason one might 
>> need to send larger messages, please speak up before the IETF LC 
>> completes. :-)
>
> Suggested modified paragraph:
>
>          "An SCTP endpoint MUST NOT send a SCTP user message with a 
> message
>           size that is larger than the maximum size indicated by the 
> peer, as it
>           cannot be assumed that the peer would accept such message."
>
> Regards,
>
> Christer


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Ben Campbell <ben@nostrum.com>
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From nobody Tue Jan 24 12:40:37 2017
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Subject: [MMUSIC] Review of draft-ietf-mmusic-4572-update-11
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Reviewer: Elwyn Davies
Review result: Ready

I am the assigned Gen-ART reviewer for this draft. The General Area
Review Team (Gen-ART) reviews all IETF documents being processed
by the IESG for the IETF Chair.  Please treat these comments just
like any other last call comments.

For more information, please see the FAQ at

<https://trac.ietf.org/trac/gen/wiki/GenArtfaq>.

Document: draft-ietf-mmusic-4572-update-11
Reviewer: Elwyn Davies
Review Date: 2017-01-24
IETF LC End Date: 2017-01-26
IESG Telechat date: Not scheduled for a telechat

Summary: Ready.  There are a couple of trivial nits mentioned below. 
Given that the draft has a minimal set of changes apart from the
addition of two sections, I have not read the unchanged pieces in
detail 

Major issues:
None

Minor issues:
None

Nits/editorial comments: 
s1, para 2: s/TLS protocol/The TLS protocol/ (as per RFC 4572)

s4, para 2: s/a new protocol identifier/the protocol identifier/ (it
isn't new any more)

s5.1: Suggest s/m- line/"m" line/  for consistency with s3.4

s5.1, para 1: s/e.g./e.g.,/

s5.1, para 4: s/that each used certificate matches/that each
certificate used matches/

s5.1, para 5: s/each used certificate matches/each certificate used
matches/

s8, para 5: ' This specification creates a new IANA registry named
"Hash Function Textual Names".'  
The registry is no longer new.  Perhaps s/creates a new IANA
registry/takes over the IANA registry from RFC 4572/



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To: Roman Shpount <roman@telurix.com>
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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Roman,

On 1/24/17 2:18 PM, Roman Shpount wrote:

>         Replacing of TCP connection or DTLS association does not break
>         the SCTP
>         association. SCTP is designed to run over unreliable transports, so
>         short term loss of connectivity when one TCP connection or one DTLS
>         association is being replaced by the other would be handled by
>         the SCTP
>         re-transmit timers. Also, when ICE is used, switching from one ICE
>         candidate to another, including switching between two ICE tcp
>         candidates
>         or switching from udp to tcp candidate, does not affect existing
>         SCTP
>         association. Existing SCTP association continues to run despite the
>         underlying candidate switch and as far as SCTP association is
>         concerned
>         it is running over the same unreliable transport. Finally, DTLS
>         association can be re-established to force re-keying the long
>         running
>         connection. This can be done without affecting SCTP association
>         running
>         on top of these DTLS associations.
>
>         This is why we are saying that the SCTP association, the DTLS
>         association and the TCP connection are managed independently
>         from each
>         other.  Each can be established and  closed without impacting
>         others.
>         Each protocol uses its own means to detect connection closure,
>         failure
>         or timeout.
>
>
>     I can see that this can be a good thing when it is what you want/need.
>
>     OTOH, if one of the two parties involved changes, and there is no
>     state sharing, you need to establish a new SCTP association. This
>     may turn out to be the case even when the offerer doesn't know that
>     the other party has changed.
>
>     So how are these two cases distinguished in the signaling? AFAICT
>     the only thing available is sctp-port. If the offerer doesn't
>     realize a change is coming, then it won't change its port. And in
>     that case the answerer won't know what port its predecessor used, so
>     it might choose the same value.
>
>     Of course, this is largely a 3pcc scenario, and if it is initiated
>     in the middle then the middle can probably know to change the port
>     somewhere.
>
>     But I'm not sure if that is the only case.
>
>
> We have considered that issue and assumed that selecting a random
> sctp-port is sufficient to determine if this is the same vs new
> connection.

I see *no* mention that the sctp-port value should be chosen randomly. 
And IMO it is unnatural to suggest port values be random. There could be 
reasons for them not being random - for instance interop with SCTP over 
IP with well known ports. Also, while there is currently no spec for 
multiple SCTP associations in the same bundle, it is not hard to imagine 
that that might be a useful thing to do in the future. In that case 
random port numbers could be more troublesome.

> There is some possibility of collisions when answering party
> accidentally picks the same port as the one which was previously used
> for SCTP communications, but this is similar to ICE ufrag
> collisions, dtls-id collisions or to-tag collisions.

But those all use values that are specified to be random.

> I do agree that
> sctp-port has less random bits then ufrag, dtls-id, or to-tag, so it is
> more likely to collide. Please let us know if this is a sufficient point
> of concern to warrant adding more randomness by changing sctp-port
> definition (making it longer)

That isn't possible, because the port number and its format are 
specified in the SCTP spec.

> or adding another attribute. Another
> solution would be to require SCTP association restart on DTLS
> association restart and use random bits in dtls-id minimize the chance
> of collision.

The most straightforward way would be to follow the pattern used with 
comedia (rfv4145), with a=coonnection=new/existing. But that attribute 
couldn't be used, because it can already be in use for a TCP connection 
below it. Also, we have already abandoned using something akin to a=setup.

Unfortunately the end result we have arrived add is, IMO, a one-off hack 
that doesn't follow established approaches. This has arisen because we 
are negotiated layered protocols and the established SDP mechanisms 
don't have provision for that. But I don't think there is the will to 
clean this up.

> In general, I am not strongly attached to not restarting SCTP
> association on underlying transport change. I am even less attached to
> restarting SCTP on DTLS restart.

Coupling SCTP restart with one of those is a simple answer.

> What I do feel strongly about is:
>
> 1. Keeping SCTP association when ICE candidates change
> 2. Keeping SCTP association on ICE restart
>
> Keeping SCTP association on ICE restart has exactly the same properties
> as keeping SCTP association on transport change (it is unclear if ICE
> restart was caused by the same endpoint or by endpoint change due to
> 3pcc). Since ICE restart and transport change are essentially the same
> thing, I proposed that SCTP association is preserved in both cases. If
> we tie SCTP association restart to DTLS association restart we will get
> the desired result for ICE only at the cost of loosing re-key ability
> for the existing SCTP association.
>
> So, my questions are:
> 1. Is possibility of sctp-port collision large enough concern to change
> when SCTP association is restarted?
> 2. Would it be acceptable to lose ability to re-key DTLS association
> while running the same SCTP assignations and force SCTP association
> restart on DTLS association restart?

I'd like to hear what others have to say.

	Thanks,
	Paul


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From: Roman Shpount <roman@telurix.com>
Date: Tue, 24 Jan 2017 17:02:39 -0500
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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--001a114f4ec450fe690546de469b
Content-Type: text/plain; charset=UTF-8

On Tue, Jan 24, 2017 at 4:10 PM, Paul Kyzivat <paul.kyzivat@comcast.net>
wrote:

> The most straightforward way would be to follow the pattern used with
> comedia (rfv4145), with a=coonnection=new/existing. But that attribute
> couldn't be used, because it can already be in use for a TCP connection
> below it. Also, we have already abandoned using something akin to a=setup.
>

We have tried this approach with DTLS and we came to the conclusion that
a=connection:new/existing does not work in combination with 3pcc. This is
why we switched to dtls-id. So, two options that we have are to define
sctp-id or restart based on dtls-id.

Regards,
_____________
Roman Shpount

--001a114f4ec450fe690546de469b
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">On Tue, Jan 24, 2017 at 4:10 PM, Paul Kyzivat <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:paul.kyzivat@comcast.net" target=3D"_blank">paul.kyzivat@co=
mcast.net</a>&gt;</span> wrote:<br></div></div><div class=3D"gmail_quote"><=
blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-l=
eft:1px solid rgb(204,204,204);padding-left:1ex">The most straightforward w=
ay would be to follow the pattern used with comedia (rfv4145), with a=3Dcoo=
nnection=3Dnew/existing. But that attribute couldn&#39;t be used, because i=
t can already be in use for a TCP connection below it. Also, we have alread=
y abandoned using something akin to a=3Dsetup.<br></blockquote><div><br></d=
iv><div>We have tried this approach with DTLS and we came to the conclusion=
 that a=3Dconnection:new/existing does not work in combination with 3pcc. T=
his is why we switched to dtls-id. So, two options that we have are to defi=
ne sctp-id or restart based on dtls-id.</div><div><br></div><div>Regards,</=
div><div><div class=3D"gmail_signature">_____________<br>Roman Shpount</div=
></div><div>=C2=A0</div></div></div></div>

--001a114f4ec450fe690546de469b--


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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 of the IETF.

        Title           : Session Description Protocol (SDP) Offer/Answer Procedures For Stream Control Transmission Protocol (SCTP) over Datagram Transport Layer Security (DTLS) Transport.
        Authors         : Christer Holmberg
                          Roman Shpount
                          Salvatore Loreto
                          Gonzalo Camarillo
	Filename        : draft-ietf-mmusic-sctp-sdp-22.txt
	Pages           : 25
	Date            : 2017-01-24

Abstract:
   The Stream Control Transmission Protocol (SCTP) is a transport
   protocol used to establish associations between two endpoints.
   draft-ietf-tsvwg-sctp-dtls-encaps-09 specifies how SCTP can be used
   on top of the Datagram Transport Layer Security (DTLS) protocol,
   referred to as SCTP-over-DTLS.

   This specification defines the following new Session Description
   Protocol (SDP) protocol identifiers (proto values):'UDP/DTLS/SCTP'
   and 'TCP/DTLS/SCTP'.  This specification also specifies how to use
   the new proto values with the SDP Offer/Answer mechanism for
   negotiating SCTP-over-DTLS associations.


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:
https://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-22

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


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From nobody Wed Jan 25 05:41:09 2017
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From: Andy Hutton <andyhutton.ietf@gmail.com>
Date: Wed, 25 Jan 2017 13:41:05 +0000
Message-ID: <CAB7PXwRV8GcH0L_v8J=dR7LdKBe-qZrj7ksmi1WFWdgsuMvvuQ@mail.gmail.com>
To: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
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Subject: Re: [MMUSIC] Fwd: I-D Action: draft-hutton-mmusic-opportunistic-negotiation-00.txt
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--001a113ee2f26569240546eb621d
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Hi Charles, thanks for the comments and yes I agree will include some
further text in section 5 and a reference to the IMTC spec in the next
update.

Andy

On Tue, Jan 24, 2017 at 3:47 PM, Charles Eckel (eckelcu) <eckelcu@cisco.com=
>
wrote:

> Hi Andy,
>
>
>
> I=E2=80=99m glad to see this draft. One initial comment after a quick rev=
iew - in
> section 5, the answerer must also be allowed to indicate the profile as A=
VP
> or SAVP yet still include the "a=3Drtcp-fb" SDP attribute. This is not
> allowed by RFC 4585, so it must be described here as well.
>
> Also, it would be good to reference the IMTC best practice document that
> popularized this relaxation of RFC 4585: IMTC 1015 =E2=80=93 =E2=80=9CSIP=
 Video Profile
> Best Practices=E2=80=9D.
>
> http://www.imtc.org/documents/official-documents/
>
>
>
> Cheers,
>
> Charles
>
>
>
> *From: *mmusic <mmusic-bounces@ietf.org> on behalf of Andy Hutton <
> andyhutton.ietf@gmail.com>
> *Date: *Monday, January 23, 2017 at 3:51 AM
> *To: *"mmusic-chairs@ietf.org" <mmusic-chairs@ietf.org>, "
> sipbrandy-chairs@ietf.org" <sipbrandy-chairs@ietf.org>, Ben Campbell <
> ben@nostrum.com>, "mmusic@ietf.org" <mmusic@ietf.org>
> *Subject: *[MMUSIC] Fwd: I-D Action: draft-hutton-mmusic-
> opportunistic-negotiation-00.txt
>
>
>
>
>
> I have just submitted this draft.
>
>
>
> The draft is the result of discussion in the SIPBrandy working group
> regarding how to proceed with https://tools.ietf.org/
> html/draft-ietf-sipbrandy-osrtp-01.  The SIPBrandy WG was not able to
> progress the opportunistic SRTP work because it required an update to RFC
> 4568 which is not covered by the SIPBrandy charter.
>
>
>
> This short draft is therefore submitted to MMUSIC with a view to updating
> the relevant RFC's and progressing the OSRTP work in SIPBrandy.
>
>
>
> Regards
>
> Andy
>
>
>
>
>
>
>
>
>
> ---------- Forwarded message ----------
> From: <internet-drafts@ietf.org>
> Date: Mon, Jan 23, 2017 at 11:08 AM
> Subject: I-D Action: draft-hutton-mmusic-opportunistic-negotiation-00.txt
> To: i-d-announce@ietf.org
>
>
>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>
>
>         Title           : Negotiating SRTP and RTCP Feedback using the
> RTP/AVP Profile
>         Authors         : Andrew Hutton
>                           Roland Jesske
>                           Alan Johnston
>                           Gonzalo Salgueiro
>                           Bernard Aboba
>         Filename        : draft-hutton-mmusic-
> opportunistic-negotiation-00.txt
>         Pages           : 6
>         Date            : 2017-01-23
>
> Abstract:
>    This document describes how the use of the Secure Real-time transport
>    protocol (SRTP) [RFC3711]. can be negotiated using the AVP (Audio
>    Video Profile) defined in [RFC3551].  Such a mechanism is used to
>    provide a means for encrypted media to be used in environments where
>    support for encryption is not known in advance, and not required.
>    The same mechanism is also applied to negotiation of the Extended RTP
>    Profile for Real-time Transport Control Protocol Based Feedback (RTP/
>    AVPF) [RFC4585].
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-hutton-mmusic-
> opportunistic-negotiation/
>
> There's also a htmlized version available at:
> https://tools.ietf.org/html/draft-hutton-mmusic-
> opportunistic-negotiation-00
>
>
> 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
> https://www.ietf.org/mailman/listinfo/i-d-announce
> Internet-Draft
> <https://www.ietf.org/mailman/listinfo/i-d-announceInternet-Draft>
> directories: http://www.ietf.org/shadow.html
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>
>
>
>

--001a113ee2f26569240546eb621d
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Hi Charles, thanks for the comments and yes I agree will i=
nclude some further text in section 5 and a reference to the IMTC spec in t=
he next update.<div><br></div><div>Andy<br><div class=3D"gmail_extra"><br><=
div class=3D"gmail_quote">On Tue, Jan 24, 2017 at 3:47 PM, Charles Eckel (e=
ckelcu) <span dir=3D"ltr">&lt;<a href=3D"mailto:eckelcu@cisco.com" target=
=3D"_blank">eckelcu@cisco.com</a>&gt;</span> wrote:<br><blockquote class=3D=
"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding=
-left:1ex">







<div bgcolor=3D"white" lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div class=3D"m_-7112962254209792869WordSection1">
<p class=3D"MsoNormal">Hi Andy,<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">I=E2=80=99m glad to see this draft. One initial comm=
ent after a quick review - in section 5, the answerer must also be allowed =
to indicate the profile as AVP or SAVP yet still include the &quot;a=3Drtcp=
-fb&quot; SDP attribute. This is not allowed by RFC 4585,
 so it must be described here as well.<u></u><u></u></p>
<p class=3D"MsoNormal">Also, it would be good to reference the IMTC best pr=
actice document that popularized this relaxation of RFC 4585: IMTC 1015 =E2=
=80=93 =E2=80=9CSIP Video Profile Best Practices=E2=80=9D.<u></u><u></u></p=
>
<p class=3D"MsoNormal"><a href=3D"http://www.imtc.org/documents/official-do=
cuments/" target=3D"_blank">http://www.imtc.org/documents/<wbr>official-doc=
uments/</a><u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">Cheers,<u></u><u></u></p>
<p class=3D"MsoNormal">Charles<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<blockquote style=3D"border:none;border-left:solid #b5c4df 4.5pt;padding:0i=
n 0in 0in 4.0pt;margin-left:3.75pt;margin-right:0in">
<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-family:Calibri;color:black">F=
rom: </span>
</b><span style=3D"font-family:Calibri;color:black">mmusic &lt;<a href=3D"m=
ailto:mmusic-bounces@ietf.org" target=3D"_blank">mmusic-bounces@ietf.org</a=
>&gt; on behalf of Andy Hutton &lt;<a href=3D"mailto:andyhutton.ietf@gmail.=
com" target=3D"_blank">andyhutton.ietf@gmail.com</a>&gt;<br>
<b>Date: </b>Monday, January 23, 2017 at 3:51 AM<br>
<b>To: </b>&quot;<a href=3D"mailto:mmusic-chairs@ietf.org" target=3D"_blank=
">mmusic-chairs@ietf.org</a>&quot; &lt;<a href=3D"mailto:mmusic-chairs@ietf=
.org" target=3D"_blank">mmusic-chairs@ietf.org</a>&gt;, &quot;<a href=3D"ma=
ilto:sipbrandy-chairs@ietf.org" target=3D"_blank">sipbrandy-chairs@ietf.org=
</a>&quot; &lt;<a href=3D"mailto:sipbrandy-chairs@ietf.org" target=3D"_blan=
k">sipbrandy-chairs@ietf.org</a>&gt;, Ben Campbell &lt;<a href=3D"mailto:be=
n@nostrum.com" target=3D"_blank">ben@nostrum.com</a>&gt;, &quot;<a href=3D"=
mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.org</a>&quot; &lt;<a =
href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.org</a>&gt;<b=
r>
<b>Subject: </b>[MMUSIC] Fwd: I-D Action: draft-hutton-mmusic-<wbr>opportun=
istic-negotiation-00.<wbr>txt<u></u><u></u></span></p>
</div><div><div class=3D"h5">
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">I have just submitted this draft.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">The draft is the result of discussion in the SIPBran=
dy working group regarding how to proceed with=C2=A0<a href=3D"https://tool=
s.ietf.org/html/draft-ietf-sipbrandy-osrtp-01" target=3D"_blank">https://to=
ols.ietf.org/<wbr>html/draft-ietf-sipbrandy-<wbr>osrtp-01</a>.=C2=A0 The SI=
PBrandy
 WG was not able to progress the opportunistic SRTP work because it require=
d an update to RFC 4568 which is not covered by the SIPBrandy charter.<u></=
u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">This short draft is therefore submitted to MMUSIC wi=
th a view to updating the relevant RFC&#39;s and progressing the OSRTP work=
 in SIPBrandy.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Regards<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Andy<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">---------- Forwarded message ----------<br>
From: &lt;<a href=3D"mailto:internet-drafts@ietf.org" target=3D"_blank">int=
ernet-drafts@ietf.org</a>&gt;<br>
Date: Mon, Jan 23, 2017 at 11:08 AM<br>
Subject: I-D Action: draft-hutton-mmusic-<wbr>opportunistic-negotiation-00.=
<wbr>txt<br>
To: <a href=3D"mailto:i-d-announce@ietf.org" target=3D"_blank">i-d-announce=
@ietf.org</a><br>
<br>
<br>
<br>
A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.<br>
<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Title=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 Negotiating SRTP and RTCP Feedback using the RTP/AVP Profile<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Authors=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0: Andr=
ew Hutton<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 Roland Jesske<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 Alan Johnston<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 Gonzalo Salgueiro<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 Bernard Aboba<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Filename=C2=A0 =C2=A0 =C2=A0 =C2=A0 : draft-hut=
ton-mmusic-<wbr>opportunistic-negotiation-00.<wbr>txt<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Pages=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 6<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Date=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :=
 2017-01-23<br>
<br>
Abstract:<br>
=C2=A0 =C2=A0This document describes how the use of the Secure Real-time tr=
ansport<br>
=C2=A0 =C2=A0protocol (SRTP) [RFC3711]. can be negotiated using the AVP (Au=
dio<br>
=C2=A0 =C2=A0Video Profile) defined in [RFC3551].=C2=A0 Such a mechanism is=
 used to<br>
=C2=A0 =C2=A0provide a means for encrypted media to be used in environments=
 where<br>
=C2=A0 =C2=A0support for encryption is not known in advance, and not requir=
ed.<br>
=C2=A0 =C2=A0The same mechanism is also applied to negotiation of the Exten=
ded RTP<br>
=C2=A0 =C2=A0Profile for Real-time Transport Control Protocol Based Feedbac=
k (RTP/<br>
=C2=A0 =C2=A0AVPF) [RFC4585].<br>
<br>
<br>
The IETF datatracker status page for this draft is:<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-hutton-mmusic-opportunist=
ic-negotiation/" target=3D"_blank">https://datatracker.ietf.org/<wbr>doc/dr=
aft-hutton-mmusic-<wbr>opportunistic-negotiation/</a><br>
<br>
There&#39;s also a htmlized version available at:<br>
<a href=3D"https://tools.ietf.org/html/draft-hutton-mmusic-opportunistic-ne=
gotiation-00" target=3D"_blank">https://tools.ietf.org/html/<wbr>draft-hutt=
on-mmusic-<wbr>opportunistic-negotiation-00</a><br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n<br>
until the htmlized version and diff are available at <a href=3D"http://tool=
s.ietf.org" target=3D"_blank">
tools.ietf.org</a>.<br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=3D"_blank">ftp://ftp=
.ietf.org/internet-<wbr>drafts/</a><br>
<br>
______________________________<wbr>_________________<br>
I-D-Announce mailing list<br>
<a href=3D"mailto:I-D-Announce@ietf.org" target=3D"_blank">I-D-Announce@iet=
f.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/i-d-announceInternet-Draft=
" target=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/i-d-announce=
<br>
Internet-Draft</a> directories: <a href=3D"http://www.ietf.org/shadow.html"=
 target=3D"_blank">
http://www.ietf.org/shadow.<wbr>html</a><br>
or <a href=3D"ftp://ftp.ietf.org/ietf/1shadow-sites.txt" target=3D"_blank">=
ftp://ftp.ietf.org/ietf/<wbr>1shadow-sites.txt</a><u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
</div></div></blockquote>
</div>
</div>

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

--001a113ee2f26569240546eb621d--


From nobody Wed Jan 25 05:41:36 2017
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Elwyn Davies <elwynd@dial.pipex.com>, "gen-art@ietf.org" <gen-art@ietf.org>
Thread-Topic: Review of draft-ietf-mmusic-4572-update-11
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Hi Elwyn,

Thanks for your review! See inline.

Nits/editorial comments:

>s1, para 2: s/TLS protocol/The TLS protocol/ (as per RFC 4572)

I will fix as suggested.


>s4, para 2: s/a new protocol identifier/the protocol identifier/ (it
>isn't new any more)

I am happy to remove =B3new=B2, but doesn=B9t =B3a=B2 still sound better th=
an =B3the=B2?


>s5.1: Suggest s/m- line/"m" line/  for consistency with s3.4

I will replace with single quotes (=8Cm=B9), for consistency with s3.4 and =
s4.


>s5.1, para 1: s/e.g./e.g.,/

I will fix as suggested.


>s5.1, para 4: s/that each used certificate matches/that each
>certificate used matches/

I will fix as suggested.


>s5.1, para 5: s/each used certificate matches/each certificate used
>matches/

I will fix as suggested.

>s8, para 5: ' This specification creates a new IANA registry named
>"Hash Function Textual Names".'
>The registry is no longer new.  Perhaps s/creates a new IANA
>registry/takes over the IANA registry from RFC 4572/

I suggest:

 "This specification takes over the IANA registry named "Hash Function
 Textual Names=B2, that was created in RFC 4572.  It will not be part of
 the SDP Parameters.=B2


Regards,

Christer


From nobody Wed Jan 25 07:14:07 2017
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From: Roman Shpount <roman@telurix.com>
Date: Wed, 25 Jan 2017 12:52:18 -0500
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To: Paul Kyzivat <paul.kyzivat@comcast.net>
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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One more clarification:

On Tue, Jan 24, 2017 at 4:10 PM, Paul Kyzivat <paul.kyzivat@comcast.net>
wrote:

> I see *no* mention that the sctp-port value should be chosen randomly. And
> IMO it is unnatural to suggest port values be random. There could be
> reasons for them not being random - for instance interop with SCTP over IP
> with well known ports. Also, while there is currently no spec for multiple
> SCTP associations in the same bundle, it is not hard to imagine that that
> might be a useful thing to do in the future. In that case random port
> numbers could be more troublesome.
>

SCTP associations described in this document are always simultaneous open
(active/active) associations. Because of this sctp-port values are
effectively emulating ephemeral ports allocated on the host system. Such
ports are randomly allocated. I do not think well known SCTP ports would
work with simultaneous open. This does not effect the ability to run
multiple SCTP associations over the same underlying transport, since all
this means that end point needs to allocate two different random sctp-port
values.

In any case, the option I was thinking about regarding sctp-port extension
is to modify it to add support for URL parameters and to be something like
a=sctp-port:6400;branch=abc3dl. The numeric part will be used in SCTP
packets and the branch parameter will be used to indicate new/existing
connection setup.

Regards,
_____________
Roman Shpount

--001a114d7daade6bbb0546eee46e
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">One =
more clarification:</div><div class=3D"gmail_quote"><br></div><div class=3D=
"gmail_quote">On Tue, Jan 24, 2017 at 4:10 PM, Paul Kyzivat <span dir=3D"lt=
r">&lt;<a href=3D"mailto:paul.kyzivat@comcast.net" target=3D"_blank">paul.k=
yzivat@comcast.net</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quot=
e" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204)=
;padding-left:1ex"><div><div class=3D"gmail-h5">I see *no* mention that the=
 sctp-port value should be chosen randomly. And IMO it is unnatural to sugg=
est port values be random. There could be reasons for them not being random=
 - for instance interop with SCTP over IP with well known ports. Also, whil=
e there is currently no spec for multiple SCTP associations in the same bun=
dle, it is not hard to imagine that that might be a useful thing to do in t=
he future. In that case random port numbers could be more troublesome.<br><=
/div></div></blockquote><div><br></div><div>SCTP associations described in =
this document are always simultaneous open (active/active) associations. Be=
cause of this sctp-port values are effectively emulating ephemeral ports al=
located on the host system. Such ports are randomly allocated. I do not thi=
nk well known SCTP ports would work with simultaneous open. This does not e=
ffect the ability to run multiple SCTP associations over the same underlyin=
g transport, since all this means that end point needs to allocate two diff=
erent random sctp-port values.</div><div><br></div><div>In any case, the op=
tion I was thinking about regarding sctp-port extension is to modify it to =
add support for URL parameters and to be something like<span style=3D"color=
:rgb(0,0,0);font-size:13.3333px">=C2=A0</span><span style=3D"color:rgb(0,0,=
0);font-size:13.3333px">a=3D</span>sctp-port<span style=3D"color:rgb(0,0,0)=
;font-size:13.3333px">:6400;branch=3Dabc3dl. The numeric part will be used =
in SCTP packets and the branch parameter will be used to indicate new/exist=
ing connection setup.</span></div><div><span style=3D"color:rgb(0,0,0);font=
-size:13.3333px"><br></span></div><div><span style=3D"color:rgb(0,0,0);font=
-size:13.3333px">Regards,</span></div><div><div class=3D"gmail_signature">_=
____________<br>Roman Shpount</div></div><div>=C2=A0</div></div></div></div=
>

--001a114d7daade6bbb0546eee46e--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Roman Shpount <roman@telurix.com>, Paul Kyzivat <paul.kyzivat@comcast.net>
Thread-Topic: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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References: <7594FB04B1934943A5C02806D1A2204B4BF9A860@ESESSMB209.ericsson.se> <18D6EC97-0A89-4139-834B-7E889CD8ECF1@nostrum.com> <CAD5OKxv_OmC+8ECNUBUjc9+o_kRaRTPnbnEf9uP_8f4AYAxEEQ@mail.gmail.com> <a0e80a3e-9754-1d1c-739b-a7f9e361ff0a@comcast.net> <CAD5OKxun818jB10g7cfA+4wMu+d4FBLAcFnt60esCFLfTCUiNQ@mail.gmail.com> <57d1cc4c-2cb4-4b01-db13-41b64b455887@comcast.net> <CAD5OKxvRwH=kLKZJ0hLrg1zjGYRMvMuv0k9HcOB43tbkViRuzw@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B4BFC06F6@ESESSMB209.ericsson.se>
From: Paul Kyzivat <paul.kyzivat@comcast.net>
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Date: Wed, 25 Jan 2017 14:12:38 -0500
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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On 1/25/17 1:18 PM, Christer Holmberg wrote:

> No. Let's not start making things complicated again. The document is in good shape in my opinion. Let's publish it.
>
> Keep in mind that nothing prevents an endpoint from closing/re-opening an SCTP association at any time. All we are saying is that a DTLS change does not automatically affect the SCTP association.

I agree that if an endpoint explicitly initiates the closing of an SCTP 
association, and the other end acknowledges that, they can then 
establish a new association. (IIUC it would take another O/A to trigger 
establishing the new one.) But it requires one of the endpoints to know 
that this is needed.

Lets consider a 3pcc scenario, where A is in call with B via controller 
P, where P isn't proxying the media. Now suppose P wants to transfer the 
call, replacing B with C. Typically it will do this by sending an 
offerless invite to A, and then forward the resulting offer from A to C.

In this case C can't resume the SCTP association that A has with B. But 
A doesn't know that a new association is needed.

How can P resolve this?

	Thanks,
	Paul


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From: Roman Shpount <roman@telurix.com>
Date: Wed, 25 Jan 2017 14:53:59 -0500
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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On Wed, Jan 25, 2017 at 2:12 PM, Paul Kyzivat <paul.kyzivat@comcast.net>
wrote:

> On 1/25/17 1:18 PM, Christer Holmberg wrote:
>
> No. Let's not start making things complicated again. The document is in
>> good shape in my opinion. Let's publish it.
>>
>> Keep in mind that nothing prevents an endpoint from closing/re-opening an
>> SCTP association at any time. All we are saying is that a DTLS change does
>> not automatically affect the SCTP association.
>>
>
> I agree that if an endpoint explicitly initiates the closing of an SCTP
> association, and the other end acknowledges that, they can then establish a
> new association. (IIUC it would take another O/A to trigger establishing
> the new one.) But it requires one of the endpoints to know that this is
> needed.
>
> Lets consider a 3pcc scenario, where A is in call with B via controller P,
> where P isn't proxying the media. Now suppose P wants to transfer the call,
> replacing B with C. Typically it will do this by sending an offerless
> invite to A, and then forward the resulting offer from A to C.
>
> In this case C can't resume the SCTP association that A has with B. But A
> doesn't know that a new association is needed.
>
> How can P resolve this?
>

The assumption is that C will allocate a random sctp-port value. Since this
value is different then the sctp-port value previously allocated by B, A
will know this is a new SCTP association. C had no SCTP association, so it
will start new SCTP association as well. The only issue here is that C
accidentally allocates the same value as it was previously allocated by B.
We have 1/65,535 chance of this happening.

My question is: do you think the probably of collision is high enough that
this needs to be fixed?

For comparison, SIP to-tag which faces the same potential for collision
(SIP call forks and ends up on two end points. Bad things happen if both
terminating end points pick the same to-tag) and requires at least 32 bit
of crypto randomness.

Regards,
_____________
Roman Shpount

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On W=
ed, Jan 25, 2017 at 2:12 PM, Paul Kyzivat <span dir=3D"ltr">&lt;<a href=3D"=
mailto:paul.kyzivat@comcast.net" target=3D"_blank">paul.kyzivat@comcast.net=
</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin=
:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"=
><span class=3D"m_-7698898092546718731gmail-">On 1/25/17 1:18 PM, Christer =
Holmberg wrote:<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left:1px solid rgb(204,204,204);padding-left:1ex">
No. Let&#39;s not start making things complicated again. The document is in=
 good shape in my opinion. Let&#39;s publish it.<br>
<br>
Keep in mind that nothing prevents an endpoint from closing/re-opening an S=
CTP association at any time. All we are saying is that a DTLS change does n=
ot automatically affect the SCTP association.<br>
</blockquote>
<br></span>
I agree that if an endpoint explicitly initiates the closing of an SCTP ass=
ociation, and the other end acknowledges that, they can then establish a ne=
w association. (IIUC it would take another O/A to trigger establishing the =
new one.) But it requires one of the endpoints to know that this is needed.=
<br>
<br>
Lets consider a 3pcc scenario, where A is in call with B via controller P, =
where P isn&#39;t proxying the media. Now suppose P wants to transfer the c=
all, replacing B with C. Typically it will do this by sending an offerless =
invite to A, and then forward the resulting offer from A to C.<br>
<br>
In this case C can&#39;t resume the SCTP association that A has with B. But=
 A doesn&#39;t know that a new association is needed.<br>
<br>
How can P resolve this?<br></blockquote><div><br></div><div>The assumption =
is that C will allocate a random sctp-port value. Since this value is diffe=
rent then the sctp-port value previously allocated by B, A will know this i=
s a new SCTP association. C had no SCTP association, so it will start new S=
CTP association as well. The only issue here is that C accidentally allocat=
es the same value as it was previously allocated by B. We have 1/65,535 cha=
nce of this happening.=C2=A0</div><div><br></div><div>My question is: do yo=
u think the probably of collision is high enough that this needs to be fixe=
d?</div><div><br></div><div>For comparison, SIP to-tag which faces the same=
 potential for collision (SIP call forks and ends up on two end points. Bad=
 things happen if both terminating end points pick the same to-tag) and req=
uires at least 32 bit of crypto randomness.</div><div><br></div><div>Regard=
s,</div><div>_____________<br></div><div><div class=3D"m_-76988980925467187=
31gmail_signature">Roman Shpount</div></div><div>=C2=A0</div></div></div></=
div>

--94eb2c05d26afd07070546f097f2--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <paul.kyzivat@comcast.net>, Roman Shpount <roman@telurix.com>
Thread-Topic: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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To: Roman Shpount <roman@telurix.com>
References: <7594FB04B1934943A5C02806D1A2204B4BF9A860@ESESSMB209.ericsson.se> <18D6EC97-0A89-4139-834B-7E889CD8ECF1@nostrum.com> <CAD5OKxv_OmC+8ECNUBUjc9+o_kRaRTPnbnEf9uP_8f4AYAxEEQ@mail.gmail.com> <a0e80a3e-9754-1d1c-739b-a7f9e361ff0a@comcast.net> <CAD5OKxun818jB10g7cfA+4wMu+d4FBLAcFnt60esCFLfTCUiNQ@mail.gmail.com> <57d1cc4c-2cb4-4b01-db13-41b64b455887@comcast.net> <CAD5OKxvRwH=kLKZJ0hLrg1zjGYRMvMuv0k9HcOB43tbkViRuzw@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B4BFC06F6@ESESSMB209.ericsson.se> <1b903af4-8d85-5d0a-b5b2-e3bdd0fd21fc@comcast.net> <CAD5OKxtPYKEYty5qJ_2ypERg5DnLMszgWdhJxwudWjMZxo1uSg@mail.gmail.com>
From: Paul Kyzivat <paul.kyzivat@comcast.net>
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On 1/25/17 2:53 PM, Roman Shpount wrote:
> On Wed, Jan 25, 2017 at 2:12 PM, Paul Kyzivat <paul.kyzivat@comcast.net
> <mailto:paul.kyzivat@comcast.net>> wrote:
>
>     On 1/25/17 1:18 PM, Christer Holmberg wrote:
>
>         No. Let's not start making things complicated again. The
>         document is in good shape in my opinion. Let's publish it.
>
>         Keep in mind that nothing prevents an endpoint from
>         closing/re-opening an SCTP association at any time. All we are
>         saying is that a DTLS change does not automatically affect the
>         SCTP association.
>
>
>     I agree that if an endpoint explicitly initiates the closing of an
>     SCTP association, and the other end acknowledges that, they can then
>     establish a new association. (IIUC it would take another O/A to
>     trigger establishing the new one.) But it requires one of the
>     endpoints to know that this is needed.
>
>     Lets consider a 3pcc scenario, where A is in call with B via
>     controller P, where P isn't proxying the media. Now suppose P wants
>     to transfer the call, replacing B with C. Typically it will do this
>     by sending an offerless invite to A, and then forward the resulting
>     offer from A to C.
>
>     In this case C can't resume the SCTP association that A has with B.
>     But A doesn't know that a new association is needed.
>
>     How can P resolve this?
>
>
> The assumption is that C will allocate a random sctp-port value. Since
> this value is different then the sctp-port value previously allocated by
> B, A will know this is a new SCTP association. C had no SCTP
> association, so it will start new SCTP association as well. The only
> issue here is that C accidentally allocates the same value as it was
> previously allocated by B. We have 1/65,535 chance of this happening.
>
> My question is: do you think the probably of collision is high enough
> that this needs to be fixed?

The chances are 1 in 2^16. That is high enough that it will happen 
regularly. Not everyone will encounter it, but there will be complaints 
- software will be accused of being "buggy".

It is hard to guess how objectionable this will be without digging 
deeper into what the implications will be. If the error is handled 
gracefully then it might not be a problem. I can't judge at the moment 
whether it will be straightforward to handle this gracefully.

	Thanks,
	Paul

> For comparison, SIP to-tag which faces the same potential for collision
> (SIP call forks and ends up on two end points. Bad things happen if both
> terminating end points pick the same to-tag) and requires at least 32
> bit of crypto randomness.
>
> Regards,
> _____________
> Roman Shpount
>


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From: Roman Shpount <roman@telurix.com>
Date: Wed, 25 Jan 2017 15:11:39 -0500
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: Paul Kyzivat <paul.kyzivat@comcast.net>, "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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--94eb2c12524632b9380546f0d7ba
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On Wed, Jan 25, 2017 at 2:54 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> >> No. Let's not start making things complicated again. The document is in
> good shape in my opinion. Let's publish it.
> >>
> >> Keep in mind that nothing prevents an endpoint from closing/re-opening
> an SCTP association at any time. All
> >> we are saying is that a DTLS change does not automatically affect the
> SCTP association.
> >
> > I agree that if an endpoint explicitly initiates the closing of an SCTP
> association, and the other end acknowledges that,
> > they can then establish a new association. (IIUC it would take another
> O/A to trigger establishing the new one.) But it
> > requires one of the endpoints to know that this is needed.
> >
> > Lets consider a 3pcc scenario, where A is in call with B via controller
> P, where P isn't proxying the media. Now suppose P
> > wants to transfer the call, replacing B with C. Typically it will do
> this by sending an offerless invite to A, and then forward
> > the resulting offer from A to C.
> >
> > In this case C can't resume the SCTP association that A has with B. But
> A doesn't know that a new association is needed.
>
> Assuming B terminates the SCTP association with A when the transfer
> occurs, A will know.
>
> But, in any case, when C gets the offer, it will try to establish an SCTP
> association with A.
>
>
It does not quite works like this. In the described case, P first
establishes a session between A and B which has an SCTP session. Then P
sends an INVITE without SDP to A. A has no indication if new SCTP
association is needed, so it continues to use current SCTP association, and
sends an offer in the response to INVITE with SDP which contains current
sctp-port. P then sends the offer received from A to C. C had no SCTP
association running at that point, so it allocates new random sctp-port,
sends an answer to P and attempts to initiate an SCTP association with A. P
sends that answer to A, and if sctp-port in the answer is different A, will
know this is a new SCTP association and attempts to initiate an SCTP
association with C. If there is a random collision, A will think this is
the existing SCTP association, C will think this is a new SCTP association,
things break.

Regards,
_____________
Roman Shpount

--94eb2c12524632b9380546f0d7ba
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">On Wed, Jan 25, 2017 at 2:54 PM, Christer Holmberg <span dir=3D"ltr">&=
lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chri=
ster.holmberg@ericsson.com</a>&gt;</span> wrote:<br></div></div><div class=
=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px =
0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">Hi,<br>
<span class=3D"gmail-"><br>
&gt;&gt; No. Let&#39;s not start making things complicated again. The docum=
ent is in good shape in my opinion. Let&#39;s publish it.<br>
&gt;&gt;<br>
&gt;&gt; Keep in mind that nothing prevents an endpoint from closing/re-ope=
ning an SCTP association at any time. All<br>
&gt;&gt; we are saying is that a DTLS change does not automatically affect =
the SCTP association.<br>
&gt;<br>
&gt; I agree that if an endpoint explicitly initiates the closing of an SCT=
P association, and the other end acknowledges that,<br>
&gt; they can then establish a new association. (IIUC it would take another=
 O/A to trigger establishing the new one.) But it<br>
&gt; requires one of the endpoints to know that this is needed.<br>
&gt;<br>
&gt; Lets consider a 3pcc scenario, where A is in call with B via controlle=
r P, where P isn&#39;t proxying the media. Now suppose P<br>
&gt; wants to transfer the call, replacing B with C. Typically it will do t=
his by sending an offerless invite to A, and then forward<br>
&gt; the resulting offer from A to C.<br>
&gt;<br>
&gt; In this case C can&#39;t resume the SCTP association that A has with B=
. But A doesn&#39;t know that a new association is needed.<br>
<br>
</span>Assuming B terminates the SCTP association with A when the transfer =
occurs, A will know.<br>
<br>
But, in any case, when C gets the offer, it will try to establish an SCTP a=
ssociation with A.<br>
<br></blockquote><div><br></div><div>It does not quite works like this. In =
the described case, P first establishes a session between A and B which has=
 an SCTP session. Then P sends an INVITE without SDP to A. A has no indicat=
ion if new SCTP association is needed, so it continues to use current SCTP =
association, and sends an offer in the response to INVITE with SDP which co=
ntains current sctp-port. P then sends the offer received from A to C. C ha=
d no SCTP association running at that point, so it allocates new random sct=
p-port, sends an answer to P and attempts to initiate an SCTP association w=
ith A. P sends that answer to A, and if sctp-port in the answer is differen=
t A, will know this is a new SCTP association and attempts to initiate an S=
CTP association with C. If there is a random collision, A will think this i=
s the existing SCTP association, C will think this is a new SCTP associatio=
n, things break.</div><div><br></div><div>Regards,</div><div><div class=3D"=
gmail_signature">_____________<br>Roman Shpount</div></div><div>=C2=A0</div=
></div></div></div>

--94eb2c12524632b9380546f0d7ba--


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From: Paul Kyzivat <paul.kyzivat@comcast.net>
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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On 1/25/17 2:54 PM, Christer Holmberg wrote:
> Hi,
>
>>> No. Let's not start making things complicated again. The document is in good shape in my opinion. Let's publish it.
>>>
>>> Keep in mind that nothing prevents an endpoint from closing/re-opening an SCTP association at any time. All
>>> we are saying is that a DTLS change does not automatically affect the SCTP association.
>>
>> I agree that if an endpoint explicitly initiates the closing of an SCTP association, and the other end acknowledges that,
>> they can then establish a new association. (IIUC it would take another O/A to trigger establishing the new one.) But it
>> requires one of the endpoints to know that this is needed.
>>
>> Lets consider a 3pcc scenario, where A is in call with B via controller P, where P isn't proxying the media. Now suppose P
>> wants to transfer the call, replacing B with C. Typically it will do this by sending an offerless invite to A, and then forward
>> the resulting offer from A to C.
>>
>> In this case C can't resume the SCTP association that A has with B. But A doesn't know that a new association is needed.
>
> Assuming B terminates the SCTP association with A when the transfer occurs, A will know.
>
> But, in any case, when C gets the offer, it will try to establish an SCTP association with A.

I agree that C will try that. I haven't done any digging into the 
details of the SCTP protocol, but it seems like A and C will be working 
at cross purposes. C will be trying to establish a new association, 
while A is trying to re-establish the existing SCTP association over a 
new DTLS association. Maybe it will just all work out, but I'm inclined 
to doubt it.

It would help to have a expert on the SCTP protocol chime in at this point.

	Thanks,
	Paul


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Roman Shpount <roman@telurix.com>
Thread-Topic: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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References: <7594FB04B1934943A5C02806D1A2204B4BF9A860@ESESSMB209.ericsson.se> <18D6EC97-0A89-4139-834B-7E889CD8ECF1@nostrum.com> <CAD5OKxv_OmC+8ECNUBUjc9+o_kRaRTPnbnEf9uP_8f4AYAxEEQ@mail.gmail.com> <a0e80a3e-9754-1d1c-739b-a7f9e361ff0a@comcast.net> <CAD5OKxun818jB10g7cfA+4wMu+d4FBLAcFnt60esCFLfTCUiNQ@mail.gmail.com> <57d1cc4c-2cb4-4b01-db13-41b64b455887@comcast.net> <CAD5OKxvRwH=kLKZJ0hLrg1zjGYRMvMuv0k9HcOB43tbkViRuzw@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B4BFC06F6@ESESSMB209.ericsson.se> <1b903af4-8d85-5d0a-b5b2-e3bdd0fd21fc@comcast.net> <7594FB04B1934943A5C02806D1A2204B4BFC0A13@ESESSMB209.ericsson.se> <CAD5OKxuyJ5GnvUbhSPb3h6Z7Wzif8ZRCrS5n-U+eD-yzweGwTg@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B4BFC0AC3@ESESSMB209.ericsson.se>
From: Roman Shpount <roman@telurix.com>
Date: Wed, 25 Jan 2017 15:24:12 -0500
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: Paul Kyzivat <paul.kyzivat@comcast.net>, "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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On Wed, Jan 25, 2017 at 3:21 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> >>> Lets consider a 3pcc scenario, where A is in call with B via
> controller P, where P isn't proxying the media. Now suppose P
> >>> wants to transfer the call, replacing B with C. Typically it will do
> this by sending an offerless invite to A, and then forward
> >>> the resulting offer from A to C.
> >>>
> >>> In this case C can't resume the SCTP association that A has with B.
> But A doesn't know that a new association is needed.
> >>
> >> Assuming B terminates the SCTP association with A when the transfer
> occurs, A will know.
> >>
> >> But, in any case, when C gets the offer, it will try to establish an
> SCTP association with A.
> >
> > It does not quite works like this. In the described case, P first
> establishes a session between A and B which has an SCTP session. Then P
> sends an INVITE without SDP to A. A > has no indication if new SCTP
> association is needed, so it continues to use current SCTP association, and
> sends an offer in the response to INVITE with SDP which contains
> > current sctp-port. P then sends the offer received from A to C. C had no
> SCTP association running at that point, so it allocates new random
> sctp-port, sends an answer to P > and attempts to initiate an SCTP
> association with A. P sends that answer to A, and if sctp-port in the
> answer is different A, will know this is a new SCTP association and
> > attempts to initiate an SCTP association with C. If there is a random
> collision, A will think this is the existing SCTP association, C will think
> this is a new SCTP association,
> > things break.
>
> I think we can assume that the verification tag from C will be different,
> can't we? A will know that the INIT does not belong to the existing SCTP
> association.
>

Upon checking with https://tools.ietf.org/html/rfc4960#section-5.2
establishing a new connection from C to A even before B have closed the
connection should work.

I would like somebody to double check this, but I think we can proceed with
the document as is.

Regards,
_____________
Roman Shpount

--94eb2c12524619177e0546f10400
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">On Wed, Jan 25, 2017 at 3:21 PM, Christer Holmberg <span dir=3D"ltr">&=
lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chri=
ster.holmberg@ericsson.com</a>&gt;</span> wrote:<br></div></div><div class=
=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px =
0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><span cl=
ass=3D"gmail-">&gt;&gt;&gt; Lets consider a 3pcc scenario, where A is in ca=
ll with B via controller P, where P isn&#39;t proxying the media. Now suppo=
se P<br>
&gt;&gt;&gt; wants to transfer the call, replacing B with C. Typically it w=
ill do this by sending an offerless invite to A, and then forward<br>
&gt;&gt;&gt; the resulting offer from A to C.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; In this case C can&#39;t resume the SCTP association that A ha=
s with B. But A doesn&#39;t know that a new association is needed.<br>
&gt;&gt;<br>
&gt;&gt; Assuming B terminates the SCTP association with A when the transfe=
r occurs, A will know.<br>
&gt;&gt;<br>
&gt;&gt; But, in any case, when C gets the offer, it will try to establish =
an SCTP association with A.<br>
&gt;<br>
&gt; It does not quite works like this. In the described case, P first esta=
blishes a session between A and B which has an SCTP session. Then P sends a=
n INVITE without SDP to A. A &gt; has no indication if new SCTP association=
 is needed, so it continues to use current SCTP association, and sends an o=
ffer in the response to INVITE with SDP which contains<br>
&gt; current sctp-port. P then sends the offer received from A to C. C had =
no SCTP association running at that point, so it allocates new random sctp-=
port, sends an answer to P &gt; and attempts to initiate an SCTP associatio=
n with A. P sends that answer to A, and if sctp-port in the answer is diffe=
rent A, will know this is a new SCTP association and<br>
&gt; attempts to initiate an SCTP association with C. If there is a random =
collision, A will think this is the existing SCTP association, C will think=
 this is a new SCTP association,<br>
&gt; things break.<br>
<br>
</span>I think we can assume that the verification tag from C will be diffe=
rent, can&#39;t we? A will know that the INIT does not belong to the existi=
ng SCTP association.<br></blockquote><div><br></div><div>Upon checking with=
 <a href=3D"https://tools.ietf.org/html/rfc4960#section-5.2">https://tools.=
ietf.org/html/rfc4960#section-5.2</a> establishing a new connection from C =
to A even before B have closed the connection should work.</div><div><br></=
div><div>I would like somebody to double check this, but I think we can pro=
ceed with the document as is.</div><div><br></div><div>Regards,=C2=A0</div>=
<div><div><div class=3D"gmail_signature">_____________<br>Roman Shpount</di=
v></div></div><div><br></div></div></div></div>

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <paul.kyzivat@comcast.net>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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Hi,

>>>> No. Let's not start making things complicated again. The document is i=
n good shape in my opinion. Let's publish it.
>>>>
>>>> Keep in mind that nothing prevents an endpoint from=20
>>>> closing/re-opening an SCTP association at any time. All we are saying =
is that a DTLS change does not automatically affect the SCTP association.
>>>
>>> I agree that if an endpoint explicitly initiates the closing of an=20
>>> SCTP association, and the other end acknowledges that, they can then=20
>>> establish a new association. (IIUC it would take another O/A to trigger=
 establishing the new one.) But it requires one of the endpoints to know th=
at this is needed.
>>>
>>> Lets consider a 3pcc scenario, where A is in call with B via=20
>>> controller P, where P isn't proxying the media. Now suppose P wants=20
>>> to transfer the call, replacing B with C. Typically it will do this by =
sending an offerless invite to A, and then forward the resulting offer from=
 A to C.
>>>
>>> In this case C can't resume the SCTP association that A has with B. But=
 A doesn't know that a new association is needed.
>>
>> Assuming B terminates the SCTP association with A when the transfer occu=
rs, A will know.
>>
>> But, in any case, when C gets the offer, it will try to establish an SCT=
P association with A.
>
> I agree that C will try that. I haven't done any digging into the details=
 of the SCTP protocol, but it seems like A and C will be working at cross p=
urposes. C will be trying to=20
> establish a new association, while A is trying to re-establish the existi=
ng SCTP association over a new DTLS association. Maybe it will just all wor=
k out, but I'm inclined to=20
> doubt it.
>
> It would help to have a expert on the SCTP protocol chime in at this poin=
t.

We also of course have to assume that A, when it receives the offerless inv=
ite, creates an offer that represents the ongoing associations (including t=
he currently used sctp-port value), and not a new one (if A offers a new sc=
tp-port value, there won't be an issue, because the INIT from C will be rec=
eived on that port). But, that is a general problem with 3pcc, not specific=
 to SCTP.

Regards,

Christer


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To: Roman Shpount <roman@telurix.com>, Christer Holmberg <christer.holmberg@ericsson.com>
References: <7594FB04B1934943A5C02806D1A2204B4BF9A860@ESESSMB209.ericsson.se> <18D6EC97-0A89-4139-834B-7E889CD8ECF1@nostrum.com> <CAD5OKxv_OmC+8ECNUBUjc9+o_kRaRTPnbnEf9uP_8f4AYAxEEQ@mail.gmail.com> <a0e80a3e-9754-1d1c-739b-a7f9e361ff0a@comcast.net> <CAD5OKxun818jB10g7cfA+4wMu+d4FBLAcFnt60esCFLfTCUiNQ@mail.gmail.com> <57d1cc4c-2cb4-4b01-db13-41b64b455887@comcast.net> <CAD5OKxvRwH=kLKZJ0hLrg1zjGYRMvMuv0k9HcOB43tbkViRuzw@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B4BFC06F6@ESESSMB209.ericsson.se> <1b903af4-8d85-5d0a-b5b2-e3bdd0fd21fc@comcast.net> <7594FB04B1934943A5C02806D1A2204B4BFC0A13@ESESSMB209.ericsson.se> <CAD5OKxuyJ5GnvUbhSPb3h6Z7Wzif8ZRCrS5n-U+eD-yzweGwTg@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B4BFC0AC3@ESESSMB209.ericsson.se> <CAD5OKxvK=V-v1m+8o5rV9h27a-MzQ2QN9BUwbXAhiUhAAHCw4Q@mail.gmail.com>
From: Paul Kyzivat <paul.kyzivat@comcast.net>
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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On 1/25/17 3:24 PM, Roman Shpount wrote:
> On Wed, Jan 25, 2017 at 3:21 PM, Christer Holmberg
> <christer.holmberg@ericsson.com <mailto:christer.holmberg@ericsson.com>>
> wrote:
>
>     >>> Lets consider a 3pcc scenario, where A is in call with B via controller P, where P isn't proxying the media. Now suppose P
>     >>> wants to transfer the call, replacing B with C. Typically it will do this by sending an offerless invite to A, and then forward
>     >>> the resulting offer from A to C.
>     >>>
>     >>> In this case C can't resume the SCTP association that A has with B. But A doesn't know that a new association is needed.
>     >>
>     >> Assuming B terminates the SCTP association with A when the transfer occurs, A will know.
>     >>
>     >> But, in any case, when C gets the offer, it will try to establish an SCTP association with A.
>     >
>     > It does not quite works like this. In the described case, P first establishes a session between A and B which has an SCTP session. Then P sends an INVITE without SDP to A. A > has no indication if new SCTP association is needed, so it continues to use current SCTP association, and sends an offer in the response to INVITE with SDP which contains
>     > current sctp-port. P then sends the offer received from A to C. C had no SCTP association running at that point, so it allocates new random sctp-port, sends an answer to P > and attempts to initiate an SCTP association with A. P sends that answer to A, and if sctp-port in the answer is different A, will know this is a new SCTP association and
>     > attempts to initiate an SCTP association with C. If there is a random collision, A will think this is the existing SCTP association, C will think this is a new SCTP association,
>     > things break.
>
>     I think we can assume that the verification tag from C will be
>     different, can't we? A will know that the INIT does not belong to
>     the existing SCTP association.
>
>
> Upon checking with https://tools.ietf.org/html/rfc4960#section-5.2
> establishing a new connection from C to A even before B have closed the
> connection should work.
>
> I would like somebody to double check this, but I think we can proceed
> with the document as is.

I don't understand SCTP sufficiently to sort out what will happen. But I 
do think we need to find out.

	Thanks,
	Paul


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From: Paul Kyzivat <paul.kyzivat@comcast.net>
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On 1/25/17 3:26 PM, Christer Holmberg wrote:
> Hi,
>
>>>>> No. Let's not start making things complicated again. The document is in good shape in my opinion. Let's publish it.
>>>>>
>>>>> Keep in mind that nothing prevents an endpoint from
>>>>> closing/re-opening an SCTP association at any time. All we are saying is that a DTLS change does not automatically affect the SCTP association.
>>>>
>>>> I agree that if an endpoint explicitly initiates the closing of an
>>>> SCTP association, and the other end acknowledges that, they can then
>>>> establish a new association. (IIUC it would take another O/A to trigger establishing the new one.) But it requires one of the endpoints to know that this is needed.
>>>>
>>>> Lets consider a 3pcc scenario, where A is in call with B via
>>>> controller P, where P isn't proxying the media. Now suppose P wants
>>>> to transfer the call, replacing B with C. Typically it will do this by sending an offerless invite to A, and then forward the resulting offer from A to C.
>>>>
>>>> In this case C can't resume the SCTP association that A has with B. But A doesn't know that a new association is needed.
>>>
>>> Assuming B terminates the SCTP association with A when the transfer occurs, A will know.
>>>
>>> But, in any case, when C gets the offer, it will try to establish an SCTP association with A.
>>
>> I agree that C will try that. I haven't done any digging into the details of the SCTP protocol, but it seems like A and C will be working at cross purposes. C will be trying to
>> establish a new association, while A is trying to re-establish the existing SCTP association over a new DTLS association. Maybe it will just all work out, but I'm inclined to
>> doubt it.
>>
>> It would help to have a expert on the SCTP protocol chime in at this point.
>
> We also of course have to assume that A, when it receives the offerless invite, creates an offer that represents the ongoing associations (including the currently used sctp-port value), and not a new one (if A offers a new sctp-port value, there won't be an issue, because the INIT from C will be received on that port). But, that is a general problem with 3pcc, not specific to SCTP.

While (AFAIK) there are no SCTP-specific requirements pertaining to 
this, it is hard to imagine anyone concluding they should do otherwise.

	Thanks,
	Paul


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To: "Paul Kyzivat" <paul.kyzivat@comcast.net>, "Christer Holmberg" <christer.holmberg@ericsson.com>, "Roman Shpount" <roman@telurix.com>, "Flemming Andreasen" <fandreas@cisco.com>
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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Hi,

Given the discussion, it seems to make sense to go ahead and start the 
IETF LC, and to ask for a TSV review team review as part of that. Does 
that make sense?

Thanks!

Ben.

On 25 Jan 2017, at 14:57, Paul Kyzivat wrote:

> On 1/25/17 3:26 PM, Christer Holmberg wrote:
>> Hi,
>>
>>>>>> No. Let's not start making things complicated again. The document 
>>>>>> is in good shape in my opinion. Let's publish it.
>>>>>>
>>>>>> Keep in mind that nothing prevents an endpoint from
>>>>>> closing/re-opening an SCTP association at any time. All we are 
>>>>>> saying is that a DTLS change does not automatically affect the 
>>>>>> SCTP association.
>>>>>
>>>>> I agree that if an endpoint explicitly initiates the closing of an
>>>>> SCTP association, and the other end acknowledges that, they can 
>>>>> then
>>>>> establish a new association. (IIUC it would take another O/A to 
>>>>> trigger establishing the new one.) But it requires one of the 
>>>>> endpoints to know that this is needed.
>>>>>
>>>>> Lets consider a 3pcc scenario, where A is in call with B via
>>>>> controller P, where P isn't proxying the media. Now suppose P 
>>>>> wants
>>>>> to transfer the call, replacing B with C. Typically it will do 
>>>>> this by sending an offerless invite to A, and then forward the 
>>>>> resulting offer from A to C.
>>>>>
>>>>> In this case C can't resume the SCTP association that A has with 
>>>>> B. But A doesn't know that a new association is needed.
>>>>
>>>> Assuming B terminates the SCTP association with A when the transfer 
>>>> occurs, A will know.
>>>>
>>>> But, in any case, when C gets the offer, it will try to establish 
>>>> an SCTP association with A.
>>>
>>> I agree that C will try that. I haven't done any digging into the 
>>> details of the SCTP protocol, but it seems like A and C will be 
>>> working at cross purposes. C will be trying to
>>> establish a new association, while A is trying to re-establish the 
>>> existing SCTP association over a new DTLS association. Maybe it will 
>>> just all work out, but I'm inclined to
>>> doubt it.
>>>
>>> It would help to have a expert on the SCTP protocol chime in at this 
>>> point.
>>
>> We also of course have to assume that A, when it receives the 
>> offerless invite, creates an offer that represents the ongoing 
>> associations (including the currently used sctp-port value), and not 
>> a new one (if A offers a new sctp-port value, there won't be an 
>> issue, because the INIT from C will be received on that port). But, 
>> that is a general problem with 3pcc, not specific to SCTP.
>
> While (AFAIK) there are no SCTP-specific requirements pertaining to 
> this, it is hard to imagine anyone concluding they should do 
> otherwise.
>
> 	Thanks,
> 	Paul
>
> _______________________________________________
> 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: Ben Campbell <ben@nostrum.com>, Paul Kyzivat <paul.kyzivat@comcast.net>, Roman Shpount <roman@telurix.com>, Flemming Andreasen <fandreas@cisco.com>
Thread-Topic: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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Hi,

>Given the discussion, it seems to make sense to go ahead and start the
>IETF LC, and to ask for a TSV review team review as part of that. Does
>that make sense?

Sounds good to me.

Regards,

Christer




>On 25 Jan 2017, at 14:57, Paul Kyzivat wrote:
>
>> On 1/25/17 3:26 PM, Christer Holmberg wrote:
>>> Hi,
>>>
>>>>>>> No. Let's not start making things complicated again. The document
>>>>>>> is in good shape in my opinion. Let's publish it.
>>>>>>>
>>>>>>> Keep in mind that nothing prevents an endpoint from
>>>>>>> closing/re-opening an SCTP association at any time. All we are
>>>>>>> saying is that a DTLS change does not automatically affect the
>>>>>>> SCTP association.
>>>>>>
>>>>>> I agree that if an endpoint explicitly initiates the closing of an
>>>>>> SCTP association, and the other end acknowledges that, they can
>>>>>> then
>>>>>> establish a new association. (IIUC it would take another O/A to
>>>>>> trigger establishing the new one.) But it requires one of the
>>>>>> endpoints to know that this is needed.
>>>>>>
>>>>>> Lets consider a 3pcc scenario, where A is in call with B via
>>>>>> controller P, where P isn't proxying the media. Now suppose P
>>>>>> wants
>>>>>> to transfer the call, replacing B with C. Typically it will do
>>>>>> this by sending an offerless invite to A, and then forward the
>>>>>> resulting offer from A to C.
>>>>>>
>>>>>> In this case C can't resume the SCTP association that A has with
>>>>>> B. But A doesn't know that a new association is needed.
>>>>>
>>>>> Assuming B terminates the SCTP association with A when the transfer
>>>>> occurs, A will know.
>>>>>
>>>>> But, in any case, when C gets the offer, it will try to establish
>>>>> an SCTP association with A.
>>>>
>>>> I agree that C will try that. I haven't done any digging into the
>>>> details of the SCTP protocol, but it seems like A and C will be
>>>> working at cross purposes. C will be trying to
>>>> establish a new association, while A is trying to re-establish the
>>>> existing SCTP association over a new DTLS association. Maybe it will
>>>> just all work out, but I'm inclined to
>>>> doubt it.
>>>>
>>>> It would help to have a expert on the SCTP protocol chime in at this
>>>> point.
>>>
>>> We also of course have to assume that A, when it receives the
>>> offerless invite, creates an offer that represents the ongoing
>>> associations (including the currently used sctp-port value), and not
>>> a new one (if A offers a new sctp-port value, there won't be an
>>> issue, because the INIT from C will be received on that port). But,
>>> that is a general problem with 3pcc, not specific to SCTP.
>>
>> While (AFAIK) there are no SCTP-specific requirements pertaining to
>> this, it is hard to imagine anyone concluding they should do
>> otherwise.
>>
>> 	Thanks,
>> 	Paul
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic


From nobody Thu Jan 26 00:28:46 2017
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-4572-update-12.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 of the IETF.

        Title           : Connection-Oriented Media Transport over TLS in SDP
        Authors         : Jonathan Lennox
                          Christer Holmberg
	Filename        : draft-ietf-mmusic-4572-update-12.txt
	Pages           : 17
	Date            : 2017-01-26

Abstract:
   This document specifies how to establish secure connection-oriented
   media transport sessions over the Transport Layer Security (TLS)
   protocol using the Session Description Protocol (SDP).  It defines
   the SDP protocol identifier, 'TCP/TLS'.  It also defines the syntax
   and semantics for an SDP 'fingerprint' attribute that identifies the
   certificate that will be presented for the TLS session.  This
   mechanism allows media transport over TLS connections to be
   established securely, so long as the integrity of session
   descriptions is assured.

   This document obsoletes RFC 4572, by clarifying the usage of multiple
   fingerprints.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-4572-update/

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-mmusic-4572-update-12

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-4572-update-12


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 Jan 26 00:29:46 2017
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From: Bernard Aboba <bernard.aboba@gmail.com>
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References: <7594FB04B1934943A5C02806D1A2204B4BF9A860@ESESSMB209.ericsson.se> <18D6EC97-0A89-4139-834B-7E889CD8ECF1@nostrum.com> <CAD5OKxv_OmC+8ECNUBUjc9+o_kRaRTPnbnEf9uP_8f4AYAxEEQ@mail.gmail.com> <a0e80a3e-9754-1d1c-739b-a7f9e361ff0a@comcast.net> <CAD5OKxun818jB10g7cfA+4wMu+d4FBLAcFnt60esCFLfTCUiNQ@mail.gmail.com> <57d1cc4c-2cb4-4b01-db13-41b64b455887@comcast.net> <CAD5OKxvRwH=kLKZJ0hLrg1zjGYRMvMuv0k9HcOB43tbkViRuzw@mail.gmail.com>
To: Roman Shpount <roman@telurix.com>
Archived-At: <https://mailarchive.ietf.org/arch/msg/mmusic/xujU_lzO9wRJVDHeW-LKdXcc-AM>
Cc: Paul Kyzivat <paul.kyzivat@comcast.net>, "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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In working through the ORTC SctpTransport implementation some of the odditie=
s of SO became apparent. AFAICT existing implementations assume that the loc=
al and remote ports are the same and defaulting to 5000, which is funky. If o=
ne assumes that the remote port needs to be provided before an SCTP Associat=
ion can be initiated, then does SO really make sense? The Answerer will have=
 the Local and remote ports first, so it will initiate, but can the Offerer r=
espond without receiving the Answer and the remote port?  This doesn't seem l=
ike active/active, really.

> On Jan 25, 2017, at 09:52, Roman Shpount <roman@telurix.com> wrote:
>=20
> One more clarification:
>=20
>> On Tue, Jan 24, 2017 at 4:10 PM, Paul Kyzivat <paul.kyzivat@comcast.net> w=
rote:
>> I see *no* mention that the sctp-port value should be chosen randomly. An=
d IMO it is unnatural to suggest port values be random. There could be reaso=
ns for them not being random - for instance interop with SCTP over IP with w=
ell known ports. Also, while there is currently no spec for multiple SCTP as=
sociations in the same bundle, it is not hard to imagine that that might be a=
 useful thing to do in the future. In that case random port numbers could be=
 more troublesome.
>=20
> SCTP associations described in this document are always simultaneous open (=
active/active) associations. Because of this sctp-port values are effectivel=
y emulating ephemeral ports allocated on the host system. Such ports are ran=
domly allocated. I do not think well known SCTP ports would work with simult=
aneous open. This does not effect the ability to run multiple SCTP associati=
ons over the same underlying transport, since all this means that end point n=
eeds to allocate two different random sctp-port values.
>=20
> In any case, the option I was thinking about regarding sctp-port extension=
 is to modify it to add support for URL parameters and to be something like a=
=3Dsctp-port:6400;branch=3Dabc3dl. The numeric part will be used in SCTP pac=
kets and the branch parameter will be used to indicate new/existing connecti=
on setup.
>=20
> Regards,
> _____________
> Roman Shpount
> =20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

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<html><head><meta http-equiv=3D"content-type" content=3D"text/html; charset=3D=
utf-8"></head><body dir=3D"auto"><div></div><div>In working through the ORTC=
 SctpTransport implementation some of the oddities of SO became apparent. AFA=
ICT existing implementations assume that the local and remote ports are the s=
ame and defaulting to 5000, which is funky. If one assumes that the remote p=
ort needs to be provided before an SCTP Association can be initiated, then d=
oes SO really make sense? The Answerer will have the Local and remote ports f=
irst, so it will initiate, but can the Offerer respond without receiving the=
 Answer and the remote port? &nbsp;This doesn't seem like active/active, rea=
lly.</div><div><br>On Jan 25, 2017, at 09:52, Roman Shpount &lt;<a href=3D"m=
ailto:roman@telurix.com">roman@telurix.com</a>&gt; wrote:<br><br></div><bloc=
kquote type=3D"cite"><div><div dir=3D"ltr"><div class=3D"gmail_extra"><div c=
lass=3D"gmail_quote">One more clarification:</div><div class=3D"gmail_quote"=
><br></div><div class=3D"gmail_quote">On Tue, Jan 24, 2017 at 4:10 PM, Paul K=
yzivat <span dir=3D"ltr">&lt;<a href=3D"mailto:paul.kyzivat@comcast.net" tar=
get=3D"_blank">paul.kyzivat@comcast.net</a>&gt;</span> wrote:<br><blockquote=
 class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px sol=
id rgb(204,204,204);padding-left:1ex"><div><div class=3D"gmail-h5">I see *no=
* mention that the sctp-port value should be chosen randomly. And IMO it is u=
nnatural to suggest port values be random. There could be reasons for them n=
ot being random - for instance interop with SCTP over IP with well known por=
ts. Also, while there is currently no spec for multiple SCTP associations in=
 the same bundle, it is not hard to imagine that that might be a useful thin=
g to do in the future. In that case random port numbers could be more troubl=
esome.<br></div></div></blockquote><div><br></div><div>SCTP associations des=
cribed in this document are always simultaneous open (active/active) associa=
tions. Because of this sctp-port values are effectively emulating ephemeral p=
orts allocated on the host system. Such ports are randomly allocated. I do n=
ot think well known SCTP ports would work with simultaneous open. This does n=
ot effect the ability to run multiple SCTP associations over the same underl=
ying transport, since all this means that end point needs to allocate two di=
fferent random sctp-port values.</div><div><br></div><div>In any case, the o=
ption I was thinking about regarding sctp-port extension is to modify it to a=
dd support for URL parameters and to be something like<span style=3D"color:r=
gb(0,0,0);font-size:13.3333px">&nbsp;</span><span style=3D"color:rgb(0,0,0);=
font-size:13.3333px">a=3D</span>sctp-port<span style=3D"color:rgb(0,0,0);fon=
t-size:13.3333px">:6400;branch=3Dabc3dl. The numeric part will be used in SC=
TP packets and the branch parameter will be used to indicate new/existing co=
nnection setup.</span></div><div><span style=3D"color:rgb(0,0,0);font-size:1=
3.3333px"><br></span></div><div><span style=3D"color:rgb(0,0,0);font-size:13=
.3333px">Regards,</span></div><div><div class=3D"gmail_signature">__________=
___<br>Roman Shpount</div></div><div>&nbsp;</div></div></div></div>
</div></blockquote><blockquote type=3D"cite"><div><span>____________________=
___________________________</span><br><span>mmusic mailing list</span><br><s=
pan><a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a></span><br><span><=
a href=3D"https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org=
/mailman/listinfo/mmusic</a></span><br></div></blockquote></body></html>=

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Bernard Aboba <bernard.aboba@gmail.com>, Roman Shpount <roman@telurix.com>
Thread-Topic: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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References: <7594FB04B1934943A5C02806D1A2204B4BF9A860@ESESSMB209.ericsson.se> <18D6EC97-0A89-4139-834B-7E889CD8ECF1@nostrum.com> <CAD5OKxv_OmC+8ECNUBUjc9+o_kRaRTPnbnEf9uP_8f4AYAxEEQ@mail.gmail.com> <a0e80a3e-9754-1d1c-739b-a7f9e361ff0a@comcast.net> <CAD5OKxun818jB10g7cfA+4wMu+d4FBLAcFnt60esCFLfTCUiNQ@mail.gmail.com> <57d1cc4c-2cb4-4b01-db13-41b64b455887@comcast.net> <CAD5OKxvRwH=kLKZJ0hLrg1zjGYRMvMuv0k9HcOB43tbkViRuzw@mail.gmail.com> <73CDCBCD-1005-4335-8A50-46DB004C4573@gmail.com>
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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--_000_D4AF8140166C9christerholmbergericssoncom_
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Hi,

>In working through the ORTC SctpTransport implementation some of the oddit=
ies of SO became apparent. AFAICT existing implementations assume that the =
local and remote ports are the same and defaulting to 5000, which is funky.=
 If
>one assumes that the remote port needs to be provided before an SCTP Assoc=
iation can be initiated, then does SO really make sense? The Answerer will =
have the Local and remote ports first, so it will initiate, but can the Off=
erer respond
>without receiving the Answer and the remote port?  This doesn't seem like =
active/active, really.

The offerer gets the port of the association initiated by the answerer in t=
he INIT sent from the answerer. And, the spec says that the same ports must=
 be used for both associations.


   "When an SCTP association is established, both SCTP endpoints MUST
   initiate the SCTP association (i.e. both SCTP endpoints take the
   'active' role), and MUST use the same SCTP port as client port and
   server port (in order to prevent two separate SCTP associations from
   being established)."

But, the offerer does need to get SOMETHING (the answer and/or the INIT) fr=
om the answerer before the offerer can initiate the association.

Regards,

Christer




One more clarification:

On Tue, Jan 24, 2017 at 4:10 PM, Paul Kyzivat <paul.kyzivat@comcast.net<mai=
lto:paul.kyzivat@comcast.net>> wrote:
I see *no* mention that the sctp-port value should be chosen randomly. And =
IMO it is unnatural to suggest port values be random. There could be reason=
s for them not being random - for instance interop with SCTP over IP with w=
ell known ports. Also, while there is currently no spec for multiple SCTP a=
ssociations in the same bundle, it is not hard to imagine that that might b=
e a useful thing to do in the future. In that case random port numbers coul=
d be more troublesome.

SCTP associations described in this document are always simultaneous open (=
active/active) associations. Because of this sctp-port values are effective=
ly emulating ephemeral ports allocated on the host system. Such ports are r=
andomly allocated. I do not think well known SCTP ports would work with sim=
ultaneous open. This does not effect the ability to run multiple SCTP assoc=
iations over the same underlying transport, since all this means that end p=
oint needs to allocate two different random sctp-port values.

In any case, the option I was thinking about regarding sctp-port extension =
is to modify it to add support for URL parameters and to be something like =
a=3Dsctp-port:6400;branch=3Dabc3dl. The numeric part will be used in SCTP p=
ackets and the branch parameter will be used to indicate new/existing conne=
ction setup.

Regards,
_____________
Roman Shpount

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

--_000_D4AF8140166C9christerholmbergericssoncom_
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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; color: rgb(0, 0, 0); font-size: 14px; font-fami=
ly: Calibri, sans-serif;">
<div>Hi,</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<div>
<div dir=3D"auto">
<div></div>
<div>&gt;In working through the ORTC SctpTransport implementation some of t=
he oddities of SO became apparent. AFAICT existing implementations assume t=
hat the local and remote ports are the same and defaulting to 5000, which i=
s funky. If</div>
</div>
</div>
</span>
<div>&gt;one assumes that the remote port needs to be provided before an SC=
TP Association can be initiated, then does SO really make sense? The Answer=
er will have the Local and remote ports first, so it will initiate, but can=
 the Offerer respond&nbsp;</div>
<div>&gt;without receiving the Answer and the remote port? &nbsp;This doesn=
't seem like active/active, really.</div>
<div><br>
</div>
<div>The offerer gets the port of the association initiated by the answerer=
 in the INIT sent from the answerer. And, the spec says that the same ports=
 must be used for both associations.</div>
<div><br>
</div>
<div>
<pre style=3D"font-variant-ligatures: normal; orphans: 2; widows: 2; word-w=
rap: break-word; white-space: pre-wrap;">   &quot;When an SCTP association =
is established, both SCTP endpoints MUST
   initiate the SCTP association (i.e. both SCTP endpoints take the
   'active' role), and MUST use the same SCTP port as client port and
   server port (in order to prevent two separate SCTP associations from
   being established).&quot;</pre>
</div>
<div><br>
</div>
<div>But, the offerer does need to get SOMETHING (the answer and/or the INI=
T) from the answerer before the offerer can initiate the association.</div>
<div><br>
</div>
<div>Regards,</div>
<div><br>
</div>
<div>Christer</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<div dir=3D"auto">
<div><br>
</div>
<blockquote type=3D"cite">
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">One more clarification:</div>
<div class=3D"gmail_quote"><br>
</div>
<div class=3D"gmail_quote">On Tue, Jan 24, 2017 at 4:10 PM, Paul Kyzivat <s=
pan dir=3D"ltr">
&lt;<a href=3D"mailto:paul.kyzivat@comcast.net" target=3D"_blank">paul.kyzi=
vat@comcast.net</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left:1px solid rgb(204,204,204);padding-left:1ex">
<div>
<div class=3D"gmail-h5">I see *no* mention that the sctp-port value should =
be chosen randomly. And IMO it is unnatural to suggest port values be rando=
m. There could be reasons for them not being random - for instance interop =
with SCTP over IP with well known
 ports. Also, while there is currently no spec for multiple SCTP associatio=
ns in the same bundle, it is not hard to imagine that that might be a usefu=
l thing to do in the future. In that case random port numbers could be more=
 troublesome.<br>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>SCTP associations described in this document are always simultaneous o=
pen (active/active) associations. Because of this sctp-port values are effe=
ctively emulating ephemeral ports allocated on the host system. Such ports =
are randomly allocated. I do not
 think well known SCTP ports would work with simultaneous open. This does n=
ot effect the ability to run multiple SCTP associations over the same under=
lying transport, since all this means that end point needs to allocate two =
different random sctp-port values.</div>
<div><br>
</div>
<div>In any case, the option I was thinking about regarding sctp-port exten=
sion is to modify it to add support for URL parameters and to be something =
like<span style=3D"color:rgb(0,0,0);font-size:13.3333px">&nbsp;</span><span=
 style=3D"color:rgb(0,0,0);font-size:13.3333px">a=3D</span>sctp-port<span s=
tyle=3D"color:rgb(0,0,0);font-size:13.3333px">:6400;branch=3Dabc3dl.
 The numeric part will be used in SCTP packets and the branch parameter wil=
l be used to indicate new/existing connection setup.</span></div>
<div><span style=3D"color:rgb(0,0,0);font-size:13.3333px"><br>
</span></div>
<div><span style=3D"color:rgb(0,0,0);font-size:13.3333px">Regards,</span></=
div>
<div>
<div class=3D"gmail_signature">_____________<br>
Roman Shpount</div>
</div>
<div>&nbsp;</div>
</div>
</div>
</div>
</div>
</blockquote>
<blockquote type=3D"cite">
<div><span>_______________________________________________</span><br>
<span>mmusic mailing list</span><br>
<span><a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a></span><br>
<span><a href=3D"https://www.ietf.org/mailman/listinfo/mmusic">https://www.=
ietf.org/mailman/listinfo/mmusic</a></span><br>
</div>
</blockquote>
</div>
</span>
</body>
</html>

--_000_D4AF8140166C9christerholmbergericssoncom_--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Flemming Andreasen <fandreas@cisco.com>, mmusic <mmusic@ietf.org>
Thread-Topic: 1 week WGLC on recent changes to draft-ietf-mmusic-dtls-sdp (-14 to -16)
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Subject: Re: [MMUSIC] 1 week WGLC on recent changes to draft-ietf-mmusic-dtls-sdp (-14 to -16)
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Hi,

Below are comments that my colleague Nevenka provided (she has subscribed
to the list, but hasn=B9t been accepted yet). I=B9ll address them in a
separate reply.

----------

Hello,

=20
I reviewed draft-ietf-mmusic-dtls-sdp-16.txt and have the following
comments:

=20
-         In Section 1 the following statement is not aligned with the
recent change which made "dtls-id" attribute globally unique i.e. the pair
of "dtls-id" attribute values (the attribute values of the offerer and the
answerer) uniquely identifies the DTLS association instead of the
"dtls-id" attribute pair in combination with "fingerprint" attribute
values (the attribute values of the offerer and the answerer). Thus the
following sentence requires modification:

"The =B9dtls-id=B9 attribute pair in combination with =B9fingerprint=B9 att=
ribute
values from offer and answer SDP uniquely identifies the DTLS association."


-        In section 4 there are contradicting statements related to a
default value of "dtls-id" attribute:
"Default Value: empty value "
=20
But the text says:
"No default value is defined for the SDP =B9dtls-id=B9 attribute. "
=20

Based on the above text  "empty value" should be replaced with "N/A=B2


=20
-        I believe that RFC 4572 should be included in 14.2 since it is
(and should be) referenced in sections 9.2 & 9.3 from old text in RFC 5763
and RFC 7345, but the new text requires modification and should refer to
draft ietf-mmusic-4572-update (but currently still has reference to RFC
4572).


=20
-        In addition I also found some editorial nits that I have provided
directly to the authors.
=20

Kind regards,

Nevenka


--------




On 19/01/17 00:24, "Flemming Andreasen" <fandreas@cisco.com> wrote:

>Greetings
>
>We previously completed WGLC on draft-ietf-mmusic-dtls-sdp, however
>there has been some updates to the draft subsequently. We are hereby
>issuing a 1 week WGLC on the the changes from -14 to -16 only, as
>indicated by the following URL:
>
>https://www.ietf.org/rfcdiff?url1=3Ddraft-ietf-mmusic-dtls-sdp-14&url2=3Dd=
raft
>-ietf-mmusic-dtls-sdp-16
>
>If you have any comments on the updates, please provide those by
>Wednesday, January 25. Comments should be sent to the document authors
>and the MMUSIC WG list.
>
>Thanks
>
>-- Flemming (as MMUSIC co-chair)
>
>
>
>


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Flemming Andreasen <fandreas@cisco.com>, mmusic <mmusic@ietf.org>
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Hi Chris=8A eeeh.. Nevenka,

Thanks for your comments! Please see inline.

>=20
>I reviewed draft-ietf-mmusic-dtls-sdp-16.txt and have the following
>comments:
>
>=20
>-         In Section 1 the following statement is not aligned with the
>recent change which made "dtls-id" attribute globally unique i.e. the pair
>of "dtls-id" attribute values (the attribute values of the offerer and the
>answerer) uniquely identifies the DTLS association instead of the
>"dtls-id" attribute pair in combination with "fingerprint" attribute
>values (the attribute values of the offerer and the answerer). Thus the
>following sentence requires modification:
>
>"The 'dtls-id' attribute pair in combination with =8Cfingerprint' attribut=
e
>values from offer and answer SDP uniquely identifies the DTLS
>association."

I will replace the sentence with the following:

   "The pair of SDP 'dtls-id' attribute values (the attribute values of
    the offerer and the answerer) uniquely identifies the DTLS
    association."



>-        In section 4 there are contradicting statements related to a
>default value of "dtls-id" attribute:
>"Default Value: empty value "
>=20
>But the text says:
>"No default value is defined for the SDP 'dtls-id' attribute. "
>=20
>
>Based on the above text  "empty value" should be replaced with "N/A"

I agree. I=B9ll replace =B3empty value=B2 with =B3N/A=B2.


>
>=20
>-        I believe that RFC 4572 should be included in 14.2 since it is
>(and should be) referenced in sections 9.2 & 9.3 from old text in RFC 5763
>and RFC 7345,=20

I will add a normative reference to RFC 4572 (even though the idnits tool
will complain about the reference not being used in the main text).


>but the new text requires modification and should refer to
>draft ietf-mmusic-4572-update (but currently still has reference to RFC
>4572).

Good catch. I=B9ll replace the reference in the new text.


>=20
>-        In addition I also found some editorial nits that I have provided
>directly to the authors.
>=20

I=B9ll fix those (some are the same also found by Christian Groves).


Thanks!

Regards,

Christer



>
>On 19/01/17 00:24, "Flemming Andreasen" <fandreas@cisco.com> wrote:
>
>>Greetings
>>
>>We previously completed WGLC on draft-ietf-mmusic-dtls-sdp, however
>>there has been some updates to the draft subsequently. We are hereby
>>issuing a 1 week WGLC on the the changes from -14 to -16 only, as
>>indicated by the following URL:
>>
>>https://www.ietf.org/rfcdiff?url1=3Ddraft-ietf-mmusic-dtls-sdp-14&url2=3D=
draf
>>t
>>-ietf-mmusic-dtls-sdp-16
>>
>>If you have any comments on the updates, please provide those by
>>Wednesday, January 25. Comments should be sent to the document authors
>>and the MMUSIC WG list.
>>
>>Thanks
>>
>>-- Flemming (as MMUSIC co-chair)
>>
>>
>>
>>
>


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Flemming Andreasen <fandreas@cisco.com>, mmusic <mmusic@ietf.org>
Thread-Topic: 1 week WGLC on recent changes to draft-ietf-mmusic-dtls-sdp (-14 to -16)
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Date: Thu, 26 Jan 2017 13:08:32 +0000
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Subject: Re: [MMUSIC] 1 week WGLC on recent changes to draft-ietf-mmusic-dtls-sdp (-14 to -16)
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From nobody Thu Jan 26 05:54:14 2017
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To: Harald Alvestrand <harald@alvestrand.no>, mmusic@ietf.org
References: <24b5eee4-29b1-33b0-ecea-58c766636311@alvestrand.no> <50b49dbf-7de1-a7c4-923f-f702ac14215e@alvestrand.no> <0df3cbb0-a3e7-011c-c4d5-63aad9350283@alvestrand.no>
From: Flemming Andreasen <fandreas@cisco.com>
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Date: Thu, 26 Jan 2017 08:54:10 -0500
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: [MMUSIC] Change Alert [Re:  Modifying an approved document: MSID]
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Hi Harald

Since the document is not yet in Auth48 and the change is relatively 
minor, it is possible to update it during Auth48.

However, from an MMUSIC chair point of view, I would like to get 
positive confirmation from some of the stakeholders that this change is 
indeed desired. Similarly, if anybody has any concerns with the 
suggested change, please send an e-mail to that effect.

Thanks

-- Flemming (as MMUSIC co-chair)



On 1/23/17 10:18 AM, Harald Alvestrand wrote:
> Den 19. jan. 2017 13:31, skrev Harald Alvestrand:
>> Ted pointed out to me that I sent this to the wrong group.
>>
>> Chairs and members, please advise.
> My proposed change is here:
>
> https://github.com/alvestrand/rtcweb-msid/pull/16
>
> The filed issue is here:
>
> https://github.com/alvestrand/rtcweb-msid/issues/15
>
> (CCing RTCWEB - please keep discussion, if any, on mmusic)
>
>>
>>
>> -------- Forwarded Message --------
>> Subject: 	[rtcweb] Modifying an approved document: MSID
>> Date: 	Wed, 18 Jan 2017 23:22:14 +0100
>> From: 	Harald Alvestrand <harald@alvestrand.no>
>> To: 	rtcweb@ietf.org <rtcweb@ietf.org>
>>
>>
>>
>> When reviewing the implications of the PeerConnection API change to
>> support AddTrack rather than AddStream, I found an issue.
>>
>> This issue concerns draft-ietf-rtcweb-msid, which is currently in
>> REF-WAIT state (I believe).
>>
>> The issue is that it is possible to add a track without specifying a
>> stream. Since the track's ID needs to be carried, we have to send an
>> "a=msid" line, but the track's ID is the *second* field on that line,
>> with the first being the stream's ID.
>>
>> This creates a problem.
>>
>> Suggested fix: Insert two lines in the document:
>>
>> 1) On SDP generation:
>>
>> "If there is no stream associated with the track, use the reserved ID
>> value '-'"
>>
>> 2) On SDP parsing
>>
>> "If the stream ID is the reserved value '-', the track is not associated
>> with a stream, and no stream is signalled or created."
>>
>> If this is OK with the community, I'll issue an updated draft with this
>> change.
>>
>>
>> -- 
>> Surveillance is pervasive. Go Dark.
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>>
>> _______________________________________________
>> 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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References: <7594FB04B1934943A5C02806D1A2204B4BF9A860@ESESSMB209.ericsson.se> <18D6EC97-0A89-4139-834B-7E889CD8ECF1@nostrum.com> <CAD5OKxv_OmC+8ECNUBUjc9+o_kRaRTPnbnEf9uP_8f4AYAxEEQ@mail.gmail.com> <a0e80a3e-9754-1d1c-739b-a7f9e361ff0a@comcast.net> <CAD5OKxun818jB10g7cfA+4wMu+d4FBLAcFnt60esCFLfTCUiNQ@mail.gmail.com> <57d1cc4c-2cb4-4b01-db13-41b64b455887@comcast.net> <CAD5OKxvRwH=kLKZJ0hLrg1zjGYRMvMuv0k9HcOB43tbkViRuzw@mail.gmail.com> <73CDCBCD-1005-4335-8A50-46DB004C4573@gmail.com> <D4AF8140.166C9%christer.holmberg@ericsson.com>
From: Paul Kyzivat <paul.kyzivat@comcast.net>
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Date: Thu, 26 Jan 2017 11:55:39 -0500
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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On 1/26/17 3:39 AM, Christer Holmberg wrote:
> Hi,
>
>>In working through the ORTC SctpTransport implementation some of the oddities of SO became apparent. AFAICT existing implementations assume that the local and remote ports are the same and defaulting to 5000, which is funky. If
>>one assumes that the remote port needs to be provided before an SCTP
> Association can be initiated, then does SO really make sense? The
> Answerer will have the Local and remote ports first, so it will
> initiate, but can the Offerer respond
>>without receiving the Answer and the remote port?  This doesn't seem
> like active/active, really.
>
> The offerer gets the port of the association initiated by the answerer
> in the INIT sent from the answerer. And, the spec says that the same
> ports must be used for both associations.
>
>    "When an SCTP association is established, both SCTP endpoints MUST
>    initiate the SCTP association (i.e. both SCTP endpoints take the
>    'active' role), and MUST use the same SCTP port as client port and
>    server port (in order to prevent two separate SCTP associations from
>    being established)."

I hadn't noticed that. Given that, section 10.3 ought to have normative 
language to that effect. Right now, all it says is:

    o  MUST associate an SDP 'sctp-port' attribute with the m- line.  If
       the offer contained a new (different than the one currently used)
       SCTP port value the answerer MUST also associate a new SCTP port
       value.  If the offer contained a zero SCTP port value, or if the
       answerer does not accept the SCTP association, the answerer MUST
       also associate a zero SCTP port value; and

Also, while this constraint is workable with UDP/DTLS/SCTP and 
TCP/DTLS/SCTP, it wouldn't be workable for SCTP directly over IP using 
well known ports for *either* end, because it would limit a given pair 
of IP addresses to one SCTP association per well known port. (SCTP ports 
9, 20,21,22 *have* been registered, so somebody thought they might be used.)

That isn't technically a problem now since the scope of the draft no 
longer includes SCTP over IP.

	Thanks,
	Paul

> But, the offerer does need to get SOMETHING (the answer and/or the INIT)
> from the answerer before the offerer can initiate the association.
>
> Regards,
>
> Christer
>
>
>
>
>> One more clarification:
>>
>> On Tue, Jan 24, 2017 at 4:10 PM, Paul Kyzivat
>> <paul.kyzivat@comcast.net <mailto:paul.kyzivat@comcast.net>> wrote:
>>
>>     I see *no* mention that the sctp-port value should be chosen
>>     randomly. And IMO it is unnatural to suggest port values be
>>     random. There could be reasons for them not being random - for
>>     instance interop with SCTP over IP with well known ports. Also,
>>     while there is currently no spec for multiple SCTP associations in
>>     the same bundle, it is not hard to imagine that that might be a
>>     useful thing to do in the future. In that case random port numbers
>>     could be more troublesome.
>>
>>
>> SCTP associations described in this document are always simultaneous
>> open (active/active) associations. Because of this sctp-port values
>> are effectively emulating ephemeral ports allocated on the host
>> system. Such ports are randomly allocated. I do not think well known
>> SCTP ports would work with simultaneous open. This does not effect the
>> ability to run multiple SCTP associations over the same underlying
>> transport, since all this means that end point needs to allocate two
>> different random sctp-port values.
>>
>> In any case, the option I was thinking about regarding sctp-port
>> extension is to modify it to add support for URL parameters and to be
>> something like a=sctp-port:6400;branch=abc3dl. The numeric part will
>> be used in SCTP packets and the branch parameter will be used to
>> indicate new/existing connection setup.
>>
>> Regards,
>> _____________
>> Roman Shpount
>>
>> _______________________________________________
>> 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
>


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <paul.kyzivat@comcast.net>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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Date: Thu, 26 Jan 2017 17:24:24 +0000
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References: <7594FB04B1934943A5C02806D1A2204B4BF9A860@ESESSMB209.ericsson.se> <18D6EC97-0A89-4139-834B-7E889CD8ECF1@nostrum.com> <CAD5OKxv_OmC+8ECNUBUjc9+o_kRaRTPnbnEf9uP_8f4AYAxEEQ@mail.gmail.com> <a0e80a3e-9754-1d1c-739b-a7f9e361ff0a@comcast.net> <CAD5OKxun818jB10g7cfA+4wMu+d4FBLAcFnt60esCFLfTCUiNQ@mail.gmail.com> <57d1cc4c-2cb4-4b01-db13-41b64b455887@comcast.net> <CAD5OKxvRwH=kLKZJ0hLrg1zjGYRMvMuv0k9HcOB43tbkViRuzw@mail.gmail.com> <73CDCBCD-1005-4335-8A50-46DB004C4573@gmail.com> <D4AF8140.166C9%christer.holmberg@ericsson.com> <d50cd67f-4593-6d03-a7d5-4576a8553f4c@comcast.net>
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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Hi,

>>>In working through the ORTC SctpTransport implementation some of the=20
>>>oddities of SO became apparent. AFAICT existing implementations assume=20
>>>that the local and remote ports are the same and defaulting to 5000,=20
>>>which is funky. If one assumes that the remote port needs to be=20
>>>provided before an SCTP
>> Association can be initiated, then does SO really make sense? The=20
>> Answerer will have the Local and remote ports first, so it will=20
>> initiate, but can the Offerer respond
>>without receiving the Answer and the remote port?  This doesn't seem
>> like active/active, really.
>>
>> The offerer gets the port of the association initiated by the answerer=20
>> in the INIT sent from the answerer. And, the spec says that the same=20
>> ports must be used for both associations.
>>
>>    "When an SCTP association is established, both SCTP endpoints MUST
>>    initiate the SCTP association (i.e. both SCTP endpoints take the
>>    'active' role), and MUST use the same SCTP port as client port and
>>    server port (in order to prevent two separate SCTP associations from
>>    being established)."
>
> I hadn't noticed that. Given that, section 10.3 ought to have normative l=
anguage to that effect. Right now, all it says is:
>
>    o  MUST associate an SDP 'sctp-port' attribute with the m- line.  If
>       the offer contained a new (different than the one currently used)
>       SCTP port value the answerer MUST also associate a new SCTP port
>       value.  If the offer contained a zero SCTP port value, or if the
>       answerer does not accept the SCTP association, the answerer MUST
>       also associate a zero SCTP port value; and

Section 10.3 is about the offer/answer procedures, including assigning the =
sctp-port value, while the other text is related to the SCTP establishment.

>Also, while this constraint is workable with UDP/DTLS/SCTP and TCP/DTLS/SC=
TP, it wouldn't be workable for SCTP directly over IP using well known port=
s for *either* end, >because it would limit a given pair of IP addresses to=
 one SCTP association per well known port. (SCTP ports 9, 20,21,22 *have* b=
een registered, so somebody thought they >might be used.)
>
>That isn't technically a problem now since the scope of the draft no longe=
r includes SCTP over IP.

Correct.=20

Regards,

Christer






> But, the offerer does need to get SOMETHING (the answer and/or the=20
> INIT) from the answerer before the offerer can initiate the association.
>
> Regards,
>
> Christer
>
>
>
>
>> One more clarification:
>>
>> On Tue, Jan 24, 2017 at 4:10 PM, Paul Kyzivat=20
>> <paul.kyzivat@comcast.net <mailto:paul.kyzivat@comcast.net>> wrote:
>>
>>     I see *no* mention that the sctp-port value should be chosen
>>     randomly. And IMO it is unnatural to suggest port values be
>>     random. There could be reasons for them not being random - for
>>     instance interop with SCTP over IP with well known ports. Also,
>>     while there is currently no spec for multiple SCTP associations in
>>     the same bundle, it is not hard to imagine that that might be a
>>     useful thing to do in the future. In that case random port numbers
>>     could be more troublesome.
>>
>>
>> SCTP associations described in this document are always simultaneous=20
>> open (active/active) associations. Because of this sctp-port values=20
>> are effectively emulating ephemeral ports allocated on the host=20
>> system. Such ports are randomly allocated. I do not think well known=20
>> SCTP ports would work with simultaneous open. This does not effect=20
>> the ability to run multiple SCTP associations over the same=20
>> underlying transport, since all this means that end point needs to=20
>> allocate two different random sctp-port values.
>>
>> In any case, the option I was thinking about regarding sctp-port=20
>> extension is to modify it to add support for URL parameters and to be=20
>> something like a=3Dsctp-port:6400;branch=3Dabc3dl. The numeric part will=
=20
>> be used in SCTP packets and the branch parameter will be used to=20
>> indicate new/existing connection setup.
>>
>> Regards,
>> _____________
>> Roman Shpount
>>
>> _______________________________________________
>> 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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From: Paul Kyzivat <paul.kyzivat@comcast.net>
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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On 1/26/17 12:24 PM, Christer Holmberg wrote:
> Hi,
>
>>>> In working through the ORTC SctpTransport implementation some of the
>>>> oddities of SO became apparent. AFAICT existing implementations assume
>>>> that the local and remote ports are the same and defaulting to 5000,
>>>> which is funky. If one assumes that the remote port needs to be
>>>> provided before an SCTP
>>> Association can be initiated, then does SO really make sense? The
>>> Answerer will have the Local and remote ports first, so it will
>>> initiate, but can the Offerer respond
>>> without receiving the Answer and the remote port?  This doesn't seem
>>> like active/active, really.
>>>
>>> The offerer gets the port of the association initiated by the answerer
>>> in the INIT sent from the answerer. And, the spec says that the same
>>> ports must be used for both associations.
>>>
>>>    "When an SCTP association is established, both SCTP endpoints MUST
>>>    initiate the SCTP association (i.e. both SCTP endpoints take the
>>>    'active' role), and MUST use the same SCTP port as client port and
>>>    server port (in order to prevent two separate SCTP associations from
>>>    being established)."
>>
>> I hadn't noticed that. Given that, section 10.3 ought to have normative language to that effect. Right now, all it says is:
>>
>>    o  MUST associate an SDP 'sctp-port' attribute with the m- line.  If
>>       the offer contained a new (different than the one currently used)
>>       SCTP port value the answerer MUST also associate a new SCTP port
>>       value.  If the offer contained a zero SCTP port value, or if the
>>       answerer does not accept the SCTP association, the answerer MUST
>>       also associate a zero SCTP port value; and
>
> Section 10.3 is about the offer/answer procedures, including assigning the sctp-port value, while the other text is related to the SCTP establishment.

Yes, but they need to be consistent. As currently written, it would be 
valid for the answerer to include a different sctp-port value in the 
answer, as long as it then ignores that and uses the value from the 
offer in the protocol.

	Thanks,
	Paul


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From: Roman Shpount <roman@telurix.com>
Date: Thu, 26 Jan 2017 13:03:29 -0500
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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--001a114f4ec4b29e8d0547032a41
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On Thu, Jan 26, 2017 at 3:39 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> >In working through the ORTC SctpTransport implementation some of the
> oddities of SO became apparent. AFAICT existing implementations assume that
> the local and remote ports are the same and defaulting to 5000, which is
> funky. If
> >one assumes that the remote port needs to be provided before an SCTP
> Association can be initiated, then does SO really make sense? The Answerer
> will have the Local and remote ports first, so it will initiate, but can
> the Offerer respond
> >without receiving the Answer and the remote port?  This doesn't seem like
> active/active, really.
>
> The offerer gets the port of the association initiated by the answerer in
> the INIT sent from the answerer. And, the spec says that the same ports
> must be used for both associations.
>
>    "When an SCTP association is established, both SCTP endpoints MUST
>    initiate the SCTP association (i.e. both SCTP endpoints take the
>    'active' role), and MUST use the same SCTP port as client port and
>    server port (in order to prevent two separate SCTP associations from
>    being established)."
>
>
I just wanted to double check, is everyone sure that SCTP client and server
ports MUST be the same? If this is the case, sctp-port cannot be used to
indicate that new SCTP association should be established. We need some
additional indicate to signal this.

Regards,
_____________
Roman Shpount

--001a114f4ec4b29e8d0547032a41
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">On Thu, Jan 26, 2017 at 3:39 AM, Christer Holmberg <span dir=3D"ltr">&=
lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chri=
ster.holmberg@ericsson.com</a>&gt;</span> wrote:<br></div></div><div class=
=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px =
0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">



<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:calibri,sans-serif">
<div>Hi,</div><span class=3D"gmail-">
<div><br>
</div>
<span id=3D"gmail-m_7130791736880652980OLK_SRC_BODY_SECTION">
<div>
<div dir=3D"auto">
<div></div>
<div>&gt;In working through the ORTC SctpTransport implementation some of t=
he oddities of SO became apparent. AFAICT existing implementations assume t=
hat the local and remote ports are the same and defaulting to 5000, which i=
s funky. If</div>
</div>
</div>
</span>
<div>&gt;one assumes that the remote port needs to be provided before an SC=
TP Association can be initiated, then does SO really make sense? The Answer=
er will have the Local and remote ports first, so it will initiate, but can=
 the Offerer respond=C2=A0</div>
<div>&gt;without receiving the Answer and the remote port?=C2=A0 This doesn=
&#39;t seem like active/active, really.</div>
<div><br>
</div>
</span><div>The offerer gets the port of the association initiated by the a=
nswerer in the INIT sent from the answerer. And, the spec says that the sam=
e ports must be used for both associations.</div>
<div><br>
</div>
<div>
<pre style=3D"font-variant-ligatures:normal;word-wrap:break-word;white-spac=
e:pre-wrap">   &quot;When an SCTP association is established, both SCTP end=
points MUST
   initiate the SCTP association (i.e. both SCTP endpoints take the
   &#39;active&#39; role), and MUST use the same SCTP port as client port a=
nd
   server port (in order to prevent two separate SCTP associations from
   being established).&quot;</pre></div></div></blockquote><div><br></div><=
div>I just wanted to double check, is everyone sure that SCTP client and se=
rver ports MUST be the same? If this is the case, sctp-port cannot be used =
to indicate that new SCTP association should be established. We need some a=
dditional indicate to signal this.</div><div><br></div><div>Regards,</div><=
div><div class=3D"gmail_signature">_____________<br>Roman Shpount</div></di=
v><div>=C2=A0</div></div></div></div>

--001a114f4ec4b29e8d0547032a41--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <paul.kyzivat@comcast.net>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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Hi,

>>>>> In working through the ORTC SctpTransport implementation some of=20
>>>>> the oddities of SO became apparent. AFAICT existing implementations=20
>>>>> assume that the local and remote ports are the same and defaulting=20
>>>>> to 5000, which is funky. If one assumes that the remote port needs=20
>>>>> to be provided before an SCTP
>>>> Association can be initiated, then does SO really make sense? The=20
>>>> Answerer will have the Local and remote ports first, so it will=20
>>>> initiate, but can the Offerer respond without receiving the Answer=20
>>>> and the remote port?  This doesn't seem like active/active, really.
>>>>
>>>> The offerer gets the port of the association initiated by the=20
>>>> answerer in the INIT sent from the answerer. And, the spec says that=20
>>>> the same ports must be used for both associations.
>>>>
>>>>    "When an SCTP association is established, both SCTP endpoints MUST
>>>>    initiate the SCTP association (i.e. both SCTP endpoints take the
>>>>    'active' role), and MUST use the same SCTP port as client port and
>>>>    server port (in order to prevent two separate SCTP associations fro=
m
>>>>    being established)."
>>>
>>> I hadn't noticed that. Given that, section 10.3 ought to have normative=
 language to that effect. Right now, all it says is:
>>>
>>>    o  MUST associate an SDP 'sctp-port' attribute with the m- line.  If
>>>       the offer contained a new (different than the one currently used)
>>>       SCTP port value the answerer MUST also associate a new SCTP port
>>>       value.  If the offer contained a zero SCTP port value, or if the
>>>       answerer does not accept the SCTP association, the answerer MUST
>>>       also associate a zero SCTP port value; and
>>
>> Section 10.3 is about the offer/answer procedures, including assigning t=
he sctp-port value, while the other text is related to the SCTP establishme=
nt.
>
> Yes, but they need to be consistent. As currently written, it would be va=
lid for the answerer to include a different sctp-port value in the answer, =
as long as it then ignores=20
> that and uses the value from the offer in the protocol.

It IS valid for the answerer the include a different sctp-port value.

What the text means is that and endpoint needs to use the same local port f=
or sending and receiving SCTP messages. But, both endpoints don't need to u=
se the same port value.

Regards,

Christer
=20


From nobody Thu Jan 26 10:19:33 2017
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Roman Shpount <roman@telurix.com>
Thread-Topic: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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--_000_7594FB04B1934943A5C02806D1A2204B4BFC4A75ESESSMB209erics_--


From nobody Thu Jan 26 10:54:03 2017
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To: Christer Holmberg <christer.holmberg@ericsson.com>, "mmusic@ietf.org" <mmusic@ietf.org>
References: <7594FB04B1934943A5C02806D1A2204B4BF9A860@ESESSMB209.ericsson.se> <18D6EC97-0A89-4139-834B-7E889CD8ECF1@nostrum.com> <CAD5OKxv_OmC+8ECNUBUjc9+o_kRaRTPnbnEf9uP_8f4AYAxEEQ@mail.gmail.com> <a0e80a3e-9754-1d1c-739b-a7f9e361ff0a@comcast.net> <CAD5OKxun818jB10g7cfA+4wMu+d4FBLAcFnt60esCFLfTCUiNQ@mail.gmail.com> <57d1cc4c-2cb4-4b01-db13-41b64b455887@comcast.net> <CAD5OKxvRwH=kLKZJ0hLrg1zjGYRMvMuv0k9HcOB43tbkViRuzw@mail.gmail.com> <73CDCBCD-1005-4335-8A50-46DB004C4573@gmail.com> <D4AF8140.166C9%christer.holmberg@ericsson.com> <d50cd67f-4593-6d03-a7d5-4576a8553f4c@comcast.net> <7594FB04B1934943A5C02806D1A2204B4BFC4960@ESESSMB209.ericsson.se> <96f2f9bf-3de9-256f-7494-9b58d4d75842@comcast.net> <7594FB04B1934943A5C02806D1A2204B4BFC4A3E@ESESSMB209.ericsson.se>
From: Paul Kyzivat <paul.kyzivat@comcast.net>
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Date: Thu, 26 Jan 2017 13:53:57 -0500
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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On 1/26/17 1:17 PM, Christer Holmberg wrote:
> Hi,
>
>>>>>> In working through the ORTC SctpTransport implementation some of
>>>>>> the oddities of SO became apparent. AFAICT existing implementations
>>>>>> assume that the local and remote ports are the same and defaulting
>>>>>> to 5000, which is funky. If one assumes that the remote port needs
>>>>>> to be provided before an SCTP
>>>>> Association can be initiated, then does SO really make sense? The
>>>>> Answerer will have the Local and remote ports first, so it will
>>>>> initiate, but can the Offerer respond without receiving the Answer
>>>>> and the remote port?  This doesn't seem like active/active, really.
>>>>>
>>>>> The offerer gets the port of the association initiated by the
>>>>> answerer in the INIT sent from the answerer. And, the spec says that
>>>>> the same ports must be used for both associations.
>>>>>
>>>>>    "When an SCTP association is established, both SCTP endpoints MUST
>>>>>    initiate the SCTP association (i.e. both SCTP endpoints take the
>>>>>    'active' role), and MUST use the same SCTP port as client port and
>>>>>    server port (in order to prevent two separate SCTP associations from
>>>>>    being established)."
>>>>
>>>> I hadn't noticed that. Given that, section 10.3 ought to have normative language to that effect. Right now, all it says is:
>>>>
>>>>    o  MUST associate an SDP 'sctp-port' attribute with the m- line.  If
>>>>       the offer contained a new (different than the one currently used)
>>>>       SCTP port value the answerer MUST also associate a new SCTP port
>>>>       value.  If the offer contained a zero SCTP port value, or if the
>>>>       answerer does not accept the SCTP association, the answerer MUST
>>>>       also associate a zero SCTP port value; and
>>>
>>> Section 10.3 is about the offer/answer procedures, including assigning the sctp-port value, while the other text is related to the SCTP establishment.
>>
>> Yes, but they need to be consistent. As currently written, it would be valid for the answerer to include a different sctp-port value in the answer, as long as it then ignores
>> that and uses the value from the offer in the protocol.
>
> It IS valid for the answerer the include a different sctp-port value.
>
> What the text means is that and endpoint needs to use the same local port for sending and receiving SCTP messages. But, both endpoints don't need to use the same port value.

You mean symmetric use of the sctp-port values?

For instance, if A offers sctp-port:5000 and B offers sctp-port:6000, 
then messages sent from A to B must have 5000 as the from-port and 6000 
as the to-port; and messages sent from B to A must have 6000 as the 
from-port and 5000 as the to-port.

Given that was your point, lets go back to what you said that triggered 
this part of the thread:

> The offerer gets the port of the association initiated by the answerer in the INIT sent from the answerer. And, the spec says that the same ports must be used for both associations.
>
>    "When an SCTP association is established, both SCTP endpoints MUST
>    initiate the SCTP association (i.e. both SCTP endpoints take the
>    'active' role), and MUST use the same SCTP port as client port and
>    server port (in order to prevent two separate SCTP associations from
>    being established)."
>
> But, the offerer does need to get SOMETHING (the answer and/or the INIT) from the answerer before the offerer can initiate the association.

You are assuming here that there is only one SCTP association at a time 
over the lower layer (in this case DTLS). If there might be more than 
one such association then the offerer can't assume that the port number 
in the incoming INIT pertains to *this* association. After all, the 
*point* of port numbers is to demux sessions over the lower layer.

Now this is a valid as long as multiple SCTP associations aren't bundled 
over the same DTLS UDP/TCP ports. Since doing such bundling is 
*currently* undefined, it can work for now. But it would make perfect 
sense to define bundlilng of multiple SCTP associations, using the 
sctp-port values for the demuxing. So in my opinion we ought not be 
spelling out procedures for SCTP association establishment that only 
work in this limited case. Hence it should be presumed that the offerer 
can't initiate the SCTP association until it receives the answer.

	Thanks,
	Paul


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <paul.kyzivat@comcast.net>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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Hi,

>>>>>>> In working through the ORTC SctpTransport implementation some of=20
>>>>>>> the oddities of SO became apparent. AFAICT existing=20
>>>>>>> implementations assume that the local and remote ports are the=20
>>>>>>> same and defaulting to 5000, which is funky. If one assumes that=20
>>>>>>> the remote port needs to be provided before an SCTP
>>>>>> Association can be initiated, then does SO really make sense? The=20
>>>>>> Answerer will have the Local and remote ports first, so it will=20
>>>>>> initiate, but can the Offerer respond without receiving the Answer=20
>>>>>> and the remote port?  This doesn't seem like active/active, really.
>>>>>>
>>>>>> The offerer gets the port of the association initiated by the=20
>>>>>> answerer in the INIT sent from the answerer. And, the spec says=20
>>>>>> that the same ports must be used for both associations.
>>>>>>
>>>>>>    "When an SCTP association is established, both SCTP endpoints MUS=
T
>>>>>>    initiate the SCTP association (i.e. both SCTP endpoints take the
>>>>>>    'active' role), and MUST use the same SCTP port as client port an=
d
>>>>>>    server port (in order to prevent two separate SCTP associations f=
rom
>>>>>>    being established)."
>>>>>
>>>>> I hadn't noticed that. Given that, section 10.3 ought to have normati=
ve language to that effect. Right now, all it says is:
>>>>>
>>>>>    o  MUST associate an SDP 'sctp-port' attribute with the m- line.  =
If
>>>>>       the offer contained a new (different than the one currently use=
d)
>>>>>       SCTP port value the answerer MUST also associate a new SCTP por=
t
>>>>>       value.  If the offer contained a zero SCTP port value, or if th=
e
>>>>>       answerer does not accept the SCTP association, the answerer MUS=
T
>>>>>       also associate a zero SCTP port value; and
>>>>
>>>> Section 10.3 is about the offer/answer procedures, including assigning=
 the sctp-port value, while the other text is related to the SCTP establish=
ment.
>>>
>>> Yes, but they need to be consistent. As currently written, it would=20
>>> be valid for the answerer to include a different sctp-port value in the=
 answer, as long as it then ignores that and uses the value from the offer =
in the protocol.
>>
>> It IS valid for the answerer the include a different sctp-port value.
>>
>> What the text means is that and endpoint needs to use the same local por=
t for sending and receiving SCTP messages. But, both endpoints don't=20
>> need to use the same port value.
>
> You mean symmetric use of the sctp-port values?

Correct.

> For instance, if A offers sctp-port:5000 and B offers sctp-port:6000, the=
n messages sent from A to B must=20
> have 5000 as the from-port and 6000 as the to-port; and messages sent fro=
m B to A must have 6000 as the from-port and 5000 as the to-port.

Yes.

> Given that was your point, lets go back to what you said that triggered t=
his part of the thread:
>
> The offerer gets the port of the association initiated by the answerer in=
 the INIT sent from the=20
> answerer. And, the spec says that the same ports must be used for both as=
sociations.
>
>    "When an SCTP association is established, both SCTP endpoints MUST
>    initiate the SCTP association (i.e. both SCTP endpoints take the
>    'active' role), and MUST use the same SCTP port as client port and
>    server port (in order to prevent two separate SCTP associations from
>    being established)."
>
> But, the offerer does need to get SOMETHING (the answer and/or the INIT) =
from the answerer before the offerer can initiate the association.
>
> You are assuming here that there is only one SCTP association at a time o=
ver the lower layer (in this case DTLS). If there=20
> might be more than one such association then the offerer can't assume tha=
t the port number in the incoming INIT pertains=20
> to *this* association. After all, the *point* of port numbers is to demux=
 sessions over the lower layer.
>
> Now this is a valid as long as multiple SCTP associations aren't bundled =
over the same DTLS UDP/TCP ports. Since doing=20
> such bundling is *currently* undefined, it can work for now. But it would=
 make perfect sense to define bundlilng of multiple=20
> SCTP associations, using the sctp-port values for the demuxing. So in my =
opinion we ought not be spelling out procedures for=20
> SCTP association establishment that only work in this limited case. Hence=
 it should be presumed that the offerer can't initiate=20
> the SCTP association until it receives the answer.

We are assuming a single SCTP association, and that is also documented in t=
he draft. Section 7 says:

   "NOTE: While [I-D.ietf-tsvwg-sctp-dtls-encaps] allows multiple SCTP
   associations on top of a single DTLS association, the procedures in
   this specification only support the negotiation of a single SCTP
   association on top of any given DTLS association."

I do not think this is the time to change that assumption.=20

Regards,

Christer


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To: Christer Holmberg <christer.holmberg@ericsson.com>, "mmusic@ietf.org" <mmusic@ietf.org>
References: <7594FB04B1934943A5C02806D1A2204B4BF9A860@ESESSMB209.ericsson.se> <18D6EC97-0A89-4139-834B-7E889CD8ECF1@nostrum.com> <CAD5OKxv_OmC+8ECNUBUjc9+o_kRaRTPnbnEf9uP_8f4AYAxEEQ@mail.gmail.com> <a0e80a3e-9754-1d1c-739b-a7f9e361ff0a@comcast.net> <CAD5OKxun818jB10g7cfA+4wMu+d4FBLAcFnt60esCFLfTCUiNQ@mail.gmail.com> <57d1cc4c-2cb4-4b01-db13-41b64b455887@comcast.net> <CAD5OKxvRwH=kLKZJ0hLrg1zjGYRMvMuv0k9HcOB43tbkViRuzw@mail.gmail.com> <73CDCBCD-1005-4335-8A50-46DB004C4573@gmail.com> <D4AF8140.166C9%christer.holmberg@ericsson.com> <d50cd67f-4593-6d03-a7d5-4576a8553f4c@comcast.net> <7594FB04B1934943A5C02806D1A2204B4BFC4960@ESESSMB209.ericsson.se> <96f2f9bf-3de9-256f-7494-9b58d4d75842@comcast.net> <7594FB04B1934943A5C02806D1A2204B4BFC4A3E@ESESSMB209.ericsson.se> <ab90cba0-3360-0b9d-eb1c-5c3e954e8e56@comcast.net> <7594FB04B1934943A5C02806D1A2204B4BFC4B28@ESESSMB209.ericsson.se>
From: Paul Kyzivat <paul.kyzivat@comcast.net>
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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On 1/26/17 2:11 PM, Christer Holmberg wrote:
> Hi,
>
>>>>>>>> In working through the ORTC SctpTransport implementation some of
>>>>>>>> the oddities of SO became apparent. AFAICT existing
>>>>>>>> implementations assume that the local and remote ports are the
>>>>>>>> same and defaulting to 5000, which is funky. If one assumes that
>>>>>>>> the remote port needs to be provided before an SCTP
>>>>>>> Association can be initiated, then does SO really make sense? The
>>>>>>> Answerer will have the Local and remote ports first, so it will
>>>>>>> initiate, but can the Offerer respond without receiving the Answer
>>>>>>> and the remote port?  This doesn't seem like active/active, really.
>>>>>>>
>>>>>>> The offerer gets the port of the association initiated by the
>>>>>>> answerer in the INIT sent from the answerer. And, the spec says
>>>>>>> that the same ports must be used for both associations.
>>>>>>>
>>>>>>>    "When an SCTP association is established, both SCTP endpoints MUST
>>>>>>>    initiate the SCTP association (i.e. both SCTP endpoints take the
>>>>>>>    'active' role), and MUST use the same SCTP port as client port and
>>>>>>>    server port (in order to prevent two separate SCTP associations from
>>>>>>>    being established)."
>>>>>>
>>>>>> I hadn't noticed that. Given that, section 10.3 ought to have normative language to that effect. Right now, all it says is:
>>>>>>
>>>>>>    o  MUST associate an SDP 'sctp-port' attribute with the m- line.  If
>>>>>>       the offer contained a new (different than the one currently used)
>>>>>>       SCTP port value the answerer MUST also associate a new SCTP port
>>>>>>       value.  If the offer contained a zero SCTP port value, or if the
>>>>>>       answerer does not accept the SCTP association, the answerer MUST
>>>>>>       also associate a zero SCTP port value; and
>>>>>
>>>>> Section 10.3 is about the offer/answer procedures, including assigning the sctp-port value, while the other text is related to the SCTP establishment.
>>>>
>>>> Yes, but they need to be consistent. As currently written, it would
>>>> be valid for the answerer to include a different sctp-port value in the answer, as long as it then ignores that and uses the value from the offer in the protocol.
>>>
>>> It IS valid for the answerer the include a different sctp-port value.
>>>
>>> What the text means is that and endpoint needs to use the same local port for sending and receiving SCTP messages. But, both endpoints don't
>>> need to use the same port value.
>>
>> You mean symmetric use of the sctp-port values?
>
> Correct.
>
>> For instance, if A offers sctp-port:5000 and B offers sctp-port:6000, then messages sent from A to B must
>> have 5000 as the from-port and 6000 as the to-port; and messages sent from B to A must have 6000 as the from-port and 5000 as the to-port.
>
> Yes.
>
>> Given that was your point, lets go back to what you said that triggered this part of the thread:
>>
>> The offerer gets the port of the association initiated by the answerer in the INIT sent from the
>> answerer. And, the spec says that the same ports must be used for both associations.
>>
>>    "When an SCTP association is established, both SCTP endpoints MUST
>>    initiate the SCTP association (i.e. both SCTP endpoints take the
>>    'active' role), and MUST use the same SCTP port as client port and
>>    server port (in order to prevent two separate SCTP associations from
>>    being established)."
>>
>> But, the offerer does need to get SOMETHING (the answer and/or the INIT) from the answerer before the offerer can initiate the association.
>>
>> You are assuming here that there is only one SCTP association at a time over the lower layer (in this case DTLS). If there
>> might be more than one such association then the offerer can't assume that the port number in the incoming INIT pertains
>> to *this* association. After all, the *point* of port numbers is to demux sessions over the lower layer.
>>
>> Now this is a valid as long as multiple SCTP associations aren't bundled over the same DTLS UDP/TCP ports. Since doing
>> such bundling is *currently* undefined, it can work for now. But it would make perfect sense to define bundlilng of multiple
>> SCTP associations, using the sctp-port values for the demuxing. So in my opinion we ought not be spelling out procedures for
>> SCTP association establishment that only work in this limited case. Hence it should be presumed that the offerer can't initiate
>> the SCTP association until it receives the answer.
>
> We are assuming a single SCTP association, and that is also documented in the draft. Section 7 says:
>
>    "NOTE: While [I-D.ietf-tsvwg-sctp-dtls-encaps] allows multiple SCTP
>    associations on top of a single DTLS association, the procedures in
>    this specification only support the negotiation of a single SCTP
>    association on top of any given DTLS association."
>
> I do not think this is the time to change that assumption.

I'm not advocating changing that assumption, I don't think we should 
make decisions that preclude relaxing it in the future.

	Thanks,
	Paul


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <paul.kyzivat@comcast.net>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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Date: Thu, 26 Jan 2017 19:38:10 +0000
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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Hi,

...

>> We are assuming a single SCTP association, and that is also documented i=
n the draft. Section 7 says:
>>
>>    "NOTE: While [I-D.ietf-tsvwg-sctp-dtls-encaps] allows multiple SCTP
>>    associations on top of a single DTLS association, the procedures in
>>    this specification only support the negotiation of a single SCTP
>>    association on top of any given DTLS association."
>>
>> I do not think this is the time to change that assumption.
>
> I'm not advocating changing that assumption, I don't think we should make=
 decisions that preclude relaxing it in the future.

If there is a need in the future, we'll deal with it. That may include some=
 kind of I-support-multiple-associations indicator, and procedures associat=
ed with that.

The only customer (that I am aware of) of this document is WebRTC, and with=
 QUIC on the horizon I am not sure there will be too many others (but, if t=
here will be, we'll deal with it).

Regards,

Christer


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From: Paul Kyzivat <paul.kyzivat@comcast.net>
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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On 1/26/17 2:38 PM, Christer Holmberg wrote:
> Hi,
>
> ...
>
>>> We are assuming a single SCTP association, and that is also documented in the draft. Section 7 says:
>>>
>>>    "NOTE: While [I-D.ietf-tsvwg-sctp-dtls-encaps] allows multiple SCTP
>>>    associations on top of a single DTLS association, the procedures in
>>>    this specification only support the negotiation of a single SCTP
>>>    association on top of any given DTLS association."
>>>
>>> I do not think this is the time to change that assumption.
>>
>> I'm not advocating changing that assumption, I don't think we should make decisions that preclude relaxing it in the future.
>
> If there is a need in the future, we'll deal with it. That may include some kind of I-support-multiple-associations indicator, and procedures associated with that.
>
> The only customer (that I am aware of) of this document is WebRTC, and with QUIC on the horizon I am not sure there will be too many others (but, if there will be, we'll deal with it).

In that case there is no need to negotiate sctp-port at all. It has 
ceased to mean anything like what it is intended to mean within the 
definition of SCTP. Pretending that we are negotiating it simply leads 
to confusion. Rather, what has been done is to agree to ignore the prt 
number as a mechanism for demultiplexing and repurpose the field in the 
protocol to carry a "establish new association" indicator.

	Thanks,
	Paul


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <paul.kyzivat@comcast.net>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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Hi,

...

>>>> We are assuming a single SCTP association, and that is also documented=
 in the draft. Section 7 says:
>>>>
>>>>    "NOTE: While [I-D.ietf-tsvwg-sctp-dtls-encaps] allows multiple SCTP
>>>>    associations on top of a single DTLS association, the procedures in
>>>>    this specification only support the negotiation of a single SCTP
>>>>    association on top of any given DTLS association."
>>>>
>>>> I do not think this is the time to change that assumption.
>>>
>>> I'm not advocating changing that assumption, I don't think we should ma=
ke decisions that preclude relaxing it in the future.
>>
>> If there is a need in the future, we'll deal with it. That may include s=
ome kind of I-support-multiple-associations indicator, and procedures assoc=
iated with that.
>>
>> The only customer (that I am aware of) of this document is WebRTC, and w=
ith QUIC on the horizon I am not sure there will be=20
>> too many others (but, if there will be, we'll deal with it).
>
> In that case there is no need to negotiate sctp-port at all. It has cease=
d to mean anything like what it is intended to mean within the
> definition of SCTP. Pretending that we are negotiating it simply leads to=
 confusion. Rather, what has been done is to agree to ignore
> the prt number as a mechanism for demultiplexing and repurpose the field =
in the protocol to carry a "establish new association" indicator.

The endpoints still need to know the port of the peer - even if they are on=
ly going to initiate a single association.

Regards,

Christer
=20


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To: Christer Holmberg <christer.holmberg@ericsson.com>, "mmusic@ietf.org" <mmusic@ietf.org>
References: <7594FB04B1934943A5C02806D1A2204B4BF9A860@ESESSMB209.ericsson.se> <CAD5OKxun818jB10g7cfA+4wMu+d4FBLAcFnt60esCFLfTCUiNQ@mail.gmail.com> <57d1cc4c-2cb4-4b01-db13-41b64b455887@comcast.net> <CAD5OKxvRwH=kLKZJ0hLrg1zjGYRMvMuv0k9HcOB43tbkViRuzw@mail.gmail.com> <73CDCBCD-1005-4335-8A50-46DB004C4573@gmail.com> <D4AF8140.166C9%christer.holmberg@ericsson.com> <d50cd67f-4593-6d03-a7d5-4576a8553f4c@comcast.net> <7594FB04B1934943A5C02806D1A2204B4BFC4960@ESESSMB209.ericsson.se> <96f2f9bf-3de9-256f-7494-9b58d4d75842@comcast.net> <7594FB04B1934943A5C02806D1A2204B4BFC4A3E@ESESSMB209.ericsson.se> <ab90cba0-3360-0b9d-eb1c-5c3e954e8e56@comcast.net> <7594FB04B1934943A5C02806D1A2204B4BFC4B28@ESESSMB209.ericsson.se> <63cbfb97-786e-2f03-ff0d-2fad2b640758@comcast.net> <7594FB04B1934943A5C02806D1A2204B4BFC4BC1@ESESSMB209.ericsson.se> <200c7ff4-dcc9-864f-80c8-6f16d8659911@comcast.net> <7594FB04B1934943A5C02806D1A2204B4BFC4E0B@ESESSMB209.ericsson.se>
From: Paul Kyzivat <paul.kyzivat@comcast.net>
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Date: Thu, 26 Jan 2017 15:35:29 -0500
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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On 1/26/17 3:12 PM, Christer Holmberg wrote:

>> In that case there is no need to negotiate sctp-port at all. It has ceased to mean anything like what it is intended to mean within the
>> definition of SCTP. Pretending that we are negotiating it simply leads to confusion. Rather, what has been done is to agree to ignore
>> the prt number as a mechanism for demultiplexing and repurpose the field in the protocol to carry a "establish new association" indicator.
>
> The endpoints still need to know the port of the peer - even if they are only going to initiate a single association.

But, IIUC, *you* were suggesting that the offerer might discover the 
port from a received INIT message rather than waiting for the answer. 
*That* only works if the answerer only has one association at a time.

As long as the offerer waits to get the sctp-port from the answer before 
sending any SCTP messages to the offerer.

	Thanks,
	Paul


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <paul.kyzivat@comcast.net>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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Hi,

>>> In that case there is no need to negotiate sctp-port at all. It has=20
>>> ceased to mean anything like what it is intended to mean within the=20
>>> definition of SCTP. Pretending that we are negotiating it simply leads =
to confusion. Rather, what has been done is to agree to ignore the prt numb=
er as a mechanism for >>> demultiplexing and repurpose the field in the pro=
tocol to carry a "establish new association" indicator.
>>
>> The endpoints still need to know the port of the peer - even if they are=
 only going to initiate a single association.
>
> But, IIUC, *you* were suggesting that the offerer might discover the port=
 from a received INIT message rather than waiting for the answer.=20

Yes. But, if the offerer for whatever reason receives the answer before the=
 INIT, it allows the offerer to send its INIT.

> *That* only works if the answerer only has one association at a time.

Correct. If we in future define support for multiple associations that won'=
t be enough. Perhaps we then need to signal multiple ports, and/or consider=
 e.g., the SCTP association ID too, etc etc etc.=20

Regards,

Christer


From nobody Thu Jan 26 13:01:43 2017
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From: Paul Kyzivat <paul.kyzivat@comcast.net>
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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On 1/26/17 3:46 PM, Christer Holmberg wrote:
> Hi,
>
>>>> In that case there is no need to negotiate sctp-port at all. It has
>>>> ceased to mean anything like what it is intended to mean within the
>>>> definition of SCTP. Pretending that we are negotiating it simply leads to confusion. Rather, what has been done is to agree to ignore the prt number as a mechanism for >>> demultiplexing and repurpose the field in the protocol to carry a "establish new association" indicator.
>>>
>>> The endpoints still need to know the port of the peer - even if they are only going to initiate a single association.
>>
>> But, IIUC, *you* were suggesting that the offerer might discover the port from a received INIT message rather than waiting for the answer.
>
> Yes. But, if the offerer for whatever reason receives the answer before the INIT, it allows the offerer to send its INIT.
>
>> *That* only works if the answerer only has one association at a time.
>
> Correct. If we in future define support for multiple associations that won't be enough. Perhaps we then need to signal multiple ports, and/or consider e.g., the SCTP association ID too, etc etc etc.

In any case, this is mostly moot, because you need the IP address and 
UDP port before you can start to establish the association, and those 
also come in the answer. (Or from ICE after having received some 
candidates in the answer.)

I guess the time when it might apply is when adding the SCTP to a 
preexisting bundle, and requiring it to be bundle-only.

I guess if somebody wants to build an optimization based on all the 
necessary special cases applying then that is their business. But I 
wouldn't like to see that written down in the specs.

	Thanks,
	Paul


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <paul.kyzivat@comcast.net>
Thread-Topic: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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Hi,

You can add an SCTP association to an existing DTLS association, that has e=
arlier been established e.g., for audio and video.

Regards,

Christer



Sent from my iPhone

> On 26 Jan 2017, at 23.01, Paul Kyzivat <paul.kyzivat@comcast.net> wrote:
>=20
>> On 1/26/17 3:46 PM, Christer Holmberg wrote:
>> Hi,
>>=20
>>>>> In that case there is no need to negotiate sctp-port at all. It has
>>>>> ceased to mean anything like what it is intended to mean within the
>>>>> definition of SCTP. Pretending that we are negotiating it simply lead=
s to confusion. Rather, what has been done is to agree to ignore the prt nu=
mber as a mechanism for >>> demultiplexing and repurpose the field in the p=
rotocol to carry a "establish new association" indicator.
>>>>=20
>>>> The endpoints still need to know the port of the peer - even if they a=
re only going to initiate a single association.
>>>=20
>>> But, IIUC, *you* were suggesting that the offerer might discover the po=
rt from a received INIT message rather than waiting for the answer.
>>=20
>> Yes. But, if the offerer for whatever reason receives the answer before =
the INIT, it allows the offerer to send its INIT.
>>=20
>>> *That* only works if the answerer only has one association at a time.
>>=20
>> Correct. If we in future define support for multiple associations that w=
on't be enough. Perhaps we then need to signal multiple ports, and/or consi=
der e.g., the SCTP association ID too, etc etc etc.
>=20
> In any case, this is mostly moot, because you need the IP address and UDP=
 port before you can start to establish the association, and those also com=
e in the answer. (Or from ICE after having received some candidates in the =
answer.)
>=20
> I guess the time when it might apply is when adding the SCTP to a preexis=
ting bundle, and requiring it to be bundle-only.
>=20
> I guess if somebody wants to build an optimization based on all the neces=
sary special cases applying then that is their business. But I wouldn't lik=
e to see that written down in the specs.
>=20
>    Thanks,
>    Paul
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From nobody Thu Jan 26 15:27:16 2017
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Subject: [MMUSIC] Last Call: <draft-ietf-mmusic-sctp-sdp-22.txt> (Session Description Protocol (SDP) Offer/Answer Procedures For Stream Control Transmission Protocol (SCTP) over Datagram Transport Layer Security (DTLS) Transport.) to Proposed Standard
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The IESG has received a request from the Multiparty Multimedia Session
Control WG (mmusic) to consider the following document:
- 'Session Description Protocol (SDP) Offer/Answer Procedures For Stream
   Control Transmission Protocol (SCTP) over Datagram Transport Layer
   Security (DTLS) Transport.'
  <draft-ietf-mmusic-sctp-sdp-22.txt> as Proposed Standard

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action. Please send substantive comments to the
ietf@ietf.org mailing lists by 2017-02-09. Exceptionally, comments may be
sent to iesg@ietf.org instead. In either case, please retain the
beginning of the Subject line to allow automated sorting.

Abstract


   The Stream Control Transmission Protocol (SCTP) is a transport
   protocol used to establish associations between two endpoints.
   draft-ietf-tsvwg-sctp-dtls-encaps-09 specifies how SCTP can be used
   on top of the Datagram Transport Layer Security (DTLS) protocol,
   referred to as SCTP-over-DTLS.

   This specification defines the following new Session Description
   Protocol (SDP) protocol identifiers (proto values):'UDP/DTLS/SCTP'
   and 'TCP/DTLS/SCTP'.  This specification also specifies how to use
   the new proto values with the SDP Offer/Answer mechanism for
   negotiating SCTP-over-DTLS associations.




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

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


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





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Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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Hi,

I just requested the IETF last call on version 22. And then I noticed a 
bunch of conversation popped up today that I hadn't seen yet. Based on a 
quick scan, I think it is reasonable to go ahead with the IETF LC, and 
if any changes result from the ongoing conversation, they can be 
addressed as part of the LC.

If anyone feels strongly otherwise, let me know and I can cancel the LC.

Thanks!

Ben.

On 26 Jan 2017, at 2:02, Christer Holmberg wrote:

> Hi,
>
>> Given the discussion, it seems to make sense to go ahead and start 
>> the
>> IETF LC, and to ask for a TSV review team review as part of that. 
>> Does
>> that make sense?
>
> Sounds good to me.
>
> Regards,
>
> Christer
>
>
>
>
>> On 25 Jan 2017, at 14:57, Paul Kyzivat wrote:
>>
>>> On 1/25/17 3:26 PM, Christer Holmberg wrote:
>>>> Hi,
>>>>
>>>>>>>> No. Let's not start making things complicated again. The 
>>>>>>>> document
>>>>>>>> is in good shape in my opinion. Let's publish it.
>>>>>>>>
>>>>>>>> Keep in mind that nothing prevents an endpoint from
>>>>>>>> closing/re-opening an SCTP association at any time. All we are
>>>>>>>> saying is that a DTLS change does not automatically affect the
>>>>>>>> SCTP association.
>>>>>>>
>>>>>>> I agree that if an endpoint explicitly initiates the closing of 
>>>>>>> an
>>>>>>> SCTP association, and the other end acknowledges that, they can
>>>>>>> then
>>>>>>> establish a new association. (IIUC it would take another O/A to
>>>>>>> trigger establishing the new one.) But it requires one of the
>>>>>>> endpoints to know that this is needed.
>>>>>>>
>>>>>>> Lets consider a 3pcc scenario, where A is in call with B via
>>>>>>> controller P, where P isn't proxying the media. Now suppose P
>>>>>>> wants
>>>>>>> to transfer the call, replacing B with C. Typically it will do
>>>>>>> this by sending an offerless invite to A, and then forward the
>>>>>>> resulting offer from A to C.
>>>>>>>
>>>>>>> In this case C can't resume the SCTP association that A has with
>>>>>>> B. But A doesn't know that a new association is needed.
>>>>>>
>>>>>> Assuming B terminates the SCTP association with A when the 
>>>>>> transfer
>>>>>> occurs, A will know.
>>>>>>
>>>>>> But, in any case, when C gets the offer, it will try to establish
>>>>>> an SCTP association with A.
>>>>>
>>>>> I agree that C will try that. I haven't done any digging into the
>>>>> details of the SCTP protocol, but it seems like A and C will be
>>>>> working at cross purposes. C will be trying to
>>>>> establish a new association, while A is trying to re-establish the
>>>>> existing SCTP association over a new DTLS association. Maybe it 
>>>>> will
>>>>> just all work out, but I'm inclined to
>>>>> doubt it.
>>>>>
>>>>> It would help to have a expert on the SCTP protocol chime in at 
>>>>> this
>>>>> point.
>>>>
>>>> We also of course have to assume that A, when it receives the
>>>> offerless invite, creates an offer that represents the ongoing
>>>> associations (including the currently used sctp-port value), and 
>>>> not
>>>> a new one (if A offers a new sctp-port value, there won't be an
>>>> issue, because the INIT from C will be received on that port). But,
>>>> that is a general problem with 3pcc, not specific to SCTP.
>>>
>>> While (AFAIK) there are no SCTP-specific requirements pertaining to
>>> this, it is hard to imagine anyone concluding they should do
>>> otherwise.
>>>
>>> 	Thanks,
>>> 	Paul
>>>
>>> _______________________________________________
>>> 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: Ben Campbell <ben@nostrum.com>
Thread-Topic: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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Date: Fri, 27 Jan 2017 03:46:24 +0000
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References: <7594FB04B1934943A5C02806D1A2204B4BF9A860@ESESSMB209.ericsson.se> <18D6EC97-0A89-4139-834B-7E889CD8ECF1@nostrum.com> <CAD5OKxv_OmC+8ECNUBUjc9+o_kRaRTPnbnEf9uP_8f4AYAxEEQ@mail.gmail.com> <a0e80a3e-9754-1d1c-739b-a7f9e361ff0a@comcast.net> <CAD5OKxun818jB10g7cfA+4wMu+d4FBLAcFnt60esCFLfTCUiNQ@mail.gmail.com> <57d1cc4c-2cb4-4b01-db13-41b64b455887@comcast.net> <CAD5OKxvRwH=kLKZJ0hLrg1zjGYRMvMuv0k9HcOB43tbkViRuzw@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B4BFC06F6@ESESSMB209.ericsson.se> <1b903af4-8d85-5d0a-b5b2-e3bdd0fd21fc@comcast.net> <7594FB04B1934943A5C02806D1A2204B4BFC0A13@ESESSMB209.ericsson.se> <74483dd6-bd2a-cb67-ba12-b5ebb96c0aca@comcast.net> <7594FB04B1934943A5C02806D1A2204B4BFC0AF1@ESESSMB209.ericsson.se> <856b91bd-27f0-8ef0-c247-be44a9a8114a@comcast.net> <9A5608A3-0AD5-4840-8852-6A166D72A7CE@nostrum.com> <D4AF79D5.166B5%christer.holmberg@ericsson.com>, <976A9EC9-F15C-4AA5-A4A4-41212DDF827E@nostrum.com>
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Cc: Flemming Andreasen <fandreas@cisco.com>, Paul Kyzivat <paul.kyzivat@comcast.net>, "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] AD Evaluation of draft-ietf-mmusic-sctp-sdp-21 - TECHNICAL comments
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Hi,

I think we can move ahead. Then, if the transport people find issues, we ma=
y have to come back.

Regards,

Christer=20

Sent from my iPhone

> On 27 Jan 2017, at 1.27, Ben Campbell <ben@nostrum.com> wrote:
>=20
> Hi,
>=20
> I just requested the IETF last call on version 22. And then I noticed a b=
unch of conversation popped up today that I hadn't seen yet. Based on a qui=
ck scan, I think it is reasonable to go ahead with the IETF LC, and if any =
changes result from the ongoing conversation, they can be addressed as part=
 of the LC.
>=20
> If anyone feels strongly otherwise, let me know and I can cancel the LC.
>=20
> Thanks!
>=20
> Ben.
>=20
>> On 26 Jan 2017, at 2:02, Christer Holmberg wrote:
>>=20
>> Hi,
>>=20
>>> Given the discussion, it seems to make sense to go ahead and start the
>>> IETF LC, and to ask for a TSV review team review as part of that. Does
>>> that make sense?
>>=20
>> Sounds good to me.
>>=20
>> Regards,
>>=20
>> Christer
>>=20
>>=20
>>=20
>>=20
>>>> On 25 Jan 2017, at 14:57, Paul Kyzivat wrote:
>>>>=20
>>>>> On 1/25/17 3:26 PM, Christer Holmberg wrote:
>>>>> Hi,
>>>>>=20
>>>>>>>>> No. Let's not start making things complicated again. The document
>>>>>>>>> is in good shape in my opinion. Let's publish it.
>>>>>>>>>=20
>>>>>>>>> Keep in mind that nothing prevents an endpoint from
>>>>>>>>> closing/re-opening an SCTP association at any time. All we are
>>>>>>>>> saying is that a DTLS change does not automatically affect the
>>>>>>>>> SCTP association.
>>>>>>>>=20
>>>>>>>> I agree that if an endpoint explicitly initiates the closing of an
>>>>>>>> SCTP association, and the other end acknowledges that, they can
>>>>>>>> then
>>>>>>>> establish a new association. (IIUC it would take another O/A to
>>>>>>>> trigger establishing the new one.) But it requires one of the
>>>>>>>> endpoints to know that this is needed.
>>>>>>>>=20
>>>>>>>> Lets consider a 3pcc scenario, where A is in call with B via
>>>>>>>> controller P, where P isn't proxying the media. Now suppose P
>>>>>>>> wants
>>>>>>>> to transfer the call, replacing B with C. Typically it will do
>>>>>>>> this by sending an offerless invite to A, and then forward the
>>>>>>>> resulting offer from A to C.
>>>>>>>>=20
>>>>>>>> In this case C can't resume the SCTP association that A has with
>>>>>>>> B. But A doesn't know that a new association is needed.
>>>>>>>=20
>>>>>>> Assuming B terminates the SCTP association with A when the transfer
>>>>>>> occurs, A will know.
>>>>>>>=20
>>>>>>> But, in any case, when C gets the offer, it will try to establish
>>>>>>> an SCTP association with A.
>>>>>>=20
>>>>>> I agree that C will try that. I haven't done any digging into the
>>>>>> details of the SCTP protocol, but it seems like A and C will be
>>>>>> working at cross purposes. C will be trying to
>>>>>> establish a new association, while A is trying to re-establish the
>>>>>> existing SCTP association over a new DTLS association. Maybe it will
>>>>>> just all work out, but I'm inclined to
>>>>>> doubt it.
>>>>>>=20
>>>>>> It would help to have a expert on the SCTP protocol chime in at this
>>>>>> point.
>>>>>=20
>>>>> We also of course have to assume that A, when it receives the
>>>>> offerless invite, creates an offer that represents the ongoing
>>>>> associations (including the currently used sctp-port value), and not
>>>>> a new one (if A offers a new sctp-port value, there won't be an
>>>>> issue, because the INIT from C will be received on that port). But,
>>>>> that is a general problem with 3pcc, not specific to SCTP.
>>>>=20
>>>> While (AFAIK) there are no SCTP-specific requirements pertaining to
>>>> this, it is hard to imagine anyone concluding they should do
>>>> otherwise.
>>>>=20
>>>>    Thanks,
>>>>    Paul
>>>>=20
>>>> _______________________________________________
>>>> mmusic mailing list
>>>> mmusic@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>=20
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic


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To: "mmusic (E-mail)" <mmusic@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>,  "draft-ietf-mmusic-sdp-bundle-negotiation@ietf.org" <draft-ietf-mmusic-sdp-bundle-negotiation@ietf.org>, <draft-ietf-rtcweb-jsep@tools.ietf.org>
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Date: Fri, 27 Jan 2017 17:43:28 +0100
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Subject: [MMUSIC] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
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MMUSIC and RTCWEB,

Here is now a more complete text proposal that also considers the RTCP. 
It also goes further than what Appendix B defines in one aspect, namely 
explicitly considering third party RTCP reporting and what to do with it.

Feedback is much appreciated. I plan to put this into a PR towards JSEP 
APPENDIX B on monday. If only to get the JSEP authors attention ;-). 
However, the text is really intended for the next version of BUNDLE.


X.  Associating RTP/RTCP With Correct SDP Media Description

    As described in [RFC3550], RTP packets are associated with RTP
    streams [RFC7656].  Each RTP stream is identified by an SSRC value,
    and each RTP packet carries an SSRC value that is used to associate
    the packet with the correct RTP stream.  RTCP packets also uses SSRCs
    to identify on which RTP streams any report or feedback relate to.
    Thus, an RTCP packet will commonly carry multiple SSRC values, and
    might therefore be providing feedback or report on multiple RTP
    streams.

    In order to be able to process received RTP/RTCP packets correctly it
    must be possible to associate an RTP stream with the correct "m="
    line, as the "m=" line and SDP attributes associated with the "m="
    line contain information needed to process the packets.

    As all RTP streams associated with a BUNDLE group are part of the
    same RTP session and using the same address:port combination for
    sending and receiving RTP/RTCP packets, the local address:port
    combination cannot be used to associate an RTP stream with the
    correct "m=" line.  In addition, multiple RTP streams might be
    associated with the same "m=" line.

    Also, as described in Section 10.1.1, the same payload type value
    might be used by multiple RTP streams, in which case the payload type
    value cannot be used to associate an RTP stream with the correct "m="
    line.  However, there are cases where each "m=" line has unique
    payload type values, and then the payload type could serve as
    indicator to the relevant "m=" line the RTP stream is associated
    with.

    An offerer and answerer can inform each other which SSRC values they
    will use for an RTP stream by using the SDP 'ssrc' attribute
    [RFC5576].  However, an offerer will not know which SSRC values the
    answerer will use until the offerer has received the answer providing



Name                      Expires July 31, 2017                 [Page 2]

Internet-Draft              Abbreviated-Title               January 2017


    that information.  Due to this, before the offerer has received the
    answer, the offerer will not be able to associate an RTP stream with
    the correct "m=" line using the SSRC value associated with the RTP
    stream.  In addition, the offerer and answerer may start using new
    SSRC values mid-session, without informing each other using the SDP
    'ssrc' attribute.

    In order for an offerer and answerer to always be able to associate
    an RTP stream with the correct "m=" line, the offerer and answerer
    using the BUNDLE extension MUST support the mechanism defined in
    Section 14, where the offerer and answerer includes the
    identification-tag (provided by the remote peer) associated with an
    "m=" line in the RTP Streams and in RTCP SDES packets part of a
    BUNDLE group.

    The mapping from an SSRC to an identification-tag is carried in RTCP
    SDES packets or in RTP header extensions (Section 14).  Since a
    compound RTCP packet can contain multiple RTCP SDES packets, and each
    RTCP SDES packet can contain multiple chunks, an RTCP packet can
    contain several SSRC to identification-tag mappings.  The offerer and
    answerer maintain tables mapping RTP streams identified by SSRC to
    "m=" lines identified by the identification-tag.  These tables are
    updated each time new information that affects how packets should be
    processed and routed are received.

    To prepare for demultiplexing RTP streams to the correct "m=" line,
    the following steps MUST be followed for each BUNDLE group based on
    the SDP signalling information.

       Construct a table mapping MID to "m=" line for each "m=" line in
       this BUNDLE group.  Note that an "m=" line may only have one MID.

       Construct a table mapping incoming RTP streams (SSRCs) to their
       "m=" line for each "m=" line in this BUNDLE group and for each RTP
       stream explicitly signalled for receiving in that "m=" line.

       Construct a table mapping payload types to "m=" line for each "m="
       line in the BUNDLE group and for each payload type configured for
       receiving in that "m=" line.  If any payload type is configured
       for receiving in more than one "m=" line in the BUNDLE group, do
       not it include it in the table.

    Note that for each of these tables, there can only be one mapping for
    any given key (MID, SSRC, or PT).  In other words, the tables are not
    multimaps.

    As "m=" lines are added or removed from the BUNDLE groups, or their
    configurations are changed, the tables above MUST also be updated.



Name                      Expires July 31, 2017                 [Page 3]

Internet-Draft              Abbreviated-Title               January 2017


    Received RTP packets that are syntactically correct are processed by
    the RTP/RTCP protocol implementation for statistics etc.  After this
    processing they need to be routed to the higher layer context
    associated with the "m=" line within the BUNDLE group.  Somewhere in
    the process where an received RTP packet is processed to be delivered
    to the higher layer by RTP the matching step below MUST be performed:

    1.  Reception of an RTP packet for an RTP stream that has an existing
        mapping in the RTP stream to m= line table.  Before proceeding in
        delivering the packet to the higher layer context according to
        the RTP stream to "m=" line mapping table the following checks
        MUST be performed:

        A.  If the packet carries an RTP header extension with a SDES MID
            value that is not in the table mapping MID to "m=" line, then
            do not deliver the RTP packet to higher layers.

        B.  If the packet carries an RTP header extension with a SDES MID
            value that is in the table mapping MID to "m=" line, and the
            value indicates a different "m=" line than the current RTP
            stream to "m=" line mapping table, then update the RTP stream
            to "m=" line mapping.

    2.  Reception of an RTP packet for an RTP stream that has no existing
        mapping to an m= line.  In this case the following actions MUST
        be performed:

        A.  If the packet carries an RTP header extension with a SDES MID
            value that is in the table mapping MID to "m=" line, then
            create an entry in the RTP stream to "m=" line mapping table
            for this RTP stream (SSRC).  Then deliver the RTP packet to
            the "m=" line context of the created mapping and stop.

        B.  If the packet carries a Payload Type that is in the payload
            type table, then create an entry in the RTP stream to "m="
            line mapping table for this RTP stream (SSRC).  Then deliver
            the RTP packet to the "m=" line context of the created
            mapping and stop.

        C.  Otherwise do not deliver the RTP packet to higher layers.
            Note, this includes unknown MID values.

    For each RTCP packet received (including each RTCP packet that is
    part of a compound RTCP packet), the RTCP packet needs to be
    processed by the RTP/RTCP implementation and relevant information and
    data from the RTCP packets needs to be routed to the appropriate
    handler for the related RTP streams.  The appropriate handler is
    determined by using the RTP stream to "m=" line mapping table.



Name                      Expires July 31, 2017                 [Page 4]

Internet-Draft              Abbreviated-Title               January 2017


    On reception of any compound RTCP packet prior to dispatching the
    received information and data, if there is an RTCP SDES packet
    included that SHOULD be processed first.  If that SDES packet
    contains SDES MID entries, this can results in updates and additions
    to the RTP stream to "m=" line mapping table.  Thus each of the SDES
    MID items are processed and the current table entries are checked if
    the corresponding MID value matches the current RTP stream to "m="
    line mapping, else the entry is updated.  If there is no RTP stream
    to "m=" line table mapping entry for the received SDES item's SSRC,
    such an entry is created.  Note, that in the process of updating the
    table entries, update flap suppression as discussed in Section 4.2.6
    of [RFC7941] should be considered.

    The various different RTCP packets as well as their various sub
    parts, such as the various RTCP Feedback message types, relates to
    the RTP streams in a couple of different ways.  The currently known
    patterns are the following:

    Reports on outgoing RTP streams:  For all RTP streams that this
       endpoint is the source of, it can expect to receive report blocks
       of several types identified as relating to an outgoing stream.
       The basic pattern for these blocks are that the RTCP packet header
       identifies the source of the reports, as identified by an SSRC,
       and containing one or more report blocks, where each report block
       identifies the RTP stream, using the SSRC, the report relates to.
       For this pattern the relevant report information is provided to
       the higher layer associated with the "m=" line the RTP Stream to
       "m=" line table identifies.  The source SSRC as identifier of the
       endpoint that the report originates are relevant for interpreting
       the information, but not necessarily for routing.  Example of this
       is pattern are:

       Sender Report (SR) and Receiver Report (RR)  The basic receiver
          report blocks from RFC3550 start with the SSRC of the RTP
          stream they report on.

       Extended Reports (XR):  RFC3611 is a framework that enables a
          large number of different reports.  However, a large number of
          these report formats are reporting on specific RTP streams and
          thus each individual report of these types contains a SSRC
          field to identify the RTP stream.

    RTCP Feedback Messages for outgoing RTP streams:  The RFC 4585 RTCP
       feedback messages allow for a number of different type of feedback
       messages.  However, the RTCP feedback message header contains the
       SSRC identifying the source of the feedback messages as well as
       the actual type of the feedback.  Some of the feedback messages
       also uses the target SSRC field in the header to identify which



Name                      Expires July 31, 2017                 [Page 5]

Internet-Draft              Abbreviated-Title               January 2017


       RTP stream the feedback is related to.  For these types all the
       FCI entries, if multiple ones are forwarded to the identified
       handler.  Examples of this pattern are:

       Picture Loss Indication (PLI):  RFC 4585 (PT=PSFB, FMT=1).

       Slice Loss Indication (SLI):  RFC 4585 (PT=PSFB, FMT=2).

       Reference Picture Selection Indication (RPSI):  RFC 4585 (PT=PSFB,
          FMT=3).

       Generic NACK:  [RFC4585] (PT=RTPFB and FMT=1).

       Other feedback messages includes the target SSRC in the Feedback
       Control Information (FCI).  Here each FCI needs to be processed
       and the SSRC field identified.  And the individual FCI combined
       with the RTCP packet header context needs to be forwarded to the
       identified handler.  Example of this pattern are:

       Full Intra Request (FIR):  [RFC5104] (PT=PSFB, FMT=4).

       Temporal-Spatial Trade-off Request (TSTR):  [RFC5104] (PT=PSFB,
          FMT=5).

       Temporal-Spatial Trade-off Notification (TSTN):  This [RFC5104]
          defined message (PT=PSFB, FMT=5) is actually a notifciation in
          response to a TSTR this endpoint sent using the SSRC the FCI
          identifies as source.

       H.271 Video Back Channel Message (VBCM):  [RFC5104] (PT=PSFB,
          FMT=7).

       Layer Refresh Request (LRR):  [I-D.ietf-avtext-lrr] (PT=PSFB,
          FMT=TBD).

    Descriptive or Notifications for an incoming RTP stream:  There exist
       some RTCP packet types that provides additional information or
       notifies about events related to the RTP stream identified.  In
       these cases the RTP stream is identified using the SSRC field
       value, and the information is provided to the higher layer
       associated with the "m=" line for the incoming RTP stream as
       identified by the current RTP stream to "m=" line table.  For this
       type of pattern it is common that the RTCP packets and information
       is repeated, either periodically (e.g.  SDES items), or for a
       duration (e.g.  BYE), thus suppression of repetitions can be
       considered.  Examples of these are:





Name                      Expires July 31, 2017                 [Page 6]

Internet-Draft              Abbreviated-Title               January 2017


       Source Description (SDES) RTCP Packet:  The base RTP/RTCP protocol
          [RFC3550] defines SDES RTCP packets as a way of providing per
          source (SSRC) specific information about the source.  An SDES
          packet contains zero or more chunks, where a chunk contains the
          SSRC for the source being described by the one or more items
          included in the chunk.  Forward the SDES items in each chunk to
          the RTP stream's handler.

       Goodbye (BYE) RTCP Packet:  This RTP/RTCP protocol [RFC3550]
          mechanism indicates that a particular RTP stream is leaving the
          RTP session.  Thus, a most relevant event to inform the handler
          for this RTP steam of.

    Third Party Targeted Reports or Feedback:  There exist some
       multiparty RTP Topologies that results in that an endpoint
       receives third party reports (SR [RFC3550], RR [RFC3550] or XR
       [RFC3611]) as well as Feedback Messages [RFC4585] that relates to
       or targets an SSRC that originates from another endpoint, and
       where the source of the RTCP packet is also another endpoint.
       This type of packets should be forwarded to the higher layer
       function dealing with the third party reporting.  And if none
       exist then they can be suppressed.  As the third party handler can
       be focused on determining the conditions for the source of the
       reports and feedback, or focused on how the RTP stream source
       progresses, or both recommendations can't be made.

    APP Packets:  The RTCP APP Packets [RFC3550] are a mechanism that
       enables experimentation.  The APP packet only specifies the source
       of the packet.  Thus this information can be related to any of the
       "m=" lines, thus deliver a copy of the packet to each "m=" line or
       an APP specific handler.

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


From nobody Fri Jan 27 09:15:07 2017
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To: Flemming Andreasen <fandreas@cisco.com>, mmusic@ietf.org
References: <24b5eee4-29b1-33b0-ecea-58c766636311@alvestrand.no> <50b49dbf-7de1-a7c4-923f-f702ac14215e@alvestrand.no> <0df3cbb0-a3e7-011c-c4d5-63aad9350283@alvestrand.no> <e4d73f68-4e70-05f9-bfc8-4429210c5313@cisco.com>
From: Harald Alvestrand <harald@alvestrand.no>
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Date: Fri, 27 Jan 2017 18:14:31 +0100
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [MMUSIC] Change Alert [Re: Modifying an approved document: MSID]
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On 01/26/2017 02:54 PM, Flemming Andreasen wrote:
> Hi Harald
>
> Since the document is not yet in Auth48 and the change is relatively
> minor, it is possible to update it during Auth48.

Would it also be OK to submit an updated draft, so that the RFC Editor
doesn't have to waste time incorporating the changes?

>
> However, from an MMUSIC chair point of view, I would like to get
> positive confirmation from some of the stakeholders that this change
> is indeed desired. Similarly, if anybody has any concerns with the
> suggested change, please send an e-mail to that effect.

I have poked a few people to give their opinion. Will poke again.

>
> Thanks
>
> -- Flemming (as MMUSIC co-chair)
>
>
>
> On 1/23/17 10:18 AM, Harald Alvestrand wrote:
>> Den 19. jan. 2017 13:31, skrev Harald Alvestrand:
>>> Ted pointed out to me that I sent this to the wrong group.
>>>
>>> Chairs and members, please advise.
>> My proposed change is here:
>>
>> https://github.com/alvestrand/rtcweb-msid/pull/16
>>
>> The filed issue is here:
>>
>> https://github.com/alvestrand/rtcweb-msid/issues/15
>>
>> (CCing RTCWEB - please keep discussion, if any, on mmusic)
>>
>>>
>>>
>>> -------- Forwarded Message --------
>>> Subject:     [rtcweb] Modifying an approved document: MSID
>>> Date:     Wed, 18 Jan 2017 23:22:14 +0100
>>> From:     Harald Alvestrand <harald@alvestrand.no>
>>> To:     rtcweb@ietf.org <rtcweb@ietf.org>
>>>
>>>
>>>
>>> When reviewing the implications of the PeerConnection API change to
>>> support AddTrack rather than AddStream, I found an issue.
>>>
>>> This issue concerns draft-ietf-rtcweb-msid, which is currently in
>>> REF-WAIT state (I believe).
>>>
>>> The issue is that it is possible to add a track without specifying a
>>> stream. Since the track's ID needs to be carried, we have to send an
>>> "a=msid" line, but the track's ID is the *second* field on that line,
>>> with the first being the stream's ID.
>>>
>>> This creates a problem.
>>>
>>> Suggested fix: Insert two lines in the document:
>>>
>>> 1) On SDP generation:
>>>
>>> "If there is no stream associated with the track, use the reserved ID
>>> value '-'"
>>>
>>> 2) On SDP parsing
>>>
>>> "If the stream ID is the reserved value '-', the track is not
>>> associated
>>> with a stream, and no stream is signalled or created."
>>>
>>> If this is OK with the community, I'll issue an updated draft with this
>>> change.
>>>
>>>
>>> -- 
>>> Surveillance is pervasive. Go Dark.
>>>
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>
>>>
>>>
>>> _______________________________________________
>>> 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
>> .
>>
>


-- 
Surveillance is pervasive. Go Dark.


From nobody Fri Jan 27 09:41:48 2017
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-dtls-sdp-17.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 of the IETF.

        Title           : Using the SDP Offer/Answer Mechanism for DTLS
        Authors         : Christer Holmberg
                          Roman Shpount
	Filename        : draft-ietf-mmusic-dtls-sdp-17.txt
	Pages           : 25
	Date            : 2017-01-27

Abstract:
   This document defines the SDP offer/answer procedures for negotiating
   and establishing a DTLS association.  The document also defines the
   criteria for when a new DTLS association must be established.  The
   document updates RFC 5763 and RFC 7345, by replacing common SDP
   offer/answer procedures with a reference to this specification.

   This document defines a new SDP media-level attribute, 'dtls-id'.


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

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

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-dtls-sdp-17


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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To: Andy Hutton <andyhutton.ietf@gmail.com>, "mmusic-chairs@ietf.org" <mmusic-chairs@ietf.org>, "sipbrandy-chairs@ietf.org" <sipbrandy-chairs@ietf.org>, Ben Campbell <ben@nostrum.com>, "mmusic@ietf.org" <mmusic@ietf.org>
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From: Bernard Aboba <bernard.aboba@gmail.com>
Date: Fri, 27 Jan 2017 10:20:08 -0800
Message-ID: <CAOW+2dsatEFie4WuaGhprxVw9z-DfMZhPOYetDr+itP5TSnoAw@mail.gmail.com>
To: Flemming Andreasen <fandreas@cisco.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, mmusic WG <mmusic@ietf.org>
Subject: Re: [MMUSIC] Change Alert [Re: Modifying an approved document: MSID]
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--94eb2c07b57a3f5ccb0547178534
Content-Type: text/plain; charset=UTF-8

I am in favor of the change.

On Thu, Jan 26, 2017 at 5:54 AM, Flemming Andreasen <fandreas@cisco.com>
wrote:

> Hi Harald
>
> Since the document is not yet in Auth48 and the change is relatively
> minor, it is possible to update it during Auth48.
>
> However, from an MMUSIC chair point of view, I would like to get positive
> confirmation from some of the stakeholders that this change is indeed
> desired. Similarly, if anybody has any concerns with the suggested change,
> please send an e-mail to that effect.
>
> Thanks
>
> -- Flemming (as MMUSIC co-chair)
>
>
>
> On 1/23/17 10:18 AM, Harald Alvestrand wrote:
>
>> Den 19. jan. 2017 13:31, skrev Harald Alvestrand:
>>
>>> Ted pointed out to me that I sent this to the wrong group.
>>>
>>> Chairs and members, please advise.
>>>
>> My proposed change is here:
>>
>> https://github.com/alvestrand/rtcweb-msid/pull/16
>>
>> The filed issue is here:
>>
>> https://github.com/alvestrand/rtcweb-msid/issues/15
>>
>> (CCing RTCWEB - please keep discussion, if any, on mmusic)
>>
>>
>>>
>>> -------- Forwarded Message --------
>>> Subject:        [rtcweb] Modifying an approved document: MSID
>>> Date:   Wed, 18 Jan 2017 23:22:14 +0100
>>> From:   Harald Alvestrand <harald@alvestrand.no>
>>> To:     rtcweb@ietf.org <rtcweb@ietf.org>
>>>
>>>
>>>
>>> When reviewing the implications of the PeerConnection API change to
>>> support AddTrack rather than AddStream, I found an issue.
>>>
>>> This issue concerns draft-ietf-rtcweb-msid, which is currently in
>>> REF-WAIT state (I believe).
>>>
>>> The issue is that it is possible to add a track without specifying a
>>> stream. Since the track's ID needs to be carried, we have to send an
>>> "a=msid" line, but the track's ID is the *second* field on that line,
>>> with the first being the stream's ID.
>>>
>>> This creates a problem.
>>>
>>> Suggested fix: Insert two lines in the document:
>>>
>>> 1) On SDP generation:
>>>
>>> "If there is no stream associated with the track, use the reserved ID
>>> value '-'"
>>>
>>> 2) On SDP parsing
>>>
>>> "If the stream ID is the reserved value '-', the track is not associated
>>> with a stream, and no stream is signalled or created."
>>>
>>> If this is OK with the community, I'll issue an updated draft with this
>>> change.
>>>
>>>
>>> --
>>> Surveillance is pervasive. Go Dark.
>>>
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>
>>>
>>>
>>> _______________________________________________
>>> 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
>

--94eb2c07b57a3f5ccb0547178534
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">I am in favor of the change.</div><div class=3D"gmail_extr=
a"><br><div class=3D"gmail_quote">On Thu, Jan 26, 2017 at 5:54 AM, Flemming=
 Andreasen <span dir=3D"ltr">&lt;<a href=3D"mailto:fandreas@cisco.com" targ=
et=3D"_blank">fandreas@cisco.com</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex">Hi Harald<br>
<br>
Since the document is not yet in Auth48 and the change is relatively minor,=
 it is possible to update it during Auth48.<br>
<br>
However, from an MMUSIC chair point of view, I would like to get positive c=
onfirmation from some of the stakeholders that this change is indeed desire=
d. Similarly, if anybody has any concerns with the suggested change, please=
 send an e-mail to that effect.<br>
<br>
Thanks<br>
<br>
-- Flemming (as MMUSIC co-chair)<br>
<br>
<br>
<br>
On 1/23/17 10:18 AM, Harald Alvestrand wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Den 19. jan. 2017 13:31, skrev Harald Alvestrand:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Ted pointed out to me that I sent this to the wrong group.<br>
<br>
Chairs and members, please advise.<br>
</blockquote>
My proposed change is here:<br>
<br>
<a href=3D"https://github.com/alvestrand/rtcweb-msid/pull/16" rel=3D"norefe=
rrer" target=3D"_blank">https://github.com/alvestrand/<wbr>rtcweb-msid/pull=
/16</a><br>
<br>
The filed issue is here:<br>
<br>
<a href=3D"https://github.com/alvestrand/rtcweb-msid/issues/15" rel=3D"nore=
ferrer" target=3D"_blank">https://github.com/alvestrand/<wbr>rtcweb-msid/is=
sues/15</a><br>
<br>
(CCing RTCWEB - please keep discussion, if any, on mmusic)<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
<br>
-------- Forwarded Message --------<br>
Subject:=C2=A0 =C2=A0 =C2=A0 =C2=A0 [rtcweb] Modifying an approved document=
: MSID<br>
Date:=C2=A0 =C2=A0Wed, 18 Jan 2017 23:22:14 +0100<br>
From:=C2=A0 =C2=A0Harald Alvestrand &lt;<a href=3D"mailto:harald@alvestrand=
.no" target=3D"_blank">harald@alvestrand.no</a>&gt;<br>
To:=C2=A0 =C2=A0 =C2=A0<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank"=
>rtcweb@ietf.org</a> &lt;<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blan=
k">rtcweb@ietf.org</a>&gt;<br>
<br>
<br>
<br>
When reviewing the implications of the PeerConnection API change to<br>
support AddTrack rather than AddStream, I found an issue.<br>
<br>
This issue concerns draft-ietf-rtcweb-msid, which is currently in<br>
REF-WAIT state (I believe).<br>
<br>
The issue is that it is possible to add a track without specifying a<br>
stream. Since the track&#39;s ID needs to be carried, we have to send an<br=
>
&quot;a=3Dmsid&quot; line, but the track&#39;s ID is the *second* field on =
that line,<br>
with the first being the stream&#39;s ID.<br>
<br>
This creates a problem.<br>
<br>
Suggested fix: Insert two lines in the document:<br>
<br>
1) On SDP generation:<br>
<br>
&quot;If there is no stream associated with the track, use the reserved ID<=
br>
value &#39;-&#39;&quot;<br>
<br>
2) On SDP parsing<br>
<br>
&quot;If the stream ID is the reserved value &#39;-&#39;, the track is not =
associated<br>
with a stream, and no stream is signalled or created.&quot;<br>
<br>
If this is OK with the community, I&#39;ll issue an updated draft with this=
<br>
change.<br>
<br>
<br>
-- <br>
Surveillance is pervasive. Go Dark.<br>
<br>
______________________________<wbr>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/rtcweb</a><br=
>
<br>
<br>
<br>
______________________________<wbr>_________________<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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/mmusic</a><br=
>
<br>
</blockquote>
______________________________<wbr>_________________<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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/mmusic</a><br=
>
.<br>
<br>
</blockquote>
<br>
______________________________<wbr>_________________<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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/mmusic</a><br=
>
</blockquote></div><br></div>

--94eb2c07b57a3f5ccb0547178534--


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In-Reply-To: <CAOW+2dsatEFie4WuaGhprxVw9z-DfMZhPOYetDr+itP5TSnoAw@mail.gmail.com>
From: Suhas Nandakumar <suhasietf@gmail.com>
Date: Sat, 28 Jan 2017 03:46:30 +0000
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To: Bernard Aboba <bernard.aboba@gmail.com>, Flemming Andreasen <fandreas@cisco.com>
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Subject: Re: [MMUSIC] Change Alert [Re: Modifying an approved document: MSID]
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--001a1147b90420ef1205471f6ede
Content-Type: text/plain; charset=UTF-8

+1
On Fri, Jan 27, 2017 at 10:20 AM Bernard Aboba <bernard.aboba@gmail.com>
wrote:

> I am in favor of the change.
>
> On Thu, Jan 26, 2017 at 5:54 AM, Flemming Andreasen <fandreas@cisco.com>
> wrote:
>
> Hi Harald
>
>
>
>
>
> Since the document is not yet in Auth48 and the change is relatively
> minor, it is possible to update it during Auth48.
>
>
>
>
>
> However, from an MMUSIC chair point of view, I would like to get positive
> confirmation from some of the stakeholders that this change is indeed
> desired. Similarly, if anybody has any concerns with the suggested change,
> please send an e-mail to that effect.
>
>
>
>
>
> Thanks
>
>
>
>
>
> -- Flemming (as MMUSIC co-chair)
>
>
>
>
>
>
>
>
>
>
>
> On 1/23/17 10:18 AM, Harald Alvestrand wrote:
>
>
>
>
> Den 19. jan. 2017 13:31, skrev Harald Alvestrand:
>
>
>
>
> Ted pointed out to me that I sent this to the wrong group.
>
>
>
>
>
> Chairs and members, please advise.
>
>
>
>
> My proposed change is here:
>
>
>
>
>
> https://github.com/alvestrand/rtcweb-msid/pull/16
>
>
>
>
>
> The filed issue is here:
>
>
>
>
>
> https://github.com/alvestrand/rtcweb-msid/issues/15
>
>
>
>
>
> (CCing RTCWEB - please keep discussion, if any, on mmusic)
>
>
>
>
>
>
>
>
>
>
>
>
>
> -------- Forwarded Message --------
>
>
> Subject:        [rtcweb] Modifying an approved document: MSID
>
>
> Date:   Wed, 18 Jan 2017 23:22:14 +0100
>
>
> From:   Harald Alvestrand <harald@alvestrand.no>
>
>
> To:     rtcweb@ietf.org <rtcweb@ietf.org>
>
>
>
>
>
>
>
>
>
>
>
> When reviewing the implications of the PeerConnection API change to
>
>
> support AddTrack rather than AddStream, I found an issue.
>
>
>
>
>
> This issue concerns draft-ietf-rtcweb-msid, which is currently in
>
>
> REF-WAIT state (I believe).
>
>
>
>
>
> The issue is that it is possible to add a track without specifying a
>
>
> stream. Since the track's ID needs to be carried, we have to send an
>
>
> "a=msid" line, but the track's ID is the *second* field on that line,
>
>
> with the first being the stream's ID.
>
>
>
>
>
> This creates a problem.
>
>
>
>
>
> Suggested fix: Insert two lines in the document:
>
>
>
>
>
> 1) On SDP generation:
>
>
>
>
>
> "If there is no stream associated with the track, use the reserved ID
>
>
> value '-'"
>
>
>
>
>
> 2) On SDP parsing
>
>
>
>
>
> "If the stream ID is the reserved value '-', the track is not associated
>
>
> with a stream, and no stream is signalled or created."
>
>
>
>
>
> If this is OK with the community, I'll issue an updated draft with this
>
>
> change.
>
>
>
>
>
>
>
>
> --
>
>
> Surveillance is pervasive. Go Dark.
>
>
>
>
>
> _______________________________________________
>
>
> rtcweb mailing list
>
>
> rtcweb@ietf.org
>
>
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
>
>
>
>
>
>
>
>
> _______________________________________________
>
>
> 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
>
>

--001a1147b90420ef1205471f6ede
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div>+1=C2=A0<br><div class=3D"gmail_quote"><div>On Fri, Jan 27, 2017 at 10=
:20 AM Bernard Aboba &lt;<a href=3D"mailto:bernard.aboba@gmail.com">bernard=
.aboba@gmail.com</a>&gt; wrote:<br></div><blockquote class=3D"gmail_quote" =
style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><di=
v class=3D"gmail_msg">I am in favor of the change.</div><div class=3D"gmail=
_extra gmail_msg"><br class=3D"gmail_msg"><div class=3D"gmail_quote gmail_m=
sg">On Thu, Jan 26, 2017 at 5:54 AM, Flemming Andreasen <span class=3D"gmai=
l_msg">&lt;<a href=3D"mailto:fandreas@cisco.com" class=3D"gmail_msg" target=
=3D"_blank">fandreas@cisco.com</a>&gt;</span> wrote:<br class=3D"gmail_msg"=
><blockquote class=3D"gmail_quote gmail_msg" style=3D"margin:0 0 0 .8ex;bor=
der-left:1px #ccc solid;padding-left:1ex">Hi Harald<br class=3D"gmail_msg">=
<br><br><br class=3D"gmail_msg"><br><br>Since the document is not yet in Au=
th48 and the change is relatively minor, it is possible to update it during=
 Auth48.<br class=3D"gmail_msg"><br><br><br class=3D"gmail_msg"><br><br>How=
ever, from an MMUSIC chair point of view, I would like to get positive conf=
irmation from some of the stakeholders that this change is indeed desired. =
Similarly, if anybody has any concerns with the suggested change, please se=
nd an e-mail to that effect.<br class=3D"gmail_msg"><br><br><br class=3D"gm=
ail_msg"><br><br>Thanks<br class=3D"gmail_msg"><br><br><br class=3D"gmail_m=
sg"><br><br>-- Flemming (as MMUSIC co-chair)<br class=3D"gmail_msg"><br><br=
><br class=3D"gmail_msg"><br><br><br class=3D"gmail_msg"><br><br><br class=
=3D"gmail_msg"><br><br>On 1/23/17 10:18 AM, Harald Alvestrand wrote:<br cla=
ss=3D"gmail_msg"><br><br><blockquote class=3D"gmail_quote gmail_msg" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><br><br>=
Den 19. jan. 2017 13:31, skrev Harald Alvestrand:<br class=3D"gmail_msg"><b=
r><br><blockquote class=3D"gmail_quote gmail_msg" style=3D"margin:0 0 0 .8e=
x;border-left:1px #ccc solid;padding-left:1ex"><br><br>Ted pointed out to m=
e that I sent this to the wrong group.<br class=3D"gmail_msg"><br><br><br c=
lass=3D"gmail_msg"><br><br>Chairs and members, please advise.<br class=3D"g=
mail_msg"><br><br></blockquote><br><br>My proposed change is here:<br class=
=3D"gmail_msg"><br><br><br class=3D"gmail_msg"><br><br><a href=3D"https://g=
ithub.com/alvestrand/rtcweb-msid/pull/16" rel=3D"noreferrer" class=3D"gmail=
_msg" target=3D"_blank">https://github.com/alvestrand/rtcweb-msid/pull/16</=
a><br class=3D"gmail_msg"><br><br><br class=3D"gmail_msg"><br><br>The filed=
 issue is here:<br class=3D"gmail_msg"><br><br><br class=3D"gmail_msg"><br>=
<br><a href=3D"https://github.com/alvestrand/rtcweb-msid/issues/15" rel=3D"=
noreferrer" class=3D"gmail_msg" target=3D"_blank">https://github.com/alvest=
rand/rtcweb-msid/issues/15</a><br class=3D"gmail_msg"><br><br><br class=3D"=
gmail_msg"><br><br>(CCing RTCWEB - please keep discussion, if any, on mmusi=
c)<br class=3D"gmail_msg"><br><br><br class=3D"gmail_msg"><br><br><blockquo=
te class=3D"gmail_quote gmail_msg" style=3D"margin:0 0 0 .8ex;border-left:1=
px #ccc solid;padding-left:1ex"><br><br><br class=3D"gmail_msg"><br><br><br=
 class=3D"gmail_msg"><br><br>-------- Forwarded Message --------<br class=
=3D"gmail_msg"><br><br>Subject:=C2=A0 =C2=A0 =C2=A0 =C2=A0 [rtcweb] Modifyi=
ng an approved document: MSID<br class=3D"gmail_msg"><br><br>Date:=C2=A0 =
=C2=A0Wed, 18 Jan 2017 23:22:14 +0100<br class=3D"gmail_msg"><br><br>From:=
=C2=A0 =C2=A0Harald Alvestrand &lt;<a href=3D"mailto:harald@alvestrand.no" =
class=3D"gmail_msg" target=3D"_blank">harald@alvestrand.no</a>&gt;<br class=
=3D"gmail_msg"><br><br>To:=C2=A0 =C2=A0 =C2=A0<a href=3D"mailto:rtcweb@ietf=
.org" class=3D"gmail_msg" target=3D"_blank">rtcweb@ietf.org</a> &lt;<a href=
=3D"mailto:rtcweb@ietf.org" class=3D"gmail_msg" target=3D"_blank">rtcweb@ie=
tf.org</a>&gt;<br class=3D"gmail_msg"><br><br><br class=3D"gmail_msg"><br><=
br><br class=3D"gmail_msg"><br><br><br class=3D"gmail_msg"><br><br>When rev=
iewing the implications of the PeerConnection API change to<br class=3D"gma=
il_msg"><br><br>support AddTrack rather than AddStream, I found an issue.<b=
r class=3D"gmail_msg"><br><br><br class=3D"gmail_msg"><br><br>This issue co=
ncerns draft-ietf-rtcweb-msid, which is currently in<br class=3D"gmail_msg"=
><br><br>REF-WAIT state (I believe).<br class=3D"gmail_msg"><br><br><br cla=
ss=3D"gmail_msg"><br><br>The issue is that it is possible to add a track wi=
thout specifying a<br class=3D"gmail_msg"><br><br>stream. Since the track&#=
39;s ID needs to be carried, we have to send an<br class=3D"gmail_msg"><br>=
<br>&quot;a=3Dmsid&quot; line, but the track&#39;s ID is the *second* field=
 on that line,<br class=3D"gmail_msg"><br><br>with the first being the stre=
am&#39;s ID.<br class=3D"gmail_msg"><br><br><br class=3D"gmail_msg"><br><br=
>This creates a problem.<br class=3D"gmail_msg"><br><br><br class=3D"gmail_=
msg"><br><br>Suggested fix: Insert two lines in the document:<br class=3D"g=
mail_msg"><br><br><br class=3D"gmail_msg"><br><br>1) On SDP generation:<br =
class=3D"gmail_msg"><br><br><br class=3D"gmail_msg"><br><br>&quot;If there =
is no stream associated with the track, use the reserved ID<br class=3D"gma=
il_msg"><br><br>value &#39;-&#39;&quot;<br class=3D"gmail_msg"><br><br><br =
class=3D"gmail_msg"><br><br>2) On SDP parsing<br class=3D"gmail_msg"><br><b=
r><br class=3D"gmail_msg"><br><br>&quot;If the stream ID is the reserved va=
lue &#39;-&#39;, the track is not associated<br class=3D"gmail_msg"><br><br=
>with a stream, and no stream is signalled or created.&quot;<br class=3D"gm=
ail_msg"><br><br><br class=3D"gmail_msg"><br><br>If this is OK with the com=
munity, I&#39;ll issue an updated draft with this<br class=3D"gmail_msg"><b=
r><br>change.<br class=3D"gmail_msg"><br><br><br class=3D"gmail_msg"><br><b=
r><br class=3D"gmail_msg"><br><br>-- <br class=3D"gmail_msg"><br><br>Survei=
llance is pervasive. Go Dark.<br class=3D"gmail_msg"><br><br><br class=3D"g=
mail_msg"><br><br>_______________________________________________<br class=
=3D"gmail_msg"><br><br>rtcweb mailing list<br class=3D"gmail_msg"><br><br><=
a href=3D"mailto:rtcweb@ietf.org" class=3D"gmail_msg" target=3D"_blank">rtc=
web@ietf.org</a><br class=3D"gmail_msg"><br><br><a href=3D"https://www.ietf=
.org/mailman/listinfo/rtcweb" rel=3D"noreferrer" class=3D"gmail_msg" target=
=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br class=3D"gm=
ail_msg"><br><br><br class=3D"gmail_msg"><br><br><br class=3D"gmail_msg"><b=
r><br><br class=3D"gmail_msg"><br><br>_____________________________________=
__________<br class=3D"gmail_msg"><br><br>mmusic mailing list<br class=3D"g=
mail_msg"><br><br><a href=3D"mailto:mmusic@ietf.org" class=3D"gmail_msg" ta=
rget=3D"_blank">mmusic@ietf.org</a><br class=3D"gmail_msg"><br><br><a href=
=3D"https://www.ietf.org/mailman/listinfo/mmusic" rel=3D"noreferrer" class=
=3D"gmail_msg" target=3D"_blank">https://www.ietf.org/mailman/listinfo/mmus=
ic</a><br class=3D"gmail_msg"><br><br><br class=3D"gmail_msg"><br><br></blo=
ckquote><br><br>_______________________________________________<br class=3D=
"gmail_msg"><br><br>mmusic mailing list<br class=3D"gmail_msg"><br><br><a h=
ref=3D"mailto:mmusic@ietf.org" class=3D"gmail_msg" target=3D"_blank">mmusic=
@ietf.org</a><br class=3D"gmail_msg"><br><br><a href=3D"https://www.ietf.or=
g/mailman/listinfo/mmusic" rel=3D"noreferrer" class=3D"gmail_msg" target=3D=
"_blank">https://www.ietf.org/mailman/listinfo/mmusic</a><br class=3D"gmail=
_msg"><br><br>.<br class=3D"gmail_msg"><br><br><br class=3D"gmail_msg"><br>=
<br></blockquote><br><br><br class=3D"gmail_msg"><br><br>__________________=
_____________________________<br class=3D"gmail_msg"><br><br>mmusic mailing=
 list<br class=3D"gmail_msg"><br><br><a href=3D"mailto:mmusic@ietf.org" cla=
ss=3D"gmail_msg" target=3D"_blank">mmusic@ietf.org</a><br class=3D"gmail_ms=
g"><br><br><a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" rel=3D"=
noreferrer" class=3D"gmail_msg" target=3D"_blank">https://www.ietf.org/mail=
man/listinfo/mmusic</a><br class=3D"gmail_msg"><br><br></blockquote></div><=
br class=3D"gmail_msg"></div><br><br>______________________________________=
_________<br class=3D"gmail_msg"><br>mmusic mailing list<br class=3D"gmail_=
msg"><br><a href=3D"mailto:mmusic@ietf.org" class=3D"gmail_msg" target=3D"_=
blank">mmusic@ietf.org</a><br class=3D"gmail_msg"><br><a href=3D"https://ww=
w.ietf.org/mailman/listinfo/mmusic" rel=3D"noreferrer" class=3D"gmail_msg" =
target=3D"_blank">https://www.ietf.org/mailman/listinfo/mmusic</a><br class=
=3D"gmail_msg"><br></blockquote></div></div>

--001a1147b90420ef1205471f6ede--


From nobody Sat Jan 28 09:10:43 2017
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-dtls-sdp-18.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 of the IETF.

        Title           : Using the SDP Offer/Answer Mechanism for DTLS
        Authors         : Christer Holmberg
                          Roman Shpount
	Filename        : draft-ietf-mmusic-dtls-sdp-18.txt
	Pages           : 25
	Date            : 2017-01-28

Abstract:
   This document defines the SDP offer/answer procedures for negotiating
   and establishing a DTLS association.  The document also defines the
   criteria for when a new DTLS association must be established.  The
   document updates RFC 5763 and RFC 7345, by replacing common SDP
   offer/answer procedures with a reference to this specification.

   This document defines a new SDP media-level attribute, 'dtls-id'.


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

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-mmusic-dtls-sdp-18

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-dtls-sdp-18


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 Mon Jan 30 05:19:04 2017
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Subject: Re: [MMUSIC] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
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Hi,

I have now generated a pull request for this text, with some very minor 
changes.

https://github.com/rtcweb-wg/jsep/pull/538

Cheers

Magnus

Den 2017-01-27 kl. 17:43, skrev Magnus Westerlund:
> MMUSIC and RTCWEB,
>
> Here is now a more complete text proposal that also considers the RTCP.
> It also goes further than what Appendix B defines in one aspect, namely
> explicitly considering third party RTCP reporting and what to do with it.
>
> Feedback is much appreciated. I plan to put this into a PR towards JSEP
> APPENDIX B on monday. If only to get the JSEP authors attention ;-).
> However, the text is really intended for the next version of BUNDLE.
>
>
> X.  Associating RTP/RTCP With Correct SDP Media Description
>
>    As described in [RFC3550], RTP packets are associated with RTP
>    streams [RFC7656].  Each RTP stream is identified by an SSRC value,
>    and each RTP packet carries an SSRC value that is used to associate
>    the packet with the correct RTP stream.  RTCP packets also uses SSRCs
>    to identify on which RTP streams any report or feedback relate to.
>    Thus, an RTCP packet will commonly carry multiple SSRC values, and
>    might therefore be providing feedback or report on multiple RTP
>    streams.
>
>    In order to be able to process received RTP/RTCP packets correctly it
>    must be possible to associate an RTP stream with the correct "m="
>    line, as the "m=" line and SDP attributes associated with the "m="
>    line contain information needed to process the packets.
>
>    As all RTP streams associated with a BUNDLE group are part of the
>    same RTP session and using the same address:port combination for
>    sending and receiving RTP/RTCP packets, the local address:port
>    combination cannot be used to associate an RTP stream with the
>    correct "m=" line.  In addition, multiple RTP streams might be
>    associated with the same "m=" line.
>
>    Also, as described in Section 10.1.1, the same payload type value
>    might be used by multiple RTP streams, in which case the payload type
>    value cannot be used to associate an RTP stream with the correct "m="
>    line.  However, there are cases where each "m=" line has unique
>    payload type values, and then the payload type could serve as
>    indicator to the relevant "m=" line the RTP stream is associated
>    with.
>
>    An offerer and answerer can inform each other which SSRC values they
>    will use for an RTP stream by using the SDP 'ssrc' attribute
>    [RFC5576].  However, an offerer will not know which SSRC values the
>    answerer will use until the offerer has received the answer providing
>
>
>
> Name                      Expires July 31, 2017                 [Page 2]
> 
> Internet-Draft              Abbreviated-Title               January 2017
>
>
>    that information.  Due to this, before the offerer has received the
>    answer, the offerer will not be able to associate an RTP stream with
>    the correct "m=" line using the SSRC value associated with the RTP
>    stream.  In addition, the offerer and answerer may start using new
>    SSRC values mid-session, without informing each other using the SDP
>    'ssrc' attribute.
>
>    In order for an offerer and answerer to always be able to associate
>    an RTP stream with the correct "m=" line, the offerer and answerer
>    using the BUNDLE extension MUST support the mechanism defined in
>    Section 14, where the offerer and answerer includes the
>    identification-tag (provided by the remote peer) associated with an
>    "m=" line in the RTP Streams and in RTCP SDES packets part of a
>    BUNDLE group.
>
>    The mapping from an SSRC to an identification-tag is carried in RTCP
>    SDES packets or in RTP header extensions (Section 14).  Since a
>    compound RTCP packet can contain multiple RTCP SDES packets, and each
>    RTCP SDES packet can contain multiple chunks, an RTCP packet can
>    contain several SSRC to identification-tag mappings.  The offerer and
>    answerer maintain tables mapping RTP streams identified by SSRC to
>    "m=" lines identified by the identification-tag.  These tables are
>    updated each time new information that affects how packets should be
>    processed and routed are received.
>
>    To prepare for demultiplexing RTP streams to the correct "m=" line,
>    the following steps MUST be followed for each BUNDLE group based on
>    the SDP signalling information.
>
>       Construct a table mapping MID to "m=" line for each "m=" line in
>       this BUNDLE group.  Note that an "m=" line may only have one MID.
>
>       Construct a table mapping incoming RTP streams (SSRCs) to their
>       "m=" line for each "m=" line in this BUNDLE group and for each RTP
>       stream explicitly signalled for receiving in that "m=" line.
>
>       Construct a table mapping payload types to "m=" line for each "m="
>       line in the BUNDLE group and for each payload type configured for
>       receiving in that "m=" line.  If any payload type is configured
>       for receiving in more than one "m=" line in the BUNDLE group, do
>       not it include it in the table.
>
>    Note that for each of these tables, there can only be one mapping for
>    any given key (MID, SSRC, or PT).  In other words, the tables are not
>    multimaps.
>
>    As "m=" lines are added or removed from the BUNDLE groups, or their
>    configurations are changed, the tables above MUST also be updated.
>
>
>
> Name                      Expires July 31, 2017                 [Page 3]
> 
> Internet-Draft              Abbreviated-Title               January 2017
>
>
>    Received RTP packets that are syntactically correct are processed by
>    the RTP/RTCP protocol implementation for statistics etc.  After this
>    processing they need to be routed to the higher layer context
>    associated with the "m=" line within the BUNDLE group.  Somewhere in
>    the process where an received RTP packet is processed to be delivered
>    to the higher layer by RTP the matching step below MUST be performed:
>
>    1.  Reception of an RTP packet for an RTP stream that has an existing
>        mapping in the RTP stream to m= line table.  Before proceeding in
>        delivering the packet to the higher layer context according to
>        the RTP stream to "m=" line mapping table the following checks
>        MUST be performed:
>
>        A.  If the packet carries an RTP header extension with a SDES MID
>            value that is not in the table mapping MID to "m=" line, then
>            do not deliver the RTP packet to higher layers.
>
>        B.  If the packet carries an RTP header extension with a SDES MID
>            value that is in the table mapping MID to "m=" line, and the
>            value indicates a different "m=" line than the current RTP
>            stream to "m=" line mapping table, then update the RTP stream
>            to "m=" line mapping.
>
>    2.  Reception of an RTP packet for an RTP stream that has no existing
>        mapping to an m= line.  In this case the following actions MUST
>        be performed:
>
>        A.  If the packet carries an RTP header extension with a SDES MID
>            value that is in the table mapping MID to "m=" line, then
>            create an entry in the RTP stream to "m=" line mapping table
>            for this RTP stream (SSRC).  Then deliver the RTP packet to
>            the "m=" line context of the created mapping and stop.
>
>        B.  If the packet carries a Payload Type that is in the payload
>            type table, then create an entry in the RTP stream to "m="
>            line mapping table for this RTP stream (SSRC).  Then deliver
>            the RTP packet to the "m=" line context of the created
>            mapping and stop.
>
>        C.  Otherwise do not deliver the RTP packet to higher layers.
>            Note, this includes unknown MID values.
>
>    For each RTCP packet received (including each RTCP packet that is
>    part of a compound RTCP packet), the RTCP packet needs to be
>    processed by the RTP/RTCP implementation and relevant information and
>    data from the RTCP packets needs to be routed to the appropriate
>    handler for the related RTP streams.  The appropriate handler is
>    determined by using the RTP stream to "m=" line mapping table.
>
>
>
> Name                      Expires July 31, 2017                 [Page 4]
> 
> Internet-Draft              Abbreviated-Title               January 2017
>
>
>    On reception of any compound RTCP packet prior to dispatching the
>    received information and data, if there is an RTCP SDES packet
>    included that SHOULD be processed first.  If that SDES packet
>    contains SDES MID entries, this can results in updates and additions
>    to the RTP stream to "m=" line mapping table.  Thus each of the SDES
>    MID items are processed and the current table entries are checked if
>    the corresponding MID value matches the current RTP stream to "m="
>    line mapping, else the entry is updated.  If there is no RTP stream
>    to "m=" line table mapping entry for the received SDES item's SSRC,
>    such an entry is created.  Note, that in the process of updating the
>    table entries, update flap suppression as discussed in Section 4.2.6
>    of [RFC7941] should be considered.
>
>    The various different RTCP packets as well as their various sub
>    parts, such as the various RTCP Feedback message types, relates to
>    the RTP streams in a couple of different ways.  The currently known
>    patterns are the following:
>
>    Reports on outgoing RTP streams:  For all RTP streams that this
>       endpoint is the source of, it can expect to receive report blocks
>       of several types identified as relating to an outgoing stream.
>       The basic pattern for these blocks are that the RTCP packet header
>       identifies the source of the reports, as identified by an SSRC,
>       and containing one or more report blocks, where each report block
>       identifies the RTP stream, using the SSRC, the report relates to.
>       For this pattern the relevant report information is provided to
>       the higher layer associated with the "m=" line the RTP Stream to
>       "m=" line table identifies.  The source SSRC as identifier of the
>       endpoint that the report originates are relevant for interpreting
>       the information, but not necessarily for routing.  Example of this
>       is pattern are:
>
>       Sender Report (SR) and Receiver Report (RR)  The basic receiver
>          report blocks from RFC3550 start with the SSRC of the RTP
>          stream they report on.
>
>       Extended Reports (XR):  RFC3611 is a framework that enables a
>          large number of different reports.  However, a large number of
>          these report formats are reporting on specific RTP streams and
>          thus each individual report of these types contains a SSRC
>          field to identify the RTP stream.
>
>    RTCP Feedback Messages for outgoing RTP streams:  The RFC 4585 RTCP
>       feedback messages allow for a number of different type of feedback
>       messages.  However, the RTCP feedback message header contains the
>       SSRC identifying the source of the feedback messages as well as
>       the actual type of the feedback.  Some of the feedback messages
>       also uses the target SSRC field in the header to identify which
>
>
>
> Name                      Expires July 31, 2017                 [Page 5]
> 
> Internet-Draft              Abbreviated-Title               January 2017
>
>
>       RTP stream the feedback is related to.  For these types all the
>       FCI entries, if multiple ones are forwarded to the identified
>       handler.  Examples of this pattern are:
>
>       Picture Loss Indication (PLI):  RFC 4585 (PT=PSFB, FMT=1).
>
>       Slice Loss Indication (SLI):  RFC 4585 (PT=PSFB, FMT=2).
>
>       Reference Picture Selection Indication (RPSI):  RFC 4585 (PT=PSFB,
>          FMT=3).
>
>       Generic NACK:  [RFC4585] (PT=RTPFB and FMT=1).
>
>       Other feedback messages includes the target SSRC in the Feedback
>       Control Information (FCI).  Here each FCI needs to be processed
>       and the SSRC field identified.  And the individual FCI combined
>       with the RTCP packet header context needs to be forwarded to the
>       identified handler.  Example of this pattern are:
>
>       Full Intra Request (FIR):  [RFC5104] (PT=PSFB, FMT=4).
>
>       Temporal-Spatial Trade-off Request (TSTR):  [RFC5104] (PT=PSFB,
>          FMT=5).
>
>       Temporal-Spatial Trade-off Notification (TSTN):  This [RFC5104]
>          defined message (PT=PSFB, FMT=5) is actually a notifciation in
>          response to a TSTR this endpoint sent using the SSRC the FCI
>          identifies as source.
>
>       H.271 Video Back Channel Message (VBCM):  [RFC5104] (PT=PSFB,
>          FMT=7).
>
>       Layer Refresh Request (LRR):  [I-D.ietf-avtext-lrr] (PT=PSFB,
>          FMT=TBD).
>
>    Descriptive or Notifications for an incoming RTP stream:  There exist
>       some RTCP packet types that provides additional information or
>       notifies about events related to the RTP stream identified.  In
>       these cases the RTP stream is identified using the SSRC field
>       value, and the information is provided to the higher layer
>       associated with the "m=" line for the incoming RTP stream as
>       identified by the current RTP stream to "m=" line table.  For this
>       type of pattern it is common that the RTCP packets and information
>       is repeated, either periodically (e.g.  SDES items), or for a
>       duration (e.g.  BYE), thus suppression of repetitions can be
>       considered.  Examples of these are:
>
>
>
>
>
> Name                      Expires July 31, 2017                 [Page 6]
> 
> Internet-Draft              Abbreviated-Title               January 2017
>
>
>       Source Description (SDES) RTCP Packet:  The base RTP/RTCP protocol
>          [RFC3550] defines SDES RTCP packets as a way of providing per
>          source (SSRC) specific information about the source.  An SDES
>          packet contains zero or more chunks, where a chunk contains the
>          SSRC for the source being described by the one or more items
>          included in the chunk.  Forward the SDES items in each chunk to
>          the RTP stream's handler.
>
>       Goodbye (BYE) RTCP Packet:  This RTP/RTCP protocol [RFC3550]
>          mechanism indicates that a particular RTP stream is leaving the
>          RTP session.  Thus, a most relevant event to inform the handler
>          for this RTP steam of.
>
>    Third Party Targeted Reports or Feedback:  There exist some
>       multiparty RTP Topologies that results in that an endpoint
>       receives third party reports (SR [RFC3550], RR [RFC3550] or XR
>       [RFC3611]) as well as Feedback Messages [RFC4585] that relates to
>       or targets an SSRC that originates from another endpoint, and
>       where the source of the RTCP packet is also another endpoint.
>       This type of packets should be forwarded to the higher layer
>       function dealing with the third party reporting.  And if none
>       exist then they can be suppressed.  As the third party handler can
>       be focused on determining the conditions for the source of the
>       reports and feedback, or focused on how the RTP stream source
>       progresses, or both recommendations can't be made.
>
>    APP Packets:  The RTCP APP Packets [RFC3550] are a mechanism that
>       enables experimentation.  The APP packet only specifies the source
>       of the packet.  Thus this information can be related to any of the
>       "m=" lines, thus deliver a copy of the packet to each "m=" line or
>       an APP specific handler.
>


-- 

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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References: <24b5eee4-29b1-33b0-ecea-58c766636311@alvestrand.no> <50b49dbf-7de1-a7c4-923f-f702ac14215e@alvestrand.no> <0df3cbb0-a3e7-011c-c4d5-63aad9350283@alvestrand.no> <e4d73f68-4e70-05f9-bfc8-4429210c5313@cisco.com> <CAOW+2dsatEFie4WuaGhprxVw9z-DfMZhPOYetDr+itP5TSnoAw@mail.gmail.com>
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Date: Mon, 30 Jan 2017 14:28:54 +0100
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Cc: Flemming Andreasen <fandreas@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, mmusic WG <mmusic@ietf.org>
Subject: Re: [MMUSIC] Change Alert [Re: Modifying an approved document: MSID]
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fine with me

On Fri, Jan 27, 2017 at 7:20 PM, Bernard Aboba <bernard.aboba@gmail.com> wr=
ote:
> I am in favor of the change.
>
> On Thu, Jan 26, 2017 at 5:54 AM, Flemming Andreasen <fandreas@cisco.com>
> wrote:
>>
>> Hi Harald
>>
>> Since the document is not yet in Auth48 and the change is relatively
>> minor, it is possible to update it during Auth48.
>>
>> However, from an MMUSIC chair point of view, I would like to get positiv=
e
>> confirmation from some of the stakeholders that this change is indeed
>> desired. Similarly, if anybody has any concerns with the suggested chang=
e,
>> please send an e-mail to that effect.
>>
>> Thanks
>>
>> -- Flemming (as MMUSIC co-chair)
>>
>>
>>
>> On 1/23/17 10:18 AM, Harald Alvestrand wrote:
>>>
>>> Den 19. jan. 2017 13:31, skrev Harald Alvestrand:
>>>>
>>>> Ted pointed out to me that I sent this to the wrong group.
>>>>
>>>> Chairs and members, please advise.
>>>
>>> My proposed change is here:
>>>
>>> https://github.com/alvestrand/rtcweb-msid/pull/16
>>>
>>> The filed issue is here:
>>>
>>> https://github.com/alvestrand/rtcweb-msid/issues/15
>>>
>>> (CCing RTCWEB - please keep discussion, if any, on mmusic)
>>>
>>>>
>>>>
>>>> -------- Forwarded Message --------
>>>> Subject:        [rtcweb] Modifying an approved document: MSID
>>>> Date:   Wed, 18 Jan 2017 23:22:14 +0100
>>>> From:   Harald Alvestrand <harald@alvestrand.no>
>>>> To:     rtcweb@ietf.org <rtcweb@ietf.org>
>>>>
>>>>
>>>>
>>>> When reviewing the implications of the PeerConnection API change to
>>>> support AddTrack rather than AddStream, I found an issue.
>>>>
>>>> This issue concerns draft-ietf-rtcweb-msid, which is currently in
>>>> REF-WAIT state (I believe).
>>>>
>>>> The issue is that it is possible to add a track without specifying a
>>>> stream. Since the track's ID needs to be carried, we have to send an
>>>> "a=3Dmsid" line, but the track's ID is the *second* field on that line=
,
>>>> with the first being the stream's ID.
>>>>
>>>> This creates a problem.
>>>>
>>>> Suggested fix: Insert two lines in the document:
>>>>
>>>> 1) On SDP generation:
>>>>
>>>> "If there is no stream associated with the track, use the reserved ID
>>>> value '-'"
>>>>
>>>> 2) On SDP parsing
>>>>
>>>> "If the stream ID is the reserved value '-', the track is not associat=
ed
>>>> with a stream, and no stream is signalled or created."
>>>>
>>>> If this is OK with the community, I'll issue an updated draft with thi=
s
>>>> change.
>>>>
>>>>
>>>> --
>>>> Surveillance is pervasive. Go Dark.
>>>>
>>>> _______________________________________________
>>>> rtcweb mailing list
>>>> rtcweb@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> 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
Victor Pascual =C3=81vila


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To: "mmusic (E-mail)" <mmusic@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>,  "draft-ietf-mmusic-sdp-bundle-negotiation@ietf.org" <draft-ietf-mmusic-sdp-bundle-negotiation@ietf.org>, <draft-ietf-rtcweb-jsep@tools.ietf.org>
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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [MMUSIC] [rtcweb] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
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Hi,

There was a request for a readable diff on github, so I have produced 
one: http://www.denstore.se/IETF/Diff-jsep-18-part.html

However, I will note that it is likely only relevant to see the changes 
up until the received RTP packet routing steps. Then the changes are so 
massive due to restructuring that the diff is not particularly useful.

I will also note that the text on github has been improved with some 
language corrections proposed by Bernard Aboba.

Cheers

Magnus

Den 2017-01-30 kl. 14:18, skrev Magnus Westerlund:
> Hi,
>
> I have now generated a pull request for this text, with some very minor
> changes.
>
> https://github.com/rtcweb-wg/jsep/pull/538
>
> Cheers
>
> Magnus
>
> Den 2017-01-27 kl. 17:43, skrev Magnus Westerlund:
>> MMUSIC and RTCWEB,
>>
>> Here is now a more complete text proposal that also considers the RTCP.
>> It also goes further than what Appendix B defines in one aspect, namely
>> explicitly considering third party RTCP reporting and what to do with it.
>>
>> Feedback is much appreciated. I plan to put this into a PR towards JSEP
>> APPENDIX B on monday. If only to get the JSEP authors attention ;-).
>> However, the text is really intended for the next version of BUNDLE.
>>
>>
>> X.  Associating RTP/RTCP With Correct SDP Media Description
>>
>>    As described in [RFC3550], RTP packets are associated with RTP
>>    streams [RFC7656].  Each RTP stream is identified by an SSRC value,
>>    and each RTP packet carries an SSRC value that is used to associate
>>    the packet with the correct RTP stream.  RTCP packets also uses SSRCs
>>    to identify on which RTP streams any report or feedback relate to.
>>    Thus, an RTCP packet will commonly carry multiple SSRC values, and
>>    might therefore be providing feedback or report on multiple RTP
>>    streams.
>>
>>    In order to be able to process received RTP/RTCP packets correctly it
>>    must be possible to associate an RTP stream with the correct "m="
>>    line, as the "m=" line and SDP attributes associated with the "m="
>>    line contain information needed to process the packets.
>>
>>    As all RTP streams associated with a BUNDLE group are part of the
>>    same RTP session and using the same address:port combination for
>>    sending and receiving RTP/RTCP packets, the local address:port
>>    combination cannot be used to associate an RTP stream with the
>>    correct "m=" line.  In addition, multiple RTP streams might be
>>    associated with the same "m=" line.
>>
>>    Also, as described in Section 10.1.1, the same payload type value
>>    might be used by multiple RTP streams, in which case the payload type
>>    value cannot be used to associate an RTP stream with the correct "m="
>>    line.  However, there are cases where each "m=" line has unique
>>    payload type values, and then the payload type could serve as
>>    indicator to the relevant "m=" line the RTP stream is associated
>>    with.
>>
>>    An offerer and answerer can inform each other which SSRC values they
>>    will use for an RTP stream by using the SDP 'ssrc' attribute
>>    [RFC5576].  However, an offerer will not know which SSRC values the
>>    answerer will use until the offerer has received the answer providing
>>
>>
>>
>> Name                      Expires July 31, 2017                 [Page 2]
>> 
>> Internet-Draft              Abbreviated-Title               January 2017
>>
>>
>>    that information.  Due to this, before the offerer has received the
>>    answer, the offerer will not be able to associate an RTP stream with
>>    the correct "m=" line using the SSRC value associated with the RTP
>>    stream.  In addition, the offerer and answerer may start using new
>>    SSRC values mid-session, without informing each other using the SDP
>>    'ssrc' attribute.
>>
>>    In order for an offerer and answerer to always be able to associate
>>    an RTP stream with the correct "m=" line, the offerer and answerer
>>    using the BUNDLE extension MUST support the mechanism defined in
>>    Section 14, where the offerer and answerer includes the
>>    identification-tag (provided by the remote peer) associated with an
>>    "m=" line in the RTP Streams and in RTCP SDES packets part of a
>>    BUNDLE group.
>>
>>    The mapping from an SSRC to an identification-tag is carried in RTCP
>>    SDES packets or in RTP header extensions (Section 14).  Since a
>>    compound RTCP packet can contain multiple RTCP SDES packets, and each
>>    RTCP SDES packet can contain multiple chunks, an RTCP packet can
>>    contain several SSRC to identification-tag mappings.  The offerer and
>>    answerer maintain tables mapping RTP streams identified by SSRC to
>>    "m=" lines identified by the identification-tag.  These tables are
>>    updated each time new information that affects how packets should be
>>    processed and routed are received.
>>
>>    To prepare for demultiplexing RTP streams to the correct "m=" line,
>>    the following steps MUST be followed for each BUNDLE group based on
>>    the SDP signalling information.
>>
>>       Construct a table mapping MID to "m=" line for each "m=" line in
>>       this BUNDLE group.  Note that an "m=" line may only have one MID.
>>
>>       Construct a table mapping incoming RTP streams (SSRCs) to their
>>       "m=" line for each "m=" line in this BUNDLE group and for each RTP
>>       stream explicitly signalled for receiving in that "m=" line.
>>
>>       Construct a table mapping payload types to "m=" line for each "m="
>>       line in the BUNDLE group and for each payload type configured for
>>       receiving in that "m=" line.  If any payload type is configured
>>       for receiving in more than one "m=" line in the BUNDLE group, do
>>       not it include it in the table.
>>
>>    Note that for each of these tables, there can only be one mapping for
>>    any given key (MID, SSRC, or PT).  In other words, the tables are not
>>    multimaps.
>>
>>    As "m=" lines are added or removed from the BUNDLE groups, or their
>>    configurations are changed, the tables above MUST also be updated.
>>
>>
>>
>> Name                      Expires July 31, 2017                 [Page 3]
>> 
>> Internet-Draft              Abbreviated-Title               January 2017
>>
>>
>>    Received RTP packets that are syntactically correct are processed by
>>    the RTP/RTCP protocol implementation for statistics etc.  After this
>>    processing they need to be routed to the higher layer context
>>    associated with the "m=" line within the BUNDLE group.  Somewhere in
>>    the process where an received RTP packet is processed to be delivered
>>    to the higher layer by RTP the matching step below MUST be performed:
>>
>>    1.  Reception of an RTP packet for an RTP stream that has an existing
>>        mapping in the RTP stream to m= line table.  Before proceeding in
>>        delivering the packet to the higher layer context according to
>>        the RTP stream to "m=" line mapping table the following checks
>>        MUST be performed:
>>
>>        A.  If the packet carries an RTP header extension with a SDES MID
>>            value that is not in the table mapping MID to "m=" line, then
>>            do not deliver the RTP packet to higher layers.
>>
>>        B.  If the packet carries an RTP header extension with a SDES MID
>>            value that is in the table mapping MID to "m=" line, and the
>>            value indicates a different "m=" line than the current RTP
>>            stream to "m=" line mapping table, then update the RTP stream
>>            to "m=" line mapping.
>>
>>    2.  Reception of an RTP packet for an RTP stream that has no existing
>>        mapping to an m= line.  In this case the following actions MUST
>>        be performed:
>>
>>        A.  If the packet carries an RTP header extension with a SDES MID
>>            value that is in the table mapping MID to "m=" line, then
>>            create an entry in the RTP stream to "m=" line mapping table
>>            for this RTP stream (SSRC).  Then deliver the RTP packet to
>>            the "m=" line context of the created mapping and stop.
>>
>>        B.  If the packet carries a Payload Type that is in the payload
>>            type table, then create an entry in the RTP stream to "m="
>>            line mapping table for this RTP stream (SSRC).  Then deliver
>>            the RTP packet to the "m=" line context of the created
>>            mapping and stop.
>>
>>        C.  Otherwise do not deliver the RTP packet to higher layers.
>>            Note, this includes unknown MID values.
>>
>>    For each RTCP packet received (including each RTCP packet that is
>>    part of a compound RTCP packet), the RTCP packet needs to be
>>    processed by the RTP/RTCP implementation and relevant information and
>>    data from the RTCP packets needs to be routed to the appropriate
>>    handler for the related RTP streams.  The appropriate handler is
>>    determined by using the RTP stream to "m=" line mapping table.
>>
>>
>>
>> Name                      Expires July 31, 2017                 [Page 4]
>> 
>> Internet-Draft              Abbreviated-Title               January 2017
>>
>>
>>    On reception of any compound RTCP packet prior to dispatching the
>>    received information and data, if there is an RTCP SDES packet
>>    included that SHOULD be processed first.  If that SDES packet
>>    contains SDES MID entries, this can results in updates and additions
>>    to the RTP stream to "m=" line mapping table.  Thus each of the SDES
>>    MID items are processed and the current table entries are checked if
>>    the corresponding MID value matches the current RTP stream to "m="
>>    line mapping, else the entry is updated.  If there is no RTP stream
>>    to "m=" line table mapping entry for the received SDES item's SSRC,
>>    such an entry is created.  Note, that in the process of updating the
>>    table entries, update flap suppression as discussed in Section 4.2.6
>>    of [RFC7941] should be considered.
>>
>>    The various different RTCP packets as well as their various sub
>>    parts, such as the various RTCP Feedback message types, relates to
>>    the RTP streams in a couple of different ways.  The currently known
>>    patterns are the following:
>>
>>    Reports on outgoing RTP streams:  For all RTP streams that this
>>       endpoint is the source of, it can expect to receive report blocks
>>       of several types identified as relating to an outgoing stream.
>>       The basic pattern for these blocks are that the RTCP packet header
>>       identifies the source of the reports, as identified by an SSRC,
>>       and containing one or more report blocks, where each report block
>>       identifies the RTP stream, using the SSRC, the report relates to.
>>       For this pattern the relevant report information is provided to
>>       the higher layer associated with the "m=" line the RTP Stream to
>>       "m=" line table identifies.  The source SSRC as identifier of the
>>       endpoint that the report originates are relevant for interpreting
>>       the information, but not necessarily for routing.  Example of this
>>       is pattern are:
>>
>>       Sender Report (SR) and Receiver Report (RR)  The basic receiver
>>          report blocks from RFC3550 start with the SSRC of the RTP
>>          stream they report on.
>>
>>       Extended Reports (XR):  RFC3611 is a framework that enables a
>>          large number of different reports.  However, a large number of
>>          these report formats are reporting on specific RTP streams and
>>          thus each individual report of these types contains a SSRC
>>          field to identify the RTP stream.
>>
>>    RTCP Feedback Messages for outgoing RTP streams:  The RFC 4585 RTCP
>>       feedback messages allow for a number of different type of feedback
>>       messages.  However, the RTCP feedback message header contains the
>>       SSRC identifying the source of the feedback messages as well as
>>       the actual type of the feedback.  Some of the feedback messages
>>       also uses the target SSRC field in the header to identify which
>>
>>
>>
>> Name                      Expires July 31, 2017                 [Page 5]
>> 
>> Internet-Draft              Abbreviated-Title               January 2017
>>
>>
>>       RTP stream the feedback is related to.  For these types all the
>>       FCI entries, if multiple ones are forwarded to the identified
>>       handler.  Examples of this pattern are:
>>
>>       Picture Loss Indication (PLI):  RFC 4585 (PT=PSFB, FMT=1).
>>
>>       Slice Loss Indication (SLI):  RFC 4585 (PT=PSFB, FMT=2).
>>
>>       Reference Picture Selection Indication (RPSI):  RFC 4585 (PT=PSFB,
>>          FMT=3).
>>
>>       Generic NACK:  [RFC4585] (PT=RTPFB and FMT=1).
>>
>>       Other feedback messages includes the target SSRC in the Feedback
>>       Control Information (FCI).  Here each FCI needs to be processed
>>       and the SSRC field identified.  And the individual FCI combined
>>       with the RTCP packet header context needs to be forwarded to the
>>       identified handler.  Example of this pattern are:
>>
>>       Full Intra Request (FIR):  [RFC5104] (PT=PSFB, FMT=4).
>>
>>       Temporal-Spatial Trade-off Request (TSTR):  [RFC5104] (PT=PSFB,
>>          FMT=5).
>>
>>       Temporal-Spatial Trade-off Notification (TSTN):  This [RFC5104]
>>          defined message (PT=PSFB, FMT=5) is actually a notifciation in
>>          response to a TSTR this endpoint sent using the SSRC the FCI
>>          identifies as source.
>>
>>       H.271 Video Back Channel Message (VBCM):  [RFC5104] (PT=PSFB,
>>          FMT=7).
>>
>>       Layer Refresh Request (LRR):  [I-D.ietf-avtext-lrr] (PT=PSFB,
>>          FMT=TBD).
>>
>>    Descriptive or Notifications for an incoming RTP stream:  There exist
>>       some RTCP packet types that provides additional information or
>>       notifies about events related to the RTP stream identified.  In
>>       these cases the RTP stream is identified using the SSRC field
>>       value, and the information is provided to the higher layer
>>       associated with the "m=" line for the incoming RTP stream as
>>       identified by the current RTP stream to "m=" line table.  For this
>>       type of pattern it is common that the RTCP packets and information
>>       is repeated, either periodically (e.g.  SDES items), or for a
>>       duration (e.g.  BYE), thus suppression of repetitions can be
>>       considered.  Examples of these are:
>>
>>
>>
>>
>>
>> Name                      Expires July 31, 2017                 [Page 6]
>> 
>> Internet-Draft              Abbreviated-Title               January 2017
>>
>>
>>       Source Description (SDES) RTCP Packet:  The base RTP/RTCP protocol
>>          [RFC3550] defines SDES RTCP packets as a way of providing per
>>          source (SSRC) specific information about the source.  An SDES
>>          packet contains zero or more chunks, where a chunk contains the
>>          SSRC for the source being described by the one or more items
>>          included in the chunk.  Forward the SDES items in each chunk to
>>          the RTP stream's handler.
>>
>>       Goodbye (BYE) RTCP Packet:  This RTP/RTCP protocol [RFC3550]
>>          mechanism indicates that a particular RTP stream is leaving the
>>          RTP session.  Thus, a most relevant event to inform the handler
>>          for this RTP steam of.
>>
>>    Third Party Targeted Reports or Feedback:  There exist some
>>       multiparty RTP Topologies that results in that an endpoint
>>       receives third party reports (SR [RFC3550], RR [RFC3550] or XR
>>       [RFC3611]) as well as Feedback Messages [RFC4585] that relates to
>>       or targets an SSRC that originates from another endpoint, and
>>       where the source of the RTCP packet is also another endpoint.
>>       This type of packets should be forwarded to the higher layer
>>       function dealing with the third party reporting.  And if none
>>       exist then they can be suppressed.  As the third party handler can
>>       be focused on determining the conditions for the source of the
>>       reports and feedback, or focused on how the RTP stream source
>>       progresses, or both recommendations can't be made.
>>
>>    APP Packets:  The RTCP APP Packets [RFC3550] are a mechanism that
>>       enables experimentation.  The APP packet only specifies the source
>>       of the packet.  Thus this information can be related to any of the
>>       "m=" lines, thus deliver a copy of the packet to each "m=" line or
>>       an APP specific handler.
>>
>
>


-- 

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 nobody Tue Jan 31 01:24:39 2017
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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 of the IETF.

        Title           : Using Simulcast in SDP and RTP Sessions
        Authors         : Bo Burman
                          Magnus Westerlund
                          Suhas Nandakumar
                          Mo Zanaty
	Filename        : draft-ietf-mmusic-sdp-simulcast-07.txt
	Pages           : 35
	Date            : 2017-01-31

Abstract:
   In some application scenarios it may be desirable to send multiple
   differently encoded versions of the same media source in different
   RTP streams.  This is called simulcast.  This document describes how
   to accomplish simulcast in RTP and how to signal it in SDP.  The
   described solution uses an RTP/RTCP identification method to identify
   RTP streams belonging to the same media source, and makes an
   extension to SDP to relate those RTP streams as being different
   simulcast formats of that media source.  The SDP extension consists
   of a new media level SDP attribute that expresses capability to send
   and/or receive simulcast RTP streams.


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

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-mmusic-sdp-simulcast-07

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-sdp-simulcast-07


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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Hi WG,

This updated version of simulcast contains some few changes as can also 
be seen from the diff:
https://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-sdp-simulcast-07

The changes are:

- A scope clarification, as result of the discussion with Roni Even.

- A Reformulation of the Identification requirements for simulcast stream.

- Correcting the statement related to source specific signalling 
[RFC5576] to address Roni Even's comment.

- Update of the last paragraph in Section 6.2 regarding simulcast 
streams differences as well as forbidding multiple instances of the same 
SCID within a single a=simulcast line.

- Removal of note in Section 6.4 as result of issue raised by Roni Even.

- Use of "m=" has been changed to media description and a few other 
editorial improvements and clarifications.

With this Bo and I think this addresses all the issues raised on the WG 
last call on -05 version. Those who had issues, please review that your 
issue is resolved.

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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Subject: [MMUSIC] Stephen Farrell's Discuss on draft-ietf-mmusic-4572-update-12: (with DISCUSS)
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Stephen Farrell has entered the following ballot position for
draft-ietf-mmusic-4572-update-12: Discuss

When responding, please keep the subject line intact and reply to all
email addresses included in the To and CC lines. (Feel free to cut this
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The document, along with other ballot positions, can be found here:
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----------------------------------------------------------------------
DISCUSS:
----------------------------------------------------------------------


I've two (or 4 depending how you count:-) things
I'd like to check here. Should be pretty easy to
handle.

(1) section 5: I'm wondering if we have the right
set of hash functions here. Deprecating md2 and md5
is great, but I have a bunch of questions about the
others:

(1.1) why not also say that sha-1 MUST NOT be used
for new things (or similar)?

(1.2) do you really need sha-224 and 384? I think
nobody uses those at all.

(1.3) I'm a bit surprised you didn't add sha3 (and
maybe remove sha-512 if that's not needed) Even if
you don't encourage use of sha3, it might be good
to include it in the abnf now in case it gets
popular.

(2) Wouldn't it be a good plan to say that TLS
as-used MUST conform to BCP195? If not, why not?





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From: Jonathan Lennox <jonathan@vidyo.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: [MMUSIC] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
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Date: Tue, 31 Jan 2017 17:43:25 +0000
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Cc: "draft-ietf-rtcweb-jsep@tools.ietf.org" <draft-ietf-rtcweb-jsep@tools.ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>, mmusic <mmusic@ietf.org>, "draft-ietf-mmusic-sdp-bundle-negotiation@ietf.org" <draft-ietf-mmusic-sdp-bundle-negotiation@ietf.org>
Subject: Re: [MMUSIC] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
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In general, this looks good.

I have one question, however.  What should happen if an RTP packet (without=
 a MID) is received for a stream that has an existing mapping to an m=3D li=
ne, but whose PT is not valid for that m=3D line?

Should it
1. Cause the stream=92s mapping to be moved to another m=3D line, if the re=
ceived PT is in the payload type table for some other m=3D line?
or=20
2. Be an error?

In other words, should it be possible for PT-mapping to cause streams to ch=
ange their mappings?

(Should this decision depend on what caused the stream to be mapped to its =
current m=3D line?)

> On Jan 27, 2017, at 11:43 AM, Magnus Westerlund <magnus.westerlund@ericss=
on.com> wrote:
>=20
> MMUSIC and RTCWEB,
>=20
> Here is now a more complete text proposal that also considers the RTCP. I=
t also goes further than what Appendix B defines in one aspect, namely expl=
icitly considering third party RTCP reporting and what to do with it.
>=20
> Feedback is much appreciated. I plan to put this into a PR towards JSEP A=
PPENDIX B on monday. If only to get the JSEP authors attention ;-). However=
, the text is really intended for the next version of BUNDLE.
>=20
>=20
> X.  Associating RTP/RTCP With Correct SDP Media Description
>=20
>   As described in [RFC3550], RTP packets are associated with RTP
>   streams [RFC7656].  Each RTP stream is identified by an SSRC value,
>   and each RTP packet carries an SSRC value that is used to associate
>   the packet with the correct RTP stream.  RTCP packets also uses SSRCs
>   to identify on which RTP streams any report or feedback relate to.
>   Thus, an RTCP packet will commonly carry multiple SSRC values, and
>   might therefore be providing feedback or report on multiple RTP
>   streams.
>=20
>   In order to be able to process received RTP/RTCP packets correctly it
>   must be possible to associate an RTP stream with the correct "m=3D"
>   line, as the "m=3D" line and SDP attributes associated with the "m=3D"
>   line contain information needed to process the packets.
>=20
>   As all RTP streams associated with a BUNDLE group are part of the
>   same RTP session and using the same address:port combination for
>   sending and receiving RTP/RTCP packets, the local address:port
>   combination cannot be used to associate an RTP stream with the
>   correct "m=3D" line.  In addition, multiple RTP streams might be
>   associated with the same "m=3D" line.
>=20
>   Also, as described in Section 10.1.1, the same payload type value
>   might be used by multiple RTP streams, in which case the payload type
>   value cannot be used to associate an RTP stream with the correct "m=3D"
>   line.  However, there are cases where each "m=3D" line has unique
>   payload type values, and then the payload type could serve as
>   indicator to the relevant "m=3D" line the RTP stream is associated
>   with.
>=20
>   An offerer and answerer can inform each other which SSRC values they
>   will use for an RTP stream by using the SDP 'ssrc' attribute
>   [RFC5576].  However, an offerer will not know which SSRC values the
>   answerer will use until the offerer has received the answer providing
>=20
>=20
>=20
> Name                      Expires July 31, 2017                 [Page 2]
>=20
> Internet-Draft              Abbreviated-Title               January 2017
>=20
>=20
>   that information.  Due to this, before the offerer has received the
>   answer, the offerer will not be able to associate an RTP stream with
>   the correct "m=3D" line using the SSRC value associated with the RTP
>   stream.  In addition, the offerer and answerer may start using new
>   SSRC values mid-session, without informing each other using the SDP
>   'ssrc' attribute.
>=20
>   In order for an offerer and answerer to always be able to associate
>   an RTP stream with the correct "m=3D" line, the offerer and answerer
>   using the BUNDLE extension MUST support the mechanism defined in
>   Section 14, where the offerer and answerer includes the
>   identification-tag (provided by the remote peer) associated with an
>   "m=3D" line in the RTP Streams and in RTCP SDES packets part of a
>   BUNDLE group.
>=20
>   The mapping from an SSRC to an identification-tag is carried in RTCP
>   SDES packets or in RTP header extensions (Section 14).  Since a
>   compound RTCP packet can contain multiple RTCP SDES packets, and each
>   RTCP SDES packet can contain multiple chunks, an RTCP packet can
>   contain several SSRC to identification-tag mappings.  The offerer and
>   answerer maintain tables mapping RTP streams identified by SSRC to
>   "m=3D" lines identified by the identification-tag.  These tables are
>   updated each time new information that affects how packets should be
>   processed and routed are received.
>=20
>   To prepare for demultiplexing RTP streams to the correct "m=3D" line,
>   the following steps MUST be followed for each BUNDLE group based on
>   the SDP signalling information.
>=20
>      Construct a table mapping MID to "m=3D" line for each "m=3D" line in
>      this BUNDLE group.  Note that an "m=3D" line may only have one MID.
>=20
>      Construct a table mapping incoming RTP streams (SSRCs) to their
>      "m=3D" line for each "m=3D" line in this BUNDLE group and for each R=
TP
>      stream explicitly signalled for receiving in that "m=3D" line.
>=20
>      Construct a table mapping payload types to "m=3D" line for each "m=
=3D"
>      line in the BUNDLE group and for each payload type configured for
>      receiving in that "m=3D" line.  If any payload type is configured
>      for receiving in more than one "m=3D" line in the BUNDLE group, do
>      not it include it in the table.
>=20
>   Note that for each of these tables, there can only be one mapping for
>   any given key (MID, SSRC, or PT).  In other words, the tables are not
>   multimaps.
>=20
>   As "m=3D" lines are added or removed from the BUNDLE groups, or their
>   configurations are changed, the tables above MUST also be updated.
>=20
>=20
>=20
> Name                      Expires July 31, 2017                 [Page 3]
>=20
> Internet-Draft              Abbreviated-Title               January 2017
>=20
>=20
>   Received RTP packets that are syntactically correct are processed by
>   the RTP/RTCP protocol implementation for statistics etc.  After this
>   processing they need to be routed to the higher layer context
>   associated with the "m=3D" line within the BUNDLE group.  Somewhere in
>   the process where an received RTP packet is processed to be delivered
>   to the higher layer by RTP the matching step below MUST be performed:
>=20
>   1.  Reception of an RTP packet for an RTP stream that has an existing
>       mapping in the RTP stream to m=3D line table.  Before proceeding in
>       delivering the packet to the higher layer context according to
>       the RTP stream to "m=3D" line mapping table the following checks
>       MUST be performed:
>=20
>       A.  If the packet carries an RTP header extension with a SDES MID
>           value that is not in the table mapping MID to "m=3D" line, then
>           do not deliver the RTP packet to higher layers.
>=20
>       B.  If the packet carries an RTP header extension with a SDES MID
>           value that is in the table mapping MID to "m=3D" line, and the
>           value indicates a different "m=3D" line than the current RTP
>           stream to "m=3D" line mapping table, then update the RTP stream
>           to "m=3D" line mapping.
>=20
>   2.  Reception of an RTP packet for an RTP stream that has no existing
>       mapping to an m=3D line.  In this case the following actions MUST
>       be performed:
>=20
>       A.  If the packet carries an RTP header extension with a SDES MID
>           value that is in the table mapping MID to "m=3D" line, then
>           create an entry in the RTP stream to "m=3D" line mapping table
>           for this RTP stream (SSRC).  Then deliver the RTP packet to
>           the "m=3D" line context of the created mapping and stop.
>=20
>       B.  If the packet carries a Payload Type that is in the payload
>           type table, then create an entry in the RTP stream to "m=3D"
>           line mapping table for this RTP stream (SSRC).  Then deliver
>           the RTP packet to the "m=3D" line context of the created
>           mapping and stop.
>=20
>       C.  Otherwise do not deliver the RTP packet to higher layers.
>           Note, this includes unknown MID values.
>=20
>   For each RTCP packet received (including each RTCP packet that is
>   part of a compound RTCP packet), the RTCP packet needs to be
>   processed by the RTP/RTCP implementation and relevant information and
>   data from the RTCP packets needs to be routed to the appropriate
>   handler for the related RTP streams.  The appropriate handler is
>   determined by using the RTP stream to "m=3D" line mapping table.
>=20
>=20
>=20
> Name                      Expires July 31, 2017                 [Page 4]
>=20
> Internet-Draft              Abbreviated-Title               January 2017
>=20
>=20
>   On reception of any compound RTCP packet prior to dispatching the
>   received information and data, if there is an RTCP SDES packet
>   included that SHOULD be processed first.  If that SDES packet
>   contains SDES MID entries, this can results in updates and additions
>   to the RTP stream to "m=3D" line mapping table.  Thus each of the SDES
>   MID items are processed and the current table entries are checked if
>   the corresponding MID value matches the current RTP stream to "m=3D"
>   line mapping, else the entry is updated.  If there is no RTP stream
>   to "m=3D" line table mapping entry for the received SDES item's SSRC,
>   such an entry is created.  Note, that in the process of updating the
>   table entries, update flap suppression as discussed in Section 4.2.6
>   of [RFC7941] should be considered.
>=20
>   The various different RTCP packets as well as their various sub
>   parts, such as the various RTCP Feedback message types, relates to
>   the RTP streams in a couple of different ways.  The currently known
>   patterns are the following:
>=20
>   Reports on outgoing RTP streams:  For all RTP streams that this
>      endpoint is the source of, it can expect to receive report blocks
>      of several types identified as relating to an outgoing stream.
>      The basic pattern for these blocks are that the RTCP packet header
>      identifies the source of the reports, as identified by an SSRC,
>      and containing one or more report blocks, where each report block
>      identifies the RTP stream, using the SSRC, the report relates to.
>      For this pattern the relevant report information is provided to
>      the higher layer associated with the "m=3D" line the RTP Stream to
>      "m=3D" line table identifies.  The source SSRC as identifier of the
>      endpoint that the report originates are relevant for interpreting
>      the information, but not necessarily for routing.  Example of this
>      is pattern are:
>=20
>      Sender Report (SR) and Receiver Report (RR)  The basic receiver
>         report blocks from RFC3550 start with the SSRC of the RTP
>         stream they report on.
>=20
>      Extended Reports (XR):  RFC3611 is a framework that enables a
>         large number of different reports.  However, a large number of
>         these report formats are reporting on specific RTP streams and
>         thus each individual report of these types contains a SSRC
>         field to identify the RTP stream.
>=20
>   RTCP Feedback Messages for outgoing RTP streams:  The RFC 4585 RTCP
>      feedback messages allow for a number of different type of feedback
>      messages.  However, the RTCP feedback message header contains the
>      SSRC identifying the source of the feedback messages as well as
>      the actual type of the feedback.  Some of the feedback messages
>      also uses the target SSRC field in the header to identify which
>=20
>=20
>=20
> Name                      Expires July 31, 2017                 [Page 5]
>=20
> Internet-Draft              Abbreviated-Title               January 2017
>=20
>=20
>      RTP stream the feedback is related to.  For these types all the
>      FCI entries, if multiple ones are forwarded to the identified
>      handler.  Examples of this pattern are:
>=20
>      Picture Loss Indication (PLI):  RFC 4585 (PT=3DPSFB, FMT=3D1).
>=20
>      Slice Loss Indication (SLI):  RFC 4585 (PT=3DPSFB, FMT=3D2).
>=20
>      Reference Picture Selection Indication (RPSI):  RFC 4585 (PT=3DPSFB,
>         FMT=3D3).
>=20
>      Generic NACK:  [RFC4585] (PT=3DRTPFB and FMT=3D1).
>=20
>      Other feedback messages includes the target SSRC in the Feedback
>      Control Information (FCI).  Here each FCI needs to be processed
>      and the SSRC field identified.  And the individual FCI combined
>      with the RTCP packet header context needs to be forwarded to the
>      identified handler.  Example of this pattern are:
>=20
>      Full Intra Request (FIR):  [RFC5104] (PT=3DPSFB, FMT=3D4).
>=20
>      Temporal-Spatial Trade-off Request (TSTR):  [RFC5104] (PT=3DPSFB,
>         FMT=3D5).
>=20
>      Temporal-Spatial Trade-off Notification (TSTN):  This [RFC5104]
>         defined message (PT=3DPSFB, FMT=3D5) is actually a notifciation i=
n
>         response to a TSTR this endpoint sent using the SSRC the FCI
>         identifies as source.
>=20
>      H.271 Video Back Channel Message (VBCM):  [RFC5104] (PT=3DPSFB,
>         FMT=3D7).
>=20
>      Layer Refresh Request (LRR):  [I-D.ietf-avtext-lrr] (PT=3DPSFB,
>         FMT=3DTBD).
>=20
>   Descriptive or Notifications for an incoming RTP stream:  There exist
>      some RTCP packet types that provides additional information or
>      notifies about events related to the RTP stream identified.  In
>      these cases the RTP stream is identified using the SSRC field
>      value, and the information is provided to the higher layer
>      associated with the "m=3D" line for the incoming RTP stream as
>      identified by the current RTP stream to "m=3D" line table.  For this
>      type of pattern it is common that the RTCP packets and information
>      is repeated, either periodically (e.g.  SDES items), or for a
>      duration (e.g.  BYE), thus suppression of repetitions can be
>      considered.  Examples of these are:
>=20
>=20
>=20
>=20
>=20
> Name                      Expires July 31, 2017                 [Page 6]
>=20
> Internet-Draft              Abbreviated-Title               January 2017
>=20
>=20
>      Source Description (SDES) RTCP Packet:  The base RTP/RTCP protocol
>         [RFC3550] defines SDES RTCP packets as a way of providing per
>         source (SSRC) specific information about the source.  An SDES
>         packet contains zero or more chunks, where a chunk contains the
>         SSRC for the source being described by the one or more items
>         included in the chunk.  Forward the SDES items in each chunk to
>         the RTP stream's handler.
>=20
>      Goodbye (BYE) RTCP Packet:  This RTP/RTCP protocol [RFC3550]
>         mechanism indicates that a particular RTP stream is leaving the
>         RTP session.  Thus, a most relevant event to inform the handler
>         for this RTP steam of.
>=20
>   Third Party Targeted Reports or Feedback:  There exist some
>      multiparty RTP Topologies that results in that an endpoint
>      receives third party reports (SR [RFC3550], RR [RFC3550] or XR
>      [RFC3611]) as well as Feedback Messages [RFC4585] that relates to
>      or targets an SSRC that originates from another endpoint, and
>      where the source of the RTCP packet is also another endpoint.
>      This type of packets should be forwarded to the higher layer
>      function dealing with the third party reporting.  And if none
>      exist then they can be suppressed.  As the third party handler can
>      be focused on determining the conditions for the source of the
>      reports and feedback, or focused on how the RTP stream source
>      progresses, or both recommendations can't be made.
>=20
>   APP Packets:  The RTCP APP Packets [RFC3550] are a mechanism that
>      enables experimentation.  The APP packet only specifies the source
>      of the packet.  Thus this information can be related to any of the
>      "m=3D" lines, thus deliver a copy of the packet to each "m=3D" line =
or
>      an APP specific handler.
>=20
> --=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>=20


From nobody Tue Jan 31 09:57:36 2017
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From: Jonathan Lennox <jonathan@vidyo.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: [MMUSIC] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
Thread-Index: AQHSe+hqNyVXjtU64EysMGTN3ByD2aFTQ7wA
Date: Tue, 31 Jan 2017 17:57:25 +0000
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References: <1f020dc6-1ac8-b71b-aee4-a711d15f1588@ericsson.com>
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Subject: Re: [MMUSIC] Text proposal for Bundle regarding Associating RTP/RTCP With Correct SDP Media Description
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And one more question.  This one has significant interop issues, I think.

Should decoding contexts be assumed to be associated with streams independe=
nt of m=3D lines, or with streams within m=3D lines?

Specifically, if a sender moves a video stream from one m=3D line to anothe=
r (maintaining its PT), must it send a fresh I-frame (keyframe) at the time=
 of the move, or can it continue sending inter frames?

> On Jan 27, 2017, at 11:43 AM, Magnus Westerlund <magnus.westerlund@ericss=
on.com> wrote:
>=20
> MMUSIC and RTCWEB,
>=20
> Here is now a more complete text proposal that also considers the RTCP. I=
t also goes further than what Appendix B defines in one aspect, namely expl=
icitly considering third party RTCP reporting and what to do with it.
>=20
> Feedback is much appreciated. I plan to put this into a PR towards JSEP A=
PPENDIX B on monday. If only to get the JSEP authors attention ;-). However=
, the text is really intended for the next version of BUNDLE.
>=20
>=20
> X.  Associating RTP/RTCP With Correct SDP Media Description
>=20
>   As described in [RFC3550], RTP packets are associated with RTP
>   streams [RFC7656].  Each RTP stream is identified by an SSRC value,
>   and each RTP packet carries an SSRC value that is used to associate
>   the packet with the correct RTP stream.  RTCP packets also uses SSRCs
>   to identify on which RTP streams any report or feedback relate to.
>   Thus, an RTCP packet will commonly carry multiple SSRC values, and
>   might therefore be providing feedback or report on multiple RTP
>   streams.
>=20
>   In order to be able to process received RTP/RTCP packets correctly it
>   must be possible to associate an RTP stream with the correct "m=3D"
>   line, as the "m=3D" line and SDP attributes associated with the "m=3D"
>   line contain information needed to process the packets.
>=20
>   As all RTP streams associated with a BUNDLE group are part of the
>   same RTP session and using the same address:port combination for
>   sending and receiving RTP/RTCP packets, the local address:port
>   combination cannot be used to associate an RTP stream with the
>   correct "m=3D" line.  In addition, multiple RTP streams might be
>   associated with the same "m=3D" line.
>=20
>   Also, as described in Section 10.1.1, the same payload type value
>   might be used by multiple RTP streams, in which case the payload type
>   value cannot be used to associate an RTP stream with the correct "m=3D"
>   line.  However, there are cases where each "m=3D" line has unique
>   payload type values, and then the payload type could serve as
>   indicator to the relevant "m=3D" line the RTP stream is associated
>   with.
>=20
>   An offerer and answerer can inform each other which SSRC values they
>   will use for an RTP stream by using the SDP 'ssrc' attribute
>   [RFC5576].  However, an offerer will not know which SSRC values the
>   answerer will use until the offerer has received the answer providing
>=20
>=20
>=20
> Name                      Expires July 31, 2017                 [Page 2]
>=20
> Internet-Draft              Abbreviated-Title               January 2017
>=20
>=20
>   that information.  Due to this, before the offerer has received the
>   answer, the offerer will not be able to associate an RTP stream with
>   the correct "m=3D" line using the SSRC value associated with the RTP
>   stream.  In addition, the offerer and answerer may start using new
>   SSRC values mid-session, without informing each other using the SDP
>   'ssrc' attribute.
>=20
>   In order for an offerer and answerer to always be able to associate
>   an RTP stream with the correct "m=3D" line, the offerer and answerer
>   using the BUNDLE extension MUST support the mechanism defined in
>   Section 14, where the offerer and answerer includes the
>   identification-tag (provided by the remote peer) associated with an
>   "m=3D" line in the RTP Streams and in RTCP SDES packets part of a
>   BUNDLE group.
>=20
>   The mapping from an SSRC to an identification-tag is carried in RTCP
>   SDES packets or in RTP header extensions (Section 14).  Since a
>   compound RTCP packet can contain multiple RTCP SDES packets, and each
>   RTCP SDES packet can contain multiple chunks, an RTCP packet can
>   contain several SSRC to identification-tag mappings.  The offerer and
>   answerer maintain tables mapping RTP streams identified by SSRC to
>   "m=3D" lines identified by the identification-tag.  These tables are
>   updated each time new information that affects how packets should be
>   processed and routed are received.
>=20
>   To prepare for demultiplexing RTP streams to the correct "m=3D" line,
>   the following steps MUST be followed for each BUNDLE group based on
>   the SDP signalling information.
>=20
>      Construct a table mapping MID to "m=3D" line for each "m=3D" line in
>      this BUNDLE group.  Note that an "m=3D" line may only have one MID.
>=20
>      Construct a table mapping incoming RTP streams (SSRCs) to their
>      "m=3D" line for each "m=3D" line in this BUNDLE group and for each R=
TP
>      stream explicitly signalled for receiving in that "m=3D" line.
>=20
>      Construct a table mapping payload types to "m=3D" line for each "m=
=3D"
>      line in the BUNDLE group and for each payload type configured for
>      receiving in that "m=3D" line.  If any payload type is configured
>      for receiving in more than one "m=3D" line in the BUNDLE group, do
>      not it include it in the table.
>=20
>   Note that for each of these tables, there can only be one mapping for
>   any given key (MID, SSRC, or PT).  In other words, the tables are not
>   multimaps.
>=20
>   As "m=3D" lines are added or removed from the BUNDLE groups, or their
>   configurations are changed, the tables above MUST also be updated.
>=20
>=20
>=20
> Name                      Expires July 31, 2017                 [Page 3]
>=20
> Internet-Draft              Abbreviated-Title               January 2017
>=20
>=20
>   Received RTP packets that are syntactically correct are processed by
>   the RTP/RTCP protocol implementation for statistics etc.  After this
>   processing they need to be routed to the higher layer context
>   associated with the "m=3D" line within the BUNDLE group.  Somewhere in
>   the process where an received RTP packet is processed to be delivered
>   to the higher layer by RTP the matching step below MUST be performed:
>=20
>   1.  Reception of an RTP packet for an RTP stream that has an existing
>       mapping in the RTP stream to m=3D line table.  Before proceeding in
>       delivering the packet to the higher layer context according to
>       the RTP stream to "m=3D" line mapping table the following checks
>       MUST be performed:
>=20
>       A.  If the packet carries an RTP header extension with a SDES MID
>           value that is not in the table mapping MID to "m=3D" line, then
>           do not deliver the RTP packet to higher layers.
>=20
>       B.  If the packet carries an RTP header extension with a SDES MID
>           value that is in the table mapping MID to "m=3D" line, and the
>           value indicates a different "m=3D" line than the current RTP
>           stream to "m=3D" line mapping table, then update the RTP stream
>           to "m=3D" line mapping.
>=20
>   2.  Reception of an RTP packet for an RTP stream that has no existing
>       mapping to an m=3D line.  In this case the following actions MUST
>       be performed:
>=20
>       A.  If the packet carries an RTP header extension with a SDES MID
>           value that is in the table mapping MID to "m=3D" line, then
>           create an entry in the RTP stream to "m=3D" line mapping table
>           for this RTP stream (SSRC).  Then deliver the RTP packet to
>           the "m=3D" line context of the created mapping and stop.
>=20
>       B.  If the packet carries a Payload Type that is in the payload
>           type table, then create an entry in the RTP stream to "m=3D"
>           line mapping table for this RTP stream (SSRC).  Then deliver
>           the RTP packet to the "m=3D" line context of the created
>           mapping and stop.
>=20
>       C.  Otherwise do not deliver the RTP packet to higher layers.
>           Note, this includes unknown MID values.
>=20
>   For each RTCP packet received (including each RTCP packet that is
>   part of a compound RTCP packet), the RTCP packet needs to be
>   processed by the RTP/RTCP implementation and relevant information and
>   data from the RTCP packets needs to be routed to the appropriate
>   handler for the related RTP streams.  The appropriate handler is
>   determined by using the RTP stream to "m=3D" line mapping table.
>=20
>=20
>=20
> Name                      Expires July 31, 2017                 [Page 4]
>=20
> Internet-Draft              Abbreviated-Title               January 2017
>=20
>=20
>   On reception of any compound RTCP packet prior to dispatching the
>   received information and data, if there is an RTCP SDES packet
>   included that SHOULD be processed first.  If that SDES packet
>   contains SDES MID entries, this can results in updates and additions
>   to the RTP stream to "m=3D" line mapping table.  Thus each of the SDES
>   MID items are processed and the current table entries are checked if
>   the corresponding MID value matches the current RTP stream to "m=3D"
>   line mapping, else the entry is updated.  If there is no RTP stream
>   to "m=3D" line table mapping entry for the received SDES item's SSRC,
>   such an entry is created.  Note, that in the process of updating the
>   table entries, update flap suppression as discussed in Section 4.2.6
>   of [RFC7941] should be considered.
>=20
>   The various different RTCP packets as well as their various sub
>   parts, such as the various RTCP Feedback message types, relates to
>   the RTP streams in a couple of different ways.  The currently known
>   patterns are the following:
>=20
>   Reports on outgoing RTP streams:  For all RTP streams that this
>      endpoint is the source of, it can expect to receive report blocks
>      of several types identified as relating to an outgoing stream.
>      The basic pattern for these blocks are that the RTCP packet header
>      identifies the source of the reports, as identified by an SSRC,
>      and containing one or more report blocks, where each report block
>      identifies the RTP stream, using the SSRC, the report relates to.
>      For this pattern the relevant report information is provided to
>      the higher layer associated with the "m=3D" line the RTP Stream to
>      "m=3D" line table identifies.  The source SSRC as identifier of the
>      endpoint that the report originates are relevant for interpreting
>      the information, but not necessarily for routing.  Example of this
>      is pattern are:
>=20
>      Sender Report (SR) and Receiver Report (RR)  The basic receiver
>         report blocks from RFC3550 start with the SSRC of the RTP
>         stream they report on.
>=20
>      Extended Reports (XR):  RFC3611 is a framework that enables a
>         large number of different reports.  However, a large number of
>         these report formats are reporting on specific RTP streams and
>         thus each individual report of these types contains a SSRC
>         field to identify the RTP stream.
>=20
>   RTCP Feedback Messages for outgoing RTP streams:  The RFC 4585 RTCP
>      feedback messages allow for a number of different type of feedback
>      messages.  However, the RTCP feedback message header contains the
>      SSRC identifying the source of the feedback messages as well as
>      the actual type of the feedback.  Some of the feedback messages
>      also uses the target SSRC field in the header to identify which
>=20
>=20
>=20
> Name                      Expires July 31, 2017                 [Page 5]
>=20
> Internet-Draft              Abbreviated-Title               January 2017
>=20
>=20
>      RTP stream the feedback is related to.  For these types all the
>      FCI entries, if multiple ones are forwarded to the identified
>      handler.  Examples of this pattern are:
>=20
>      Picture Loss Indication (PLI):  RFC 4585 (PT=3DPSFB, FMT=3D1).
>=20
>      Slice Loss Indication (SLI):  RFC 4585 (PT=3DPSFB, FMT=3D2).
>=20
>      Reference Picture Selection Indication (RPSI):  RFC 4585 (PT=3DPSFB,
>         FMT=3D3).
>=20
>      Generic NACK:  [RFC4585] (PT=3DRTPFB and FMT=3D1).
>=20
>      Other feedback messages includes the target SSRC in the Feedback
>      Control Information (FCI).  Here each FCI needs to be processed
>      and the SSRC field identified.  And the individual FCI combined
>      with the RTCP packet header context needs to be forwarded to the
>      identified handler.  Example of this pattern are:
>=20
>      Full Intra Request (FIR):  [RFC5104] (PT=3DPSFB, FMT=3D4).
>=20
>      Temporal-Spatial Trade-off Request (TSTR):  [RFC5104] (PT=3DPSFB,
>         FMT=3D5).
>=20
>      Temporal-Spatial Trade-off Notification (TSTN):  This [RFC5104]
>         defined message (PT=3DPSFB, FMT=3D5) is actually a notifciation i=
n
>         response to a TSTR this endpoint sent using the SSRC the FCI
>         identifies as source.
>=20
>      H.271 Video Back Channel Message (VBCM):  [RFC5104] (PT=3DPSFB,
>         FMT=3D7).
>=20
>      Layer Refresh Request (LRR):  [I-D.ietf-avtext-lrr] (PT=3DPSFB,
>         FMT=3DTBD).
>=20
>   Descriptive or Notifications for an incoming RTP stream:  There exist
>      some RTCP packet types that provides additional information or
>      notifies about events related to the RTP stream identified.  In
>      these cases the RTP stream is identified using the SSRC field
>      value, and the information is provided to the higher layer
>      associated with the "m=3D" line for the incoming RTP stream as
>      identified by the current RTP stream to "m=3D" line table.  For this
>      type of pattern it is common that the RTCP packets and information
>      is repeated, either periodically (e.g.  SDES items), or for a
>      duration (e.g.  BYE), thus suppression of repetitions can be
>      considered.  Examples of these are:
>=20
>=20
>=20
>=20
>=20
> Name                      Expires July 31, 2017                 [Page 6]
>=20
> Internet-Draft              Abbreviated-Title               January 2017
>=20
>=20
>      Source Description (SDES) RTCP Packet:  The base RTP/RTCP protocol
>         [RFC3550] defines SDES RTCP packets as a way of providing per
>         source (SSRC) specific information about the source.  An SDES
>         packet contains zero or more chunks, where a chunk contains the
>         SSRC for the source being described by the one or more items
>         included in the chunk.  Forward the SDES items in each chunk to
>         the RTP stream's handler.
>=20
>      Goodbye (BYE) RTCP Packet:  This RTP/RTCP protocol [RFC3550]
>         mechanism indicates that a particular RTP stream is leaving the
>         RTP session.  Thus, a most relevant event to inform the handler
>         for this RTP steam of.
>=20
>   Third Party Targeted Reports or Feedback:  There exist some
>      multiparty RTP Topologies that results in that an endpoint
>      receives third party reports (SR [RFC3550], RR [RFC3550] or XR
>      [RFC3611]) as well as Feedback Messages [RFC4585] that relates to
>      or targets an SSRC that originates from another endpoint, and
>      where the source of the RTCP packet is also another endpoint.
>      This type of packets should be forwarded to the higher layer
>      function dealing with the third party reporting.  And if none
>      exist then they can be suppressed.  As the third party handler can
>      be focused on determining the conditions for the source of the
>      reports and feedback, or focused on how the RTP stream source
>      progresses, or both recommendations can't be made.
>=20
>   APP Packets:  The RTCP APP Packets [RFC3550] are a mechanism that
>      enables experimentation.  The APP packet only specifies the source
>      of the packet.  Thus this information can be related to any of the
>      "m=3D" lines, thus deliver a copy of the packet to each "m=3D" line =
or
>      an APP specific handler.
>=20
> --=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>=20


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From: Martin Thomson <martin.thomson@gmail.com>
Date: Wed, 1 Feb 2017 09:59:03 +1100
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To: Jonathan Lennox <jonathan@vidyo.com>
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Cc: "mmusic-chairs@ietf.org" <mmusic-chairs@ietf.org>, mmusic <mmusic@ietf.org>, Flemming Andreasen <fandreas@cisco.com>, The The IESG <iesg@ietf.org>, "draft-ietf-mmusic-4572-update@ietf.org" <draft-ietf-mmusic-4572-update@ietf.org>, Stephen Farrell <stephen.farrell@cs.tcd.ie>
Subject: Re: [MMUSIC] Stephen Farrell's Discuss on draft-ietf-mmusic-4572-update-12: (with DISCUSS)
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On 1 February 2017 at 04:23, Jonathan Lennox <jonathan@vidyo.com> wrote:
>> (2) Wouldn't it be a good plan to say that TLS
>> as-used MUST conform to BCP195? If not, why not?
>
> My inclination would be to say yes, but I=E2=80=99ll let people with more=
 of the existing implementations comment further in case there=E2=80=99s so=
me issue.

If we said MUST, I suspect that we'd end up making a whole lot of
implementations non-compliant.  There's a lot of DTLS 1.0 out there.
That said, the same applies to SHA-1.

I guess it depends on your stance regarding what is as opposed to what
should be.

FWIW, we didn't ask this sort of question because the intent was to
clarify hash usage only, avoiding touching the rest of the document.


From nobody Tue Jan 31 23:19:26 2017
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From: Eric Rescorla <ekr@rtfm.com>
Date: Wed, 1 Feb 2017 08:18:41 +0100
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To: Stephen Farrell <stephen.farrell@cs.tcd.ie>
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Cc: Flemming Andreasen <fandreas@cisco.com>, mmusic-chairs@ietf.org, draft-ietf-mmusic-4572-update@ietf.org, The IESG <iesg@ietf.org>, mmusic WG <mmusic@ietf.org>
Subject: Re: [MMUSIC] Stephen Farrell's Discuss on draft-ietf-mmusic-4572-update-12: (with DISCUSS)
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On Tue, Jan 31, 2017 at 5:08 PM, Stephen Farrell <stephen.farrell@cs.tcd.ie>
wrote:

> Stephen Farrell has entered the following ballot position for
> draft-ietf-mmusic-4572-update-12: Discuss
>
> When responding, please keep the subject line intact and reply to all
> email addresses included in the To and CC lines. (Feel free to cut this
> introductory paragraph, however.)
>
>
> Please refer to https://www.ietf.org/iesg/statement/discuss-criteria.html
> for more information about IESG DISCUSS and COMMENT positions.
>
>
> The document, along with other ballot positions, can be found here:
> https://datatracker.ietf.org/doc/draft-ietf-mmusic-4572-update/
>
>
>
> ----------------------------------------------------------------------
> DISCUSS:
> ----------------------------------------------------------------------
>
>
> I've two (or 4 depending how you count:-) things
> I'd like to check here. Should be pretty easy to
> handle.
>
> (1) section 5: I'm wondering if we have the right
> set of hash functions here. Deprecating md2 and md5
> is great, but I have a bunch of questions about the
> others:
>
> (1.1) why not also say that sha-1 MUST NOT be used
> for new things (or similar)?
>
> (1.2) do you really need sha-224 and 384? I think
> nobody uses those at all.
>

It's certainly not correct that nobody uses SHA-384.

In fact, for TLS 1.3, you can't sign anything with P-384 without using
SHA-384.

-Ekr


> (1.3) I'm a bit surprised you didn't add sha3 (and
> maybe remove sha-512 if that's not needed) Even if
> you don't encourage use of sha3, it might be good
> to include it in the abnf now in case it gets
> popular.
>
> (2) Wouldn't it be a good plan to say that TLS
> as-used MUST conform to BCP195? If not, why not?
>
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>

--94eb2c1292922284f9054772de37
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Tue, Jan 31, 2017 at 5:08 PM, Stephen Farrell <span dir=3D"ltr">&lt;=
<a href=3D"mailto:stephen.farrell@cs.tcd.ie" target=3D"_blank">stephen.farr=
ell@cs.tcd.ie</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" st=
yle=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Steph=
en Farrell has entered the following ballot position for<br>
draft-ietf-mmusic-4572-update-<wbr>12: Discuss<br>
<br>
When responding, please keep the subject line intact and reply to all<br>
email addresses included in the To and CC lines. (Feel free to cut this<br>
introductory paragraph, however.)<br>
<br>
<br>
Please refer to <a href=3D"https://www.ietf.org/iesg/statement/discuss-crit=
eria.html" rel=3D"noreferrer" target=3D"_blank">https://www.ietf.org/iesg/<=
wbr>statement/discuss-criteria.<wbr>html</a><br>
for more information about IESG DISCUSS and COMMENT positions.<br>
<br>
<br>
The document, along with other ballot positions, can be found here:<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-mmusic-4572-update/"=
 rel=3D"noreferrer" target=3D"_blank">https://datatracker.ietf.org/<wbr>doc=
/draft-ietf-mmusic-4572-<wbr>update/</a><br>
<br>
<br>
<br>
------------------------------<wbr>------------------------------<wbr>-----=
-----<br>
DISCUSS:<br>
------------------------------<wbr>------------------------------<wbr>-----=
-----<br>
<br>
<br>
I&#39;ve two (or 4 depending how you count:-) things<br>
I&#39;d like to check here. Should be pretty easy to<br>
handle.<br>
<br>
(1) section 5: I&#39;m wondering if we have the right<br>
set of hash functions here. Deprecating md2 and md5<br>
is great, but I have a bunch of questions about the<br>
others:<br>
<br>
(1.1) why not also say that sha-1 MUST NOT be used<br>
for new things (or similar)?<br>
<br>
(1.2) do you really need sha-224 and 384? I think<br>
nobody uses those at all.<br></blockquote><div><br></div><div>It&#39;s cert=
ainly not correct that nobody uses SHA-384.</div><div><br></div><div>In fac=
t, for TLS 1.3, you can&#39;t sign anything with P-384 without using SHA-38=
4.</div><div><br></div><div>-Ekr</div><div><br></div><blockquote class=3D"g=
mail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-l=
eft:1ex">
<br>
(1.3) I&#39;m a bit surprised you didn&#39;t add sha3 (and<br>
maybe remove sha-512 if that&#39;s not needed) Even if<br>
you don&#39;t encourage use of sha3, it might be good<br>
to include it in the abnf now in case it gets<br>
popular.<br>
<br>
(2) Wouldn&#39;t it be a good plan to say that TLS<br>
as-used MUST conform to BCP195? If not, why not?<br>
<br>
<br>
<br>
<br>
______________________________<wbr>_________________<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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/mmusic</a><br=
>
</blockquote></div><br></div></div>

--94eb2c1292922284f9054772de37--

