
From randell-ietf@jesup.org  Fri Jul  1 20:52:26 2011
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Date: Fri, 01 Jul 2011 23:49:16 -0400
From: Randell Jesup <randell-ietf@jesup.org>
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Subject: Re: [MMUSIC] Revised ice-updated-offer draft
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On 6/30/2011 3:51 AM, Elwell, John wrote:
> I received no comments on -01. To prepare for a meaningful discussion in Quebec, it would be good to start that discussion now, on the list. In particular, the Conclusions section ends with the statement:
> " According to discussions in section Section 4, it seems to be the
>     case that the updated offer is needed, in practice, in very few
>     environments, and therefore consideration should be given to relaxing
>     the requirement in [RFC5245]."
> What would be people's reactions to updating RFC 5245 to relax this requirement?
>
> John
>

One comment on the draft:

      Switching codec or switching from audio to image requires an SDP offer-answer cycle.

Maybe, yes, but switching codecs from compressed to uncompressed may or may not
require a re-invite:

I.e. with this re-invite isn't needed for the codec change:

m=audio 3456 RTP/AVP 18 0
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000

Of course, if you don't specify both (why not?), that won't work.  (If you support both
you should specify both and probably accept both, so this *should* be the norm for this usecase)

Switching to image could be a re-invite to add m=image or to make it active (and make m=audio
inactive or turn off audio via m=audio 0 RTP/AVP 0).  In theory you could have both "active"
but only send data on one, though that wouldn't be normal I believe.


-- 
Randell Jesup
randell-ietf@jesup.org


From Martin.Stiemerling@neclab.eu  Mon Jul  4 02:09:03 2011
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From: Martin Stiemerling <Martin.Stiemerling@neclab.eu>
To: "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: I-D Action: draft-schmidt-mmusic-sdp-local-identifier-01.txt
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Subject: [MMUSIC] FW: I-D Action: draft-schmidt-mmusic-sdp-local-identifier-01.txt
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Dear all,

The draft below describes how dynamic Policy Controllers can identify media=
 receivers which are located behind a NAT/home gateway for the resource and=
 admission process.=20

We are looking forward to your comments

  Martin=20

> -----Original Message-----
> From: i-d-announce-bounces@ietf.org [mailto:i-d-announce-
> bounces@ietf.org] On Behalf Of internet-drafts@ietf.org
> Sent: Wednesday, June 01, 2011 3:38 PM
> To: i-d-announce@ietf.org
> Subject: I-D Action: draft-schmidt-mmusic-sdp-local-identifier-01.txt
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts direct=
ories.
>=20
> 	Title           : SDP Local Identifier
> 	Author(s)       : Mischa Schmidt
>                           Hans-Joerg Kolbe
>                           Martin Stiemerling
> 	Filename        : draft-schmidt-mmusic-sdp-local-identifier-01.txt
> 	Pages           : 12
> 	Date            : 2011-06-01
>=20
>    Dynamic Policy Controllers like resource and admission control
>    systems need to be able to locate media receivers in each segment of
>    the network to determine the path of the media flows prior to
>    allowing or rejecting the session setup.  This is fine as long as
>    this is handled within an operator&#39;s network but limited when
>    extending this to home network.  Current definitions of SDP are not
>    clear about conveying a local identifier (e.g., a private IP address)
>    of the client that is destined to receive media streams after
>    successful SDP negotiation.  In this draft we outline the need for
>    carrying such information over SDP and discuss a set of possible
>    solutions in a scenario with a home network and a operator network
>    domain.
>=20
>=20
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-schmidt-mmusic-sdp-local-
> identifier-01.txt
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> This Internet-Draft can be retrieved at:
> ftp://ftp.ietf.org/internet-drafts/draft-schmidt-mmusic-sdp-local-identif=
ier-
> 01.txt
> _______________________________________________
> I-D-Announce mailing list
> I-D-Announce@ietf.org
> https://www.ietf.org/mailman/listinfo/i-d-announce
> Internet-Draft directories: http://www.ietf.org/shadow.html or
> ftp://ftp.ietf.org/ietf/1shadow-sites.txt

From ari.keranen@nomadiclab.com  Mon Jul  4 09:24:21 2011
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Subject: [MMUSIC] ICE TCP ice-options?
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Hi all,

I have now updated the ICE TCP draft [1] according to the WGLC comments, =
but noticed one more issue worth considering:

Should we have ice-options tag for ICE TCP?

The ICE RFC says [2]:

   Each extension
   or change to ICE is associated with a token.  When an agent
   supporting such an extension or change generates an offer or an
   answer, it MUST include the token for that extension in this
   attribute.  This allows each side to know what the other side is
   doing.=20

However, at least my impression is that TCP candidates are not an =
extension in the sense described in the RFC. TCP candidates were even =
originally part of the basic ICE. Also the RFC says [3]:

   If its peer is using ICE options (present in
   an ice-options attribute from the peer) that the agent does not
   understand, the agent MUST use a regular nomination algorithm. =20

So, having "tcp" ice-option would unnecessarily result in forcing =
regular nomination even if not supporting the TCP candidates (or some =
other transport protocol) would simply result in not pairing them with =
any local candidate.

Hence, I think we do not need "tcp" options tag (and can keep the draft =
as it is). Opinions?


Cheers,
Ari

[1] =
http://users.piuha.net/akeranen/drafts/draft-ietf-mmusic-ice-tcp-14.txt
[2] http://tools.ietf.org/html/rfc5245#section-14
[3] http://tools.ietf.org/html/rfc5245#section-8.1.1=

From john.elwell@siemens-enterprise.com  Mon Jul  4 09:33:15 2011
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Randell Jesup <randell-ietf@jesup.org>, "mmusic@ietf.org" <mmusic@ietf.org>
Date: Mon, 4 Jul 2011 18:33:10 +0200
Thread-Topic: [MMUSIC] Revised ice-updated-offer draft
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Randell,

Yes, you are correct that if you specify two codecs to start with, as trans=
mitter you can switch at will. However, if you want the peer endpoint also =
to switch the codec it transmits, would you not need to do an updated offer=
? So endpoint A detects fax and switches from G.729 to G.711, but wants end=
point B (which may not be in a position to detect the fax) also to switch f=
rom G.729 to G.711.

Anyway, certainly the draft could do with some clarification in this area.

John
=20

> -----Original Message-----
> From: mmusic-bounces@ietf.org=20
> [mailto:mmusic-bounces@ietf.org] On Behalf Of Randell Jesup
> Sent: 02 July 2011 04:49
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] Revised ice-updated-offer draft
>=20
> On 6/30/2011 3:51 AM, Elwell, John wrote:
> > I received no comments on -01. To prepare for a meaningful=20
> discussion in Quebec, it would be good to start that=20
> discussion now, on the list. In particular, the Conclusions=20
> section ends with the statement:
> > " According to discussions in section Section 4, it seems to be the
> >     case that the updated offer is needed, in practice, in very few
> >     environments, and therefore consideration should be=20
> given to relaxing
> >     the requirement in [RFC5245]."
> > What would be people's reactions to updating RFC 5245 to=20
> relax this requirement?
> >
> > John
> >
>=20
> One comment on the draft:
>=20
>       Switching codec or switching from audio to image=20
> requires an SDP offer-answer cycle.
>=20
> Maybe, yes, but switching codecs from compressed to=20
> uncompressed may or may not
> require a re-invite:
>=20
> I.e. with this re-invite isn't needed for the codec change:
>=20
> m=3Daudio 3456 RTP/AVP 18 0
> a=3Drtpmap:18 G729/8000
> a=3Dfmtp:18 annexb=3Dno
> a=3Drtpmap:0 PCMU/8000
>=20
> Of course, if you don't specify both (why not?), that won't=20
> work.  (If you support both
> you should specify both and probably accept both, so this=20
> *should* be the norm for this usecase)
>=20
> Switching to image could be a re-invite to add m=3Dimage or to=20
> make it active (and make m=3Daudio
> inactive or turn off audio via m=3Daudio 0 RTP/AVP 0).  In=20
> theory you could have both "active"
> but only send data on one, though that wouldn't be normal I believe.
>=20
>=20
> --=20
> Randell Jesup
> randell-ietf@jesup.org
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
> =

From randell-ietf@jesup.org  Mon Jul  4 20:27:20 2011
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Subject: Re: [MMUSIC] Revised ice-updated-offer draft
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On 7/4/2011 12:33 PM, Elwell, John wrote:
> Randell,
>
> Yes, you are correct that if you specify two codecs to start with, as transmitter you can switch at will. However, if you want the peer endpoint also to switch the codec it transmits, would you not need to do an updated offer? So endpoint A detects fax and switches from G.729 to G.711, but wants endpoint B (which may not be in a position to detect the fax) also to switch from G.729 to G.711.

Ok, it's true that's undefined (though many endpoints will try to or 
must send the same codec they receive, and so would switch if you 
switch).  The only way to force a codec is to offer only 1 - and that 
causes problems if the other side doesn't support it, plus it makes 
switching impossible or only by reinvite.

That's a more general problem, however.  If you assume a single codec or 
the need to force the other side to switch, then yes, a reinvite is needed.

-- 
Randell Jesup
randell-ietf@jesup.org


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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-00.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the Multiparty Multimedia Session Control=
 Working Group of the IETF.

	Title           : Stream Control Transmission Protocol (SCTP)-Based Media =
Transport in the Session Description Protocol (SDP)
	Author(s)       : Salvatore Loreto
                          Gonzalo Camarillo
	Filename        : draft-ietf-mmusic-sctp-sdp-00.txt
	Pages           : 10
	Date            : 2011-07-04

   SCTP (Stream Control Transmission Protocol) is a transport protocol
   used to establish associations between two endpoints.  This document
   describes how to express media transport over SCTP in SDP (Session
   Description Protocol).  This document defines the &#39;SCTP&#39; and &#3=
9;SCTP/
   DTLS&#39; protocol identifiers for SDP.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sctp-sdp-00.txt

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

This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-mmusic-sctp-sdp-00.txt

From fandreas@cisco.com  Tue Jul  5 16:59:49 2011
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Cc: draft-ietf-mmusic-media-loopback@tools.ietf.org
Subject: [MMUSIC] Media loopback draft and General Need for ICE alternative ?
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Hi MMUSIC

We haven't seen much list discussion on this one. It would be really 
useful if the WG participants could provide some feedback on the media 
loopback points raised below so we can determine the best way to proceed 
here. In particular, we would like to know if the WG believes that the 
alternative NAT traversal mechanism (latching) proposed in the draft is 
generally applicable outside of the media loopback use case, and/or if 
we in general need an alternative to ICE for NAT traversal.

Please send your comments to the list.

Thanks

-- Flemming
(as MMUSIC co-chair)




-------- Original Message --------
Subject: 	RE: [MMUSIC] Media loopback 
revisited(draft-ietf-mmusic-media-loopback-15.txt)
Date: 	Wed, 1 Jun 2011 07:02:45 -0500
From: 	Kaynam Hedayat <kaynam.hedayat@exfo.com>
To: 	Flemming Andreasen (fandreas) <fandreas@cisco.com>, 
<draft-ietf-mmusic-media-loopback@tools.ietf.org>
CC: 	mmusic <mmusic@ietf.org>



Flemming,

Thank you for reviewing the new version of the draft. Some comments from 
the authors below.

Regards,

Kaynam

*From:* mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] *On 
Behalf Of *Flemming Andreasen
*Sent:* Thursday, May 12, 2011 8:06 PM
*To:* draft-ietf-mmusic-media-loopback@tools.ietf.org
*Cc:* mmusic
*Subject:* [MMUSIC] Media loopback 
revisited(draft-ietf-mmusic-media-loopback-15.txt)

Hi

I have taken a look at the updated media-loopback draft

     
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-media-loopback-15.txt 
<http://www.ietf.org/internet-drafts/draft-ietf-mmusic-media-loopback-15.txt>

and I consider it to be a subtantial improvement over v14. I still have 
two signficant comments, both of which probably require discussion with 
our ADs and/or the Payload WG chairs, however it would be useful for the 
MMUSIC WG participants to give their view on these as well:

1) Alternative NAT traversal mechanism (Section 5.7)
One general question/comment for the group and our ADs, and two 
technical comments as well
1a) The draft defines an alternative NAT traversal mechanism instead of 
ICE, and this mechanism will only work in certain deployment scenarios 
under certain assumptions. The draft does say that ICE is the preferred 
mechanism and the alternative specified here should only be used if ICE 
is not available and you fulfill the deployment scenario requirements.  
Considering the intended status of the document is Proposed Standard, is 
this acceptable ?

*/[KH] Looking forward to feedback from the group. Please note that 
there are implementations of this mechanism in existing products and in 
use by operators. All implementers are aware that ICE is the preferred 
mechanism and plan to use this mechanism only if ICE is NOT available./*



1b) The draft allows for the answerer to add this NAT traversal 
mechanism as a media format to the answer, even if the offer did not 
include it. The draft furthermore requires the offerer to accept use of 
this mechanism then. This breaks normal O/A behavior and I don't see any 
good reason to do so. If the offerer supports and is willing to use this 
mechanism, he should include it in the offer.

*/[KH] /**/The authors will remove the requirement for the offerer to 
accept the Loopback Pump media format if it did not include it in the 
offer. If the offerer does not have the Loopback Pump media format in 
the offer, the answerer may stay still include it in the response. But 
in this case the offerer is not obligated to accept the call./**//*



1c) The draft seems to assume what is commonly referred to as latching 
behavior by the loopback-source inasmuch as in order for the mechanism 
to work, the loopback-source must send packets to the observed source 
IP-address and port rather than what was negotiated in the SDP. This 
operation however is not explained in the draft, and it needs to be 
(with a MUST).

*/[KH] The authors will explain this behavior in the draft with a MUST./*




2) Definition of several new RTP payload formats (Sections 7, 8, and 14.2)
The draft defines several new RTP payload formats, however, as far as I 
know, the draft has not been reviewed in the PAYLOAD WG (or AVT prior to 
that). Ideally, there would be two different drafts, with the RTP 
payload part handled in PAYLOAD, and the other part handled in MMUSIC, 
however I'm sympathetic to the fact that we are on v15 of this draft.

*/[KH] The authors prefer to keep one draft. Please advise on when is 
the right time to send for PAYLOAD WG review./*




I have a few detailed comments as well that should be relatively easy to 
resolve, however I'll defer those until we figure out a path forward on 
the above.

Thanks

-- Flemming


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Hi MMUSIC <br>
<br>
We haven't seen much list discussion on this one. It would be really
useful if the WG participants could provide some feedback on the media
loopback points raised below so we can determine the best way to
proceed here. In particular, we would like to know if the WG believes
that the alternative NAT traversal mechanism (latching) proposed in the
draft is generally applicable outside of the media loopback use case,
and/or if we in general need an alternative to ICE for NAT traversal. <br>
<br>
Please send your comments to the list. <br>
<br>
Thanks <br>
<br>
-- Flemming <br>
(as MMUSIC co-chair)<br>
<br>
<br>
<br>
<br>
-------- Original Message --------
<table class="moz-email-headers-table" border="0" cellpadding="0"
 cellspacing="0">
  <tbody>
    <tr>
      <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject: </th>
      <td>RE: [MMUSIC] Media loopback
revisited(draft-ietf-mmusic-media-loopback-15.txt)</td>
    </tr>
    <tr>
      <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date: </th>
      <td>Wed, 1 Jun 2011 07:02:45 -0500</td>
    </tr>
    <tr>
      <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From: </th>
      <td>Kaynam Hedayat <a class="moz-txt-link-rfc2396E" href="mailto:kaynam.hedayat@exfo.com">&lt;kaynam.hedayat@exfo.com&gt;</a></td>
    </tr>
    <tr>
      <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To: </th>
      <td>Flemming Andreasen (fandreas) <a class="moz-txt-link-rfc2396E" href="mailto:fandreas@cisco.com">&lt;fandreas@cisco.com&gt;</a>,
<a class="moz-txt-link-rfc2396E" href="mailto:draft-ietf-mmusic-media-loopback@tools.ietf.org">&lt;draft-ietf-mmusic-media-loopback@tools.ietf.org&gt;</a></td>
    </tr>
    <tr>
      <th align="RIGHT" nowrap="nowrap" valign="BASELINE">CC: </th>
      <td>mmusic <a class="moz-txt-link-rfc2396E" href="mailto:mmusic@ietf.org">&lt;mmusic@ietf.org&gt;</a></td>
    </tr>
  </tbody>
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<div class="Section1">
<p class="MsoNormal"><span
 style="font-size: 11pt; font-family: &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">Flemming,<o:p></o:p></span></p>
<p class="MsoNormal"><span
 style="font-size: 11pt; font-family: &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal"><span
 style="font-size: 11pt; font-family: &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">Thank
you for reviewing the new version of the draft. Some
comments from the authors below.<o:p></o:p></span></p>
<p class="MsoNormal"><span
 style="font-size: 11pt; font-family: &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal"><span
 style="font-size: 11pt; font-family: &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">Regards,<o:p></o:p></span></p>
<p class="MsoNormal"><span
 style="font-size: 11pt; font-family: &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">Kaynam<o:p></o:p></span></p>
<p class="MsoNormal"><span
 style="font-size: 11pt; font-family: &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal"><span
 style="font-size: 11pt; font-family: &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);"><o:p>&nbsp;</o:p></span></p>
<div>
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 style="border-style: solid none none; border-color: rgb(181, 196, 223) -moz-use-text-color -moz-use-text-color; border-width: 1pt medium medium; padding: 3pt 0in 0in;">
<p class="MsoNormal"><b><span
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 style="font-size: 10pt; font-family: &quot;Tahoma&quot;,&quot;sans-serif&quot;; color: windowtext;">
<a class="moz-txt-link-abbreviated" href="mailto:mmusic-bounces@ietf.org">mmusic-bounces@ietf.org</a>
[<a class="moz-txt-link-freetext" href="mailto:mmusic-bounces@ietf.org">mailto:mmusic-bounces@ietf.org</a>] <b>On Behalf Of </b>Flemming
Andreasen<br>
<b>Sent:</b> Thursday, May 12, 2011 8:06 PM<br>
<b>To:</b> <a class="moz-txt-link-abbreviated" href="mailto:draft-ietf-mmusic-media-loopback@tools.ietf.org">draft-ietf-mmusic-media-loopback@tools.ietf.org</a><br>
<b>Cc:</b> mmusic<br>
<b>Subject:</b> [MMUSIC] Media loopback
revisited(draft-ietf-mmusic-media-loopback-15.txt)<o:p></o:p></span></p>
</div>
</div>
<p class="MsoNormal"><o:p>&nbsp;</o:p></p>
<p class="MsoNormal">Hi <br>
<br>
I have taken a look at the updated media-loopback draft<br>
&nbsp;&nbsp;&nbsp; <br>
<span style="font-size: 10pt;"><a moz-do-not-send="true"
 href="http://www.ietf.org/internet-drafts/draft-ietf-mmusic-media-loopback-15.txt">&nbsp;&nbsp;&nbsp;
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-media-loopback-15.txt</a></span><br>
<br>
<span style="font-size: 10pt;">and I consider it to be a subtantial
improvement
over v14. I still have two signficant comments, both of which probably
require
discussion with our ADs and/or the Payload WG chairs, however it would
be
useful for the MMUSIC WG participants to give their view on these as
well:<br>
<br>
1) Alternative NAT traversal mechanism (Section 5.7)<br>
One general question/comment for the group and our ADs, and two
technical
comments as well<br>
1a) The draft defines an alternative NAT traversal mechanism instead of
ICE,
and this mechanism will only work in certain deployment scenarios under
certain
assumptions. The draft does say that ICE is the preferred mechanism and
the
alternative specified here should only be used if ICE is not available
and you
fulfill the deployment scenario requirements.&nbsp; Considering the intended
status of the document is Proposed Standard, is this acceptable ? </span><span
 style="font-size: 10pt; color: rgb(31, 73, 125);"><o:p></o:p></span></p>
<p class="MsoNormal"><b><i><span
 style="font-size: 11pt; font-family: &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">[KH]
Looking forward to feedback from the group. Please note
that there are implementations of this mechanism in existing products
and in
use by operators. All implementers are aware that ICE is the preferred
mechanism and plan to use this mechanism only if ICE is NOT available.<o:p></o:p></span></i></b></p>
<p class="MsoNormal"><span style="font-size: 10pt;"><br>
<br>
1b) The draft allows for the answerer to add this NAT traversal
mechanism as a
media format to the answer, even if the offer did not include it. The
draft
furthermore requires the offerer to accept use of this mechanism then.
This
breaks normal O/A behavior and I don't see any good reason to do so. If
the
offerer supports and is willing to use this mechanism, he should
include it in
the offer. </span><span
 style="font-size: 10pt; color: rgb(31, 73, 125);"><o:p></o:p></span></p>
<p class="MsoNormal"><b><i><span
 style="font-size: 11pt; font-family: &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">[KH]
</span></i></b><b><i><span
 style="font-size: 11pt; color: rgb(31, 73, 125);">The authors will
remove the requirement for the offerer to
accept the Loopback Pump media format if it did not include it in the
offer. If
the offerer does not have the Loopback Pump media format in the offer,
the
answerer may stay still include it in the response. But in this case
the
offerer is not obligated to accept the call.</span></i></b><b><i><span
 style="font-size: 11pt; font-family: &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);"><o:p></o:p></span></i></b></p>
<p class="MsoNormal"><span style="font-size: 10pt;"><br>
<br>
1c) The draft seems to assume what is commonly referred to as latching
behavior
by the loopback-source inasmuch as in order for the mechanism to work,
the
loopback-source must send packets to the observed source IP-address and
port
rather than what was negotiated in the SDP. This operation however is
not
explained in the draft, and it needs to be (with a MUST). </span><span
 style="font-size: 10pt; color: rgb(31, 73, 125);"><o:p></o:p></span></p>
<p class="MsoNormal"><b><i><span
 style="font-size: 11pt; font-family: &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">[KH]
The authors will explain this behavior in the draft with a
MUST.<o:p></o:p></span></i></b></p>
<p class="MsoNormal"><span style="font-size: 10pt;"><br>
<br>
<br>
2) Definition of several new RTP payload formats (Sections 7, 8, and
14.2)<br>
The draft defines several new RTP payload formats, however, as far as I
know,
the draft has not been reviewed in the PAYLOAD WG (or AVT prior to
that).
Ideally, there would be two different drafts, with the RTP payload part
handled
in PAYLOAD, and the other part handled in MMUSIC, however I'm
sympathetic to
the fact that we are on v15 of this draft. </span><span
 style="font-size: 10pt; color: rgb(31, 73, 125);"><o:p></o:p></span></p>
<p class="MsoNormal"><b><i><span
 style="font-size: 11pt; font-family: &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">[KH]
The authors prefer to keep one draft. Please advise on when
is the right time to send for PAYLOAD WG review.<o:p></o:p></span></i></b></p>
<p class="MsoNormal"><span style="font-size: 10pt;"><br>
<br>
<br>
I have a few detailed comments as well that should be relatively easy
to
resolve, however I'll defer those until we figure out a path forward on
the
above. <br>
<br>
Thanks <br>
<br>
-- Flemming </span><o:p></o:p></p>
</div>
</body>
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On Tue, 2011-07-05 at 19:59 -0400, Flemming Andreasen wrote:
> In particular, we would like to know if the WG believes that the
> alternative NAT traversal mechanism (latching) proposed in the draft
> is generally applicable outside of the media loopback use case, and/or
> if we in general need an alternative to ICE for NAT traversal.=20

As someone who works on VoIP clients for use over the real Internet (not
a closed network), I'm not so happy by any draft proposing workaround
for people who are too lazy to implement ICE+TURN. Because everything
else is annoyingly unreliable. The proposed latching mechanism will not
work in every case and seems really a poor workaround for not having
ICE. Since any client using media loopback will need both sides to be
updated, there is no reason to not require ICE for any case where a NAT
can exist, obviously ICE may not be needed in a purely client-server
model or in a closed controlled network.

--=20
Olivier Cr=C3=AAte
olivier.crete@collabora.com

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From emil@sip-communicator.org  Wed Jul  6 10:36:32 2011
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Subject: Re: [MMUSIC] Media loopback draft and General Need for ICE alternative ?
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=D0=9D=D0=B0 06.07.11 02:23, Olivier Cr=C3=AAte =D0=BD=D0=B0=D0=BF=D0=B8=D1=
=81=D0=B0:
> On Tue, 2011-07-05 at 19:59 -0400, Flemming Andreasen wrote:
>> In particular, we would like to know if the WG believes that the
>> alternative NAT traversal mechanism (latching) proposed in the draft
>> is generally applicable outside of the media loopback use case, and/or=

>> if we in general need an alternative to ICE for NAT traversal.=20
>=20
> As someone who works on VoIP clients for use over the real Internet (no=
t
> a closed network), I'm not so happy by any draft proposing workaround
> for people who are too lazy to implement ICE+TURN. Because everything
> else is annoyingly unreliable. The proposed latching mechanism

I don't see a proposed latching mechanism in the document. What I saw
was a fallback mechanism in case ICE is killed by a proxy that does
latching (which is what the overwhelming majority of the SIP proxies on
the real Internet do).

> will not
> work in every case and seems really a poor workaround for not having
> ICE.

Sure, but if you can't use ICE because someone in the middle mangled
your SDP, then I suppose a poor workaround is better than nothing.

> Since any client using media loopback will need both sides to be
> updated,

The problem here does not come from the sides that need to support
loopback. The issue is caused by middle boxes that are loopback
agnostic. As much as I appreciate ICE and what it brings to RTC, and as
much as I'd like to believe that everyone would end up adopting it, it
simply sounds unreasonable to pretend that everyone is already using it.

On a side note, "the chicken and egg" problem described in the draft is
similar to what happens with unidirectional streams (e.g. where one side
starts transmitting video while the other is in recvonly). The solution
proposed in this draft would also be handy in such a situation so it
would probably make sense to have it in a separate document. Of course
such a document would have to be make it clear it only exists for
backward compatibility and that it should only be used if ICE is blocked.=


Emil

--
http://jitsi.org



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On 7/6/11 1:36 PM, Emil Ivov wrote:
>
> На 06.07.11 02:23, Olivier Crête написа:
> > On Tue, 2011-07-05 at 19:59 -0400, Flemming Andreasen wrote:
> >> In particular, we would like to know if the WG believes that the
> >> alternative NAT traversal mechanism (latching) proposed in the draft
> >> is generally applicable outside of the media loopback use case, and/or
> >> if we in general need an alternative to ICE for NAT traversal.
> >
> > As someone who works on VoIP clients for use over the real Internet (not
> > a closed network), I'm not so happy by any draft proposing workaround
> > for people who are too lazy to implement ICE+TURN. Because everything
> > else is annoyingly unreliable. The proposed latching mechanism
>
> I don't see a proposed latching mechanism in the document. What I saw
> was a fallback mechanism in case ICE is killed by a proxy that does
> latching (which is what the overwhelming majority of the SIP proxies on
> the real Internet do).
>
Please see Section 5.7 in the loopback draft, which defines a latching 
mechanism as an alternative to ICE, not a fallback if ICE is 
blocked/doesn't work.

> > will not
> > work in every case and seems really a poor workaround for not having
> > ICE.
>
> Sure, but if you can't use ICE because someone in the middle mangled
> your SDP, then I suppose a poor workaround is better than nothing.
>
The proposed latching mechanism depends on SDP extensions as well, so it 
would seem equally vulnerable to SDP mangling by an intermediary.


> > Since any client using media loopback will need both sides to be
> > updated,
>
> The problem here does not come from the sides that need to support
> loopback. The issue is caused by middle boxes that are loopback
> agnostic. As much as I appreciate ICE and what it brings to RTC, and as
> much as I'd like to believe that everyone would end up adopting it, it
> simply sounds unreasonable to pretend that everyone is already using it.
>
It's not a question of whether everybody is using it. It's a question of 
whether there is a problem with ICE and we need an alternative like this 
proposed latching mechanism.


> On a side note, "the chicken and egg" problem described in the draft is
> similar to what happens with unidirectional streams (e.g. where one side
> starts transmitting video while the other is in recvonly). The solution
> proposed in this draft would also be handy in such a situation so it
> would probably make sense to have it in a separate document. Of course
> such a document would have to be make it clear it only exists for
> backward compatibility and that it should only be used if ICE is blocked.
>
If I read your comments right, the only issue you have with ICE is that 
it may be blocked whereas another solution (like latching) might not be. 
Is that an accurate summary ?

Thanks

-- Flemming



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

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  <p><font size="2">На 06.07.11 02:23, Olivier Crête написа:<br>
&gt; On Tue, 2011-07-05 at 19:59 -0400, Flemming Andreasen wrote:<br>
&gt;&gt; In particular, we would like to know if the WG believes that
the<br>
&gt;&gt; alternative NAT traversal mechanism (latching) proposed in the
draft<br>
&gt;&gt; is generally applicable outside of the media loopback use
case, and/or<br>
&gt;&gt; if we in general need an alternative to ICE for NAT traversal.<br>
&gt;<br>
&gt; As someone who works on VoIP clients for use over the real
Internet (not<br>
&gt; a closed network), I'm not so happy by any draft proposing
workaround<br>
&gt; for people who are too lazy to implement ICE+TURN. Because
everything<br>
&gt; else is annoyingly unreliable. The proposed latching mechanism<br>
  <br>
I don't see a proposed latching mechanism in the document. What I saw<br>
was a fallback mechanism in case ICE is killed by a proxy that does<br>
latching (which is what the overwhelming majority of the SIP proxies on<br>
the real Internet do).<br>
  <br>
  </font></p>
</blockquote>
Please see Section 5.7 in the loopback draft, which defines a latching
mechanism as an alternative to ICE, not a fallback if ICE is
blocked/doesn't work. <br>
<br>
<blockquote cite="mid:4E149D0E.2040809@jitsi.org" type="cite">
  <p><font size="2">&gt; will not<br>
&gt; work in every case and seems really a poor workaround for not
having<br>
&gt; ICE.<br>
  <br>
Sure, but if you can't use ICE because someone in the middle mangled<br>
your SDP, then I suppose a poor workaround is better than nothing.<br>
  <br>
  </font></p>
</blockquote>
The proposed latching mechanism depends on SDP extensions as well, so
it would seem equally vulnerable to SDP mangling by an intermediary. <br>
<br>
<br>
<blockquote cite="mid:4E149D0E.2040809@jitsi.org" type="cite">
  <p><font size="2">&gt; Since any client using media loopback will
need both sides to be<br>
&gt; updated,<br>
  <br>
The problem here does not come from the sides that need to support<br>
loopback. The issue is caused by middle boxes that are loopback<br>
agnostic. As much as I appreciate ICE and what it brings to RTC, and as<br>
much as I'd like to believe that everyone would end up adopting it, it<br>
simply sounds unreasonable to pretend that everyone is already using it.<br>
  </font></p>
</blockquote>
It's not a question of whether everybody is using it. It's a question
of whether there is a problem with ICE and we need an alternative like
this proposed latching mechanism. <br>
<br>
<br>
<blockquote cite="mid:4E149D0E.2040809@jitsi.org" type="cite">
  <p><font size="2">On a side note, "the chicken and egg" problem
described in the draft is<br>
similar to what happens with unidirectional streams (e.g. where one side<br>
starts transmitting video while the other is in recvonly). The solution<br>
proposed in this draft would also be handy in such a situation so it<br>
would probably make sense to have it in a separate document. Of course<br>
such a document would have to be make it clear it only exists for<br>
backward compatibility and that it should only be used if ICE is
blocked.<br>
  <br>
  </font></p>
</blockquote>
If I read your comments right, the only issue you have with ICE is that
it may be blocked whereas another solution (like latching) might not
be. Is that an accurate summary ? <br>
<br>
Thanks <br>
<br>
-- Flemming <br>
<br>
<br>
<br>
<blockquote cite="mid:4E149D0E.2040809@jitsi.org" type="cite">
  <p><font size="2">Emil<br>
  <br>
--<br>
  <a moz-do-not-send="true" href="http://jitsi.org">http://jitsi.org</a><br>
  <br>
  <br>
_______________________________________________<br>
mmusic mailing list<br>
<a class="moz-txt-link-abbreviated" href="mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
  <a moz-do-not-send="true"
 href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a><br>
  </font>
  </p>
</blockquote>
</body>
</html>

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From emil@sip-communicator.org  Wed Jul  6 16:07:51 2011
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Date: Thu, 07 Jul 2011 01:07:45 +0200
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Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] Media loopback draft and General Need for ICE alternative ?
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=D0=9D=D0=B0 06.07.11 22:44, Flemming Andreasen =D0=BD=D0=B0=D0=BF=D0=B8=D1=
=81=D0=B0:
>> I don't see a proposed latching mechanism in the document. What I saw
>> was a fallback mechanism in case ICE is killed by a proxy that does
>> latching (which is what the overwhelming majority of the SIP proxies o=
n
>> the real Internet do).
>>
> Please see Section 5.7 in the loopback draft, which defines a latching
> mechanism as an alternative to ICE, not a fallback if ICE is
> blocked/doesn't work.

I suppose I am missing something because in section 5 I read:

    ICE/STUN/TURN provide a general solution ...
    ... Loopback sessions ... SHOULD use these general solutions
    wherever possible ... In scenarios where ICE/STUN/TURN is not
    available [and C1] ... the following solution MAY be adapted.

(I am not sure why C1, requiring that the loopback source has a public
IP address, is necessary here but we can probably come back to this later=
).

So, the way I understand this is that loopback clients need to have and
use ICE. If ICE is not there they could try to compensate using the
mechanism described in 5.7.

As to the mechanism itself it basically says that mirrors shouldn't wait
to hear from the source and that they should start streaming as soon as
they are ready. To me this sounds quite reasonable. It's basic hole
punching and, as I said in my previous mail, it could also be of use in
other cases.

So here's a scenario where I see this reasonable. S (source) supports
ICE and calls M (mirror) which also supports ICE. They are both behind
NATs and they are both connected to a common SIP server, say iptel.org.
When S's INVITE goes through the iptel.org proxy it replaces the
connection address in the SDP with that of its RTP proxy. This of course
means that even though it would see the entire candidate list M would
have to disable ICE. So in this case, which is not at all uncommon, the
technique in 5.7 would be the only way to get media flowing both ways.

Also, in this case, the only one doing latching is the iptel.org service
(not the endpoints that do loopback) and the mechanism from 5.7 allows
to work around it.

>> > will not
>> > work in every case and seems really a poor workaround for not having=

>> > ICE.
>>
>> Sure, but if you can't use ICE because someone in the middle mangled
>> your SDP, then I suppose a poor workaround is better than nothing.
>>
> The proposed latching mechanism depends on SDP extensions as well, so i=
t
> would seem equally vulnerable to SDP mangling by an intermediary.

Well I guess I must be missing something because I only see regular SDP
in this scenario and nothing that would be vulnerable to a middle box.

>> On a side note, "the chicken and egg" problem described in the draft i=
s
>> similar to what happens with unidirectional streams (e.g. where one si=
de
>> starts transmitting video while the other is in recvonly). The solutio=
n
>> proposed in this draft would also be handy in such a situation so it
>> would probably make sense to have it in a separate document. Of course=

>> such a document would have to be make it clear it only exists for
>> backward compatibility and that it should only be used if ICE is block=
ed.
>>
> If I read your comments right, the only issue you have with ICE is that=

> it may be blocked whereas another solution (like latching) might not be=
=2E
> Is that an accurate summary ?

My point is that everyone is using latching and latching disables ICE
and in such cases it would be nice for clients to have something to fall
back to.

I don't see anything in this document being an actual alternative to ICE.=


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

--=20
Emil Ivov, Ph.D.                       67000 Strasbourg,
Project Lead                           France
Jitsi
emcho@jitsi.org                        PHONE: +33.1.77.62.43.30
http://jitsi.org                       FAX:   +33.1.77.62.47.31


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Subject: Re: [MMUSIC] Media loopback draft and General Need for ICE alternative ?
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On 7/6/11 7:07 PM, Emil Ivov wrote:
>
>
> На 06.07.11 22:44, Flemming Andreasen написа:
> >> I don't see a proposed latching mechanism in the document. What I saw
> >> was a fallback mechanism in case ICE is killed by a proxy that does
> >> latching (which is what the overwhelming majority of the SIP proxies on
> >> the real Internet do).
> >>
> > Please see Section 5.7 in the loopback draft, which defines a latching
> > mechanism as an alternative to ICE, not a fallback if ICE is
> > blocked/doesn't work.
>
> I suppose I am missing something because in section 5 I read:
>
>     ICE/STUN/TURN provide a general solution ...
>     ... Loopback sessions ... SHOULD use these general solutions
>     wherever possible ... In scenarios where ICE/STUN/TURN is not
>     available [and C1] ... the following solution MAY be adapted.
>
> (I am not sure why C1, requiring that the loopback source has a public
> IP address, is necessary here but we can probably come back to this 
> later).
>
> So, the way I understand this is that loopback clients need to have and
> use ICE. If ICE is not there they could try to compensate using the
> mechanism described in 5.7.
>
ICE is at SHOULD strength, not MUST, and there are no fallback 
procedures defined that says to use ICE first and if it fails, then use 
latching. The way it's written currently is basically an alternative to 
implementations without ICE.

>
> As to the mechanism itself it basically says that mirrors shouldn't wait
> to hear from the source and that they should start streaming as soon as
> they are ready.
>
No - that is not what the draft says. The draft defines a new payload 
type called "loopbkprimer" and associated procedures for using it to 
punch holes and associated latching operation on the RTP endpoint itself 
(latching details are not spelled out in detail in the draft currently).



> To me this sounds quite reasonable. It's basic hole
> punching and, as I said in my previous mail, it could also be of use in
> other cases.
>
> So here's a scenario where I see this reasonable. S (source) supports
> ICE and calls M (mirror) which also supports ICE. They are both behind
> NATs and they are both connected to a common SIP server, say iptel.org.
> When S's INVITE goes through the iptel.org proxy it replaces the
> connection address in the SDP with that of its RTP proxy. This of course
> means that even though it would see the entire candidate list M would
> have to disable ICE.
>
- How does the iptel.org proxy know it needs to insert a RTP proxy ?
- How would M know an RTP proxy was inserted and that M needs to disable 
ICE ?
- Why does M need to disable ICE in this case rather than let it run its 
course - is there an assumption that this RTP proxy will not pass ICE 
packets through and trying to do ICE will somehow incur problems ?



> So in this case, which is not at all uncommon, the
> technique in 5.7 would be the only way to get media flowing both ways.
>
ICE enables two endpoints to be behind a NAT as well. It may require 
TURN instead of an RTP proxy as is suggested in this example. The 
advantage of a TURN based solution is that it will work for other types 
of media streams than RTP.

> Also, in this case, the only one doing latching is the iptel.org service
> (not the endpoints that do loopback) and the mechanism from 5.7 allows
> to work around it.
>
That's not clear based on your description. The mechanism in section 5.7 
would require both S and M to be loopback-mirrors and send the loopback 
primer packets to the RTP proxy, which in turn needs to be a 
loopback-source for both endpoints and at this point is no longer a 
proxy (nor is the iptel.org "proxy").


> >> > will not
> >> > work in every case and seems really a poor workaround for not having
> >> > ICE.
> >>
> >> Sure, but if you can't use ICE because someone in the middle mangled
> >> your SDP, then I suppose a poor workaround is better than nothing.
> >>
> > The proposed latching mechanism depends on SDP extensions as well, so it
> > would seem equally vulnerable to SDP mangling by an intermediary.
>
> Well I guess I must be missing something because I only see regular SDP
> in this scenario and nothing that would be vulnerable to a middle box.
>
If you only see regular SDP, then you are not following the mechanism 
defined in the loopback draft (loopback, mirror and source SDP 
attributes and loopbkprimer payload format).


> >> On a side note, "the chicken and egg" problem described in the draft is
> >> similar to what happens with unidirectional streams (e.g. where one 
> side
> >> starts transmitting video while the other is in recvonly). The solution
> >> proposed in this draft would also be handy in such a situation so it
> >> would probably make sense to have it in a separate document. Of course
> >> such a document would have to be make it clear it only exists for
> >> backward compatibility and that it should only be used if ICE is 
> blocked.
> >>
> > If I read your comments right, the only issue you have with ICE is that
> > it may be blocked whereas another solution (like latching) might not be.
> > Is that an accurate summary ?
>
> My point is that everyone is using latching and latching disables ICE
> and in such cases it would be nice for clients to have something to fall
> back to.
>
> I don't see anything in this document being an actual alternative to ICE.
>
It defines a NAT traversal solution different from ICE, and that will 
only work under certain assumptions. The question is whether the WG 
believes we need such a mechanism in general, and if so why.

Thanks

-- Flemming



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On 7/6/11 7:07 PM, Emil Ivov wrote:
<blockquote cite="mid:4E14EAC1.4040203@jitsi.org" type="cite">
  <meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
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  <title>Re: [MMUSIC] Media loopback draft and General Need for ICE
alternative ?</title>
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  <br>
  <br>
  <p><font size="2">На 06.07.11 22:44, Flemming Andreasen написа:<br>
&gt;&gt; I don't see a proposed latching mechanism in the document.
What I saw<br>
&gt;&gt; was a fallback mechanism in case ICE is killed by a proxy that
does<br>
&gt;&gt; latching (which is what the overwhelming majority of the SIP
proxies on<br>
&gt;&gt; the real Internet do).<br>
&gt;&gt;<br>
&gt; Please see Section 5.7 in the loopback draft, which defines a
latching<br>
&gt; mechanism as an alternative to ICE, not a fallback if ICE is<br>
&gt; blocked/doesn't work.<br>
  <br>
I suppose I am missing something because in section 5 I read:<br>
  <br>
    ICE/STUN/TURN provide a general solution ...<br>
    ... Loopback sessions ... SHOULD use these general solutions<br>
    wherever possible ... In scenarios where ICE/STUN/TURN is not<br>
    available [and C1] ... the following solution MAY be adapted.<br>
  <br>
(I am not sure why C1, requiring that the loopback source has a public<br>
IP address, is necessary here but we can probably come back to this
later).<br>
  <br>
So, the way I understand this is that loopback clients need to have and<br>
use ICE. If ICE is not there they could try to compensate using the<br>
mechanism described in 5.7.<br>
  </font></p>
</blockquote>
ICE is at SHOULD strength, not MUST, and there are no fallback
procedures defined that says to use ICE first and if it fails, then use
latching. The way it's written currently is basically an alternative to
implementations without ICE. <br>
<br>
<blockquote cite="mid:4E14EAC1.4040203@jitsi.org" type="cite">
  <p><font size="2"><br>
As to the mechanism itself it basically says that mirrors shouldn't wait<br>
to hear from the source and that they should start streaming as soon as<br>
they are ready. </font></p>
</blockquote>
No - that is not what the draft says. The draft defines a new payload
type called "loopbkprimer" and associated procedures for using it to
punch holes and associated latching operation on the RTP endpoint
itself (latching details are not spelled out in detail in the draft
currently). <br>
<br>
<br>
<br>
<blockquote cite="mid:4E14EAC1.4040203@jitsi.org" type="cite">
  <p><font size="2">To me this sounds quite reasonable. It's basic hole<br>
punching and, as I said in my previous mail, it could also be of use in<br>
other cases.<br>
  <br>
So here's a scenario where I see this reasonable. S (source) supports<br>
ICE and calls M (mirror) which also supports ICE. They are both behind<br>
NATs and they are both connected to a common SIP server, say iptel.org.<br>
When S's INVITE goes through the iptel.org proxy it replaces the<br>
connection address in the SDP with that of its RTP proxy. This of course<br>
means that even though it would see the entire candidate list M would<br>
have to disable ICE. </font></p>
</blockquote>
- How does the iptel.org proxy know it needs to insert a RTP proxy ? <br>
- How would M know an RTP proxy was inserted and that M needs to
disable ICE ? <br>
- Why does M need to disable ICE in this case rather than let it run
its course - is there an assumption that this RTP proxy will not pass
ICE packets through and trying to do ICE will somehow incur problems ? <br>
<br>
<br>
<br>
<blockquote cite="mid:4E14EAC1.4040203@jitsi.org" type="cite">
  <p><font size="2">So in this case, which is not at all uncommon, the<br>
technique in 5.7 would be the only way to get media flowing both ways.<br>
  <br>
  </font></p>
</blockquote>
ICE enables two endpoints to be behind a NAT as well. It may require
TURN instead of an RTP proxy as is suggested in this example. The
advantage of a TURN based solution is that it will work for other types
of media streams than RTP. <br>
<br>
<blockquote cite="mid:4E14EAC1.4040203@jitsi.org" type="cite">
  <p><font size="2">Also, in this case, the only one doing latching is
the iptel.org service<br>
(not the endpoints that do loopback) and the mechanism from 5.7 allows<br>
to work around it.<br>
  <br>
  </font></p>
</blockquote>
That's not clear based on your description. The mechanism in section
5.7 would require both S and M to be loopback-mirrors and send the
loopback primer packets to the RTP proxy, which in turn needs to be a
loopback-source for both endpoints and at this point is no longer a
proxy (nor is the iptel.org "proxy"). <br>
<br>
<br>
<blockquote cite="mid:4E14EAC1.4040203@jitsi.org" type="cite">
  <p><font size="2">&gt;&gt; &gt; will not<br>
&gt;&gt; &gt; work in every case and seems really a poor workaround for
not having<br>
&gt;&gt; &gt; ICE.<br>
&gt;&gt;<br>
&gt;&gt; Sure, but if you can't use ICE because someone in the middle
mangled<br>
&gt;&gt; your SDP, then I suppose a poor workaround is better than
nothing.<br>
&gt;&gt;<br>
&gt; The proposed latching mechanism depends on SDP extensions as well,
so it<br>
&gt; would seem equally vulnerable to SDP mangling by an intermediary.<br>
  <br>
Well I guess I must be missing something because I only see regular SDP<br>
in this scenario and nothing that would be vulnerable to a middle box.<br>
  <br>
  </font></p>
</blockquote>
If you only see regular SDP, then you are not following the mechanism
defined in the loopback draft (loopback, mirror and source SDP
attributes and loopbkprimer payload format). <br>
<br>
<br>
<blockquote cite="mid:4E14EAC1.4040203@jitsi.org" type="cite">
  <p><font size="2">&gt;&gt; On a side note, "the chicken and egg"
problem described in the draft is<br>
&gt;&gt; similar to what happens with unidirectional streams (e.g.
where one side<br>
&gt;&gt; starts transmitting video while the other is in recvonly). The
solution<br>
&gt;&gt; proposed in this draft would also be handy in such a situation
so it<br>
&gt;&gt; would probably make sense to have it in a separate document.
Of course<br>
&gt;&gt; such a document would have to be make it clear it only exists
for<br>
&gt;&gt; backward compatibility and that it should only be used if ICE
is blocked.<br>
&gt;&gt;<br>
&gt; If I read your comments right, the only issue you have with ICE is
that<br>
&gt; it may be blocked whereas another solution (like latching) might
not be.<br>
&gt; Is that an accurate summary ?<br>
  <br>
My point is that everyone is using latching and latching disables ICE<br>
and in such cases it would be nice for clients to have something to fall<br>
back to.<br>
  <br>
I don't see anything in this document being an actual alternative to
ICE.<br>
  <br>
  </font></p>
</blockquote>
It defines a NAT traversal solution different from ICE, and that will
only work under certain assumptions. The question is whether the WG
believes we need such a mechanism in general, and if so why. <br>
<br>
Thanks <br>
<br>
-- Flemming <br>
<font size="2"><br>
</font><br>
<blockquote cite="mid:4E14EAC1.4040203@jitsi.org" type="cite">
</blockquote>
</body>
</html>

--------------090601010200090703070205--

From emil@sip-communicator.org  Wed Jul  6 22:44:35 2011
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Date: Thu, 07 Jul 2011 07:44:28 +0200
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Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] Media loopback draft and General Need for ICE alternative ?
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=D0=9D=D0=B0 07.07.11 02:39, Flemming Andreasen =D0=BD=D0=B0=D0=BF=D0=B8=D1=
=81=D0=B0:
>> =D0=9D=D0=B0 06.07.11 22:44, Flemming Andreasen =D0=BD=D0=B0=D0=BF=D0=B8=
=D1=81=D0=B0:
>> >> I don't see a proposed latching mechanism in the document. What I s=
aw
>> >> was a fallback mechanism in case ICE is killed by a proxy that does=

>> >> latching (which is what the overwhelming majority of the SIP proxie=
s on
>> >> the real Internet do).
>> >>
>> > Please see Section 5.7 in the loopback draft, which defines a latchi=
ng
>> > mechanism as an alternative to ICE, not a fallback if ICE is
>> > blocked/doesn't work.
>>
>> I suppose I am missing something because in section 5 I read:
>>
>>     ICE/STUN/TURN provide a general solution ...
>>     ... Loopback sessions ... SHOULD use these general solutions
>>     wherever possible ... In scenarios where ICE/STUN/TURN is not
>>     available [and C1] ... the following solution MAY be adapted.
>>
>> (I am not sure why C1, requiring that the loopback source has a public=

>> IP address, is necessary here but we can probably come back to this
>> later).
>>
>> So, the way I understand this is that loopback clients need to have an=
d
>> use ICE. If ICE is not there they could try to compensate using the
>> mechanism described in 5.7.
>>
> ICE is at SHOULD strength, not MUST, and there are no fallback
> procedures defined that says to use ICE first and if it fails, then use=

> latching.

Definitely not. I never meant to imply there were. It is however a
fallback in the case endpoints are prevented from even starting ICE.

>> As to the mechanism itself it basically says that mirrors shouldn't wa=
it
>> to hear from the source and that they should start streaming as soon a=
s
>> they are ready.
>>
> No - that is not what the draft says. The draft defines a new payload
> type called "loopbkprimer" and associated procedures for using it to
> punch holes and associated latching operation on the RTP endpoint itsel=
f
> (latching details are not spelled out in detail in the draft currently)=
=2E

Indeed and this is exactly what I was referring to. The "loopbkprimer"
is the payload type that mirrors "should start streaming as soon as they
are ready" rather than "wait[ing] to hear from the source first".

>> To me this sounds quite reasonable. It's basic hole
>> punching and, as I said in my previous mail, it could also be of use i=
n
>> other cases.
>>
>> So here's a scenario where I see this reasonable. S (source) supports
>> ICE and calls M (mirror) which also supports ICE. They are both behind=

>> NATs and they are both connected to a common SIP server, say iptel.org=
=2E
>> When S's INVITE goes through the iptel.org proxy it replaces the
>> connection address in the SDP with that of its RTP proxy. This of cour=
se
>> means that even though it would see the entire candidate list M would
>> have to disable ICE.
>>
> - How does the iptel.org proxy know it needs to insert a RTP proxy ?

It either does so unconditionally or it recognizes the addresses in the
SDP as non-public so it replaces them. This how the overwhelming
majority of the Internet VoIP providers out there handle NAT traversal.

> - How would M know an RTP proxy was inserted and that M needs to disabl=
e
> ICE ?

Because the default candidate (the one inserted by the proxy) is not in
the candidate list.

> - Why does M need to disable ICE in this case rather than let it run it=
s
> course=20

Because 5245 says so (Section 5.1):

   The agent will proceed with the ICE procedures defined in this
   specification if, for each media stream in the SDP it received, the
   default destination for each component of that media stream appears
   in a candidate attribute.

   ...

   If this condition is not met, the agent MUST process the SDP based on
   normal RFC 3264 procedures


> is there an assumption that this RTP proxy will not pass ICE
> packets through and trying to do ICE will somehow incur problems ?

I suppose this would be one of the reasons yes. I didn't follow the
discussions that led to this text back in the time but I'd guess the
assumption was that if someone in the signalling path is taking control
then it would be best not to get in the way.

>> So in this case, which is not at all uncommon, the
>> technique in 5.7 would be the only way to get media flowing both ways.=

>>
> ICE enables two endpoints to be behind a NAT as well. It may require
> TURN instead of an RTP proxy as is suggested in this example. The
> advantage of a TURN based solution is that it will work for other types=

> of media streams than RTP.


>=20
>> Also, in this case, the only one doing latching is the iptel.org servi=
ce
>> (not the endpoints that do loopback) and the mechanism from 5.7 allows=

>> to work around it.
>>
> That's not clear based on your description. The mechanism in section 5.=
7
> would require both S and M to be loopback-mirrors and send the loopback=

> primer packets to the RTP proxy, which in turn needs to be a
> loopback-source for both endpoints and at this point is no longer a
> proxy (nor is the iptel.org "proxy").

I am not following. Why would both S and M need to be mirrors? S would
start streaming ... it has no reason not to. The RTP proxy in the middle
however won't be able to relay its packets until it hears from M. Until
then, it simply wouldn't know where to relay them to.

>> >> > will not
>> >> > work in every case and seems really a poor workaround for not hav=
ing
>> >> > ICE.
>> >>
>> >> Sure, but if you can't use ICE because someone in the middle mangle=
d
>> >> your SDP, then I suppose a poor workaround is better than nothing.
>> >>
>> > The proposed latching mechanism depends on SDP extensions as well, s=
o it
>> > would seem equally vulnerable to SDP mangling by an intermediary.
>>
>> Well I guess I must be missing something because I only see regular SD=
P
>> in this scenario and nothing that would be vulnerable to a middle box.=

>>
> If you only see regular SDP, then you are not following the mechanism
> defined in the loopback draft (loopback, mirror and source SDP
> attributes and loopbkprimer payload format).

How are these "irregular" SDP and what reason would a middle box have to
remove them?

>> >> On a side note, "the chicken and egg" problem described in the draf=
t is
>> >> similar to what happens with unidirectional streams (e.g. where one=

>> side
>> >> starts transmitting video while the other is in recvonly). The solu=
tion
>> >> proposed in this draft would also be handy in such a situation so i=
t
>> >> would probably make sense to have it in a separate document. Of cou=
rse
>> >> such a document would have to be make it clear it only exists for
>> >> backward compatibility and that it should only be used if ICE is
>> blocked.
>> >>
>> > If I read your comments right, the only issue you have with ICE is t=
hat
>> > it may be blocked whereas another solution (like latching) might not=
 be.
>> > Is that an accurate summary ?
>>
>> My point is that everyone is using latching and latching disables ICE
>> and in such cases it would be nice for clients to have something to fa=
ll
>> back to.
>>
>> I don't see anything in this document being an actual alternative to I=
CE.
>>
> It defines a NAT traversal solution different from ICE, and that will
> only work under certain assumptions. The question is whether the WG
> believes we need such a mechanism in general, and if so why.

Well again, I don't see anything in this draft that would allow two
clients, both of which are behind a NAT, to talk to each other. The
relay in the middle does. In that respect I don't see how anyone can
pretend that the loopback primer is an "alternative to ICE". It is not.
 It does not provide means for clients to discover a direct path to each
other. It does not provide relaying. It only helps to resolve a
starvation in the echo use case and in a scenario where a regular
bi-directional voip call would have worked anyway because of the
latching that middle boxes are doing.

Emil

--
http://jitsi.org


From magnus.westerlund@ericsson.com  Thu Jul  7 01:46:38 2011
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Subject: [MMUSIC] Issues with bandwidth modifiers in SDP
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Hi,

Bo Burman and I have written an I-D that discusses a number of issues
around conferencing, including handling of multiple SSRCs in the same
RTP session. This also has brought forward an issue with the current
bandwidth modifiers. They are very ill suited when you have both
asymmetric conditions based on which direction the flows goes and how to
indicate individual streams versus limits for the whole aggregate. This
include the issue of having payload specific per stream upper bounds.

https://datatracker.ietf.org/doc/draft-westerlund-avtcore-multistream-and-simulcast/

So please review this and comment on the list. We have requested agenda
time but don't know if we will get any. So if you are interested in this
topic, commenting on the list will increase our chances for face to face
discussion of this i Quebec.

cheers

Magnus Westerlund

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


From miguel.a.garcia@ericsson.com  Thu Jul  7 05:21:59 2011
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Subject: [MMUSIC] Draft agenda for MMUSIC at IETF 81 in Quebec
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Hi,

we have just uploaded a draft agenda for the MMUSIC meeting in Quebec. 
Available at:

http://www.ietf.org/proceedings/81/agenda/mmusic.htm

As usually, it is subject to suffer changes until the very last minute.

BR,

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

From fandreas@cisco.com  Thu Jul  7 07:50:30 2011
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Date: Thu, 07 Jul 2011 10:49:53 -0400
From: Flemming Andreasen <fandreas@cisco.com>
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Subject: Re: [MMUSIC] Media loopback draft and General Need for ICE alternative ?
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Thanks - I now understand your point about ice-mismatch and how latching 
may be a fallback in that case (I'll note that this is not how the 
loopback draft currently describes it though). More below (snipped some 
stuff).



On 7/7/11 1:44 AM, Emil Ivov wrote:
>
> На 07.07.11 02:39, Flemming Andreasen написа:
> >> На 06.07.11 22:44, Flemming Andreasen написа:
>
<snip>

>
> >> So in this case, which is not at all uncommon, the
> >> technique in 5.7 would be the only way to get media flowing both ways.
> >>
> > ICE enables two endpoints to be behind a NAT as well. It may require
> > TURN instead of an RTP proxy as is suggested in this example. The
> > advantage of a TURN based solution is that it will work for other types
> > of media streams than RTP.
>
>
> >
> >> Also, in this case, the only one doing latching is the iptel.org 
> service
> >> (not the endpoints that do loopback) and the mechanism from 5.7 allows
> >> to work around it.
> >>
> > That's not clear based on your description. The mechanism in section 5.7
> > would require both S and M to be loopback-mirrors and send the loopback
> > primer packets to the RTP proxy, which in turn needs to be a
> > loopback-source for both endpoints and at this point is no longer a
> > proxy (nor is the iptel.org "proxy").
>
> I am not following. Why would both S and M need to be mirrors? S would
> start streaming ... it has no reason not to. The RTP proxy in the middle
> however won't be able to relay its packets until it hears from M. Until
> then, it simply wouldn't know where to relay them to.
>
You're right. So:
1) M starts sending loopbkprimer packets to R (RTP Proxy), which has no 
idea where to send them yet (since S is behind a NAT). RTP proxy can 
however do latching operation for M at this point.
2) S starts sending loopback packets to R (RTP Proxy), which can forward 
them to M because of the latching operation done in step 1).
3) R can now also do latching operation for S and hence we have a 
two-way path between S and M via R
4) S receives loopback packet and stops sending loopback primer packets 
at this point
5) Loopback between S and M is verified (via R)



> >> >> > will not
> >> >> > work in every case and seems really a poor workaround for not 
> having
> >> >> > ICE.
> >> >>
> >> >> Sure, but if you can't use ICE because someone in the middle mangled
> >> >> your SDP, then I suppose a poor workaround is better than nothing.
> >> >>
> >> > The proposed latching mechanism depends on SDP extensions as 
> well, so it
> >> > would seem equally vulnerable to SDP mangling by an intermediary.
> >>
> >> Well I guess I must be missing something because I only see regular SDP
> >> in this scenario and nothing that would be vulnerable to a middle box.
> >>
> > If you only see regular SDP, then you are not following the mechanism
> > defined in the loopback draft (loopback, mirror and source SDP
> > attributes and loopbkprimer payload format).
>
> How are these "irregular" SDP and what reason would a middle box have to
> remove them?
>
My point is that they are just as regular or irregular as any other SDP, 
including ICE. The one difference I do see at this point is that by the 
middle box changing address information in the SDP, it will effectively 
turn off ICE based on the ice-mismatch procedure you pointed out. The 
loopback mechanism will not suffer from this problem, as long as the 
middle box passes through the SDP with whatever parameters it contains 
(i.e. the middle box does not insist on understanding the SDP itself; if 
it did, it would have to support the loopback mechanism including its 
defined SDP extensions and loopback primer payload).



> >> >> On a side note, "the chicken and egg" problem described in the 
> draft is
> >> >> similar to what happens with unidirectional streams (e.g. where one
> >> side
> >> >> starts transmitting video while the other is in recvonly). The 
> solution
> >> >> proposed in this draft would also be handy in such a situation so it
> >> >> would probably make sense to have it in a separate document. Of 
> course
> >> >> such a document would have to be make it clear it only exists for
> >> >> backward compatibility and that it should only be used if ICE is
> >> blocked.
> >> >>
> >> > If I read your comments right, the only issue you have with ICE 
> is that
> >> > it may be blocked whereas another solution (like latching) might 
> not be.
> >> > Is that an accurate summary ?
> >>
> >> My point is that everyone is using latching and latching disables ICE
> >> and in such cases it would be nice for clients to have something to 
> fall
> >> back to.
> >>
> >> I don't see anything in this document being an actual alternative 
> to ICE.
> >>
> > It defines a NAT traversal solution different from ICE, and that will
> > only work under certain assumptions. The question is whether the WG
> > believes we need such a mechanism in general, and if so why.
>
> Well again, I don't see anything in this draft that would allow two
> clients, both of which are behind a NAT, to talk to each other. The
> relay in the middle does. In that respect I don't see how anyone can
> pretend that the loopback primer is an "alternative to ICE". It is not.
>  It does not provide means for clients to discover a direct path to each
> other. It does not provide relaying. It only helps to resolve a
> starvation in the echo use case and in a scenario where a regular
> bi-directional voip call would have worked anyway because of the
> latching that middle boxes are doing.
>
I agree with your point above, however it does also provide a limited 
alternative to ICE inasmuch as it defines an alternative NAT traversal 
mechanism that can be used in certain limited deployment scenarios 
(without relays), and the way the draft is written currently, that's the 
way this mechanim is positioned as well. Again, you are bringing up 
another real-life use case that has merit IMO. It would be good to know 
if there are others.

Thanks

-- Flemming



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Thanks - I now understand your point about ice-mismatch and how
latching may be a fallback in that case (I'll note that this is not how
the loopback draft currently describes it though). More below (snipped
some stuff). <br>
<br>
<br>
<br>
On 7/7/11 1:44 AM, Emil Ivov wrote:
<blockquote cite="mid:4E1547BC.8030902@jitsi.org" type="cite">
  <meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
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  <title>Re: [MMUSIC] Media loopback draft and General Need for ICE
alternative ?</title>
<!-- Converted from text/plain format -->
  <p><font size="2">На 07.07.11 02:39, Flemming Andreasen написа:<br>
&gt;&gt; На 06.07.11 22:44, Flemming Andreasen написа:<br>
  <br>
  </font></p>
</blockquote>
&lt;snip&gt;<br>
<br>
<blockquote cite="mid:4E1547BC.8030902@jitsi.org" type="cite">
  <p><font size="2"><br>
&gt;&gt; So in this case, which is not at all uncommon, the<br>
&gt;&gt; technique in 5.7 would be the only way to get media flowing
both ways.<br>
&gt;&gt;<br>
&gt; ICE enables two endpoints to be behind a NAT as well. It may
require<br>
&gt; TURN instead of an RTP proxy as is suggested in this example. The<br>
&gt; advantage of a TURN based solution is that it will work for other
types<br>
&gt; of media streams than RTP.<br>
  <br>
  <br>
&gt;<br>
&gt;&gt; Also, in this case, the only one doing latching is the
iptel.org service<br>
&gt;&gt; (not the endpoints that do loopback) and the mechanism from
5.7 allows<br>
&gt;&gt; to work around it.<br>
&gt;&gt;<br>
&gt; That's not clear based on your description. The mechanism in
section 5.7<br>
&gt; would require both S and M to be loopback-mirrors and send the
loopback<br>
&gt; primer packets to the RTP proxy, which in turn needs to be a<br>
&gt; loopback-source for both endpoints and at this point is no longer a<br>
&gt; proxy (nor is the iptel.org "proxy").<br>
  <br>
I am not following. Why would both S and M need to be mirrors? S would<br>
start streaming ... it has no reason not to. The RTP proxy in the middle<br>
however won't be able to relay its packets until it hears from M. Until<br>
then, it simply wouldn't know where to relay them to.<br>
  <br>
  </font></p>
</blockquote>
You're right. So:<br>
1) M starts sending loopbkprimer packets to R (RTP Proxy), which has no
idea where to send them yet (since S is behind a NAT). RTP proxy can
however do latching operation for M at this point. <br>
2) S starts sending loopback packets to R (RTP Proxy), which can
forward them to M because of the latching operation done in step 1). <br>
3) R can now also do latching operation for S and hence we have a
two-way path between S and M via R<br>
4) S receives loopback packet and stops sending loopback primer packets
at this point<br>
5) Loopback between S and M is verified (via R)<br>
<br>
<br>
<br>
<blockquote cite="mid:4E1547BC.8030902@jitsi.org" type="cite">
  <p><font size="2">&gt;&gt; &gt;&gt; &gt; will not<br>
&gt;&gt; &gt;&gt; &gt; work in every case and seems really a poor
workaround for not having<br>
&gt;&gt; &gt;&gt; &gt; ICE.<br>
&gt;&gt; &gt;&gt;<br>
&gt;&gt; &gt;&gt; Sure, but if you can't use ICE because someone in the
middle mangled<br>
&gt;&gt; &gt;&gt; your SDP, then I suppose a poor workaround is better
than nothing.<br>
&gt;&gt; &gt;&gt;<br>
&gt;&gt; &gt; The proposed latching mechanism depends on SDP extensions
as well, so it<br>
&gt;&gt; &gt; would seem equally vulnerable to SDP mangling by an
intermediary.<br>
&gt;&gt;<br>
&gt;&gt; Well I guess I must be missing something because I only see
regular SDP<br>
&gt;&gt; in this scenario and nothing that would be vulnerable to a
middle box.<br>
&gt;&gt;<br>
&gt; If you only see regular SDP, then you are not following the
mechanism<br>
&gt; defined in the loopback draft (loopback, mirror and source SDP<br>
&gt; attributes and loopbkprimer payload format).<br>
  <br>
How are these "irregular" SDP and what reason would a middle box have to<br>
remove them?<br>
  <br>
  </font></p>
</blockquote>
My point is that they are just as regular or irregular as any other
SDP, including ICE. The one difference I do see at this point is that
by the middle box changing address information in the SDP, it will
effectively turn off ICE based on the ice-mismatch procedure you
pointed out. The loopback mechanism will not suffer from this problem,
as long as the middle box passes through the SDP with whatever
parameters it contains (i.e. the middle box does not insist on
understanding the SDP itself; if it did, it would have to support the
loopback mechanism including its defined SDP extensions and loopback
primer payload). <br>
<br>
<br>
<br>
<blockquote cite="mid:4E1547BC.8030902@jitsi.org" type="cite">
  <p><font size="2">&gt;&gt; &gt;&gt; On a side note, "the chicken and
egg" problem described in the draft is<br>
&gt;&gt; &gt;&gt; similar to what happens with unidirectional streams
(e.g. where one<br>
&gt;&gt; side<br>
&gt;&gt; &gt;&gt; starts transmitting video while the other is in
recvonly). The solution<br>
&gt;&gt; &gt;&gt; proposed in this draft would also be handy in such a
situation so it<br>
&gt;&gt; &gt;&gt; would probably make sense to have it in a separate
document. Of course<br>
&gt;&gt; &gt;&gt; such a document would have to be make it clear it
only exists for<br>
&gt;&gt; &gt;&gt; backward compatibility and that it should only be
used if ICE is<br>
&gt;&gt; blocked.<br>
&gt;&gt; &gt;&gt;<br>
&gt;&gt; &gt; If I read your comments right, the only issue you have
with ICE is that<br>
&gt;&gt; &gt; it may be blocked whereas another solution (like
latching) might not be.<br>
&gt;&gt; &gt; Is that an accurate summary ?<br>
&gt;&gt;<br>
&gt;&gt; My point is that everyone is using latching and latching
disables ICE<br>
&gt;&gt; and in such cases it would be nice for clients to have
something to fall<br>
&gt;&gt; back to.<br>
&gt;&gt;<br>
&gt;&gt; I don't see anything in this document being an actual
alternative to ICE.<br>
&gt;&gt;<br>
&gt; It defines a NAT traversal solution different from ICE, and that
will<br>
&gt; only work under certain assumptions. The question is whether the WG<br>
&gt; believes we need such a mechanism in general, and if so why.<br>
  <br>
Well again, I don't see anything in this draft that would allow two<br>
clients, both of which are behind a NAT, to talk to each other. The<br>
relay in the middle does. In that respect I don't see how anyone can<br>
pretend that the loopback primer is an "alternative to ICE". It is not.<br>
 It does not provide means for clients to discover a direct path to each<br>
other. It does not provide relaying. It only helps to resolve a<br>
starvation in the echo use case and in a scenario where a regular<br>
bi-directional voip call would have worked anyway because of the<br>
latching that middle boxes are doing.<br>
  <br>
  </font></p>
</blockquote>
I agree with your point above, however it does also provide a limited
alternative to ICE inasmuch as it defines an alternative NAT traversal
mechanism that can be used in certain limited deployment scenarios
(without relays), and the way the draft is written currently, that's
the way this mechanim is positioned as well. Again, you are bringing up
another real-life use case that has merit IMO. It would be good to know
if there are others. <br>
<br>
Thanks <br>
<br>
-- Flemming <br>
<font size="2"><br>
</font><br>
<blockquote cite="mid:4E1547BC.8030902@jitsi.org" type="cite">
</blockquote>
</body>
</html>

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On 7/7/2011 4:46 AM, Magnus Westerlund wrote:
> Hi,
>
> Bo Burman and I have written an I-D that discusses a number of issues
> around conferencing, including handling of multiple SSRCs in the same
> RTP session. This also has brought forward an issue with the current
> bandwidth modifiers. They are very ill suited when you have both
> asymmetric conditions based on which direction the flows goes and how to
> indicate individual streams versus limits for the whole aggregate. This
> include the issue of having payload specific per stream upper bounds.
>
> https://datatracker.ietf.org/doc/draft-westerlund-avtcore-multistream-and-simulcast/

Up front: realize I'm commenting as I read, but that means that it's 
also an indication if
something wasn't clear up-front without reading more of later parts of 
the spec.


The purpose of the document is a little fuzzy - is it for simulcast (as 
stated and defined
in the doc) or for both multiple-source and simulcast (as implied in the 
document and its
title).  It  seems to be the former, but the title and some early 
statements imply it's both.

Also it should be made clear (if it isn't; I'm reading fast) that these 
mixer cases specifically
would include one-to-one communication through a server.  Also: should 
this discuss
non-mixer cases (point-to-point simulcast)?  I think so.

    Thus there has been perceived little need for
    handling multiple SSRCs in implementations.  This has resulted in an
    installed legacy base that isn't fully RTP specification compliant
    and will have different issues if they receive multiple SSRCs of
    media, either simultaneously or in sequence.  These issues will
    manifest themselves in various ways, either by software crashes, or
    simply in limited functionality, like only decoding and playing back
    the first or latest SSRC received and discarding any other SSRCs.


As an implementer of a point-to-point videophone, we handled SSRC changes,
but only were set up to decode one stream at a time (especially of 
video).  (A later
phone could decode up to 3 and encode 1 when acting as a bridge for an
ad-hoc video conference.)

Our algorithm was to immediately react to an SSRC change, but then lock 
out changes
for a (short) period because since we only played one stream at a time 
we didn't want to reset the
codec on every other incoming packet.  This also helped us with a few 
other usecases involving
various B2BUA-ish servers and server-mediated call transfers, etc.

Not truly compliant per-se, of course, but worked for the usecases we 
needed to deal with.

    Simulcasting has the benefit that it is conceptually simple.  It
    enables use of any media codec that the participants agree on,
    allowing the mixer to be codec-agnostic.  Considering today's video
    encoders, it is less bit-rate efficient in the path from the sending
    client to the mixer but more efficient in the mixer to receiver path
    compared to Scalable Video Coding.


If you're targeting a single lower-bitrate encoding, yes.  If the 
bandwidth down to each receiver
is adaptive to their available bandwidth (think mobile reception, or 
WiFi) then giving the mixer
the ability to subset differently for each receiver without having to 
re-encode.  (And it avoids
having to tell the sender to change bitrates if one of the receivers 
needs a lower rate than the
sender is sending, and it avoids giving all the other recipients a 
lowest-common-denominator
stream).  Of course this is part of the entire value-proposition of SVC 
- that it enables this sort of
optimized reception with a single encoding and a low-horsepower mixer.  
Yes, I realize I'm saying
fairly obvious things here.

    If only resolution and temporal variations
    are needed, this can be implemented using H.264, as each simulcast
    version provides the different resolution, and each media stream
    within a simulcast encoding has temporal scalability using no-
    reference frames.


Note that this implies that the mixer parses the H.264 to the point 
where it knows if the frame
will be used as a reference frame.  Not a problem, and probably doesn't 
require any change to the text.

Even within SVC (and you may have stated or implied this) you could make 
multiple encodings
with differently-targeted layers - for example a "low" set targeting 
bandwidths <500K, and a
high set for 500K and up.  This might help avoid the problem of too many 
layers degrading
the bit-efficiency of the encoding.

     3.2.  Simulcasting to Consuming End-Point


This section is clearly based on the assumption (up front) that the 
streams are in fact identical
content with different encodings.  Not a problem, especially if the 
target of the document is
a little clearer.

3.3 and 3.4: These assume no mixer, right?

4.3: negotiation of transmission side - how does an endpoint indicate 
these are "equivalent"
streams and not "alternative" streams (different camera/mic, etc).  How 
are the differing
encodings negotiated, or is that out of scope?  (I'd think it would be 
in-scope, because the mixer
is the one who knows who's connected to it and their capabilities.)

4.5: SDP signaling and bandwidth - b=AS could be used in theory for 
either direction, however
the general offer/answer semantic is that you define what you're 
willing/ready to receive, not send,
so b=AS would be the downstream direction, not the highest of upstream 
or downstream - I don't
know of any that use it that way today.  Some use upstream (perhaps 
most), some use downstream.
Also there should be discussion of how RTCP TMMBR fits into this.

nit in 6.2:

    When one have multiple actual media sources in a session,

have->has

7.1.2: SSRC Multiplexing

Note that "legacy" receivers will often be much happier with NO SSRC change when the
mixer switches streams.  While in theory they should handle them, not all do.  And even
if they do, they may cause glitches at the transition (decoder resets, etc).  Yes, they
shouldn't do that.


   NAT Traversal Failure Rate:  Due to that one need more than a single
       flow to be established through the NAT there is some risk that one
       succeed in establishing the first flow but fails with one or more
       of the additional flows.  The risk that this happens are hard to
       quantify.  However, that risk should be fairly low as one has just
       prior successfully established one flow from the same interfaces.
       Thus only rare events as NAT resource overload, or selecting
       particular port numbers that are filtered etc, should be reasons
       for failure.

As the number of flows increases, the odds of a lost packet (or congestion)
increase.  The pacing timer should help with this, but even so I believe the
odds of problems will increase above a linear rate.  However, this is an
off-the-cuff assessment and I haven't followed the ICE discussions (N zillion MB
of them...).  :-)

8.2.  Mixer Requests of Client streams

I'd guess TMMBR of 0 may not go over well with some existing clients, or may be ignored.

8.4  Multiplexing sessions

This touches on the discussions currently in rtcweb

8.5.1.2 Multiple sources (telepresence)

Ok, this confuses me as the discussion I thought was of simultaneous encodings
of the same data, not true different sources.

At this point I'm afraid my eyes have glazed over reading the (very nicely detailed
and explained) SDP examples.  :-)


-- 
Randell Jesup
randell-ietf@jesup.org


From emil@sip-communicator.org  Thu Jul  7 09:31:21 2011
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Subject: Re: [MMUSIC] Media loopback draft and General Need for ICE alternative ?
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Hey Flemming,

=D0=9D=D0=B0 07.07.11 16:49, Flemming Andreasen =D0=BD=D0=B0=D0=BF=D0=B8=D1=
=81=D0=B0:

<snip>

>> How are these "irregular" SDP and what reason would a middle box have =
to
>> remove them?
>>
> My point is that they are just as regular or irregular as any other SDP=
,
> including ICE. The one difference I do see at this point is that by the=

> middle box changing address information in the SDP, it will effectively=

> turn off ICE based on the ice-mismatch procedure you pointed out. The
> loopback mechanism will not suffer from this problem, as long as the
> middle box passes through the SDP with whatever parameters it contains
> (i.e. the middle box does not insist on understanding the SDP itself; i=
f
> it did, it would have to support the loopback mechanism including its
> defined SDP extensions and loopback primer payload).

Wel does it really need to support the payload or any loopback-specific
mechanisms? I apologise if I am missing something but, as long as the
middle box is not planning on absolutely decoding media, it should be
OK. According to my experience, it is reasonable to expect that most SIP
proxies would handle this properly.

I am of course only talking about the NAT traversal/latching aspects of
the draft. Obviously the echo-related params may be more exposed to
pruning.

<snip>

>> Well again, I don't see anything in this draft that would allow two
>> clients, both of which are behind a NAT, to talk to each other. The
>> relay in the middle does. In that respect I don't see how anyone can
>> pretend that the loopback primer is an "alternative to ICE". It is not=
=2E
>>  It does not provide means for clients to discover a direct path to ea=
ch
>> other. It does not provide relaying. It only helps to resolve a
>> starvation in the echo use case and in a scenario where a regular
>> bi-directional voip call would have worked anyway because of the
>> latching that middle boxes are doing.
>>
> I agree with your point above, however it does also provide a limited
> alternative to ICE inasmuch as it defines an alternative NAT traversal
> mechanism that can be used in certain limited deployment scenarios
> (without relays), and the way the draft is written currently, that's th=
e
> way this mechanim is positioned as well. Again, you are bringing up
> another real-life use case that has merit IMO. It would be good to know=

> if there are others.

Well, there's also the case of UDP-only SIP proxies. I have yet to see a
real-life audio/video INVITE that can fit into a single ethernet MTU
together with an ICE candidate lists. In such cases clients may again
wish to fall back to an ICEless leap of faith and hope for latching.

Of course, UDP-only proxies being prohibited by 3261, this case may be
considered irrelevant. Still, I thought I' mention it anyway given that
many providers out there choose not to enable TCP regardless of what
3261 says.

Cheers,
Emil


--=20
http://jitsi.org


From magnus.westerlund@ericsson.com  Fri Jul  8 01:21:59 2011
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Hi Randell,

Thanks for the feedback. See inline for feedback on your comments.

On 2011-07-07 17:13, Randell Jesup wrote:
> On 7/7/2011 4:46 AM, Magnus Westerlund wrote:
>> Hi,
>>
>> Bo Burman and I have written an I-D that discusses a number of issues
>> around conferencing, including handling of multiple SSRCs in the same
>> RTP session. This also has brought forward an issue with the current
>> bandwidth modifiers. They are very ill suited when you have both
>> asymmetric conditions based on which direction the flows goes and how to
>> indicate individual streams versus limits for the whole aggregate. This
>> include the issue of having payload specific per stream upper bounds.
>>
>> https://datatracker.ietf.org/doc/draft-westerlund-avtcore-multistream-and-simulcast/
> 
> Up front: realize I'm commenting as I read, but that means that it's 
> also an indication if
> something wasn't clear up-front without reading more of later parts of 
> the spec.
> 
> 
> The purpose of the document is a little fuzzy - is it for simulcast (as 
> stated and defined
> in the doc) or for both multiple-source and simulcast (as implied in the 
> document and its
> title).  It  seems to be the former, but the title and some early 
> statements imply it's both.


Yes, this is document that has grown while we wrote it. I actually
expect that we start splitting up the different pieces into individual
documents and tries to clarify what is applicable in what case.

But our intention from the start is to address both cases. But yes we
started with simulcast and for that we do need to resolve the multiple
SSRC in RTP session problems. So simulcast requires multiple SSRCs,
however multiple SSRC has many usages beyond simulcast.


> 
> Also it should be made clear (if it isn't; I'm reading fast) that these 
> mixer cases specifically
> would include one-to-one communication through a server.  Also: should 
> this discuss
> non-mixer cases (point-to-point simulcast)?  I think so.

We tried to be clear that our main focus for simulcast is the
centralized case using a RTP mixer that selects which version to
forward. But Section 3 does look at the other usages of simulcast, like
point to point.

> 
>     Thus there has been perceived little need for
>     handling multiple SSRCs in implementations.  This has resulted in an
>     installed legacy base that isn't fully RTP specification compliant
>     and will have different issues if they receive multiple SSRCs of
>     media, either simultaneously or in sequence.  These issues will
>     manifest themselves in various ways, either by software crashes, or
>     simply in limited functionality, like only decoding and playing back
>     the first or latest SSRC received and discarding any other SSRCs.
> 
> 
> As an implementer of a point-to-point videophone, we handled SSRC changes,
> but only were set up to decode one stream at a time (especially of 
> video).  (A later
> phone could decode up to 3 and encode 1 when acting as a bridge for an
> ad-hoc video conference.)
> 
> Our algorithm was to immediately react to an SSRC change, but then lock 
> out changes
> for a (short) period because since we only played one stream at a time 
> we didn't want to reset the
> codec on every other incoming packet.  This also helped us with a few 
> other usecases involving
> various B2BUA-ish servers and server-mediated call transfers, etc.
> 
> Not truly compliant per-se, of course, but worked for the usecases we 
> needed to deal with.

Yes, and what you say is a confirmation of the claim we are making.
There is a number of implementations that has limited support for
multiple SSRCs but they don't express that in signalling. Therefore to
make these multiple SSRC applications work well going forward we need
explicit signalling of the number of simultanous SSRCs one support.

> 
>     Simulcasting has the benefit that it is conceptually simple.  It
>     enables use of any media codec that the participants agree on,
>     allowing the mixer to be codec-agnostic.  Considering today's video
>     encoders, it is less bit-rate efficient in the path from the sending
>     client to the mixer but more efficient in the mixer to receiver path
>     compared to Scalable Video Coding.
> 
> 
> If you're targeting a single lower-bitrate encoding, yes.  If the 
> bandwidth down to each receiver
> is adaptive to their available bandwidth (think mobile reception, or 
> WiFi) then giving the mixer
> the ability to subset differently for each receiver without having to 
> re-encode.  (And it avoids
> having to tell the sender to change bitrates if one of the receivers 
> needs a lower rate than the
> sender is sending, and it avoids giving all the other recipients a 
> lowest-common-denominator
> stream).  Of course this is part of the entire value-proposition of SVC 
> - that it enables this sort of
> optimized reception with a single encoding and a low-horsepower mixer.  
> Yes, I realize I'm saying
> fairly obvious things here.

Exactly, this is the whole point of simulcasting. And our research hows
that this has some advantages over SVC.

> 
>     If only resolution and temporal variations
>     are needed, this can be implemented using H.264, as each simulcast
>     version provides the different resolution, and each media stream
>     within a simulcast encoding has temporal scalability using no-
>     reference frames.
> 
> 
> Note that this implies that the mixer parses the H.264 to the point 
> where it knows if the frame
> will be used as a reference frame.  Not a problem, and probably doesn't 
> require any change to the text.

Yes. you need to look at the NRI bits in the H.264 RTP payload header to
make this determination. But you need to do similar operations in an SVC
MANE if you single stream transport (SST) payload mode. So no real
difference for the temporal scalability operation.

> 
> Even within SVC (and you may have stated or implied this) you could make 
> multiple encodings
> with differently-targeted layers - for example a "low" set targeting 
> bandwidths <500K, and a
> high set for 500K and up.  This might help avoid the problem of too many 
> layers degrading
> the bit-efficiency of the encoding.

I try to make clear that I see combining SVC for medium and fine-grained
scalability while using different SVC encodings for the large steps and
send them as simulcasts to improve the efficiency of the system as an
option.

> 
>      3.2.  Simulcasting to Consuming End-Point
> 
> 
> This section is clearly based on the assumption (up front) that the 
> streams are in fact identical
> content with different encodings.  Not a problem, especially if the 
> target of the document is
> a little clearer.

Yes, we always target simulcast where the original source of the content
is the same, only the encoded representation is different.

> 
> 3.3 and 3.4: These assume no mixer, right?

Yes.

> 
> 4.3: negotiation of transmission side - how does an endpoint indicate 
> these are "equivalent"
> streams and not "alternative" streams (different camera/mic, etc).  How 
> are the differing
> encodings negotiated, or is that out of scope?  (I'd think it would be 
> in-scope, because the mixer
> is the one who knows who's connected to it and their capabilities.)

I hope you got this explained when you read on. We have the SCS and SCR
grouping semantics to indicate that media lines are simulcast
alternatives to each other. The actual SSRCs in the different RTP
sessions are paired as alternatives using the SRCNAME SDES item. In
practice SRCNAME that are identical for media streams of the same media
type are alternatives to each other. The negotiation of what alternative
stream is done as regular offer/answer for each media line, thus
allowing us to avoid re-inventing that mechanism.

> 
> 4.5: SDP signaling and bandwidth - b=AS could be used in theory for 
> either direction, however
> the general offer/answer semantic is that you define what you're 
> willing/ready to receive, not send,
> so b=AS would be the downstream direction, not the highest of upstream 
> or downstream - I don't
> know of any that use it that way today.  Some use upstream (perhaps 
> most), some use downstream.

Yes, I think there is great confusion around how the bandwidth modifier
actually are used. Our examples in the end do use b=AS as receive
direction only. The discussion in this section may not be as consistent.

I think the main point from our side is that there is need for
additional bandwidth signalling attributes to allow one to express
assymetric capabilities vs desire and do O/A negotiation of it. And do
that both for individual steams and encoding proposals and for the whole
session aggregate.

> Also there should be discussion of how RTCP TMMBR fits into this.

Likely, from my perspective TMMBR has quite clear semantics as it
applies to a particular SSRC. Thus the stream and direction for a given
TMMBR limiation is clear. But, I guess it is good to understand what the
upper limit for that stream may be.

> 
> nit in 6.2:
> 
>     When one have multiple actual media sources in a session,
> 
> have->has

Ok

> 
> 7.1.2: SSRC Multiplexing
> 
> Note that "legacy" receivers will often be much happier with NO SSRC change when the
> mixer switches streams.  While in theory they should handle them, not all do.  And even
> if they do, they may cause glitches at the transition (decoder resets, etc).  Yes, they
> shouldn't do that.

Full agree, and as this is an RTP mixer I assume that the mixer will
have its own SSRC value that it uses to send what ever stream it mixes
or selects from the incomming the content of the media, but sent as a
consistent media stream from the mixers SSRC. Thus we don't have SSRC
switching. The issue I bring up is that of the CSRC field. If one
include CSRC then that changes.

But, in fact I think this is mostly a mote argument. Also for Session
multiplexing we likely will have this issue as using different SSRCs in
the different RTP sessions have advantages.

> 
> 
>    NAT Traversal Failure Rate:  Due to that one need more than a single
>        flow to be established through the NAT there is some risk that one
>        succeed in establishing the first flow but fails with one or more
>        of the additional flows.  The risk that this happens are hard to
>        quantify.  However, that risk should be fairly low as one has just
>        prior successfully established one flow from the same interfaces.
>        Thus only rare events as NAT resource overload, or selecting
>        particular port numbers that are filtered etc, should be reasons
>        for failure.
> 
> As the number of flows increases, the odds of a lost packet (or congestion)
> increase.  The pacing timer should help with this, but even so I believe the
> odds of problems will increase above a linear rate.  However, this is an
> off-the-cuff assessment and I haven't followed the ICE discussions (N zillion MB
> of them...).  :-)

Maybe, and I would love to know how real the issue is. Also, to a
certain degree this is an orthogonal problem if it is real. Because it
will occur also for SSRC multiplexing as even that case has multiple
underlying UDP flows, just a few less than Session multiplexed Simulcast.


> 
> 8.2.  Mixer Requests of Client streams
> 
> I'd guess TMMBR of 0 may not go over well with some existing clients, or may be ignored.

Very much possible. We plan to follow up on this topic in the future.

> 
> 8.4  Multiplexing sessions
> 
> This touches on the discussions currently in rtcweb

Yes, it does. It has also been voiced in discussion around CLUE interest
of multiplexing multiple sessions on top one underlying transport flow.

My main point is that it is separate problem that needs to be addressed
somehow. If you are interested in this problem I would recommend that
one do read the next version of:
https://datatracker.ietf.org/doc/draft-perkins-rtcweb-rtp-usage/

It will be out on Monday. Which tries to make clear the issues which the
Rosenberg's proposal and discuss possible ways forward for that topic.
https://datatracker.ietf.org/doc/draft-rosenberg-rtcweb-rtpmux/


> 
> 8.5.1.2 Multiple sources (telepresence)
> 
> Ok, this confuses me as the discussion I thought was of simultaneous encodings
> of the same data, not true different sources.

Well, this is the case that tries to make it clear how you can combine
multiple real media source, like several cameras in a telepresence room,
with simulcast.

> 
> At this point I'm afraid my eyes have glazed over reading the (very nicely detailed
> and explained) SDP examples.  :-)

Yes, those examples are quite beefy. But, I do hope that they will help
people understand, it also helped us making clear that we had a coherent
story for many of the different usages we see for what we propose.

Cheers

Magnus Westerlund

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


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Subject: Re: [MMUSIC] Reviewing draft-ietf-dccp-udpencap
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On 15 Jun 2011, at 14:32, Miguel A. Garcia wrote:
> A quick review of Section 5 of this document:
>=20
> - I think it is a good idea to clearly indicate which SDP attributes =
are newly created and which ones are reused from other RFCs. I therefore =
recommend to add the following sentences somewhere to the corresponding =
sections:
>=20
> 5.3:
> RFC 4145 [RFC4145] defines the "setup" attribute whose purpose is to =
indcate which of the end points should initiate the connection =
establishment. This document reuses the "setup" attribute to similarly =
indicate which end point initiates the DCCP-UDP connection =
establishment.
>=20
> 5.4:
> RFC 5245 [RFC5245] defines the "candidate" attribute whose purpose is =
to provide one of many possible candidate addresses for communication. =
This document reuses the "candidate" attribute to indicate native or =
encapsulated candidate addresses

Agree. These both seem good additions.

> - In 5.2 last paragraph, I am missing some normative statements, for =
example:
>=20
> If RTCP is multiplexed with RTP, endpoints MUST encode the DCCP port =
used for RTCP in the "rtcp" attribute specified in RFC 3605 [RFC3605]. =
An SDP offerer MAY indicate its willingnes to multiplex RTP and RTCP =
onto a single DCCP port by adding an "rtcp-mux" attribute as specified =
in RFC 5761 [RFC5761]. If the answer also includes the "rtcp-mux" =
attribute (as per RFC 5761 [RFC5761]), then RTP and RTCP are multiplexed =
onto a single DCCP port, otherwise separate DCCP ports are used for RTP =
and RTCP.  In each case, only a single UDP port is used for the DCCP-UDP =
encapsulation.

Makes sense.

> - I didn't find any description of the "dccp-service-code". However, =
it is written in the examples in Section 5.5. Is this a leftover from a =
previous version of the document?

It's from RFC 5762. We should add a reference.

> - I agree with your comment at the end of Section 5.5 indicating that =
an example using ICE would be beneficial.

Yep.

> - Section 7.3. There are two references pointing to Section 5.1 in the =
document, but they should actually point to Section 5.2
>=20
> - I think the following references should be made normative: ICE-TCP, =
RFC3264, RFC 4566, RFC5245, RFC5761

I'm happy with all of these being made normative.

Cheers,
Colin



> BR,
>=20
>       Miguel
>=20
>=20
> On 15/06/2011 9:42, Pasi Sarolahti wrote:
>> Hi,
>>=20
>> In DCCP WG we are soon concluding draft-ietf-dccp-udpencap ("Datagram =
Congestion Control Protocol (DCCP) Encapsulation for NAT Traversal =
(DCCP-UDP)"). This draft has parts related to the work done in MMUSIC =
(especially Section 5), and it was presented in the MMUSIC meeting in =
last IETF. The authors have modified the text based on the input =
received, and we are now looking for volunteer(s) from MMUSIC to review =
whether the new text (mostly in Section 5) looks ok, before moving =
forward with the draft.
>>=20
>> The draft can be found at =
http://tools.ietf.org/html/draft-ietf-dccp-udpencap-08
>>=20
>> - Pasi
>>=20
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>>=20
>=20
> --=20
> Miguel A. Garcia
> +34-91-339-3608
> Ericsson Spain
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic



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




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Subject: Re: [MMUSIC] Comment on draft-jennings-mmusic-adjacent-grouping-03
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On Mar 18, 2011, at 5:46 , Christer Holmberg wrote:

>=20
> Hi Cullen/Ali,
>=20
> There were some comments given on this draft in Beijing, but I am not =
sure they have been addressed in the new version.
>=20
> Q1: Declaration vs negotiation
> ------------------------------

I've decided the work declarative in the context of SDP just creates =
confusion. I removed all this and tried to make it clear what one can =
and can not do with it.=20

>=20
> Q3: Audio/video association
> -----------------------------
>=20
> The draft does not provide a mechanism to e.g. associate audio and =
video streams. I GUESS that could be done by using the same mid value =
for two streams (e.g. an audio and a video stream). IF that's the case, =
I think it needs to be described.

I don't really have any good use cases to drive this but imagine we had =
a two screen video conferencing system where each screen had a camera =
above it and had microphones on both sides of the screen. Roughly =
speaking going from left to right we have Mic1 Cam1 Mic2 Mic3 Cam2 Mic4. =
If theses were the mid labels I could imagine a few things one could do=20=


1)
Create an ADJ group that had mixed audio and and video such as
a=3Dgroup:ADJ Mic1 Cam1 Mic2 Mic3 Cam2 Mic4

2)
Create two ADJ groups, one for the audio and one for the video=20
a=3Dgroup:ADJ Mic1 Mic2 Mic3 Mic4
a=3Dgroup:ADJ Cam1 Cam2

3) Probably a better solution than either of these is use ADJ for the =
cameras then put the audio into some surround sound audio stream.=20

I'm not really two worked up about things like this as I have not seen =
uses case for it.=20

>=20
> Q4: Telepresence
> ------------------
>=20
> (Assuming this is intended to be used also for negotiation)
>=20
> The draft also talks about telepresence, where I agree this kind of =
feature very likely will be needed. But, my suggestion would be for CLUE =
to agree on requirements before taking on this draft, to make sure that =
it can be used to meet those requirements.
>=20
> This was also mentioned in Beijing.
>=20
>        "Magnus Westerlund <magnus.westerlund@ericsson.com> would like =
to see where the telepresence work goes, but was concerned that
>        we not have two solutions to the same problem. Peter Musgrave =
<peter.musgrave@magorcorp.com> supported him,..."
>=20
>        "Mary Barnes <mary.ietf.barnes@gmail.com> asked if anyone else =
supported the work. Why not wait for telepresence?"
>=20
> So, again, it needs to be clarified what the scope of the propsed =
mechanism is.

I think the proposed mechanism is pretty clear about what it can and can =
not do

>=20
>=20
> Q5: Editorial
> -------------
>=20
> The SDP example shows audio streams (m=3Daudio), but I assume it's =
meant to be video (m=3Dvideo)?

Fixed - thanks

>=20


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Subject: Re: [MMUSIC] draft-jennings-mmusic-adjacent-grouping-03
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On Apr 1, 2011, at 24:43 , Stephen Botzko wrote:

> I am wondering about the following:
>=20
> Section 3.1 "This is a declarative SDP parameter and is not =
negotiated."

this whole topic of declarative has created tons of confusion. I just =
removed this and tried to make is clear that this can be used SIP offer =
/ answer or in RTSP style SDP. However, the sender puts in how it thinks =
the streams it is sending should be displayed and there is nothing the =
receiver can do to change that - they can use it or ignore it.=20

>=20
>=20
> Also, I do not think the arrangement of audio streams in an MxN grid =
is very compelling - do you have a good use case for this? =20

I agree - the only uses cases I have for a grid layout is video. The two =
cases I am dealing with are layout of signage and layout of security =
video cameras.=20=

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thanks - makes sense - I fixed up the latest version to support multiple =
grids.=20

On Mar 23, 2011, at 20:01 , Paul Kyzivat wrote:

> (Just catching up on my reading...)
>=20
> One observation about this draft:
>=20
> The grouping framework permits multiple groups to be formed with the =
same semantics. (E.g. you could define two audio/video groups, each to =
be lip synchronized.)
>=20
> Analogously, one could presumably use the ADJ grouping semantic to =
form two independent groups of video streams.
>=20
> OTOH, you might want to form two different groupings of m-lines that =
both map onto the same grid of devices. (For instance, a grouping of =
video m-lines and a corresponding grouping of audio m-lines.)
>=20
> But the a=3Dmedia-grid-dims attribute doesn't provide any way to =
identify which group it applies to if there is more than one.
>=20
> At the very least, the issue of what to do when there are multiple =
adjacency groupings ought to be discussed.
>=20
> 	Thanks,
> 	Paul


Cullen Jennings
For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html



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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-ice-tcp-14.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the Multiparty Multimedia Session Control=
 Working Group of the IETF.

	Title           : TCP Candidates with Interactive Connectivity Establishme=
nt (ICE)
	Author(s)       : Jonathan Rosenberg
                          Ari Keranen
                          Bruce B. Lowekamp
                          Adam Roach
	Filename        : draft-ietf-mmusic-ice-tcp-14.txt
	Pages           : 30
	Date            : 2011-07-11

   Interactive Connectivity Establishment (ICE) defines a mechanism for
   NAT traversal for multimedia communication protocols based on the
   offer/answer model of session negotiation.  ICE works by providing a
   set of candidate transport addresses for each media stream, which are
   then validated with peer-to-peer connectivity checks based on Session
   Traversal Utilities for NAT (STUN).  ICE provides a general framework
   for describing candidates, but only defines UDP-based transport
   protocols.  This specification extends ICE to TCP-based media,
   including the ability to offer a mix of TCP and UDP-based candidates
   for a single stream.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-ice-tcp-14.txt

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

This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-mmusic-ice-tcp-14.txt

From fluffy@cisco.com  Mon Jul 11 09:16:23 2011
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On Jul 11, 2011, at 9:10 , Cullen Jennings wrote:

>=20
> thanks
>=20
> On Mar 31, 2011, at 23:20 , Peter Musgrave wrote:
>=20
>> Hi Cullen/Ali,=20
>>=20
>> Overall, I think this is clean and usable. It solves enough useful =
cases without trying to "solve world hunger".=20
>>=20
>> Is it worth having an example in which the video is SSRC muxed as =
described in section 3.2?
>=20
> added=20
>=20
>>=20
>> Also, I believe your example in 4.2 wants to have all the m lines be =
video (they are audio in the sample).=20
>=20
> fixed - thanks and thanks to other people that caught this too
>=20
>>=20
>> Thanks,=20
>>=20
>> Peter Musgrave
>=20
>=20
> Cullen Jennings
> For corporate legal information go to:
> http://www.cisco.com/web/about/doing_business/legal/cri/index.html
>=20
>=20


Cullen Jennings
For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html



From ari.keranen@nomadiclab.com  Mon Jul 11 09:49:33 2011
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On Jul 11, 2011, at 7:14 PM, Internet-Drafts@ietf.org wrote:
> A New Internet-Draft is available from the on-line Internet-Drafts =
directories. This draft is a work item of the Multiparty Multimedia =
Session Control Working Group of the IETF.
>=20
> 	Title           : TCP Candidates with Interactive Connectivity =
Establishment (ICE)
> 	Author(s)       : Jonathan Rosenberg
>                          Ari Keranen
>                          Bruce B. Lowekamp
>                          Adam Roach
> 	Filename        : draft-ietf-mmusic-ice-tcp-14.txt
> 	Pages           : 30
> 	Date            : 2011-07-11
>=20
>   Interactive Connectivity Establishment (ICE) defines a mechanism for
>   NAT traversal for multimedia communication protocols based on the
>   offer/answer model of session negotiation.  ICE works by providing a
>   set of candidate transport addresses for each media stream, which =
are
>   then validated with peer-to-peer connectivity checks based on =
Session
>   Traversal Utilities for NAT (STUN).  ICE provides a general =
framework
>   for describing candidates, but only defines UDP-based transport
>   protocols.  This specification extends ICE to TCP-based media,
>   including the ability to offer a mix of TCP and UDP-based candidates
>   for a single stream.
>=20
>=20
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-ice-tcp-14.txt


This new revision addresses all the fixes and changes proposed during =
the WGLC. This version is identical to the one posted earlier to the =
list. Unless someone thinks we really need the TCP ice-option, I believe =
we can move forward with this version.


Cheers,
Ari


From simon.perreault@viagenie.ca  Tue Jul 12 12:04:39 2011
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Subject: Re: [MMUSIC] ICE TCP ice-options?
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Ari Keranen wrote, on 07/04/2011 12:24 PM:
> Hence, I think we do not need "tcp" options tag (and can keep the draft as it is). Opinions?

Makes sense. Seems in line with the spirit of ICE extensibility. +1

Go ahead and publish this RFC! :)

Simon
-- 
DTN made easy, lean, and smart --> http://postellation.viagenie.ca
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From jmpolk@cisco.com  Wed Jul 13 14:36:50 2011
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Date: Wed, 13 Jul 2011 16:36:46 -0500
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Cc: "Subha Dhesikan \(sdhesika\)" <sdhesika@cisco.com>, Paul Jones <paulej@cisco.com>
Subject: [MMUSIC] New rev of the Traffic Class attribute draft
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MMUSIC

We have recently submitted the next version of the traffic class 
attribute, here

http://www.ietf.org/internet-drafts/draft-polk-mmusic-traffic-class-for-sdp-02.txt

Most of the work in this version has to do with a newly inserted 
Section 2 (i.e., the old section 2 is now section 3).

This new section 2 creates and defines the framework for this 
attribute (something Christer mentioned he would like to see.

Also, several mentioned they wanted to see term or string definitions 
we were proposing that the last version lacked. This is in the new version.

With the addition to this new section, the biggest changes are as follows:

- we changed the attribute string have this format

parent.application(.adjective)(.adjective)(.admitted/non-admitted)

(this isn't the exact ABNF, which is in Section 3 now)

- We added that the parent and application components are a pairing.

- we added that there can be zero or more adjective components 
describing the traffic class

- we provide a list of each component for we review, i.e., separate 
lists of parent, application, adjective and cac-class components. 
Some of the application components are settle (i.e., we have one for 
Netflix type of traffic, but don't have an actual string picked yet).

Most of the rest of the changes were minor nits or changes to the new 
attribute structure.

Comments are really appreciated.

James, Paul & Subha


From Martin.Stiemerling@neclab.eu  Thu Jul 14 02:31:45 2011
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From: Martin Stiemerling <Martin.Stiemerling@neclab.eu>
To: "Roy Yue(PeiYu)" <yuepeiyu@huawei.com>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: Review Report of RTSP 2.0 draft
Thread-Index: AcwZ6O+T9jCll7vaQYC1JLTWCU/oVQoH4b+Q
Date: Thu, 14 Jul 2011 09:31:22 +0000
Message-ID: <E84E7B8FF3F2314DA16E48EC89AB49F01CEE088F@DAPHNIS.office.hd>
References: <E1BDDFCD18CF9748BAB4B7FAF2D532D925419C@SZXEML503-MBX.china.huawei.com>
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Subject: Re: [MMUSIC] Review Report of RTSP 2.0 draft
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<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#1F497D">Hi Ro=
y,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#1F497D"><o:p>=
&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#1F497D">Thank=
s for your thorough review of the RTSP 2.0 spec. The fixes have been applie=
d, as noted in the attached excel file.
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#1F497D"><o:p>=
&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#1F497D">There=
 will be an update of the RTSP 2.0 draft available as soon as the I-D submi=
ssion is open again.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#1F497D"><o:p>=
&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#1F497D">&nbsp=
; Martin<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#1F497D"><o:p>=
&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#1F497D"><o:p>=
&nbsp;</o:p></span></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0mm 0mm 0mm =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0mm =
0mm 0mm">
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><b><span st=
yle=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quo=
t;">From:</span></b><span style=3D"font-size:10.0pt;font-family:&quot;Tahom=
a&quot;,&quot;sans-serif&quot;"> mmusic-bounces@ietf.org [mailto:mmusic-bou=
nces@ietf.org]
<b>On Behalf Of </b>Roy Yue(PeiYu)<br>
<b>Sent:</b> Tuesday, May 24, 2011 10:03 AM<br>
<b>To:</b> mmusic@ietf.org<br>
<b>Cc:</b> Flemming Andreasen<br>
<b>Subject:</b> [MMUSIC] Review Report of RTSP 2.0 draft<o:p></o:p></span><=
/p>
</div>
</div>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><o:p>&nbsp;=
</o:p></p>
<p class=3D"MsoNormal">Hello Miguel, Flemming, et al,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">As promised during IETF #80, I have finished reviewi=
ng the whole RTSP 2.0 draft (i.e. draft-ietf-mmusic-rfc2326bis-27).<o:p></o=
:p></p>
<p class=3D"MsoNormal">Enclosed is the comment and editorial proposal I hav=
e, in a simple excel sheet.<o:p></o:p></p>
<p class=3D"MsoNormal">PS: should I put it in any specific format instead?<=
o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Among the comments, typos, bugs, the first 14 commen=
ts are more significant, while others are more obvious.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Best regards,<o:p></o:p></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><span style=
=3D"font-size:10.0pt;font-family:STXihei;color:black">Roy(</span><span lang=
=3D"ZH-CN" style=3D"font-size:10.0pt;font-family:SimSun;color:black">=C0=D6=
=C5=E0=D3=F1</span><span style=3D"font-size:10.0pt;font-family:STXihei;colo=
r:black">)<br>
</span><span style=3D"font-size:10.0pt;font-family:STXihei;color:gray">Huaw=
ei Software Technologies Co., Ltd
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pes=3D"ridImg"><![endif]><span style=3D"font-size:12.0pt;font-family:SimSun=
"><o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:12.0pt;font-family:SimSun">=
<br>
</span><span style=3D"font-size:7.5pt;font-family:STXihei;color:gray">N5, N=
O. 101, Huawei Base<br>
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Tel: &#43;86-25-56622058<br>
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<a href=3D"http://www.huawei.com">http://www.huawei.com</a><br>
---------------------------------------------------------------------------=
----------------------------------------------------------<br>
</span><span lang=3D"ZH-CN" style=3D"font-size:7.5pt;font-family:SimSun;col=
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From miguel.a.garcia@ericsson.com  Mon Jul 18 05:39:18 2011
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Cc: Flemming Andreasen <fandreas@cisco.com>
Subject: Re: [MMUSIC] Unofficial WGLC for	draft-salgueiro-mmusic-image-iana-registration-07.txt
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The 4-week unofficial WGLC period of this draft has ended.

We will ask the author to proceed with the individual submission 
(AD-sponsored).

/Miguel

On 16/06/2011 8:09, Miguel A. Garcia wrote:
> This is to start a 4-week unofficial WGLC of the IANA registration of the
> image media type in SDP,
>
> http://datatracker.ietf.org/doc/draft-salgueiro-mmusic-image-iana-registration/
>
> This document is going to proceed through the individually submitted,
> AD-sponsored track. However, the draft came as a requirements of the
> MMUSIC WG, has been substantially reviewed in MMUSIC, therefore the
> chairs want to have a last review in the format of an unofficial WGLC.
>
> Please review the draft, and send comments to the authors and the list.
> Even if you don't have comments because you are happy with the current
> document, please say so.
>
> The deadline for receiving comments is July 14th.
>
> /Miguel and Flemming

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

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For your information

Magnus Westerlund

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Subject: [MMUSIC] RFC 6336 on IANA Registry for Interactive Connectivity Establishment (ICE) Options
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A new Request for Comments is now available in online RFC libraries.

        
        RFC 6336

        Title:      IANA Registry for Interactive Connectivity 
                    Establishment (ICE) Options 
        Author:     M. Westerlund, C. Perkins
        Status:     Standards Track
        Stream:     IETF
        Date:       July 2011
        Mailbox:    magnus.westerlund@ericsson.com, 
                    csp@csperkins.org
        Pages:      5
        Characters: 8228
        Updates:    RFC5245

        I-D Tag:    draft-ietf-mmusic-ice-options-registry-02.txt

        URL:        http://www.rfc-editor.org/rfc/rfc6336.txt

It has been identified that "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT) Traversal for
Offer/Answer Protocols" (RFC 5245) is missing a registry for ICE
options.  This document defines this missing IANA registry and
updates RFC 5245.  [STANDARDS-TRACK]

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

This is now a Proposed Standard Protocol.

STANDARDS TRACK: This document specifies an Internet standards track
protocol for the Internet community,and requests discussion and suggestions
for improvements.  Please refer to the current edition of the Internet
Official Protocol Standards (STD 1) for the standardization state and
status of this protocol.  Distribution of this memo is unlimited.

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From fluffy@cisco.com  Wed Jul 20 07:36:10 2011
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Subject: [MMUSIC] Comment on draft-westerlund-avtcore-multistream-and-simulcast
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In the case were we want to use SDP to indicate video streams are =
simulcast of the same thing, I think it would be better to use the SDP =
grouping framework instead of using SSRC to indicate this.=20

I also wonder, why not just use layered codecs, seems like outperform =
the simulcast approach. I read section 3.1 but don't really find it =
convincing that this simulcast is better than layered codecs.=20


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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
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> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On =
Behalf Of Cullen Jennings (fluffy)
> Sent: Wednesday, July 20, 2011 4:36 PM
> To: IETF AVT WG; IETF MMUSIC WG
> Subject: [MMUSIC] Comment =
ondraft-westerlund-avtcore-multistream-and-simulcast
>=20
>=20
> In the case were we want to use SDP to indicate video streams are =
simulcast of the same thing, I think it would be better to
> use the SDP grouping framework instead of using SSRC to indicate this.

As in:
https://datatracker.ietf.org/doc/draft-begen-mmusic-redundancy-grouping/ =


> I also wonder, why not just use layered codecs, seems like outperform =
the simulcast approach. I read section 3.1 but don't
> really find it convincing that this simulcast is better than layered =
codecs.

The draft above provides some motivation why people would wanna transmit =
the same thing twice or more.
=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

From stewe@stewe.org  Wed Jul 20 08:26:50 2011
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From: Stephan Wenger <stewe@stewe.org>
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On 7.20.2011 07:36 , "Cullen Jennings" <fluffy@cisco.com> wrote:

> 
> In the case were we want to use SDP to indicate video streams are simulcast of
> the same thing, I think it would be better to use the SDP grouping framework
> instead of using SSRC to indicate this.
> 
> I also wonder, why not just use layered codecs, seems like outperform the
> simulcast approach. I read section 3.1 but don't really find it convincing
> that this simulcast is better than layered codecs.
> 


I read the word "scalable" -- can't help to chip in...

>From Magnus' draft, section 3.1, second-to-last paragraph characterizing
scalable coding, with inline comments:

> The scalable video encoding requires a more complex encoder compared
>    to non-scalable encoding.
> 
This is true from an implementation complexity viewpoint.  It is not
necessarily true from a cycle viewpoint, at least not in the encoder.  I
would argue that, in most practical cases, highly optimized scalable and
simulcast encoders require about the same number of cycles to code similar
resolutions.  Perhaps within a +/- 10% range.
In the decoder, modern standards such as H.264 SVC allow for something
called "single loop decoding", which keep the computational complexity at
only marginally higher levels than for non-scalable decoding of the highest
simulcasted solution.  There are cases where layered decoding actually takes
fewer cycles (at very high bitrates, where the overhead due to entropy
decoding of the less efficient simulcasted stream outweighs the complexity
through cross-layer prediction).
> 
> Also, if the resolution difference is big,
>    the scalable codec may in fact be only marginally more bandwidth
>    efficient, between the encoding client and the mixer, than a
>    simulcast that sends the resolutions in separate streams, assuming
>    equivalent video quality.
> 
This is correct, but it's only half of the story.  One operation point for
which most scalable codecs today are optimized is 2:1 spatial scalability,
meaning that the base layer is half the resolution in both dimensions or 1/4
of the pixel count.  Another one is (roughly) 1.5 scalability, with 720p to
1080p in mind.  SVC works still reasonably well at 1:3, although this is not
a common operation point.
The second half of the story is that SVC (as well as previous layered coding
standards) do allow for multilayer scalability.  If you know how to
implement this (it's not trivial), you get quite a bit of the gain, with the
additional advantage of having intermediate operation points that you can
use when video pictures are zoomed under user control.
In either case, modern scalable video standards, well implemented, are
always better than simulcast from a bandwidth viewpoint.  And should be.
That's information theory 101.

> 
> At the same time, with scalable video
>    encoding, the transmission of all but the lowest resolution will
>    definitely consume more bandwidth from the mixer to the other
>    participants than a non-scalable encoding, again assuming equivalent
>    video quality.

Again, this is correct.  Once again, though, only part of the story.
First, he overhead of scalable coding over non-scalable coding of the
highest resolution has decreased to levels that may not quite be background
noise, but not much higher either.  You may wish to take a look at a
document Alex Eleftheriadis and me wrote for rtcweb,
http://tools.ietf.org/html/draft-wenger-rtcweb-layered-codec-00.  It
contains a reference to the characterization tests conducted by MPEG.  Most
tests performed in academia roughly mirror those results.  Second, most of
us sit behind highly asymmetric links, such as cable modem or DSL.  In such
environments, an overhead (even a large one) for the largest picture
resolution is less penalizing than an overhead on the sending side.  Sending
bits are more expensive than receiving bits.

What the draft has omitted are SVC's superior error resilience properties
and congestion control options, when being used in an appropriately designed
system environment.  Some of that stems from temporal scalability (which
modern non-scalable video coding standards include in an occasionally
crippled form), but a lot more comes from spatial scalability.  Alex and my
draft cited above provides some input.

Regards,
Stephan
  




> ___________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
> 



--B_3393995204_1741896
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	charset="US-ASCII"
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<html><head></head><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: s=
pace; -webkit-line-break: after-white-space; color: rgb(0, 0, 0); font-size:=
 14px; font-family: Calibri, sans-serif; "><div>On 7.20.2011 07:36 , "Cullen=
 Jennings" &lt;<a href=3D"mailto:fluffy@cisco.com">fluffy@cisco.com</a>&gt; wr=
ote:</div><div><br></div><blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE"=
 style=3D"border-left-color: rgb(181, 196, 223); border-left-width: 5px; borde=
r-left-style: solid; padding-top: 0px; padding-right: 0px; padding-bottom: 0=
px; padding-left: 5px; margin-top: 0px; margin-right: 0px; margin-bottom: 0p=
x; margin-left: 5px; "><div><br></div><div>In the case were we want to use S=
DP to indicate video streams are simulcast of the same thing, I think it wou=
ld be better to use the SDP grouping framework instead of using SSRC to indi=
cate this.</div><div><br></div><div>I also wonder, why not just use layered =
codecs, seems like outperform the simulcast approach. I read section 3.1 but=
 don't really find it convincing that this simulcast is better than layered =
codecs.</div><div><br></div></blockquote><div><hr align=3D"center" size=3D"&quot=
;3&quot;" width=3D"&quot;95%&quot;"></div><div><br></div><div>I read the word =
"scalable" -- can't help to chip in...</div><div><br></div><div>From Magnus'=
 draft, section 3.1, second-to-last paragraph characterizing scalable coding=
, with inline comments:</div><div><br></div><div><div><blockquote style=3D"mar=
gin:0 0 0 40px; border:none; padding:0px;"><div>The scalable video encoding =
requires a more complex encoder compared</div><div>   to non-scalable encodi=
ng.&nbsp;</div><div><br></div></blockquote><div>This is true from an impleme=
ntation complexity viewpoint. &nbsp;It is not necessarily true from a cycle =
viewpoint, at least not in the encoder. &nbsp;I would argue that, in most pr=
actical cases, highly optimized scalable and simulcast encoders require abou=
t the same number of cycles to code similar resolutions. &nbsp;Perhaps withi=
n a +/- 10% range.</div><div>In the decoder, modern standards such as H.264 =
SVC allow for something called "single loop decoding", which keep the comput=
ational complexity at only marginally higher levels than for non-scalable de=
coding of the highest simulcasted solution. &nbsp;There are cases where laye=
red decoding actually takes fewer cycles (at very high bitrates, where the o=
verhead due to entropy decoding of the less efficient simulcasted stream out=
weighs the complexity through cross-layer prediction).</div><blockquote styl=
e=3D"margin:0 0 0 40px; border:none; padding:0px;"><div><br></div><div>Also, i=
f the resolution difference is big,</div><div>   the scalable codec may in f=
act be only marginally more bandwidth</div><div>   efficient, between the en=
coding client and the mixer, than a</div><div>   simulcast that sends the re=
solutions in separate streams, assuming</div><div>   equivalent video qualit=
y.&nbsp;</div><div><br></div></blockquote><div>This is correct, but it's onl=
y half of the story. &nbsp;One operation point for which most scalable codec=
s today are optimized is 2:1 spatial scalability, meaning that the base laye=
r is half the resolution in both dimensions or 1/4 of the pixel count. &nbsp=
;Another one is (roughly) 1.5 scalability, with 720p to 1080p in mind. &nbsp=
;SVC works still reasonably well at 1:3, although this is not a common opera=
tion point.</div><div>The second half of the story is that SVC (as well as p=
revious layered coding standards) do allow for multilayer scalability. &nbsp=
;If you know how to implement this (it's not trivial), you get quite a bit o=
f the gain, with the additional advantage of having intermediate operation p=
oints that you can use when video pictures are zoomed under user control.</d=
iv><div>In either case, modern scalable video standards, well implemented, a=
re always better than simulcast from a bandwidth viewpoint. &nbsp;And should=
 be. &nbsp;That's information theory 101.</div><div><br></div><blockquote st=
yle=3D"margin:0 0 0 40px; border:none; padding:0px;"><div><br></div><div>At th=
e same time, with scalable video</div><div>   encoding, the transmission of =
all but the lowest resolution will</div><div>   definitely consume more band=
width from the mixer to the other</div><div>   participants than a non-scala=
ble encoding, again assuming equivalent</div><div>   video quality.</div></b=
lockquote><div><br></div><div>Again, this is correct. &nbsp;Once again, thou=
gh, only part of the story.</div><div>First, he overhead of scalable coding =
over non-scalable coding of the highest resolution has decreased to levels t=
hat may not quite be background noise, but not much higher either. &nbsp;You=
 may wish to take a look at a document Alex Eleftheriadis and me wrote for r=
tcweb,&nbsp;http://tools.ietf.org/html/draft-wenger-rtcweb-layered-codec-00.=
 &nbsp;It contains a reference to the characterization tests conducted by MP=
EG. &nbsp;Most tests performed in academia roughly mirror those results. &nb=
sp;Second, most of us sit behind highly asymmetric links, such as cable mode=
m or DSL. &nbsp;In such environments, an overhead (even a large one) for the=
 largest picture resolution is less penalizing than an overhead on the sendi=
ng side. &nbsp;Sending bits are more expensive than receiving bits.</div><di=
v><br></div><div>What the draft has omitted are SVC's superior error resilie=
nce properties and congestion control options, when being used in an appropr=
iately designed system environment. &nbsp;Some of that stems from temporal s=
calability (which modern non-scalable video coding standards include in an o=
ccasionally crippled form), but a lot more comes from spatial scalability. &=
nbsp;Alex and my draft cited above provides some input.</div><div><br></div>=
<div>Regards,</div><div>Stephan</div><div>&nbsp;&nbsp;</div></div><div><br><=
/div></div><div><br></div><div><br></div><div><br></div><blockquote id=3D"MAC_=
OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT: #b5c4df 5 solid; PADDING=
:0 0 0 5; MARGIN:0 0 0 5;"><div>___________________________</div><div>Audio/=
Video Transport Core Maintenance</div><div><a href=3D"mailto:avt@ietf.org">avt=
@ietf.org</a></div><div><a href=3D"https://www.ietf.org/mailman/listinfo/avt">=
https://www.ietf.org/mailman/listinfo/avt</a></div><div><br></div></blockquo=
te></body></html>

--B_3393995204_1741896--



From magnus.westerlund@ericsson.com  Thu Jul 21 02:24:48 2011
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Subject: Re: [MMUSIC] Comment on draft-westerlund-avtcore-multistream-and-simulcast
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On 2011-07-20 16:36, Cullen Jennings wrote:
> 
> In the case were we want to use SDP to indicate video streams are
> simulcast of the same thing, I think it would be better to use the
> SDP grouping framework instead of using SSRC to indicate this.

Cullen, I recommend that you go read the draft again.

We are proposing to use SDP grouping framework. We suggest that one
defines two grouping semantics SCS and SCR. One for grouping media lines
where one want to send different simulcast versions, another for
indicating the willingness to receive simulcast over a group of media
lines.

This is to enable simulcast only in one direction and to ensure that we
get offer/answer behavior. The grouping semantics is what is used to
negotiate the actual usage of simulcast or not.

We do have the SRCNAME but that is for actually indicating how different
streams map to different sources. This becomes important when an
end-point has more than one source of the same media type it want to
simulcast, like for example multiple cameras in a telepresence room.

The SRCNAME SDES item doesn't need to be in the SDP, it is only included
in the examples to show that it is possible to have them to speed up
acquisition of this information.

> 
> I also wonder, why not just use layered codecs, seems like outperform
> the simulcast approach. I read section 3.1 but don't really find it
> convincing that this simulcast is better than layered codecs.

Layered coding is less efficient to reach a particular quality level
than a stream that is encoded for that particular level without any
additional quality levels.

This means that layered encoding is guaranteed to be less efficient in
the leg from a central mixer to a receiver for selecting a given media
quality.

Layered encoding is however a bit more efficient in the leg from the
media source to the mixer than simulcasting.

The exact levels of gain varies with the target points, the number of
qualities etc.

We see that a clear use case for combining simulcast with codecs that
has certain scalability features. For example using H.264/AVC or SVC to
provide temporal scalability, but use simulcast for resolution changes.


Cheers

Magnus Westerlund

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


From stephen.botzko@gmail.com  Thu Jul 21 04:12:46 2011
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Subject: Re: [MMUSIC] [AVTCORE] Comment on draft-westerlund-avtcore-multistream-and-simulcast
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in-line
Stephen Botzko

On Thu, Jul 21, 2011 at 5:24 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> On 2011-07-20 16:36, Cullen Jennings wrote:
> >
> > In the case were we want to use SDP to indicate video streams are
> > simulcast of the same thing, I think it would be better to use the
> > SDP grouping framework instead of using SSRC to indicate this.
>
> Cullen, I recommend that you go read the draft again.
>
> We are proposing to use SDP grouping framework. We suggest that one
> defines two grouping semantics SCS and SCR. One for grouping media lines
> where one want to send different simulcast versions, another for
> indicating the willingness to receive simulcast over a group of media
> lines.
>
> This is to enable simulcast only in one direction and to ensure that we
> get offer/answer behavior. The grouping semantics is what is used to
> negotiate the actual usage of simulcast or not.
>
> We do have the SRCNAME but that is for actually indicating how different
> streams map to different sources. This becomes important when an
> end-point has more than one source of the same media type it want to
> simulcast, like for example multiple cameras in a telepresence room.
>
> The SRCNAME SDES item doesn't need to be in the SDP, it is only included
> in the examples to show that it is possible to have them to speed up
> acquisition of this information.
>
> >
> > I also wonder, why not just use layered codecs, seems like outperform
> > the simulcast approach. I read section 3.1 but don't really find it
> > convincing that this simulcast is better than layered codecs.
>
> Layered coding is less efficient to reach a particular quality level
> than a stream that is encoded for that particular level without any
> additional quality levels.
>
> This means that layered encoding is guaranteed to be less efficient in
> the leg from a central mixer to a receiver for selecting a given media
> quality.
>
> Layered encoding is however a bit more efficient in the leg from the
> media source to the mixer than simulcasting.
>
> The exact levels of gain varies with the target points, the number of
> qualities etc.
>
> We see that a clear use case for combining simulcast with codecs that
> has certain scalability features. For example using H.264/AVC or SVC to
> provide temporal scalability, but use simulcast for resolution changes.
>
> I agree that simulcast is important, including the case where two differe=
nt
codecs are used.  In that scenario, the resolution might be the same.

>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
>

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in-line<br>Stephen Botzko<br><br><div class=3D"gmail_quote">On Thu, Jul 21,=
 2011 at 5:24 AM, Magnus Westerlund <span dir=3D"ltr">&lt;<a href=3D"mailto=
:magnus.westerlund@ericsson.com">magnus.westerlund@ericsson.com</a>&gt;</sp=
an> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;"><div class=3D"im">On 2011-07-20 16:36, Cull=
en Jennings wrote:<br>
&gt;<br>
&gt; In the case were we want to use SDP to indicate video streams are<br>
&gt; simulcast of the same thing, I think it would be better to use the<br>
&gt; SDP grouping framework instead of using SSRC to indicate this.<br>
<br>
</div>Cullen, I recommend that you go read the draft again.<br>
<br>
We are proposing to use SDP grouping framework. We suggest that one<br>
defines two grouping semantics SCS and SCR. One for grouping media lines<br=
>
where one want to send different simulcast versions, another for<br>
indicating the willingness to receive simulcast over a group of media<br>
lines.<br>
<br>
This is to enable simulcast only in one direction and to ensure that we<br>
get offer/answer behavior. The grouping semantics is what is used to<br>
negotiate the actual usage of simulcast or not.<br>
<br>
We do have the SRCNAME but that is for actually indicating how different<br=
>
streams map to different sources. This becomes important when an<br>
end-point has more than one source of the same media type it want to<br>
simulcast, like for example multiple cameras in a telepresence room.<br>
<br>
The SRCNAME SDES item doesn&#39;t need to be in the SDP, it is only include=
d<br>
in the examples to show that it is possible to have them to speed up<br>
acquisition of this information.<br>
<div class=3D"im"><br>
&gt;<br>
&gt; I also wonder, why not just use layered codecs, seems like outperform<=
br>
&gt; the simulcast approach. I read section 3.1 but don&#39;t really find i=
t<br>
&gt; convincing that this simulcast is better than layered codecs.<br>
<br>
</div>Layered coding is less efficient to reach a particular quality level<=
br>
than a stream that is encoded for that particular level without any<br>
additional quality levels.<br>
<br>
This means that layered encoding is guaranteed to be less efficient in<br>
the leg from a central mixer to a receiver for selecting a given media<br>
quality.<br>
<br>
Layered encoding is however a bit more efficient in the leg from the<br>
media source to the mixer than simulcasting.<br>
<br>
The exact levels of gain varies with the target points, the number of<br>
qualities etc.<br>
<br>
We see that a clear use case for combining simulcast with codecs that<br>
has certain scalability features. For example using H.264/AVC or SVC to<br>
provide temporal scalability, but use simulcast for resolution changes.<br>
<br></blockquote><div>I agree that simulcast is important, including the ca=
se where two different codecs are used.=A0 In that scenario, the resolution=
 might be the same. <br></div><blockquote class=3D"gmail_quote" style=3D"ma=
rgin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding=
-left: 1ex;">

<br>
Cheers<br>
<br>
Magnus Westerlund<br>
<br>
----------------------------------------------------------------------<br>
Multimedia Technologies, Ericsson Research EAB/TVM<br>
----------------------------------------------------------------------<br>
Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0<a href=3D"tel:%2B46%=
2010%207148287" value=3D"+46107148287">+46 10 7148287</a><br>
F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile <a href=3D"tel:%2B4=
6%2073%200949079" value=3D"+46730949079">+46 73 0949079</a><br>
SE-164 80 Stockholm, Sweden| mailto: <a href=3D"mailto:magnus.westerlund@er=
icsson.com">magnus.westerlund@ericsson.com</a><br>
----------------------------------------------------------------------<br>
<br>
_______________________________________________<br>
<div class=3D"im">Audio/Video Transport Core Maintenance<br>
</div><a href=3D"mailto:avt@ietf.org">avt@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/avt" target=3D"_blank">htt=
ps://www.ietf.org/mailman/listinfo/avt</a><br>
</blockquote></div><br>

--bcaec548aa5799741904a8926dc6--

From magnus.westerlund@ericsson.com  Sat Jul 23 03:58:32 2011
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Subject: [MMUSIC] Comments on draft-begen-mmusic-redundancy-grouping-01
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Hi,

(Primarily as individual, but the WG home question is as WG chair)

I have reviewed this draft and some comments on it.

1. If this solution proposal is to be taken forward I think it needs to
be discussed in AVTCORE. The reason is that it has several RTP details
that needs to be figured out to make stream duplication issue free in a
general usage. I will take up a few of these issues in the below comments.

2. One aspect this document has is a relation to what is discussed in
draft-westerlund-avtcore-multistream-and-simulcast-00. Stream
duplication both results in a certain form of multi-stream and also has
a similar but slightly different semantics to simulcast in that the
streams are two representations of a single original media source. In
this case they are clones, but not true duplicates due to that RTP
doesn't handle try duplicates well.

3. Another issue that also the authors has identified is the question of
what one do when one has multiple media streams from the same end-point.
And in a general case one can't use the CNAME to separate the sources.
The reason is that two media sources that has a common timing
(synchronization) context do needs to indicate that by having the same
CNAME. Thus there are need for some additional mechanism to ensure that
SSRC carrying RTP media streams that are duplicates of each other do in
fact be labeled explicitly. I also note that a SDP based solution is not
sufficient to resolve this issue. I think we need something in RTP, so
that SSRC collisions can be handled.

4. In addition there is a few questions how this duplication really
should set some RTP fields. Is the duplication only working on Sequence
number level? There seem to be additional questions, for example in the
need to have RTP timestamp considerations between the streams. They need
to point identically into the joint timeline. Otherwise outages, or
receiver taking synch from one and not the other stream can end up with
different synchronizations. Becomes especially interesting to discuss
when the time shifted transmission.

5. The draft also discussed Payload Type differences and have them in
the SDP examples. I don't see how that is appropriate. If the streams
are true encoding duplicates then they should use the same payload type
configuration, and thus a single payload type should be sufficient and
in fact recommended, unless the source encoding truly changes.

6. I think it is worth discussing also the benefits of SSRC multiplexed
versus the session multiplexed. In multicast context there is clear
benefits from a control and congestion perspective to have session
multiplexed versions and these mapped onto different multicast groups
enabling a receiver to choose one or both streams.

7. Thus I think the signalling aspects of this is the minor part.
Although there clearly are some aspects of this needing discussion. For
example clearer offer/answer rules for handling these.

Cheers

Magnus Westerlund

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


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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>, "mmusic (E-mail)" <mmusic@ietf.org>, "IETF AVTCore WG" <avt@ietf.org>
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Subject: Re: [MMUSIC] [AVTCORE] Comments on draft-begen-mmusic-redundancy-grouping-01
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> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of =
Magnus Westerlund
> Sent: Friday, July 22, 2011 8:39 AM
> To: mmusic (E-mail); IETF AVTCore WG
> Subject: [AVTCORE] Comments on =
draft-begen-mmusic-redundancy-grouping-01
>=20
> Hi,
>=20
> (Primarily as individual, but the WG home question is as WG chair)
>=20
> I have reviewed this draft and some comments on it.

Thanks for the review.
=20
> 1. If this solution proposal is to be taken forward I think it needs =
to
> be discussed in AVTCORE. The reason is that it has several RTP details
> that needs to be figured out to make stream duplication issue free in =
a
> general usage. I will take up a few of these issues in the below =
comments.

That is why we asked for a slot in avtcore on wed.
=20
> 2. One aspect this document has is a relation to what is discussed in
> draft-westerlund-avtcore-multistream-and-simulcast-00. Stream
> duplication both results in a certain form of multi-stream and also =
has
> a similar but slightly different semantics to simulcast in that the
> streams are two representations of a single original media source. In
> this case they are clones, but not true duplicates due to that RTP
> doesn't handle try duplicates well.

The data carried in the redundant streams are identical but the rtp =
header differs due to the different SSRCs. That is all. SSRC muxing =
nicely fits the use case we are tackling with. Simulcasting is quite =
different from our use case because in simulcasting different RTP =
streams most likely carry different data.
=20
> 3. Another issue that also the authors has identified is the question =
of
> what one do when one has multiple media streams from the same =
end-point.

>From the same end-point (e.g., source) there could of course of multiple =
media streams. You can use SSRC muxing for example to identify your =
streams and use ssrc-group to group them if needed.

The concern in that editorial note was what if there were multiple =
streams in each m line but what if you wanted grouping/association among =
only a subset.

For example,

        m=3Dvideo 30000 RTP/AVP 101 102
        ....
        m=3Dvideo 30000 RTP/AVP 103 104

I can easily associate 101 and 102 if I want to, but what if 101 and 103 =
were associated with each other. Can I use a=3Dgroup line? Probably not. =
But, I don't know why you would need that kind of relation in the first =
place (even if you need, there are probably less messy ways to express =
them). If that had been a very desirable feature, SDP and grouping =
semantics along with RTP would not have defined in their current form I =
guess.

> And in a general case one can't use the CNAME to separate the sources.
> The reason is that two media sources that has a common timing
> (synchronization) context do needs to indicate that by having the same
> CNAME. Thus there are need for some additional mechanism to ensure =
that

If 101 and 103 are associated, why can't I use the same CNAME for those =
streams? If 102 and 104 are associated/related, I would use a common =
CNAME for them, too (but theirs would be separate from 101 and 103's).

> SSRC carrying RTP media streams that are duplicates of each other do =
in
> fact be labeled explicitly. I also note that a SDP based solution is =
not
> sufficient to resolve this issue. I think we need something in RTP, so
> that SSRC collisions can be handled.

I can't say I am following.

> 4. In addition there is a few questions how this duplication really
> should set some RTP fields. Is the duplication only working on =
Sequence

That is the easiest way for a middle box or endpoint doing the stream =
merging (duplicate suppression). If seqnums are different, what would =
one do? look into the payload?

> number level? There seem to be additional questions, for example in =
the
> need to have RTP timestamp considerations between the streams. They =
need
> to point identically into the joint timeline. Otherwise outages, or
> receiver taking synch from one and not the other stream can end up =
with
> different synchronizations. Becomes especially interesting to discuss
> when the time shifted transmission.

The duplicate streams are coming from the same source, so they have the =
same timestamp as well as seqnums. It has to be this way because the =
node merging the streams will not re-write anything at all. It is a =
simple suppression operation. Beauty of it is that things stay pretty =
simple to use and implement without any changes to RTP/RTCP plane at =
all.

As for the time-shifted transmission, I suppose you are referring to the =
temporal diversity/delayed duplication =
(draft-begen-mmusic-temporal-interleaving-02). Well, in that case, =
timestamps are still the same for the reasons above. Changing the =
timestamps for the delayed packets does not bring any benefit (actually =
it complicates things) since the ultimate RTP application will still =
receive one and only one RTP stream.

> 5. The draft also discussed Payload Type differences and have them in
> the SDP examples. I don't see how that is appropriate. If the streams

When you are doing SSRC muxing, you put multiple streams in the same m =
line and they end up having different PT numbers. What is wrong with =
that? If they are in separate m lines, they could have the same PT =
number.

> are true encoding duplicates then they should use the same payload =
type
> configuration, and thus a single payload type should be sufficient and
> in fact recommended, unless the source encoding truly changes.

See above.

> 6. I think it is worth discussing also the benefits of SSRC =
multiplexed
> versus the session multiplexed. In multicast context there is clear
> benefits from a control and congestion perspective to have session
> multiplexed versions and these mapped onto different multicast groups
> enabling a receiver to choose one or both streams.

This is what we have in section 5.2. Different streams can be sent in =
different multicast sessions but it does not mean that they are separate =
RTP sessions. I discussed this in detail with Colin. Thanks to him for =
trying to teach me over and over what an RTP session actually means.
=20
> 7. Thus I think the signalling aspects of this is the minor part.
> Although there clearly are some aspects of this needing discussion. =
For
> example clearer offer/answer rules for handling these.

We have some text around O/A stuff. But frankly, I don=92t see any use =
case where duplication will be used in a unicast application. I mean if =
redundancy is needed, one should use FEC instead for unicast transport. =
Duplication is a nice use case for multicast transport where the bw =
overhead is easier to justify and the implementation is easier. And one =
would naturally implement this spec in networks where he already knows =
that the network elements support this spec. So, maybe we should remove =
the whole O/A text anyway and write that this will only be used =
declaratively.

-acbegen
=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt

From magnus.westerlund@ericsson.com  Mon Jul 25 04:54:35 2011
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Subject: Re: [MMUSIC] [AVTCORE] Comments on draft-begen-mmusic-redundancy-grouping-01
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On 2011-07-24 23:21, Ali C. Begen (abegen) wrote:
> 
> 
>> -----Original Message----- From: avt-bounces@ietf.org
>> [mailto:avt-bounces@ietf.org] On Behalf Of Magnus Westerlund Sent:
>> Friday, July 22, 2011 8:39 AM To: mmusic (E-mail); IETF AVTCore WG 
>> Subject: [AVTCORE] Comments on
>> draft-begen-mmusic-redundancy-grouping-01
>> 
>> Hi,
>> 
>> (Primarily as individual, but the WG home question is as WG chair)
>> 
>> I have reviewed this draft and some comments on it.
> 
> Thanks for the review.
> 
>> 1. If this solution proposal is to be taken forward I think it
>> needs to be discussed in AVTCORE. The reason is that it has several
>> RTP details that needs to be figured out to make stream duplication
>> issue free in a general usage. I will take up a few of these issues
>> in the below comments.
> 
> That is why we asked for a slot in avtcore on wed.
> 
>> 2. One aspect this document has is a relation to what is discussed
>> in draft-westerlund-avtcore-multistream-and-simulcast-00. Stream 
>> duplication both results in a certain form of multi-stream and also
>> has a similar but slightly different semantics to simulcast in that
>> the streams are two representations of a single original media
>> source. In this case they are clones, but not true duplicates due
>> to that RTP doesn't handle try duplicates well.
> 
> The data carried in the redundant streams are identical but the rtp
> header differs due to the different SSRCs. That is all. SSRC muxing
> nicely fits the use case we are tackling with. Simulcasting is quite
> different from our use case because in simulcasting different RTP
> streams most likely carry different data.

I agree that the purpose for multiple RTP streams from the same original
media source are different from the primary case I set out to solve. But
as Jonathan Lennox pointed out, redundancy and FEC have very similar
behaviors to simulcast with different encodings of the same original
source.

Thus, I do see a need to discuss for example how you identify the
streams that are a duplicate of each other.

> 
>> 3. Another issue that also the authors has identified is the
>> question of what one do when one has multiple media streams from
>> the same end-point.
> 
>> From the same end-point (e.g., source) there could of course of
>> multiple media streams. You can use SSRC muxing for example to
>> identify your streams and use ssrc-group to group them if needed.
> 
> The concern in that editorial note was what if there were multiple
> streams in each m line but what if you wanted grouping/association
> among only a subset.
> 
> For example,
> 
> m=video 30000 RTP/AVP 101 102 .... m=video 30000 RTP/AVP 103 104
> 
> I can easily associate 101 and 102 if I want to, but what if 101 and
> 103 were associated with each other. Can I use a=group line? Probably
> not. But, I don't know why you would need that kind of relation in
> the first place (even if you need, there are probably less messy ways
> to express them). If that had been a very desirable feature, SDP and
> grouping semantics along with RTP would not have defined in their
> current form I guess.

Yes, that is a very valid question also for simulcast.

And I think we can provide a story for doing that. But I don't think it
should be based on using the RTP payload type to identify either of the
versions.

> 
>> And in a general case one can't use the CNAME to separate the
>> sources. The reason is that two media sources that has a common
>> timing (synchronization) context do needs to indicate that by
>> having the same CNAME. Thus there are need for some additional
>> mechanism to ensure that
> 
> If 101 and 103 are associated, why can't I use the same CNAME for
> those streams? If 102 and 104 are associated/related, I would use a
> common CNAME for them, too (but theirs would be separate from 101 and
> 103's).

If SSRC A has a duplciation in SSRC C and SSRC B has a duplication in
SSRC D and both A and B comes from the same end-point and have a common
synchronization context, then all of A-D needs to have the same CNAME.

And I get the impression that you are using a payload type based
association scheme here. And I think that is a bad idea. For example it
can't support particular many streams. It also looks semantically wrong,
as if Stream A uses PT 96, then SSRC C should use PT 96 also for the
corresponding packet duplicates. And if A switches from one PT to
another the other stream needs to follow.

Thus I think a SDES item based approach for indicating that it is in
fact A and C that are duplicates, rather than A and B or A and D.

> 
>> SSRC carrying RTP media streams that are duplicates of each other
>> do in fact be labeled explicitly. I also note that a SDP based
>> solution is not sufficient to resolve this issue. I think we need
>> something in RTP, so that SSRC collisions can be handled.
> 
> I can't say I am following.

I suggest that you use something like the SRCNAME SDES item extension I
propose in my draft.

> 
>> 4. In addition there is a few questions how this duplication
>> really should set some RTP fields. Is the duplication only working
>> on Sequence
> 
> That is the easiest way for a middle box or endpoint doing the stream
> merging (duplicate suppression). If seqnums are different, what would
> one do? look into the payload?

It was simply that you had not documented what needs to be the same.
Sequence number being the same is likely the easiest, but one do have to
raise some warning flags if one has processing that changes these
streams. For example a translator modifying the packetization must do
the exact same manipulation on the other stream.

But, I don't think anything else really works.

> 
>> number level? There seem to be additional questions, for example in
>> the need to have RTP timestamp considerations between the streams.
>> They need to point identically into the joint timeline. Otherwise
>> outages, or receiver taking synch from one and not the other stream
>> can end up with different synchronizations. Becomes especially
>> interesting to discuss when the time shifted transmission.
> 
> The duplicate streams are coming from the same source, so they have
> the same timestamp as well as seqnums. It has to be this way because
> the node merging the streams will not re-write anything at all. It is
> a simple suppression operation. Beauty of it is that things stay
> pretty simple to use and implement without any changes to RTP/RTCP
> plane at all.

Yes, but you hadn't made it clear, and you have both SSRC and implied
change of PT. So I didn't want to assume anything.

The timestamp field also have implcations due to its binding with RTCP
SR information.

> 
> As for the time-shifted transmission, I suppose you are referring to
> the temporal diversity/delayed duplication
> (draft-begen-mmusic-temporal-interleaving-02). Well, in that case,
> timestamps are still the same for the reasons above. Changing the
> timestamps for the delayed packets does not bring any benefit
> (actually it complicates things) since the ultimate RTP application
> will still receive one and only one RTP stream.

Ok, but you will need to make clear on how one should set the RTCP SR
timestamp NTP mapping to be clear. And what that implies for the RTCP
jitter calculation.

> 
>> 5. The draft also discussed Payload Type differences and have them
>> in the SDP examples. I don't see how that is appropriate. If the
>> streams
> 
> When you are doing SSRC muxing, you put multiple streams in the same
> m line and they end up having different PT numbers. What is wrong
> with that? If they are in separate m lines, they could have the same
> PT number.

I am still asking why, there is no reason to use different PTs, unless
you are doing the association between the different streams by them
having different PTs. Which I believe is a Bad Idea (TM). The encoding
in the packet are the same, so why don't they have the same PT.


> 
>> 6. I think it is worth discussing also the benefits of SSRC
>> multiplexed versus the session multiplexed. In multicast context
>> there is clear benefits from a control and congestion perspective
>> to have session multiplexed versions and these mapped onto
>> different multicast groups enabling a receiver to choose one or
>> both streams.
> 
> This is what we have in section 5.2. Different streams can be sent in
> different multicast sessions but it does not mean that they are
> separate RTP sessions. I discussed this in detail with Colin. Thanks
> to him for trying to teach me over and over what an RTP session
> actually means.

Yes, you have an example, but not a discussion of why an implementor
should choose Session multiplexing over SSRC. I think that should be
included, or I think you can wait for the architectural document I think
will result from my multi-stream and simulcast draft.

> 
>> 7. Thus I think the signalling aspects of this is the minor part. 
>> Although there clearly are some aspects of this needing discussion.
>> For example clearer offer/answer rules for handling these.
> 
> We have some text around O/A stuff. But frankly, I don’t see any use
> case where duplication will be used in a unicast application. I mean
> if redundancy is needed, one should use FEC instead for unicast
> transport. Duplication is a nice use case for multicast transport
> where the bw overhead is easier to justify and the implementation is
> easier. And one would naturally implement this spec in networks where
> he already knows that the network elements support this spec. So,
> maybe we should remove the whole O/A text anyway and write that this
> will only be used declaratively.

Then, I think we should slap an application statement to the beginning
of the draft to ensure that it may only be used in multicast. For
example b=TIAS was only intended to used in declarative SDPs, but it has
become common in O/A, but it doesn't have clear rules.

So I think it imporant that one don't assume to much. Yes, it might not
be the best idea for unicast, but that is not going to prevent it from
being used.

Cheers

Magnus Westerlund

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


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Date: Mon, 25 Jul 2011 15:26:05 +0300
From: Varun Singh <vsingh.ietf@gmail.com>
To: IETF MMUSIC WG <mmusic@ietf.org>
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Subject: [MMUSIC] Multipath RTP (interface advertisement in SDP)
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Greetings,

The draft is part of AVTCore but we have recently introduced an "interface" attribute in SDP, as an alternative to in-band interface advertisement. The draft also contains some O/A examples and these changes can be found in the SDP Considerations section. http://tools.ietf.org/rfcdiff?url2=draft-singh-avtcore-mprtp-02.txt

Comments and suggestions are welcome from MMUSIC. 

Regards,
Varun

Begin forwarded message:
> 
> A new version of I-D, draft-singh-avtcore-mprtp-02.txt has been successfully submitted by Varun Singh and posted to the IETF repository.
> 
> Filename: draft-singh-avtcore-mprtp
> Revision: 02
> Title: Multipath RTP (MPRTP)
> Creation date: 2011-07-08
> WG ID: Individual Submission
> Number of pages: 34
> 
> Abstract:
> The Real-time Transport Protocol (RTP) is used to deliver real-time
> content and, along with the RTP Control Protocol (RTCP), forms the
> control channel between the sender and receiver. However, RTP and
> RTCP assume a single delivery path between the sender and receiver
> and make decisions based on the measured characteristics of this
> single path. Increasingly, endpoints are becoming multi-homed, which
> means that they are connected via multiple Internet paths. Network
> utilization can be improved when endpoints use multiple parallel
> paths for communication. The resulting increase in reliability and
> throughput can also enhance the user experience. This document
> extends the Real-time Transport Protocol (RTP) so that a single
> session can take advantage of the availability of multiple paths
> between two endpoints.
> 
> 
> 
> 
> The IETF Secretariat
-- 

http://www.netlab.tkk.fi/~varun



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Subject: Re: [MMUSIC] [AVTCORE] Comments on draft-begen-mmusic-redundancy-grouping-01
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One small comment, basically agreeing with Magnus point about payload =
types.

Usually when people use payload identifiers to do demultiplexing they =
later regret it. Payload identifiers (in RTP or any other protocol for =
that matter) should only be used to decide which parser to hand the data =
to, not which state variable to bind the data in the packet to.=20

(further aside: this is one of the few things the OSI guys actually got =
right when they had both payload ids and SAPs at every layer).

DaveO.


On Jul 25, 2011, at 6:54 AM, Magnus Westerlund wrote:

> On 2011-07-24 23:21, Ali C. Begen (abegen) wrote:
>>=20
>>=20
>>> -----Original Message----- From: avt-bounces@ietf.org
>>> [mailto:avt-bounces@ietf.org] On Behalf Of Magnus Westerlund Sent:
>>> Friday, July 22, 2011 8:39 AM To: mmusic (E-mail); IETF AVTCore WG=20=

>>> Subject: [AVTCORE] Comments on
>>> draft-begen-mmusic-redundancy-grouping-01
>>>=20
>>> Hi,
>>>=20
>>> (Primarily as individual, but the WG home question is as WG chair)
>>>=20
>>> I have reviewed this draft and some comments on it.
>>=20
>> Thanks for the review.
>>=20
>>> 1. If this solution proposal is to be taken forward I think it
>>> needs to be discussed in AVTCORE. The reason is that it has several
>>> RTP details that needs to be figured out to make stream duplication
>>> issue free in a general usage. I will take up a few of these issues
>>> in the below comments.
>>=20
>> That is why we asked for a slot in avtcore on wed.
>>=20
>>> 2. One aspect this document has is a relation to what is discussed
>>> in draft-westerlund-avtcore-multistream-and-simulcast-00. Stream=20
>>> duplication both results in a certain form of multi-stream and also
>>> has a similar but slightly different semantics to simulcast in that
>>> the streams are two representations of a single original media
>>> source. In this case they are clones, but not true duplicates due
>>> to that RTP doesn't handle try duplicates well.
>>=20
>> The data carried in the redundant streams are identical but the rtp
>> header differs due to the different SSRCs. That is all. SSRC muxing
>> nicely fits the use case we are tackling with. Simulcasting is quite
>> different from our use case because in simulcasting different RTP
>> streams most likely carry different data.
>=20
> I agree that the purpose for multiple RTP streams from the same =
original
> media source are different from the primary case I set out to solve. =
But
> as Jonathan Lennox pointed out, redundancy and FEC have very similar
> behaviors to simulcast with different encodings of the same original
> source.
>=20
> Thus, I do see a need to discuss for example how you identify the
> streams that are a duplicate of each other.
>=20
>>=20
>>> 3. Another issue that also the authors has identified is the
>>> question of what one do when one has multiple media streams from
>>> the same end-point.
>>=20
>>> =46rom the same end-point (e.g., source) there could of course of
>>> multiple media streams. You can use SSRC muxing for example to
>>> identify your streams and use ssrc-group to group them if needed.
>>=20
>> The concern in that editorial note was what if there were multiple
>> streams in each m line but what if you wanted grouping/association
>> among only a subset.
>>=20
>> For example,
>>=20
>> m=3Dvideo 30000 RTP/AVP 101 102 .... m=3Dvideo 30000 RTP/AVP 103 104
>>=20
>> I can easily associate 101 and 102 if I want to, but what if 101 and
>> 103 were associated with each other. Can I use a=3Dgroup line? =
Probably
>> not. But, I don't know why you would need that kind of relation in
>> the first place (even if you need, there are probably less messy ways
>> to express them). If that had been a very desirable feature, SDP and
>> grouping semantics along with RTP would not have defined in their
>> current form I guess.
>=20
> Yes, that is a very valid question also for simulcast.
>=20
> And I think we can provide a story for doing that. But I don't think =
it
> should be based on using the RTP payload type to identify either of =
the
> versions.
>=20
>>=20
>>> And in a general case one can't use the CNAME to separate the
>>> sources. The reason is that two media sources that has a common
>>> timing (synchronization) context do needs to indicate that by
>>> having the same CNAME. Thus there are need for some additional
>>> mechanism to ensure that
>>=20
>> If 101 and 103 are associated, why can't I use the same CNAME for
>> those streams? If 102 and 104 are associated/related, I would use a
>> common CNAME for them, too (but theirs would be separate from 101 and
>> 103's).
>=20
> If SSRC A has a duplciation in SSRC C and SSRC B has a duplication in
> SSRC D and both A and B comes from the same end-point and have a =
common
> synchronization context, then all of A-D needs to have the same CNAME.
>=20
> And I get the impression that you are using a payload type based
> association scheme here. And I think that is a bad idea. For example =
it
> can't support particular many streams. It also looks semantically =
wrong,
> as if Stream A uses PT 96, then SSRC C should use PT 96 also for the
> corresponding packet duplicates. And if A switches from one PT to
> another the other stream needs to follow.
>=20
> Thus I think a SDES item based approach for indicating that it is in
> fact A and C that are duplicates, rather than A and B or A and D.
>=20
>>=20
>>> SSRC carrying RTP media streams that are duplicates of each other
>>> do in fact be labeled explicitly. I also note that a SDP based
>>> solution is not sufficient to resolve this issue. I think we need
>>> something in RTP, so that SSRC collisions can be handled.
>>=20
>> I can't say I am following.
>=20
> I suggest that you use something like the SRCNAME SDES item extension =
I
> propose in my draft.
>=20
>>=20
>>> 4. In addition there is a few questions how this duplication
>>> really should set some RTP fields. Is the duplication only working
>>> on Sequence
>>=20
>> That is the easiest way for a middle box or endpoint doing the stream
>> merging (duplicate suppression). If seqnums are different, what would
>> one do? look into the payload?
>=20
> It was simply that you had not documented what needs to be the same.
> Sequence number being the same is likely the easiest, but one do have =
to
> raise some warning flags if one has processing that changes these
> streams. For example a translator modifying the packetization must do
> the exact same manipulation on the other stream.
>=20
> But, I don't think anything else really works.
>=20
>>=20
>>> number level? There seem to be additional questions, for example in
>>> the need to have RTP timestamp considerations between the streams.
>>> They need to point identically into the joint timeline. Otherwise
>>> outages, or receiver taking synch from one and not the other stream
>>> can end up with different synchronizations. Becomes especially
>>> interesting to discuss when the time shifted transmission.
>>=20
>> The duplicate streams are coming from the same source, so they have
>> the same timestamp as well as seqnums. It has to be this way because
>> the node merging the streams will not re-write anything at all. It is
>> a simple suppression operation. Beauty of it is that things stay
>> pretty simple to use and implement without any changes to RTP/RTCP
>> plane at all.
>=20
> Yes, but you hadn't made it clear, and you have both SSRC and implied
> change of PT. So I didn't want to assume anything.
>=20
> The timestamp field also have implcations due to its binding with RTCP
> SR information.
>=20
>>=20
>> As for the time-shifted transmission, I suppose you are referring to
>> the temporal diversity/delayed duplication
>> (draft-begen-mmusic-temporal-interleaving-02). Well, in that case,
>> timestamps are still the same for the reasons above. Changing the
>> timestamps for the delayed packets does not bring any benefit
>> (actually it complicates things) since the ultimate RTP application
>> will still receive one and only one RTP stream.
>=20
> Ok, but you will need to make clear on how one should set the RTCP SR
> timestamp NTP mapping to be clear. And what that implies for the RTCP
> jitter calculation.
>=20
>>=20
>>> 5. The draft also discussed Payload Type differences and have them
>>> in the SDP examples. I don't see how that is appropriate. If the
>>> streams
>>=20
>> When you are doing SSRC muxing, you put multiple streams in the same
>> m line and they end up having different PT numbers. What is wrong
>> with that? If they are in separate m lines, they could have the same
>> PT number.
>=20
> I am still asking why, there is no reason to use different PTs, unless
> you are doing the association between the different streams by them
> having different PTs. Which I believe is a Bad Idea (TM). The encoding
> in the packet are the same, so why don't they have the same PT.
>=20
>=20
>>=20
>>> 6. I think it is worth discussing also the benefits of SSRC
>>> multiplexed versus the session multiplexed. In multicast context
>>> there is clear benefits from a control and congestion perspective
>>> to have session multiplexed versions and these mapped onto
>>> different multicast groups enabling a receiver to choose one or
>>> both streams.
>>=20
>> This is what we have in section 5.2. Different streams can be sent in
>> different multicast sessions but it does not mean that they are
>> separate RTP sessions. I discussed this in detail with Colin. Thanks
>> to him for trying to teach me over and over what an RTP session
>> actually means.
>=20
> Yes, you have an example, but not a discussion of why an implementor
> should choose Session multiplexing over SSRC. I think that should be
> included, or I think you can wait for the architectural document I =
think
> will result from my multi-stream and simulcast draft.
>=20
>>=20
>>> 7. Thus I think the signalling aspects of this is the minor part.=20
>>> Although there clearly are some aspects of this needing discussion.
>>> For example clearer offer/answer rules for handling these.
>>=20
>> We have some text around O/A stuff. But frankly, I don=92t see any =
use
>> case where duplication will be used in a unicast application. I mean
>> if redundancy is needed, one should use FEC instead for unicast
>> transport. Duplication is a nice use case for multicast transport
>> where the bw overhead is easier to justify and the implementation is
>> easier. And one would naturally implement this spec in networks where
>> he already knows that the network elements support this spec. So,
>> maybe we should remove the whole O/A text anyway and write that this
>> will only be used declaratively.
>=20
> Then, I think we should slap an application statement to the beginning
> of the draft to ensure that it may only be used in multicast. For
> example b=3DTIAS was only intended to used in declarative SDPs, but it =
has
> become common in O/A, but it doesn't have clear rules.
>=20
> So I think it imporant that one don't assume to much. Yes, it might =
not
> be the best idea for unicast, but that is not going to prevent it from
> being used.
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt


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> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Monday, July 25, 2011 6:54 AM
> To: Ali C. Begen (abegen)
> Cc: mmusic (E-mail); IETF AVTCore WG
> Subject: Re: [AVTCORE] Comments on =
draft-begen-mmusic-redundancy-grouping-01
>=20
> On 2011-07-24 23:21, Ali C. Begen (abegen) wrote:
> >
> >
> >> -----Original Message----- From: avt-bounces@ietf.org
> >> [mailto:avt-bounces@ietf.org] On Behalf Of Magnus Westerlund Sent:
> >> Friday, July 22, 2011 8:39 AM To: mmusic (E-mail); IETF AVTCore WG
> >> Subject: [AVTCORE] Comments on
> >> draft-begen-mmusic-redundancy-grouping-01
> >>
> >> Hi,
> >>
> >> (Primarily as individual, but the WG home question is as WG chair)
> >>
> >> I have reviewed this draft and some comments on it.
> >
> > Thanks for the review.
> >
> >> 1. If this solution proposal is to be taken forward I think it
> >> needs to be discussed in AVTCORE. The reason is that it has several
> >> RTP details that needs to be figured out to make stream duplication
> >> issue free in a general usage. I will take up a few of these issues
> >> in the below comments.
> >
> > That is why we asked for a slot in avtcore on wed.
> >
> >> 2. One aspect this document has is a relation to what is discussed
> >> in draft-westerlund-avtcore-multistream-and-simulcast-00. Stream
> >> duplication both results in a certain form of multi-stream and also
> >> has a similar but slightly different semantics to simulcast in that
> >> the streams are two representations of a single original media
> >> source. In this case they are clones, but not true duplicates due
> >> to that RTP doesn't handle try duplicates well.
> >
> > The data carried in the redundant streams are identical but the rtp
> > header differs due to the different SSRCs. That is all. SSRC muxing
> > nicely fits the use case we are tackling with. Simulcasting is quite
> > different from our use case because in simulcasting different RTP
> > streams most likely carry different data.
>=20
> I agree that the purpose for multiple RTP streams from the same =
original
> media source are different from the primary case I set out to solve. =
But
> as Jonathan Lennox pointed out, redundancy and FEC have very similar
> behaviors to simulcast with different encodings of the same original
> source.
>=20
> Thus, I do see a need to discuss for example how you identify the
> streams that are a duplicate of each other.

Ok, lets discuss it. My current proposal is to use different SSRCs. PTs =
turn out to be different since I designate these streams in the same m =
line. Unless I am mistakes, you cannot put two different streams with =
the same PT in a single m line. Remember, we do not use PT to demux the =
RTP streams, we use SSRC for that purpose.
=20
> >
> >> 3. Another issue that also the authors has identified is the
> >> question of what one do when one has multiple media streams from
> >> the same end-point.
> >
> >> From the same end-point (e.g., source) there could of course of
> >> multiple media streams. You can use SSRC muxing for example to
> >> identify your streams and use ssrc-group to group them if needed.
> >
> > The concern in that editorial note was what if there were multiple
> > streams in each m line but what if you wanted grouping/association
> > among only a subset.
> >
> > For example,
> >
> > m=3Dvideo 30000 RTP/AVP 101 102 .... m=3Dvideo 30000 RTP/AVP 103 104
> >
> > I can easily associate 101 and 102 if I want to, but what if 101 and
> > 103 were associated with each other. Can I use a=3Dgroup line? =
Probably
> > not. But, I don't know why you would need that kind of relation in
> > the first place (even if you need, there are probably less messy =
ways
> > to express them). If that had been a very desirable feature, SDP and
> > grouping semantics along with RTP would not have defined in their
> > current form I guess.
>=20
> Yes, that is a very valid question also for simulcast.
>=20
> And I think we can provide a story for doing that. But I don't think =
it
> should be based on using the RTP payload type to identify either of =
the
> versions.
>=20
> >
> >> And in a general case one can't use the CNAME to separate the
> >> sources. The reason is that two media sources that has a common
> >> timing (synchronization) context do needs to indicate that by
> >> having the same CNAME. Thus there are need for some additional
> >> mechanism to ensure that
> >
> > If 101 and 103 are associated, why can't I use the same CNAME for
> > those streams? If 102 and 104 are associated/related, I would use a
> > common CNAME for them, too (but theirs would be separate from 101 =
and
> > 103's).
>=20
> If SSRC A has a duplciation in SSRC C and SSRC B has a duplication in
> SSRC D and both A and B comes from the same end-point and have a =
common
> synchronization context, then all of A-D needs to have the same CNAME.

Well, above implies that all streams are coming from the same source and =
yes, they will end up having the same CNAME since you required A will be =
synched with B. But, still I think this is workable without any =
additional mechanism. A and C would be in the same m line and B and D =
would be in the same (but separate from the first) m line. That would =
give me what I want.
=20
> And I get the impression that you are using a payload type based
> association scheme here. And I think that is a bad idea. For example =
it
> can't support particular many streams. It also looks semantically =
wrong,
> as if Stream A uses PT 96, then SSRC C should use PT 96 also for the
> corresponding packet duplicates. And if A switches from one PT to
> another the other stream needs to follow.

No, I am not suggesting to use PT for demuxing. I use SSRC for that =
purpose. As I asked above, if I can use the same PT for multiple streams =
in the same m line, then I will. But I don=92t think I can.

> Thus I think a SDES item based approach for indicating that it is in
> fact A and C that are duplicates, rather than A and B or A and D.
>=20
> >
> >> SSRC carrying RTP media streams that are duplicates of each other
> >> do in fact be labeled explicitly. I also note that a SDP based
> >> solution is not sufficient to resolve this issue. I think we need
> >> something in RTP, so that SSRC collisions can be handled.
> >
> > I can't say I am following.
>=20
> I suggest that you use something like the SRCNAME SDES item extension =
I
> propose in my draft.
>=20
> >
> >> 4. In addition there is a few questions how this duplication
> >> really should set some RTP fields. Is the duplication only working
> >> on Sequence
> >
> > That is the easiest way for a middle box or endpoint doing the =
stream
> > merging (duplicate suppression). If seqnums are different, what =
would
> > one do? look into the payload?
>=20
> It was simply that you had not documented what needs to be the same.

I think I did, but I will check.

> Sequence number being the same is likely the easiest, but one do have =
to
> raise some warning flags if one has processing that changes these
> streams. For example a translator modifying the packetization must do
> the exact same manipulation on the other stream.

Yes, obviously. We can put this warning.
=20
> But, I don't think anything else really works.

Right, seqnum is the easiest and possibly the only way in practice.
=20
> >
> >> number level? There seem to be additional questions, for example in
> >> the need to have RTP timestamp considerations between the streams.
> >> They need to point identically into the joint timeline. Otherwise
> >> outages, or receiver taking synch from one and not the other stream
> >> can end up with different synchronizations. Becomes especially
> >> interesting to discuss when the time shifted transmission.
> >
> > The duplicate streams are coming from the same source, so they have
> > the same timestamp as well as seqnums. It has to be this way because
> > the node merging the streams will not re-write anything at all. It =
is
> > a simple suppression operation. Beauty of it is that things stay
> > pretty simple to use and implement without any changes to RTP/RTCP
> > plane at all.
>=20
> Yes, but you hadn't made it clear, and you have both SSRC and implied
> change of PT. So I didn't want to assume anything.

OK, will make this clear, as well.
=20
> The timestamp field also have implcations due to its binding with RTCP
> SR information.
>=20
> >
> > As for the time-shifted transmission, I suppose you are referring to
> > the temporal diversity/delayed duplication
> > (draft-begen-mmusic-temporal-interleaving-02). Well, in that case,
> > timestamps are still the same for the reasons above. Changing the
> > timestamps for the delayed packets does not bring any benefit
> > (actually it complicates things) since the ultimate RTP application
> > will still receive one and only one RTP stream.
>=20
> Ok, but you will need to make clear on how one should set the RTCP SR
> timestamp NTP mapping to be clear. And what that implies for the RTCP
> jitter calculation.

The fact that you are using delayed duplication or spatial diversity =
means that you will likely introduce some jitter into your stream unless =
the node merging the streams smooths out that jitter in the first place. =
There is not nothing special for the ultimate endpoint for preparing the =
reports, it just does  what it is supposed to do. In most systems, the =
ultimate RTP receivers might not be even aware of the fact that they are =
receiving a stream merged from two identical streams. That is a design =
feature.

> >
> >> 5. The draft also discussed Payload Type differences and have them
> >> in the SDP examples. I don't see how that is appropriate. If the
> >> streams
> >
> > When you are doing SSRC muxing, you put multiple streams in the same
> > m line and they end up having different PT numbers. What is wrong
> > with that? If they are in separate m lines, they could have the same
> > PT number.
>=20
> I am still asking why, there is no reason to use different PTs, unless
> you are doing the association between the different streams by them
> having different PTs. Which I believe is a Bad Idea (TM). The encoding
> in the packet are the same, so why don't they have the same PT.

Well again, I am not using it for demuxing purposes. I just believe that =
is how the m line works. Tell me otherwise and I will be happy to make =
PTs the same.
=20
>=20
> >
> >> 6. I think it is worth discussing also the benefits of SSRC
> >> multiplexed versus the session multiplexed. In multicast context
> >> there is clear benefits from a control and congestion perspective
> >> to have session multiplexed versions and these mapped onto
> >> different multicast groups enabling a receiver to choose one or
> >> both streams.
> >
> > This is what we have in section 5.2. Different streams can be sent =
in
> > different multicast sessions but it does not mean that they are
> > separate RTP sessions. I discussed this in detail with Colin. Thanks
> > to him for trying to teach me over and over what an RTP session
> > actually means.
>=20
> Yes, you have an example, but not a discussion of why an implementor
> should choose Session multiplexing over SSRC. I think that should be
> included, or I think you can wait for the architectural document I =
think
> will result from my multi-stream and simulcast draft.

Lets discuss in the session.
=20
> >
> >> 7. Thus I think the signalling aspects of this is the minor part.
> >> Although there clearly are some aspects of this needing discussion.
> >> For example clearer offer/answer rules for handling these.
> >
> > We have some text around O/A stuff. But frankly, I don=92t see any =
use
> > case where duplication will be used in a unicast application. I mean
> > if redundancy is needed, one should use FEC instead for unicast
> > transport. Duplication is a nice use case for multicast transport
> > where the bw overhead is easier to justify and the implementation is
> > easier. And one would naturally implement this spec in networks =
where
> > he already knows that the network elements support this spec. So,
> > maybe we should remove the whole O/A text anyway and write that this
> > will only be used declaratively.
>=20
> Then, I think we should slap an application statement to the beginning
> of the draft to ensure that it may only be used in multicast. For

Wfm.

> example b=3DTIAS was only intended to used in declarative SDPs, but it =
has
> become common in O/A, but it doesn't have clear rules.
>=20
> So I think it imporant that one don't assume to much. Yes, it might =
not
> be the best idea for unicast, but that is not going to prevent it from
> being used.

If one really uses this for unicast, let them deal with all the =
potential O/A problems that might surface. It is more like they will =
deserve it :)

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


From magnus.westerlund@ericsson.com  Mon Jul 25 09:12:36 2011
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Subject: Re: [MMUSIC] [AVTCORE] Comments on draft-begen-mmusic-redundancy-grouping-01
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On 2011-07-25 10:07, Ali C. Begen (abegen) wrote:

> Ok, lets discuss it. My current proposal is to use different SSRCs.
PTs turn out to be different since I designate these streams in the same
m line. Unless I am mistakes, you cannot put two different streams with
the same PT in a single m line. Remember, we do not use PT to demux the
RTP streams, we use SSRC for that purpose.

No, that is wrong. The PT is completely unrelated to to which SSRC that
use them. The set of configured PTs can be used by any of the sources
that transmit in the direction for which they are configured. I do
include the direction because of the O/A aspects of having different
configurations for receving in different directions.


>> If SSRC A has a duplciation in SSRC C and SSRC B has a duplication
>> in SSRC D and both A and B comes from the same end-point and have a
>> common synchronization context, then all of A-D needs to have the
>> same CNAME.
> 
> Well, above implies that all streams are coming from the same source
> and yes, they will end up having the same CNAME since you required A
> will be synched with B. But, still I think this is workable without
> any additional mechanism. A and C would be in the same m line and B
> and D would be in the same (but separate from the first) m line. That
> would give me what I want.

I am not agreeing that you would put A and B in different sessions. This
doesn't make sense if you actually like to provide two related media
streams of the same type, for example two cameras covering different
parts of a telepresence room, or two different camera angels in a movie.

What you are suggesting seems to be a hack to just get around the issue
that if you have streams that are some form of alternatives of the
actual same original source then you need to identify, like the SRCNAME
proposal.


> 
>> And I get the impression that you are using a payload type based 
>> association scheme here. And I think that is a bad idea. For
>> example it can't support particular many streams. It also looks
>> semantically wrong, as if Stream A uses PT 96, then SSRC C should
>> use PT 96 also for the corresponding packet duplicates. And if A
>> switches from one PT to another the other stream needs to follow.
> 
> No, I am not suggesting to use PT for demuxing. I use SSRC for that
> purpose. As I asked above, if I can use the same PT for multiple
> streams in the same m line, then I will. But I don’t think I can.
> 

Yes, understand this is a different type of missunderstanding of RTP.


>> 
>> Ok, but you will need to make clear on how one should set the RTCP
>> SR timestamp NTP mapping to be clear. And what that implies for the
>> RTCP jitter calculation.
> 
> The fact that you are using delayed duplication or spatial diversity
> means that you will likely introduce some jitter into your stream
> unless the node merging the streams smooths out that jitter in the
> first place. There is not nothing special for the ultimate endpoint
> for preparing the reports, it just does  what it is supposed to do.
> In most systems, the ultimate RTP receivers might not be even aware
> of the fact that they are receiving a stream merged from two
> identical streams. That is a design feature.
> 

When you talk about jitter here, you are talking about small transport
level jitter, rather than jitter on the level of frame intervals.

Sure, if you have a translator that merge these, then you clearly should
be hidden to the end-point behind translator.

>>>> 7. Thus I think the signalling aspects of this is the minor
>>>> part. Although there clearly are some aspects of this needing
>>>> discussion. For example clearer offer/answer rules for handling
>>>> these.
>>> 
>>> We have some text around O/A stuff. But frankly, I don’t see any
>>> use case where duplication will be used in a unicast application.
>>> I mean if redundancy is needed, one should use FEC instead for
>>> unicast transport. Duplication is a nice use case for multicast
>>> transport where the bw overhead is easier to justify and the
>>> implementation is easier. And one would naturally implement this
>>> spec in networks where he already knows that the network elements
>>> support this spec. So, maybe we should remove the whole O/A text
>>> anyway and write that this will only be used declaratively.
>> 
>> Then, I think we should slap an application statement to the
>> beginning of the draft to ensure that it may only be used in
>> multicast. For
> 
> Wfm.
> 
>> example b=TIAS was only intended to used in declarative SDPs, but
>> it has become common in O/A, but it doesn't have clear rules.
>> 
>> So I think it imporant that one don't assume to much. Yes, it might
>> not be the best idea for unicast, but that is not going to prevent
>> it from being used.
> 
> If one really uses this for unicast, let them deal with all the
> potential O/A problems that might surface. It is more like they will
> deserve it :)
> 

:-)

Cheers

Magnus Westerlund

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


From adam@nostrum.com  Tue Jul 26 10:29:19 2011
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Cc: "MMUSIC \(Multiparty Multimedia Session Control\) WG" <mmusic@ietf.org>
Subject: [MMUSIC] SDP for SCTP
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I have a couple of comments on the SDP for SCTP draft.

In looking through the current version of the document, it appears that 
datagram is being assumed (based on, for example, the mention of DTLS 
without TLS). Surely we want to consider using stream-oriented uses as 
well. You might consider that the earlier foray into this arena 
(draft-fairlie-mmusic-sdp-sctp) used different protocol identifiers for 
stream and datagram uses (SCTP and USCTP).

Also, something that sticks out is that the current mechanism does not 
have any way to play nicely with middleboxes and multihomed connections. 
As the draft notes, existing middleboxes do not yet have support for 
SCTP. It would be nice if we gave them the tools to handle this properly 
(should they choose to do so) inside the SDP. Is there a reason we're 
not adding a new set of a= lines to indicate additional IP addresses 
associated with a connection?

/a

From dwing@cisco.com  Tue Jul 26 13:20:59 2011
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From: "Dan Wing" <dwing@cisco.com>
To: "'Varun Singh'" <vsingh.ietf@gmail.com>, "'IETF MMUSIC WG'" <mmusic@ietf.org>
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Subject: Re: [MMUSIC] Multipath RTP (interface advertisement in SDP)
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> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Varun Singh
> Sent: Monday, July 25, 2011 8:26 AM
> To: IETF MMUSIC WG
> Subject: [MMUSIC] Multipath RTP (interface advertisement in SDP)
> 
> Greetings,
> 
> The draft is part of AVTCore but we have recently introduced an
> "interface" attribute in SDP, as an alternative to in-band interface
> advertisement. The draft also contains some O/A examples and these
> changes can be found in the SDP Considerations section.
> http://tools.ietf.org/rfcdiff?url2=draft-singh-avtcore-mprtp-02.txt
> 
> Comments and suggestions are welcome from MMUSIC.

ICE's STUN connectivity checks, which use the password exchanged
in signaling as HMACs for the connectivity check, feel like a 
natural way to add new MPRTP addresses.  Have you analyzed using
such a technique and discarded the idea?

-d


> Regards,
> Varun
> 
> Begin forwarded message:
> >
> > A new version of I-D, draft-singh-avtcore-mprtp-02.txt has been
> successfully submitted by Varun Singh and posted to the IETF
> repository.
> >
> > Filename: draft-singh-avtcore-mprtp
> > Revision: 02
> > Title: Multipath RTP (MPRTP)
> > Creation date: 2011-07-08
> > WG ID: Individual Submission
> > Number of pages: 34
> >
> > Abstract:
> > The Real-time Transport Protocol (RTP) is used to deliver real-time
> > content and, along with the RTP Control Protocol (RTCP), forms the
> > control channel between the sender and receiver. However, RTP and
> > RTCP assume a single delivery path between the sender and receiver
> > and make decisions based on the measured characteristics of this
> > single path. Increasingly, endpoints are becoming multi-homed, which
> > means that they are connected via multiple Internet paths. Network
> > utilization can be improved when endpoints use multiple parallel
> > paths for communication. The resulting increase in reliability and
> > throughput can also enhance the user experience. This document
> > extends the Real-time Transport Protocol (RTP) so that a single
> > session can take advantage of the availability of multiple paths
> > between two endpoints.
> >
> >
> >
> >
> > The IETF Secretariat
> --
> 
> http://www.netlab.tkk.fi/~varun
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From stewe@stewe.org  Wed Jul 27 06:52:25 2011
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> This message is in MIME format. Since your mail reader does not understand
this format, some or all of this message may not be legible.

--B_3394605142_5344166
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	charset="US-ASCII"
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Hi all,
We have received a liaison statement from the MPEG meeting that took place
last week.  No action is requested from us.
They inform us that they have issued a second DIS of their HTTP streaming
project known as "DASH", to be voted on according to ISO/IEC rules in the
next couple of months.  I heard that there was a hiccup during the first
ballot.  They must have addressed the ballot comments.  That said, I'm not
aware of earth-shattering changes between the new version and the previous
one, but I'm not an insider in Dash.
The liaison statement will shortly appear on the IETF's tracker.  If you
need a version urgently, please contact me in private.
Thanks,
Stephan



--B_3394605142_5344166
Content-type: text/html;
	charset="US-ASCII"
Content-transfer-encoding: quoted-printable

<html><head></head><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: s=
pace; -webkit-line-break: after-white-space; color: rgb(0, 0, 0); font-size:=
 14px; font-family: Calibri, sans-serif; "><div><div>Hi all,</div><div>We ha=
ve received a liaison statement from the MPEG meeting that took place last w=
eek. &nbsp;No action is requested from us.</div><div>They inform us that the=
y have issued a second DIS of their HTTP streaming project known as "DASH", =
to be voted on according to ISO/IEC rules in the next couple of months. &nbs=
p;I heard that there was a hiccup during the first ballot. &nbsp;They must h=
ave addressed the ballot comments. &nbsp;That said, I'm not aware of earth-s=
hattering changes between the new version and the previous one, but I'm not =
an insider in Dash. &nbsp;</div><div>The liaison statement will shortly appe=
ar on the IETF's tracker. &nbsp;If you need a version urgently, please conta=
ct me in private.</div><div>Thanks,</div><div>Stephan</div></div></body></ht=
ml>

--B_3394605142_5344166--



From yuepeiyu@huawei.com  Wed Jul 27 07:11:16 2011
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Date: Wed, 27 Jul 2011 14:10:38 +0000
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<html dir=3D"ltr">
<head>
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<style id=3D"owaParaStyle">P {
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<div style=3D"direction: ltr;font-family: Tahoma;color: #000000;font-size: =
10pt;">
<p>Hi all,</p>
<p>&nbsp;</p>
<p>I was in the DASH meeting last week. Here is some backgroud information =
FYI.</p>
<p>The ballot result is not approval because of the new ballot rules define=
d by ISO: &nbsp;a couple of nationa bodies voted &quot;Disapprove with comm=
ents ...&quot; which was meaning &quot;if the comments are addressed then&n=
bsp;may change to Approve&quot;. However, according to the new
 rule, &quot;Disapprove with comments&quot; means &quot;Disapprove&quot;, w=
hatever the comments are addressed. This is the first time the new rule app=
lies.</p>
<p>&nbsp;</p>
<p>Regards, Roy</p>
<p>&nbsp;</p>
<div style=3D"FONT-FAMILY: Times New Roman; COLOR: #000000; FONT-SIZE: 16px=
">
<hr tabindex=3D"-1">
<div style=3D"DIRECTION: ltr" id=3D"divRpF456339"><font color=3D"#000000" s=
ize=3D"2" face=3D"Tahoma"><b>=B7=A2=BC=FE=C8=CB:</b> fecframe-bounces@ietf.=
org [fecframe-bounces@ietf.org] =B4=FA=B1=ED Stephan Wenger [stewe@stewe.or=
g]<br>
<b>=B7=A2=CB=CD=CA=B1=BC=E4:</b> 2011=C4=EA7=D4=C227=C8=D5 21:52<br>
<b>=B5=BD:</b> mmusic@ietf.org; avtcore@ietf.org; fecframe@ietf.org; rai@ie=
tf.org<br>
<b>=D6=F7=CC=E2:</b> [Fecframe] Incoming liaison statement from MPEG<br>
</font><br>
</div>
<div></div>
<div>
<div>
<div>Hi all,</div>
<div>We have received a liaison statement from the MPEG meeting that took p=
lace last week. &nbsp;No action is requested from us.</div>
<div>They inform us that they have issued a second DIS of their HTTP stream=
ing project known as &quot;DASH&quot;, to be voted on according to ISO/IEC =
rules in the next couple of months. &nbsp;I heard that there was a hiccup d=
uring the first ballot. &nbsp;They must have addressed
 the ballot comments. &nbsp;That said, I'm not aware of earth-shattering ch=
anges between the new version and the previous one, but I'm not an insider =
in Dash. &nbsp;</div>
<div>The liaison statement will shortly appear on the IETF's tracker. &nbsp=
;If you need a version urgently, please contact me in private.</div>
<div>Thanks,</div>
<div>Stephan</div>
</div>
</div>
</div>
</div>
</body>
</html>

--Boundary_(ID_QBcnPUxs2/VU2v4DD02AbQ)--

From aallen@rim.com  Wed Jul 27 08:17:54 2011
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From: Andrew Allen <aallen@rim.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: resurecting the CCAP attribute
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Subject: [MMUSIC] resurecting the CCAP attribute
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Yesterday several MMUSIC participants including Jonathan Lennox and Miguel G=
arcia got together to discuss what to do regarding the CCAP attribute which=
 is currently used in 3GPP specifications but was removed from the misc-cap=
 draft (which then became the sdp-icap-bcap draft without the CCAP attribute=
). The CCAP attribute was removed after earlier comments on the list from Jo=
nathan regarding the possibility that CCAP might be used as an alternative t=
o ICE for negotiating IP addresses for the C line (see email thread below).=
 3GPP currently utilize the CCAP attribute with SDP-cap-neg for negotiation=
 of use of a circuit switched call for the audio media as an alternative to=
 an audio RTP stream over packet switched access. CCAP is used to offer the=
 PSTN net type and E.164 number used to set up that call.

After a thorough discussion the conclusion of those involved in the discussi=
on (including Jonathan) was that the best way forward given the constraints=
 of the 3GPP architecture is to resurrect the CCAP attribute but with some w=
ording that places constraints on its usage so that it isn't used to negotia=
te alternative addresses for a network type for which ICE can be used to neg=
otiate addresses. This means that CCAP could be used to negotiate the possib=
ility to use PSTN net type as an alternative to the IN net type but that usi=
ng alternative IP address candidates within the IN net type would need to be=
 negotiated using ICE not using CCAP.

Are the any concerns from other MMUSIC participants if CCAP is resurrected (=
probably in an again renamed version of the sdp-icap-bcap draft that would a=
gain include CCAP) with appropriate text to address the ICE issue along the=
 lines of above?

Andrew

  *   To: "Simo.Veikkolainen at nokia.com<mailto:Simo.Veikkolainen@DOMAIN.HI=
DDEN>" <Simo.Veikkolainen at nokia.com<mailto:Simo.Veikkolainen@DOMAIN.HIDDE=
N>>, "mmusic at ietf.org<mailto:mmusic@DOMAIN.HIDDEN>" <mmusic at ietf.org<m=
ailto:mmusic@DOMAIN.HIDDEN>>
  *   Subject: Re: [MMUSIC] What to do with the ccap attribute?
  *   From: Jonathan Lennox <jonathan at vidyo.com<mailto:jonathan@DOMAIN.HI=
DDEN>>
  *   Date: Mon, 10 May 2010 11:24:36 -0400
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  *   Thread-index: AcrwJWqZpvLJ586JTjK5XQiTSB6kSAALoU1w
  *   Thread-topic: What to do with the ccap attribute?

________________________________

I think it'd be best to unify the requirements of this use case

(separate IP and PSTN addresses) with the requirements of the

IPv4/IPv6 transition use cases, and come up with a common solution.



Having two mechanisms (ANAT and ICE) to specify multiple connection

addresses is a standard but annoying consequence of a technology

transition. Having three (ANAT, ICE, and ALTC) gets unmanageable.

Having four (ANAT, ICE, ALTC, and CCAP) is just silly.



--

Jonathan Lennox

Vidyo, Inc

jonathan at vidyo.com



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

> From: Simo.Veikkolainen at nokia.com [mailto:Simo.Veikkolainen at nokia.co=
m]

> Sent: Monday, May 10, 2010 5:45 AM

> To: mmusic at ietf.org

> Subject: [MMUSIC] What to do with the ccap attribute?

>

> Now that the discussion on ICE and v6 transition has quieted down, I

> would like to ask for the list's opinion on how to proceed with the

> ccap attribute defined in http://tools.ietf.org/html/draft-ietf-mmusic-

> sdp-misc-cap

>

> Note that this draft will be re-submitted as an individual document

> according to the WG decision in IETF#77. Nevertheless it contains the

> most up-to-date text, hence the reference.

>

> As a short reminder, the misc-cap document defines capability

> attributes for "i=3D", "b"=3D and "c=3D" lines (icap, bcap and ccap,

> respectively). The ccap capability attribute is used to indicate

> alternative connection addresses. Our use case for this has been the

> ability to indicate alternative IP and PSTN addresses; a feature which

> is needed in some 3GPP scenarios.

>

> As ccap is currently specified, it includes also the port number in

> addition to the contents of a "c=3D" line (<nettype>, <addrtype> and the

> <connection-address>). The port number is needed in order to indicate a

> port number for internet addresses, the PSTN addresses don't have a

> similar concept.

>

> Now, as a side effect, the ccap attribute could also be used to

> circumvent ICE, if one used IP4 and IP6 addrtypes as alternatives.

>

> In order to move forward with the misc-cap document I'm now asking how

> do folks feel we should handle this issue. Possible alternatives are at

> least:

>

> 1. Leave the current text as is (see [1]).

>

> 2. Make a more strict statement that ccap SHOULD NOT be used for v4/v6

> address pairs.

>

> 3. Come up with altogether different way of indicating alternative

> connection addresses that fulfill our use case (PSTN/IP addresses).

>

> Regards,

> Simo

>

> [1] The text from Section 3.1.2 from  http://tools.ietf.org/html/draft-

> ietf-mmusic-sdp-misc-cap currently says

> "

>    The connection information capability can be used to negotiate the

>    use of IPv4 or IPv6 addresses without resort to Interactive

>    Connectivity Establishment(ICE) [I-D.ietf-mmusic-ice].  Note,

>    however, that ICE provides for real-time reachability testing of

>    multiple addresses, whereas use of the connection capability forces

>    an early choice of connection address.

>

>    [I-D.boucadair-mmusic-altc] describes a simple method of specifying

>    alternative network addresses when the transport protocol (<proto>)

>    is the same.

> "

> _______________________________________________

> mmusic mailing list

> mmusic at ietf.org

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




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<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Yesterday several MMUSIC participants including Jonat=
han Lennox and Miguel Garcia got together to discuss what to do regarding th=
e CCAP attribute which is currently used in 3GPP specifications but was remo=
ved from the misc-cap draft (which
 then became the sdp-icap-bcap draft without the CCAP attribute). The CCAP a=
ttribute was removed after earlier comments on the list from Jonathan regard=
ing the possibility that CCAP might be used as an alternative to ICE for neg=
otiating IP addresses for the
 C line (see email thread below). 3GPP currently utilize the CCAP attribute=
 with SDP-cap-neg for negotiation of use of a circuit switched call for the=
 audio media as an alternative to an audio RTP stream over packet switched a=
ccess. CCAP is used to offer the
 PSTN net type and E.164 number used to set up that call.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">After a thorough discussion the conclusion of those i=
nvolved in the discussion (including Jonathan) was that the best way forward=
 given the constraints of the 3GPP architecture is to resurrect the CCAP att=
ribute but with some wording that
 places constraints on its usage so that it isn&#8217;t used to negotiate al=
ternative addresses for a network type for which ICE can be used to negotiat=
e addresses. This means that CCAP could be used to negotiate the possibility=
 to use PSTN net type as an alternative
 to the IN net type but that using alternative IP address candidates within=
 the IN net type would need to be negotiated using ICE not using CCAP.<o:p><=
/o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Are the any concerns from other MMUSIC participants i=
f CCAP is resurrected (probably in an again renamed version of the sdp-icap-=
bcap draft that would again include CCAP) with appropriate text to address t=
he ICE issue along the lines of
 above?<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Andrew<o:p></o:p></p>
<ul type=3D"disc">
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o</span></em>: &quot;<a href=3D"mailto:Simo.Veikkolainen@DOMAIN.HIDDEN">Simo=
.Veikkolainen at nokia.com</a>&quot; &lt;<a href=3D"mailto:Simo.Veikkolainen=
@DOMAIN.HIDDEN">Simo.Veikkolainen at nokia.com</a>&gt;, &quot;<a href=3D"mai=
lto:mmusic@DOMAIN.HIDDEN">mmusic
 at ietf.org</a>&quot; &lt;<a href=3D"mailto:mmusic@DOMAIN.HIDDEN">mmusic at=
 ietf.org</a>&gt; <o:p>
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<em><span style=3D"font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">S=
ubject</span></em>: Re: [MMUSIC] What to do with the ccap attribute?
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<em><span style=3D"font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">F=
rom</span></em>: Jonathan Lennox &lt;<a href=3D"mailto:jonathan@DOMAIN.HIDDE=
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st@ietf.org?subject=3Dsubscribe</a>&gt;
<o:p></o:p></li><li class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso=
-margin-bottom-alt:auto;
     mso-list:l0 level1 lfo1">
<em><span style=3D"font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">L=
ist-unsubscribe</span></em>: &lt;<a href=3D"https://www.ietf.org/mailman/lis=
tinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a>&gt;, &lt;<a h=
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-margin-bottom-alt:auto;
     mso-list:l0 level1 lfo1">
<em><span style=3D"font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">R=
eferences</span></em>: &lt;<a href=3D"mailto:B23B311878A0B4438F5F09F47E01AAE=
050870CAD88@DOMAIN.HIDDEN">B23B311878A0B4438F5F09F47E01AAE050870CAD88 at NOK=
-EUMSG-01.mgdnok.nokia.com</a>&gt;
<o:p></o:p></li><li class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso=
-margin-bottom-alt:auto;
     mso-list:l0 level1 lfo1">
<em><span style=3D"font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">T=
hread-index</span></em>: AcrwJWqZpvLJ586JTjK5XQiTSB6kSAALoU1w
<o:p></o:p></li><li class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso=
-margin-bottom-alt:auto;
     mso-list:l0 level1 lfo1">
<em><span style=3D"font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">T=
hread-topic</span></em>: What to do with the ccap attribute?
<o:p></o:p></li></ul>
<div class=3D"MsoNormal" align=3D"center" style=3D"text-align:center">
<hr size=3D"2" width=3D"100%" align=3D"center">
</div>
<pre>I think it'd be best to unify the requirements of this use case <o:p></=
o:p></pre>
<pre>(separate IP and PSTN addresses) with the requirements of the <o:p></o:=
p></pre>
<pre>IPv4/IPv6 transition use cases, and come up with a common solution.<o:p=
></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>Having two mechanisms (ANAT and ICE) to specify multiple connection <o:=
p></o:p></pre>
<pre>addresses is a standard but annoying consequence of a technology <o:p><=
/o:p></pre>
<pre>transition. Having three (ANAT, ICE, and ALTC) gets unmanageable. <o:p>=
</o:p></pre>
<pre>Having four (ANAT, ICE, ALTC, and CCAP) is just silly.<o:p></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>-- <o:p></o:p></pre>
<pre>Jonathan Lennox<o:p></o:p></pre>
<pre>Vidyo, Inc<o:p></o:p></pre>
<pre>jonathan at vidyo.com<o:p></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>&gt; -----Original Message-----<o:p></o:p></pre>
<pre>&gt; From: Simo.Veikkolainen at nokia.com [<a href=3D"mailto:Simo.Veikk=
olainen">mailto:Simo.Veikkolainen</a> at nokia.com]<o:p></o:p></pre>
<pre>&gt; Sent: Monday, May 10, 2010 5:45 AM<o:p></o:p></pre>
<pre>&gt; To: mmusic at ietf.org<o:p></o:p></pre>
<pre>&gt; Subject: [MMUSIC] What to do with the ccap attribute?<o:p></o:p></=
pre>
<pre>&gt; <o:p></o:p></pre>
<pre>&gt; Now that the discussion on ICE and v6 transition has quieted down,=
 I<o:p></o:p></pre>
<pre>&gt; would like to ask for the list's opinion on how to proceed with th=
e<o:p></o:p></pre>
<pre>&gt; ccap attribute defined in <a href=3D"http://tools.ietf.org/html/dr=
aft-ietf-mmusic-">http://tools.ietf.org/html/draft-ietf-mmusic-</a><o:p></o:=
p></pre>
<pre>&gt; sdp-misc-cap<o:p></o:p></pre>
<pre>&gt; <o:p></o:p></pre>
<pre>&gt; Note that this draft will be re-submitted as an individual documen=
t<o:p></o:p></pre>
<pre>&gt; according to the WG decision in IETF#77. Nevertheless it contains=
 the<o:p></o:p></pre>
<pre>&gt; most up-to-date text, hence the reference.<o:p></o:p></pre>
<pre>&gt; <o:p></o:p></pre>
<pre>&gt; As a short reminder, the misc-cap document defines capability<o:p>=
</o:p></pre>
<pre>&gt; attributes for &quot;i=3D&quot;, &quot;b&quot;=3D and &quot;c=3D&q=
uot; lines (icap, bcap and ccap,<o:p></o:p></pre>
<pre>&gt; respectively). The ccap capability attribute is used to indicate<o=
:p></o:p></pre>
<pre>&gt; alternative connection addresses. Our use case for this has been t=
he<o:p></o:p></pre>
<pre>&gt; ability to indicate alternative IP and PSTN addresses; a feature w=
hich<o:p></o:p></pre>
<pre>&gt; is needed in some 3GPP scenarios.<o:p></o:p></pre>
<pre>&gt; <o:p></o:p></pre>
<pre>&gt; As ccap is currently specified, it includes also the port number i=
n<o:p></o:p></pre>
<pre>&gt; addition to the contents of a &quot;c=3D&quot; line (&lt;nettype&g=
t;, &lt;addrtype&gt; and the<o:p></o:p></pre>
<pre>&gt; &lt;connection-address&gt;). The port number is needed in order to=
 indicate a<o:p></o:p></pre>
<pre>&gt; port number for internet addresses, the PSTN addresses don't have=
 a<o:p></o:p></pre>
<pre>&gt; similar concept.<o:p></o:p></pre>
<pre>&gt; <o:p></o:p></pre>
<pre>&gt; Now, as a side effect, the ccap attribute could also be used to<o:=
p></o:p></pre>
<pre>&gt; circumvent ICE, if one used IP4 and IP6 addrtypes as alternatives.=
<o:p></o:p></pre>
<pre>&gt; <o:p></o:p></pre>
<pre>&gt; In order to move forward with the misc-cap document I'm now asking=
 how<o:p></o:p></pre>
<pre>&gt; do folks feel we should handle this issue. Possible alternatives a=
re at<o:p></o:p></pre>
<pre>&gt; least:<o:p></o:p></pre>
<pre>&gt; <o:p></o:p></pre>
<pre>&gt; 1. Leave the current text as is (see [1]).<o:p></o:p></pre>
<pre>&gt; <o:p></o:p></pre>
<pre>&gt; 2. Make a more strict statement that ccap SHOULD NOT be used for v=
4/v6<o:p></o:p></pre>
<pre>&gt; address pairs.<o:p></o:p></pre>
<pre>&gt; <o:p></o:p></pre>
<pre>&gt; 3. Come up with altogether different way of indicating alternative=
<o:p></o:p></pre>
<pre>&gt; connection addresses that fulfill our use case (PSTN/IP addresses)=
.<o:p></o:p></pre>
<pre>&gt; <o:p></o:p></pre>
<pre>&gt; Regards,<o:p></o:p></pre>
<pre>&gt; Simo<o:p></o:p></pre>
<pre>&gt; <o:p></o:p></pre>
<pre>&gt; [1] The text from Section 3.1.2 from&nbsp; <a href=3D"http://tools=
.ietf.org/html/draft-">http://tools.ietf.org/html/draft-</a><o:p></o:p></pre=
>
<pre>&gt; ietf-mmusic-sdp-misc-cap currently says<o:p></o:p></pre>
<pre>&gt; &quot;<o:p></o:p></pre>
<pre>&gt;&nbsp;&nbsp;&nbsp; The connection information capability can be use=
d to negotiate the<o:p></o:p></pre>
<pre>&gt;&nbsp;&nbsp;&nbsp; use of IPv4 or IPv6 addresses without resort to=
 Interactive<o:p></o:p></pre>
<pre>&gt;&nbsp;&nbsp;&nbsp; Connectivity Establishment(ICE) [I-D.ietf-mmusic=
-ice].&nbsp; Note,<o:p></o:p></pre>
<pre>&gt;&nbsp;&nbsp;&nbsp; however, that ICE provides for real-time reachab=
ility testing of<o:p></o:p></pre>
<pre>&gt;&nbsp;&nbsp;&nbsp; multiple addresses, whereas use of the connectio=
n capability forces<o:p></o:p></pre>
<pre>&gt;&nbsp;&nbsp;&nbsp; an early choice of connection address.<o:p></o:p=
></pre>
<pre>&gt; <o:p></o:p></pre>
<pre>&gt;&nbsp;&nbsp;&nbsp; [I-D.boucadair-mmusic-altc] describes a simple m=
ethod of specifying<o:p></o:p></pre>
<pre>&gt;&nbsp;&nbsp;&nbsp; alternative network addresses when the transport=
 protocol (&lt;proto&gt;)<o:p></o:p></pre>
<pre>&gt;&nbsp;&nbsp;&nbsp; is the same.<o:p></o:p></pre>
<pre>&gt; &quot;<o:p></o:p></pre>
<pre>&gt; _______________________________________________<o:p></o:p></pre>
<pre>&gt; mmusic mailing list<o:p></o:p></pre>
<pre>&gt; mmusic at ietf.org<o:p></o:p></pre>
<pre>&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/mmusic">https://w=
ww.ietf.org/mailman/listinfo/mmusic</a><o:p></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
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So what happens if a UAC puts both an IPv4 and IPv6 addresses as ccap - is =
the UAS mandated to ignore it, or reject the request, or may it use it?

I ask because we've taken the altc draft outside of the IETF as an independ=
ent submission to the RFC editor, using the altc syntax.  If it would be be=
tter, we could instead change our independent submission to say "implement =
the sdp-icap-bcap using CCAP but ignore the MUST NOT =85". I don't know if =
that would be better or worse, so just asking.

-hadriel


On Jul 27, 2011, at 11:17 AM, Andrew Allen wrote:


Yesterday several MMUSIC participants including Jonathan Lennox and Miguel =
Garcia got together to discuss what to do regarding the CCAP attribute whic=
h is currently used in 3GPP specifications but was removed from the misc-ca=
p draft (which then became the sdp-icap-bcap draft without the CCAP attribu=
te). The CCAP attribute was removed after earlier comments on the list from=
 Jonathan regarding the possibility that CCAP might be used as an alternati=
ve to ICE for negotiating IP addresses for the C line (see email thread bel=
ow). 3GPP currently utilize the CCAP attribute with SDP-cap-neg for negotia=
tion of use of a circuit switched call for the audio media as an alternativ=
e to an audio RTP stream over packet switched access. CCAP is used to offer=
 the PSTN net type and E.164 number used to set up that call.

After a thorough discussion the conclusion of those involved in the discuss=
ion (including Jonathan) was that the best way forward given the constraint=
s of the 3GPP architecture is to resurrect the CCAP attribute but with some=
 wording that places constraints on its usage so that it isn=92t used to ne=
gotiate alternative addresses for a network type for which ICE can be used =
to negotiate addresses. This means that CCAP could be used to negotiate the=
 possibility to use PSTN net type as an alternative to the IN net type but =
that using alternative IP address candidates within the IN net type would n=
eed to be negotiated using ICE not using CCAP.

Are the any concerns from other MMUSIC participants if CCAP is resurrected =
(probably in an again renamed version of the sdp-icap-bcap draft that would=
 again include CCAP) with appropriate text to address the ICE issue along t=
he lines of above?

Andrew

 *   To: "Simo.Veikkolainen at nokia.com<mailto:Simo.Veikkolainen@DOMAIN.HI=
DDEN>" <Simo.Veikkolainen at nokia.com<mailto:Simo.Veikkolainen@DOMAIN.HIDD=
EN>>, "mmusic at ietf.org<mailto:mmusic@DOMAIN.HIDDEN>" <mmusic at ietf.org=
<mailto:mmusic@DOMAIN.HIDDEN>>
 *   Subject: Re: [MMUSIC] What to do with the ccap attribute?
 *   From: Jonathan Lennox <jonathan at vidyo.com<mailto:jonathan@DOMAIN.HI=
DDEN>>
 *   Date: Mon, 10 May 2010 11:24:36 -0400
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 *   Thread-topic: What to do with the ccap attribute?

________________________________

I think it'd be best to unify the requirements of this use case

(separate IP and PSTN addresses) with the requirements of the

IPv4/IPv6 transition use cases, and come up with a common solution.



Having two mechanisms (ANAT and ICE) to specify multiple connection

addresses is a standard but annoying consequence of a technology

transition. Having three (ANAT, ICE, and ALTC) gets unmanageable.

Having four (ANAT, ICE, ALTC, and CCAP) is just silly.



--

Jonathan Lennox

Vidyo, Inc

jonathan at vidyo.com<http://vidyo.com>



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

> From: Simo.Veikkolainen at nokia.com<http://nokia.com> [mailto:Simo.Veikk=
olainen at nokia.com<http://nokia.com>]

> Sent: Monday, May 10, 2010 5:45 AM

> To: mmusic at ietf.org<http://ietf.org>

> Subject: [MMUSIC] What to do with the ccap attribute?

>

> Now that the discussion on ICE and v6 transition has quieted down, I

> would like to ask for the list's opinion on how to proceed with the

> ccap attribute defined in http://tools.ietf.org/html/draft-ietf-mmusic-

> sdp-misc-cap

>

> Note that this draft will be re-submitted as an individual document

> according to the WG decision in IETF#77. Nevertheless it contains the

> most up-to-date text, hence the reference.

>

> As a short reminder, the misc-cap document defines capability

> attributes for "i=3D", "b"=3D and "c=3D" lines (icap, bcap and ccap,

> respectively). The ccap capability attribute is used to indicate

> alternative connection addresses. Our use case for this has been the

> ability to indicate alternative IP and PSTN addresses; a feature which

> is needed in some 3GPP scenarios.

>

> As ccap is currently specified, it includes also the port number in

> addition to the contents of a "c=3D" line (<nettype>, <addrtype> and the

> <connection-address>). The port number is needed in order to indicate a

> port number for internet addresses, the PSTN addresses don't have a

> similar concept.

>

> Now, as a side effect, the ccap attribute could also be used to

> circumvent ICE, if one used IP4 and IP6 addrtypes as alternatives.

>

> In order to move forward with the misc-cap document I'm now asking how

> do folks feel we should handle this issue. Possible alternatives are at

> least:

>

> 1. Leave the current text as is (see [1]).

>

> 2. Make a more strict statement that ccap SHOULD NOT be used for v4/v6

> address pairs.

>

> 3. Come up with altogether different way of indicating alternative

> connection addresses that fulfill our use case (PSTN/IP addresses).

>

> Regards,

> Simo

>

> [1] The text from Section 3.1.2 from  http://tools.ietf.org/html/draft-

> ietf-mmusic-sdp-misc-cap currently says

> "

>    The connection information capability can be used to negotiate the

>    use of IPv4 or IPv6 addresses without resort to Interactive

>    Connectivity Establishment(ICE) [I-D.ietf-mmusic-ice].  Note,

>    however, that ICE provides for real-time reachability testing of

>    multiple addresses, whereas use of the connection capability forces

>    an early choice of connection address.

>

>    [I-D.boucadair-mmusic-altc] describes a simple method of specifying

>    alternative network addresses when the transport protocol (<proto>)

>    is the same.

> "

> _______________________________________________

> mmusic mailing list

> mmusic at ietf.org<http://ietf.org>

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




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<html><head><base href=3D"x-msg://335/"></head><body style=3D"word-wrap: br=
eak-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
"><div>So what happens if a UAC puts both an IPv4 and IPv6 addresses as cca=
p - is the UAS mandated to ignore it, or reject the request, or may it use =
it?</div><div><br></div><div>I ask because we've taken the altc draft outsi=
de of the IETF as an independent submission to the RFC editor, using the al=
tc syntax. &nbsp;If it would be better, we could instead change our indepen=
dent submission to say "implement the sdp-icap-bcap using CCAP but ignore t=
he MUST NOT =85". I don't know if that would be better or worse, so just as=
king.</div><div><br></div><div>-hadriel</div><div><br></div><br><div><div>O=
n Jul 27, 2011, at 11:17 AM, Andrew Allen wrote:</div><br class=3D"Apple-in=
terchange-newline"><blockquote type=3D"cite"><span class=3D"Apple-style-spa=
n" style=3D"border-collapse: separate; font-family: Helvetica; font-style: =
normal; font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: 0px=
; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0p=
x; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto=
; -webkit-text-stroke-width: 0px; font-size: medium; "><div lang=3D"EN-US" =
link=3D"blue" vlink=3D"purple"><div class=3D"WordSection1" style=3D"page: W=
ordSection1; "><div style=3D"margin-top: 0in; margin-right: 0in; margin-lef=
t: 0in; margin-bottom: 0.0001pt; font-size: 11pt; font-family: Calibri, san=
s-serif; "><o:p>&nbsp;</o:p></div><div style=3D"margin-top: 0in; margin-rig=
ht: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 11pt; font-f=
amily: Calibri, sans-serif; ">Yesterday several MMUSIC participants includi=
ng Jonathan Lennox and Miguel Garcia got together to discuss what to do reg=
arding the CCAP attribute which is currently used in 3GPP specifications bu=
t was removed from the misc-cap draft (which then became the sdp-icap-bcap =
draft without the CCAP attribute). The CCAP attribute was removed after ear=
lier comments on the list from Jonathan regarding the possibility that CCAP=
 might be used as an alternative to ICE for negotiating IP addresses for th=
e C line (see email thread below). 3GPP currently utilize the CCAP attribut=
e with SDP-cap-neg for negotiation of use of a circuit switched call for th=
e audio media as an alternative to an audio RTP stream over packet switched=
 access. CCAP is used to offer the PSTN net type and E.164 number used to s=
et up that call.<o:p></o:p></div><div style=3D"margin-top: 0in; margin-righ=
t: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 11pt; font-fa=
mily: Calibri, sans-serif; "><o:p>&nbsp;</o:p></div><div style=3D"margin-to=
p: 0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-=
size: 11pt; font-family: Calibri, sans-serif; ">After a thorough discussion=
 the conclusion of those involved in the discussion (including Jonathan) wa=
s that the best way forward given the constraints of the 3GPP architecture =
is to resurrect the CCAP attribute but with some wording that places constr=
aints on its usage so that it isn=92t used to negotiate alternative address=
es for a network type for which ICE can be used to negotiate addresses. Thi=
s means that CCAP could be used to negotiate the possibility to use PSTN ne=
t type as an alternative to the IN net type but that using alternative IP a=
ddress candidates within the IN net type would need to be negotiated using =
ICE not using CCAP.<o:p></o:p></div><div style=3D"margin-top: 0in; margin-r=
ight: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 11pt; font=
-family: Calibri, sans-serif; "><o:p>&nbsp;</o:p></div><div style=3D"margin=
-top: 0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; fo=
nt-size: 11pt; font-family: Calibri, sans-serif; ">Are the any concerns fro=
m other MMUSIC participants if CCAP is resurrected (probably in an again re=
named version of the sdp-icap-bcap draft that would again include CCAP) wit=
h appropriate text to address the ICE issue along the lines of above?<o:p><=
/o:p></div><div style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0=
in; margin-bottom: 0.0001pt; font-size: 11pt; font-family: Calibri, sans-se=
rif; "><o:p>&nbsp;</o:p></div><div style=3D"margin-top: 0in; margin-right: =
0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 11pt; font-famil=
y: Calibri, sans-serif; ">Andrew<o:p></o:p></div><ul type=3D"disc" style=3D=
"margin-bottom: 0in; "><li class=3D"MsoNormal" style=3D"margin-top: 0in; ma=
rgin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 11pt=
; font-family: Calibri, sans-serif; "><em><span style=3D"font-family: Calib=
ri, sans-serif; ">To</span></em>: "<a href=3D"mailto:Simo.Veikkolainen@DOMA=
IN.HIDDEN" style=3D"color: blue; text-decoration: underline; ">Simo.Veikkol=
ainen at nokia.com</a>" &lt;<a href=3D"mailto:Simo.Veikkolainen@DOMAIN.HIDD=
EN" style=3D"color: blue; text-decoration: underline; ">Simo.Veikkolainen a=
t nokia.com</a>&gt;, "<a href=3D"mailto:mmusic@DOMAIN.HIDDEN" style=3D"colo=
r: blue; text-decoration: underline; ">mmusic at ietf.org</a>" &lt;<a href=
=3D"mailto:mmusic@DOMAIN.HIDDEN" style=3D"color: blue; text-decoration: und=
erline; ">mmusic at ietf.org</a>&gt;<o:p></o:p></li><li class=3D"MsoNormal"=
 style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; margin-bott=
om: 0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif; "><em><spa=
n style=3D"font-family: Calibri, sans-serif; ">Subject</span></em>: Re: [MM=
USIC] What to do with the ccap attribute?<o:p></o:p></li><li class=3D"MsoNo=
rmal" style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; margin=
-bottom: 0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif; "><em=
><span style=3D"font-family: Calibri, sans-serif; ">From</span></em>: Jonat=
han Lennox &lt;<a href=3D"mailto:jonathan@DOMAIN.HIDDEN" style=3D"color: bl=
ue; text-decoration: underline; ">jonathan at vidyo.com</a>&gt;<o:p></o:p><=
/li><li class=3D"MsoNormal" style=3D"margin-top: 0in; margin-right: 0in; ma=
rgin-left: 0in; margin-bottom: 0.0001pt; font-size: 11pt; font-family: Cali=
bri, sans-serif; "><em><span style=3D"font-family: Calibri, sans-serif; ">D=
ate</span></em>: Mon, 10 May 2010 11:24:36 -0400<o:p></o:p></li><li class=
=3D"MsoNormal" style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0i=
n; margin-bottom: 0.0001pt; font-size: 11pt; font-family: Calibri, sans-ser=
if; "><em><span style=3D"font-family: Calibri, sans-serif; ">Accept-languag=
e</span></em>: en-US<o:p></o:p></li><li class=3D"MsoNormal" style=3D"margin=
-top: 0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; fo=
nt-size: 11pt; font-family: Calibri, sans-serif; "><em><span style=3D"font-=
family: Calibri, sans-serif; ">Acceptlanguage</span></em>: en-US<o:p></o:p>=
</li><li class=3D"MsoNormal" style=3D"margin-top: 0in; margin-right: 0in; m=
argin-left: 0in; margin-bottom: 0.0001pt; font-size: 11pt; font-family: Cal=
ibri, sans-serif; "><em><span style=3D"font-family: Calibri, sans-serif; ">=
Delivered-to</span></em>:<span class=3D"Apple-converted-space">&nbsp;</span=
><a href=3D"mailto:mmusic@DOMAIN.HIDDEN" style=3D"color: blue; text-decorat=
ion: underline; ">mmusic at core3.amsl.com</a><o:p></o:p></li><li class=3D"=
MsoNormal" style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; m=
argin-bottom: 0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif; =
"><em><span style=3D"font-family: Calibri, sans-serif; ">In-reply-to</span>=
</em>: &lt;<a href=3D"mailto:B23B311878A0B4438F5F09F47E01AAE050870CAD88@DOM=
AIN.HIDDEN" style=3D"color: blue; text-decoration: underline; ">B23B311878A=
0B4438F5F09F47E01AAE050870CAD88 at NOK-EUMSG-01.mgdnok.nokia.com</a>&gt;<o:=
p></o:p></li><li class=3D"MsoNormal" style=3D"margin-top: 0in; margin-right=
: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 11pt; font-fam=
ily: Calibri, sans-serif; "><em><span style=3D"font-family: Calibri, sans-s=
erif; ">List-archive</span></em>: &lt;<a href=3D"http://www.ietf.org/mail-a=
rchive/web/mmusic" style=3D"color: blue; text-decoration: underline; ">http=
://www.ietf.org/mail-archive/web/mmusic</a>&gt;<o:p></o:p></li><li class=3D=
"MsoNormal" style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; =
margin-bottom: 0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif;=
 "><em><span style=3D"font-family: Calibri, sans-serif; ">List-help</span><=
/em>: &lt;<a href=3D"mailto:mmusic-request@ietf.org?subject=3Dhelp" style=
=3D"color: blue; text-decoration: underline; ">mailto:mmusic-request@ietf.o=
rg?subject=3Dhelp</a>&gt;<o:p></o:p></li><li class=3D"MsoNormal" style=3D"m=
argin-top: 0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001p=
t; font-size: 11pt; font-family: Calibri, sans-serif; "><em><span style=3D"=
font-family: Calibri, sans-serif; ">List-id</span></em>: Multiparty Multime=
dia Session Control Working Group &lt;<a href=3D"http://mmusic.ietf.org" st=
yle=3D"color: blue; text-decoration: underline; ">mmusic.ietf.org</a>&gt;<o=
:p></o:p></li><li class=3D"MsoNormal" style=3D"margin-top: 0in; margin-righ=
t: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 11pt; font-fa=
mily: Calibri, sans-serif; "><em><span style=3D"font-family: Calibri, sans-=
serif; ">List-post</span></em>: &lt;<a href=3D"mailto:mmusic@ietf.org" styl=
e=3D"color: blue; text-decoration: underline; ">mailto:mmusic@ietf.org</a>&=
gt;<o:p></o:p></li><li class=3D"MsoNormal" style=3D"margin-top: 0in; margin=
-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 11pt; fo=
nt-family: Calibri, sans-serif; "><em><span style=3D"font-family: Calibri, =
sans-serif; ">List-subscribe</span></em>: &lt;<a href=3D"https://www.ietf.o=
rg/mailman/listinfo/mmusic" style=3D"color: blue; text-decoration: underlin=
e; ">https://www.ietf.org/mailman/listinfo/mmusic</a>&gt;, &lt;<a href=3D"m=
ailto:mmusic-request@ietf.org?subject=3Dsubscribe" style=3D"color: blue; te=
xt-decoration: underline; ">mailto:mmusic-request@ietf.org?subject=3Dsubscr=
ibe</a>&gt;<o:p></o:p></li><li class=3D"MsoNormal" style=3D"margin-top: 0in=
; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
11pt; font-family: Calibri, sans-serif; "><em><span style=3D"font-family: C=
alibri, sans-serif; ">List-unsubscribe</span></em>: &lt;<a href=3D"https://=
www.ietf.org/mailman/listinfo/mmusic" style=3D"color: blue; text-decoration=
: underline; ">https://www.ietf.org/mailman/listinfo/mmusic</a>&gt;, &lt;<a=
 href=3D"mailto:mmusic-request@ietf.org?subject=3Dunsubscribe" style=3D"col=
or: blue; text-decoration: underline; ">mailto:mmusic-request@ietf.org?subj=
ect=3Dunsubscribe</a>&gt;<o:p></o:p></li><li class=3D"MsoNormal" style=3D"m=
argin-top: 0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001p=
t; font-size: 11pt; font-family: Calibri, sans-serif; "><em><span style=3D"=
font-family: Calibri, sans-serif; ">References</span></em>: &lt;<a href=3D"=
mailto:B23B311878A0B4438F5F09F47E01AAE050870CAD88@DOMAIN.HIDDEN" style=3D"c=
olor: blue; text-decoration: underline; ">B23B311878A0B4438F5F09F47E01AAE05=
0870CAD88 at NOK-EUMSG-01.mgdnok.nokia.com</a>&gt;<o:p></o:p></li><li class=
=3D"MsoNormal" style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0i=
n; margin-bottom: 0.0001pt; font-size: 11pt; font-family: Calibri, sans-ser=
if; "><em><span style=3D"font-family: Calibri, sans-serif; ">Thread-index</=
span></em>: AcrwJWqZpvLJ586JTjK5XQiTSB6kSAALoU1w<o:p></o:p></li><li class=
=3D"MsoNormal" style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0i=
n; margin-bottom: 0.0001pt; font-size: 11pt; font-family: Calibri, sans-ser=
if; "><em><span style=3D"font-family: Calibri, sans-serif; ">Thread-topic</=
span></em>: What to do with the ccap attribute?<o:p></o:p></li></ul><div cl=
ass=3D"MsoNormal" align=3D"center" style=3D"margin-top: 0in; margin-right: =
0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 11pt; font-famil=
y: Calibri, sans-serif; text-align: center; "><hr size=3D"2" width=3D"100%"=
 align=3D"center"></div><pre style=3D"margin-top: 0in; margin-right: 0in; m=
argin-left: 0in; margin-bottom: 0.0001pt; font-size: 10pt; font-family: 'Co=
urier New'; ">I think it'd be best to unify the requirements of this use ca=
se <o:p></o:p></pre><pre style=3D"margin-top: 0in; margin-right: 0in; margi=
n-left: 0in; margin-bottom: 0.0001pt; font-size: 10pt; font-family: 'Courie=
r New'; ">(separate IP and PSTN addresses) with the requirements of the <o:=
p></o:p></pre><pre style=3D"margin-top: 0in; margin-right: 0in; margin-left=
: 0in; margin-bottom: 0.0001pt; font-size: 10pt; font-family: 'Courier New'=
; ">IPv4/IPv6 transition use cases, and come up with a common solution.<o:p=
></o:p></pre><pre style=3D"margin-top: 0in; margin-right: 0in; margin-left:=
 0in; margin-bottom: 0.0001pt; font-size: 10pt; font-family: 'Courier New';=
 "><o:p>&nbsp;</o:p></pre><pre style=3D"margin-top: 0in; margin-right: 0in;=
 margin-left: 0in; margin-bottom: 0.0001pt; font-size: 10pt; font-family: '=
Courier New'; ">Having two mechanisms (ANAT and ICE) to specify multiple co=
nnection <o:p></o:p></pre><pre style=3D"margin-top: 0in; margin-right: 0in;=
 margin-left: 0in; margin-bottom: 0.0001pt; font-size: 10pt; font-family: '=
Courier New'; ">addresses is a standard but annoying consequence of a techn=
ology <o:p></o:p></pre><pre style=3D"margin-top: 0in; margin-right: 0in; ma=
rgin-left: 0in; margin-bottom: 0.0001pt; font-size: 10pt; font-family: 'Cou=
rier New'; ">transition. Having three (ANAT, ICE, and ALTC) gets unmanageab=
le. <o:p></o:p></pre><pre style=3D"margin-top: 0in; margin-right: 0in; marg=
in-left: 0in; margin-bottom: 0.0001pt; font-size: 10pt; font-family: 'Couri=
er New'; ">Having four (ANAT, ICE, ALTC, and CCAP) is just silly.<o:p></o:p=
></pre><pre style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; =
margin-bottom: 0.0001pt; font-size: 10pt; font-family: 'Courier New'; "><o:=
p>&nbsp;</o:p></pre><pre style=3D"margin-top: 0in; margin-right: 0in; margi=
n-left: 0in; margin-bottom: 0.0001pt; font-size: 10pt; font-family: 'Courie=
r New'; ">-- <o:p></o:p></pre><pre style=3D"margin-top: 0in; margin-right: =
0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 10pt; font-famil=
y: 'Courier New'; ">Jonathan Lennox<o:p></o:p></pre><pre style=3D"margin-to=
p: 0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-=
size: 10pt; font-family: 'Courier New'; ">Vidyo, Inc<o:p></o:p></pre><pre s=
tyle=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; margin-bottom=
: 0.0001pt; font-size: 10pt; font-family: 'Courier New'; ">jonathan at <a h=
ref=3D"http://vidyo.com" style=3D"color: blue; text-decoration: underline; =
">vidyo.com</a><o:p></o:p></pre><pre style=3D"margin-top: 0in; margin-right=
: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 10pt; font-fam=
ily: 'Courier New'; "><o:p>&nbsp;</o:p></pre><pre style=3D"margin-top: 0in;=
 margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 1=
0pt; font-family: 'Courier New'; ">&gt; -----Original Message-----<o:p></o:=
p></pre><pre style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in;=
 margin-bottom: 0.0001pt; font-size: 10pt; font-family: 'Courier New'; ">&g=
t; From: Simo.Veikkolainen at <a href=3D"http://nokia.com" style=3D"color: =
blue; text-decoration: underline; ">nokia.com</a> [<a href=3D"mailto:Simo.V=
eikkolainen" style=3D"color: blue; text-decoration: underline; ">mailto:Sim=
o.Veikkolainen</a> at <a href=3D"http://nokia.com" style=3D"color: blue; te=
xt-decoration: underline; ">nokia.com</a>]<o:p></o:p></pre><pre style=3D"ma=
rgin-top: 0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt=
; font-size: 10pt; font-family: 'Courier New'; ">&gt; Sent: Monday, May 10,=
 2010 5:45 AM<o:p></o:p></pre><pre style=3D"margin-top: 0in; margin-right: =
0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 10pt; font-famil=
y: 'Courier New'; ">&gt; To: mmusic at <a href=3D"http://ietf.org" style=3D=
"color: blue; text-decoration: underline; ">ietf.org</a><o:p></o:p></pre><p=
re style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; margin-bo=
ttom: 0.0001pt; font-size: 10pt; font-family: 'Courier New'; ">&gt; Subject=
: [MMUSIC] What to do with the ccap attribute?<o:p></o:p></pre><pre style=
=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.=
0001pt; font-size: 10pt; font-family: 'Courier New'; ">&gt; <o:p></o:p></pr=
e><pre style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; margi=
n-bottom: 0.0001pt; font-size: 10pt; font-family: 'Courier New'; ">&gt; Now=
 that the discussion on ICE and v6 transition has quieted down, I<o:p></o:p=
></pre><pre style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; =
margin-bottom: 0.0001pt; font-size: 10pt; font-family: 'Courier New'; ">&gt=
; would like to ask for the list's opinion on how to proceed with the<o:p><=
/o:p></pre><pre style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0=
in; margin-bottom: 0.0001pt; font-size: 10pt; font-family: 'Courier New'; "=
>&gt; ccap attribute defined in <a href=3D"http://tools.ietf.org/html/draft=
-ietf-mmusic-" style=3D"color: blue; text-decoration: underline; ">http://t=
ools.ietf.org/html/draft-ietf-mmusic-</a><o:p></o:p></pre><pre style=3D"mar=
gin-top: 0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt;=
 font-size: 10pt; font-family: 'Courier New'; ">&gt; sdp-misc-cap<o:p></o:p=
></pre><pre style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; =
margin-bottom: 0.0001pt; font-size: 10pt; font-family: 'Courier New'; ">&gt=
; <o:p></o:p></pre><pre style=3D"margin-top: 0in; margin-right: 0in; margin=
-left: 0in; margin-bottom: 0.0001pt; font-size: 10pt; font-family: 'Courier=
 New'; ">&gt; Note that this draft will be re-submitted as an individual do=
cument<o:p></o:p></pre><pre style=3D"margin-top: 0in; margin-right: 0in; ma=
rgin-left: 0in; margin-bottom: 0.0001pt; font-size: 10pt; font-family: 'Cou=
rier New'; ">&gt; according to the WG decision in IETF#77. Nevertheless it =
contains the<o:p></o:p></pre><pre style=3D"margin-top: 0in; margin-right: 0=
in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 10pt; font-family=
: 'Courier New'; ">&gt; most up-to-date text, hence the reference.<o:p></o:=
p></pre><pre style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in;=
 margin-bottom: 0.0001pt; font-size: 10pt; font-family: 'Courier New'; ">&g=
t; <o:p></o:p></pre><pre style=3D"margin-top: 0in; margin-right: 0in; margi=
n-left: 0in; margin-bottom: 0.0001pt; font-size: 10pt; font-family: 'Courie=
r New'; ">&gt; As a short reminder, the misc-cap document defines capabilit=
y<o:p></o:p></pre><pre style=3D"margin-top: 0in; margin-right: 0in; margin-=
left: 0in; margin-bottom: 0.0001pt; font-size: 10pt; font-family: 'Courier =
New'; ">&gt; attributes for "i=3D", "b"=3D and "c=3D" lines (icap, bcap and=
 ccap,<o:p></o:p></pre><pre style=3D"margin-top: 0in; margin-right: 0in; ma=
rgin-left: 0in; margin-bottom: 0.0001pt; font-size: 10pt; font-family: 'Cou=
rier New'; ">&gt; respectively). The ccap capability attribute is used to i=
ndicate<o:p></o:p></pre><pre style=3D"margin-top: 0in; margin-right: 0in; m=
argin-left: 0in; margin-bottom: 0.0001pt; font-size: 10pt; font-family: 'Co=
urier New'; ">&gt; alternative connection addresses. Our use case for this =
has been the<o:p></o:p></pre><pre style=3D"margin-top: 0in; margin-right: 0=
in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 10pt; font-family=
: 'Courier New'; ">&gt; ability to indicate alternative IP and PSTN address=
es; a feature which<o:p></o:p></pre><pre style=3D"margin-top: 0in; margin-r=
ight: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 10pt; font=
-family: 'Courier New'; ">&gt; is needed in some 3GPP scenarios.<o:p></o:p>=
</pre><pre style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; m=
argin-bottom: 0.0001pt; font-size: 10pt; font-family: 'Courier New'; ">&gt;=
 <o:p></o:p></pre><pre style=3D"margin-top: 0in; margin-right: 0in; margin-=
left: 0in; margin-bottom: 0.0001pt; font-size: 10pt; font-family: 'Courier =
New'; ">&gt; As ccap is currently specified, it includes also the port numb=
er in<o:p></o:p></pre><pre style=3D"margin-top: 0in; margin-right: 0in; mar=
gin-left: 0in; margin-bottom: 0.0001pt; font-size: 10pt; font-family: 'Cour=
ier New'; ">&gt; addition to the contents of a "c=3D" line (&lt;nettype&gt;=
, &lt;addrtype&gt; and the<o:p></o:p></pre><pre style=3D"margin-top: 0in; m=
argin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 10p=
t; font-family: 'Courier New'; ">&gt; &lt;connection-address&gt;). The port=
 number is needed in order to indicate a<o:p></o:p></pre><pre style=3D"marg=
in-top: 0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; =
font-size: 10pt; font-family: 'Courier New'; ">&gt; port number for interne=
t addresses, the PSTN addresses don't have a<o:p></o:p></pre><pre style=3D"=
margin-top: 0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001=
pt; font-size: 10pt; font-family: 'Courier New'; ">&gt; similar concept.<o:=
p></o:p></pre><pre style=3D"margin-top: 0in; margin-right: 0in; margin-left=
: 0in; margin-bottom: 0.0001pt; font-size: 10pt; font-family: 'Courier New'=
; ">&gt; <o:p></o:p></pre><pre style=3D"margin-top: 0in; margin-right: 0in;=
 margin-left: 0in; margin-bottom: 0.0001pt; font-size: 10pt; font-family: '=
Courier New'; ">&gt; Now, as a side effect, the ccap attribute could also b=
e used to<o:p></o:p></pre><pre style=3D"margin-top: 0in; margin-right: 0in;=
 margin-left: 0in; margin-bottom: 0.0001pt; font-size: 10pt; font-family: '=
Courier New'; ">&gt; circumvent ICE, if one used IP4 and IP6 addrtypes as a=
lternatives.<o:p></o:p></pre><pre style=3D"margin-top: 0in; margin-right: 0=
in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 10pt; font-family=
: 'Courier New'; ">&gt; <o:p></o:p></pre><pre style=3D"margin-top: 0in; mar=
gin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 10pt;=
 font-family: 'Courier New'; ">&gt; In order to move forward with the misc-=
cap document I'm now asking how<o:p></o:p></pre><pre style=3D"margin-top: 0=
in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size=
: 10pt; font-family: 'Courier New'; ">&gt; do folks feel we should handle t=
his issue. Possible alternatives are at<o:p></o:p></pre><pre style=3D"margi=
n-top: 0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; f=
ont-size: 10pt; font-family: 'Courier New'; ">&gt; least:<o:p></o:p></pre><=
pre style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; margin-b=
ottom: 0.0001pt; font-size: 10pt; font-family: 'Courier New'; ">&gt; <o:p><=
/o:p></pre><pre style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0=
in; margin-bottom: 0.0001pt; font-size: 10pt; font-family: 'Courier New'; "=
>&gt; 1. Leave the current text as is (see [1]).<o:p></o:p></pre><pre style=
=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.=
0001pt; font-size: 10pt; font-family: 'Courier New'; ">&gt; <o:p></o:p></pr=
e><pre style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; margi=
n-bottom: 0.0001pt; font-size: 10pt; font-family: 'Courier New'; ">&gt; 2. =
Make a more strict statement that ccap SHOULD NOT be used for v4/v6<o:p></o=
:p></pre><pre style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in=
; margin-bottom: 0.0001pt; font-size: 10pt; font-family: 'Courier New'; ">&=
gt; address pairs.<o:p></o:p></pre><pre style=3D"margin-top: 0in; margin-ri=
ght: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 10pt; font-=
family: 'Courier New'; ">&gt; <o:p></o:p></pre><pre style=3D"margin-top: 0i=
n; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size:=
 10pt; font-family: 'Courier New'; ">&gt; 3. Come up with altogether differ=
ent way of indicating alternative<o:p></o:p></pre><pre style=3D"margin-top:=
 0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-si=
ze: 10pt; font-family: 'Courier New'; ">&gt; connection addresses that fulf=
ill our use case (PSTN/IP addresses).<o:p></o:p></pre><pre style=3D"margin-=
top: 0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; fon=
t-size: 10pt; font-family: 'Courier New'; ">&gt; <o:p></o:p></pre><pre styl=
e=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0=
.0001pt; font-size: 10pt; font-family: 'Courier New'; ">&gt; Regards,<o:p><=
/o:p></pre><pre style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0=
in; margin-bottom: 0.0001pt; font-size: 10pt; font-family: 'Courier New'; "=
>&gt; Simo<o:p></o:p></pre><pre style=3D"margin-top: 0in; margin-right: 0in=
; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 10pt; font-family: =
'Courier New'; ">&gt; <o:p></o:p></pre><pre style=3D"margin-top: 0in; margi=
n-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 10pt; f=
ont-family: 'Courier New'; ">&gt; [1] The text from Section 3.1.2 from&nbsp=
; <a href=3D"http://tools.ietf.org/html/draft-" style=3D"color: blue; text-=
decoration: underline; ">http://tools.ietf.org/html/draft-</a><o:p></o:p></=
pre><pre style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; mar=
gin-bottom: 0.0001pt; font-size: 10pt; font-family: 'Courier New'; ">&gt; i=
etf-mmusic-sdp-misc-cap currently says<o:p></o:p></pre><pre style=3D"margin=
-top: 0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; fo=
nt-size: 10pt; font-family: 'Courier New'; ">&gt; "<o:p></o:p></pre><pre st=
yle=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; margin-bottom:=
 0.0001pt; font-size: 10pt; font-family: 'Courier New'; ">&gt;&nbsp;&nbsp;&=
nbsp; The connection information capability can be used to negotiate the<o:=
p></o:p></pre><pre style=3D"margin-top: 0in; margin-right: 0in; margin-left=
: 0in; margin-bottom: 0.0001pt; font-size: 10pt; font-family: 'Courier New'=
; ">&gt;&nbsp;&nbsp;&nbsp; use of IPv4 or IPv6 addresses without resort to =
Interactive<o:p></o:p></pre><pre style=3D"margin-top: 0in; margin-right: 0i=
n; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 10pt; font-family:=
 'Courier New'; ">&gt;&nbsp;&nbsp;&nbsp; Connectivity Establishment(ICE) [I=
-D.ietf-mmusic-ice].&nbsp; Note,<o:p></o:p></pre><pre style=3D"margin-top: =
0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-siz=
e: 10pt; font-family: 'Courier New'; ">&gt;&nbsp;&nbsp;&nbsp; however, that=
 ICE provides for real-time reachability testing of<o:p></o:p></pre><pre st=
yle=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; margin-bottom:=
 0.0001pt; font-size: 10pt; font-family: 'Courier New'; ">&gt;&nbsp;&nbsp;&=
nbsp; multiple addresses, whereas use of the connection capability forces<o=
:p></o:p></pre><pre style=3D"margin-top: 0in; margin-right: 0in; margin-lef=
t: 0in; margin-bottom: 0.0001pt; font-size: 10pt; font-family: 'Courier New=
'; ">&gt;&nbsp;&nbsp;&nbsp; an early choice of connection address.<o:p></o:=
p></pre><pre style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in;=
 margin-bottom: 0.0001pt; font-size: 10pt; font-family: 'Courier New'; ">&g=
t; <o:p></o:p></pre><pre style=3D"margin-top: 0in; margin-right: 0in; margi=
n-left: 0in; margin-bottom: 0.0001pt; font-size: 10pt; font-family: 'Courie=
r New'; ">&gt;&nbsp;&nbsp;&nbsp; [I-D.boucadair-mmusic-altc] describes a si=
mple method of specifying<o:p></o:p></pre><pre style=3D"margin-top: 0in; ma=
rgin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 10pt=
; font-family: 'Courier New'; ">&gt;&nbsp;&nbsp;&nbsp; alternative network =
addresses when the transport protocol (&lt;proto&gt;)<o:p></o:p></pre><pre =
style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; margin-botto=
m: 0.0001pt; font-size: 10pt; font-family: 'Courier New'; ">&gt;&nbsp;&nbsp=
;&nbsp; is the same.<o:p></o:p></pre><pre style=3D"margin-top: 0in; margin-=
right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 10pt; fon=
t-family: 'Courier New'; ">&gt; "<o:p></o:p></pre><pre style=3D"margin-top:=
 0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-si=
ze: 10pt; font-family: 'Courier New'; ">&gt; ______________________________=
_________________<o:p></o:p></pre><pre style=3D"margin-top: 0in; margin-rig=
ht: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 10pt; font-f=
amily: 'Courier New'; ">&gt; mmusic mailing list<o:p></o:p></pre><pre style=
=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.=
0001pt; font-size: 10pt; font-family: 'Courier New'; ">&gt; mmusic at <a hr=
ef=3D"http://ietf.org" style=3D"color: blue; text-decoration: underline; ">=
ietf.org</a><o:p></o:p></pre><pre style=3D"margin-top: 0in; margin-right: 0=
in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 10pt; font-family=
: 'Courier New'; ">&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/mm=
usic" style=3D"color: blue; text-decoration: underline; ">https://www.ietf.=
org/mailman/listinfo/mmusic</a><o:p></o:p></pre><pre style=3D"margin-top: 0=
in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size=
: 10pt; font-family: 'Courier New'; "><o:p>&nbsp;</o:p></pre><div style=3D"=
margin-top: 0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001=
pt; font-size: 11pt; font-family: Calibri, sans-serif; "><o:p>&nbsp;</o:p><=
/div></div>----------------------------------------------------------------=
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on (including any attachments) may contain confidential information, privil=
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r applicable privileges), or constitute non-public information. Any use of =
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 If you have received this transmission in error, please immediately reply =
to the sender and delete this information from your system. Use, disseminat=
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space">&nbsp;</span><span>&lt;ATT00001..c&gt;</span></div></span></blockquo=
te></div><br></body></html>=

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From miguel.a.garcia@ericsson.com  Wed Jul 27 11:18:04 2011
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Date: Wed, 27 Jul 2011 20:17:58 +0200
From: "Miguel A. Garcia" <Miguel.A.Garcia@ericsson.com>
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To: Hadriel Kaplan <HKaplan@acmepacket.com>
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] resurecting the CCAP attribute
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As an individual....

Hadriel,

As far as I understand, the requirement from 3GPP is to be able to offer 
an audio stream of circuit-switched or over regular IP. As such, the 
authors of the misc-cap draft do not have a strong requirement for 
letting a mixture of different network types (3GPP requirement) or the 
same (your requirement, the altc draft).

I would say that we need to clarify whether we can come up with a text 
that indicates what is the intended usage and the allowed usage.

/Miguel



On 27/07/2011 19:33, Hadriel Kaplan wrote:
> So what happens if a UAC puts both an IPv4 and IPv6 addresses as ccap -
> is the UAS mandated to ignore it, or reject the request, or may it use it?
>
> I ask because we've taken the altc draft outside of the IETF as an
> independent submission to the RFC editor, using the altc syntax. If it
> would be better, we could instead change our independent submission to
> say "implement the sdp-icap-bcap using CCAP but ignore the MUST NOT …". I
> don't know if that would be better or worse, so just asking.
>
> -hadriel
>
>
> On Jul 27, 2011, at 11:17 AM, Andrew Allen wrote:
>
>> Yesterday several MMUSIC participants including Jonathan Lennox and
>> Miguel Garcia got together to discuss what to do regarding the CCAP
>> attribute which is currently used in 3GPP specifications but was
>> removed from the misc-cap draft (which then became the sdp-icap-bcap
>> draft without the CCAP attribute). The CCAP attribute was removed after
>> earlier comments on the list from Jonathan regarding the possibility
>> that CCAP might be used as an alternative to ICE for negotiating IP
>> addresses for the C line (see email thread below). 3GPP currently
>> utilize the CCAP attribute with SDP-cap-neg for negotiation of use of a
>> circuit switched call for the audio media as an alternative to an audio
>> RTP stream over packet switched access. CCAP is used to offer the PSTN
>> net type and E.164 number used to set up that call.
>> After a thorough discussion the conclusion of those involved in the
>> discussion (including Jonathan) was that the best way forward given the
>> constraints of the 3GPP architecture is to resurrect the CCAP attribute
>> but with some wording that places constraints on its usage so that it
>> isn’t used to negotiate alternative addresses for a network type for
>> which ICE can be used to negotiate addresses. This means that CCAP
>> could be used to negotiate the possibility to use PSTN net type as an
>> alternative to the IN net type but that using alternative IP address
>> candidates within the IN net type would need to be negotiated using ICE
>> not using CCAP.
>> Are the any concerns from other MMUSIC participants if CCAP is
>> resurrected (probably in an again renamed version of the sdp-icap-bcap
>> draft that would again include CCAP) with appropriate text to address
>> the ICE issue along the lines of above?
>> Andrew
>>
>>   * /To/: "Simo.Veikkolainen at nokia.com
>>     <mailto:Simo.Veikkolainen@DOMAIN.HIDDEN>" <Simo.Veikkolainen at
>>     nokia.com <mailto:Simo.Veikkolainen@DOMAIN.HIDDEN>>, "mmusic at
>>     ietf.org <mailto:mmusic@DOMAIN.HIDDEN>" <mmusic at ietf.org
>>     <mailto:mmusic@DOMAIN.HIDDEN>>
>>   * /Subject/: Re: [MMUSIC] What to do with the ccap attribute?
>>   * /From/: Jonathan Lennox <jonathan at vidyo.com
>>     <mailto:jonathan@DOMAIN.HIDDEN>>
>>   * /Date/: Mon, 10 May 2010 11:24:36 -0400
>>   * /Accept-language/: en-US
>>   * /Acceptlanguage/: en-US
>>   * /Delivered-to/:mmusic at core3.amsl.com <mailto:mmusic@DOMAIN.HIDDEN>
>>   * /In-reply-to/: <B23B311878A0B4438F5F09F47E01AAE050870CAD88 at
>>     NOK-EUMSG-01.mgdnok.nokia.com
>>     <mailto:B23B311878A0B4438F5F09F47E01AAE050870CAD88@DOMAIN.HIDDEN>>
>>   * /List-archive/: <http://www.ietf.org/mail-archive/web/mmusic>
>>   * /List-help/: <mailto:mmusic-request@ietf.org?subject=help>
>>   * /List-id/: Multiparty Multimedia Session Control Working Group
>>     <mmusic.ietf.org <http://mmusic.ietf.org>>
>>   * /List-post/: <mailto:mmusic@ietf.org>
>>   * /List-subscribe/: <https://www.ietf.org/mailman/listinfo/mmusic>,
>>     <mailto:mmusic-request@ietf.org?subject=subscribe>
>>   * /List-unsubscribe/: <https://www.ietf.org/mailman/listinfo/mmusic>,
>>     <mailto:mmusic-request@ietf.org?subject=unsubscribe>
>>   * /References/: <B23B311878A0B4438F5F09F47E01AAE050870CAD88 at
>>     NOK-EUMSG-01.mgdnok.nokia.com
>>     <mailto:B23B311878A0B4438F5F09F47E01AAE050870CAD88@DOMAIN.HIDDEN>>
>>   * /Thread-index/: AcrwJWqZpvLJ586JTjK5XQiTSB6kSAALoU1w
>>   * /Thread-topic/: What to do with the ccap attribute?
>>
>> -------------------------------------------------------------------------
>> I think it'd be best to unify the requirements of this use case
>> (separate IP and PSTN addresses) with the requirements of the
>> IPv4/IPv6 transition use cases, and come up with a common solution.
>>
>> Having two mechanisms (ANAT and ICE) to specify multiple connection
>> addresses is a standard but annoying consequence of a technology
>> transition. Having three (ANAT, ICE, and ALTC) gets unmanageable.
>> Having four (ANAT, ICE, ALTC, and CCAP) is just silly.
>>
>> --
>> Jonathan Lennox
>> Vidyo, Inc
>> jonathan atvidyo.com  <http://vidyo.com>
>>
>> >  -----Original Message-----
>> >  From: Simo.Veikkolainen atnokia.com  <http://nokia.com>  [mailto:Simo.Veikkolainen  atnokia.com  <http://nokia.com>]
>> >  Sent: Monday, May 10, 2010 5:45 AM
>> >  To: mmusic atietf.org  <http://ietf.org>
>> >  Subject: [MMUSIC] What to do with the ccap attribute?
>> >
>> >  Now that the discussion on ICE and v6 transition has quieted down, I
>> >  would like to ask for the list's opinion on how to proceed with the
>> >  ccap attribute defined inhttp://tools.ietf.org/html/draft-ietf-mmusic-
>> >  sdp-misc-cap
>> >
>> >  Note that this draft will be re-submitted as an individual document
>> >  according to the WG decision in IETF#77. Nevertheless it contains the
>> >  most up-to-date text, hence the reference.
>> >
>> >  As a short reminder, the misc-cap document defines capability
>> >  attributes for "i=", "b"= and "c=" lines (icap, bcap and ccap,
>> >  respectively). The ccap capability attribute is used to indicate
>> >  alternative connection addresses. Our use case for this has been the
>> >  ability to indicate alternative IP and PSTN addresses; a feature which
>> >  is needed in some 3GPP scenarios.
>> >
>> >  As ccap is currently specified, it includes also the port number in
>> >  addition to the contents of a "c=" line (<nettype>,<addrtype>  and the
>> >  <connection-address>). The port number is needed in order to indicate a
>> >  port number for internet addresses, the PSTN addresses don't have a
>> >  similar concept.
>> >
>> >  Now, as a side effect, the ccap attribute could also be used to
>> >  circumvent ICE, if one used IP4 and IP6 addrtypes as alternatives.
>> >
>> >  In order to move forward with the misc-cap document I'm now asking how
>> >  do folks feel we should handle this issue. Possible alternatives are at
>> >  least:
>> >
>> >  1. Leave the current text as is (see [1]).
>> >
>> >  2. Make a more strict statement that ccap SHOULD NOT be used for v4/v6
>> >  address pairs.
>> >
>> >  3. Come up with altogether different way of indicating alternative
>> >  connection addresses that fulfill our use case (PSTN/IP addresses).
>> >
>> >  Regards,
>> >  Simo
>> >
>> >  [1] The text from Section 3.1.2 from    http://tools.ietf.org/html/draft-
>> >  ietf-mmusic-sdp-misc-cap currently says
>> >  "
>> >        The connection information capability can be used to negotiate the
>> >        use of IPv4 or IPv6 addresses without resort to Interactive
>> >        Connectivity Establishment(ICE) [I-D.ietf-mmusic-ice].    Note,
>> >        however, that ICE provides for real-time reachability testing of
>> >        multiple addresses, whereas use of the connection capability forces
>> >        an early choice of connection address.
>> >
>> >        [I-D.boucadair-mmusic-altc] describes a simple method of specifying
>> >        alternative network addresses when the transport protocol (<proto>)
>> >        is the same.
>> >  "
>> >  _______________________________________________
>> >  mmusic mailing list
>> >  mmusic atietf.org  <http://ietf.org>
>> >  https://www.ietf.org/mailman/listinfo/mmusic
>>
>> ---------------------------------------------------------------------
>> This transmission (including any attachments) may contain confidential
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>> transmission in error, please immediately reply to the sender and
>> delete this information from your system. Use, dissemination,
>> distribution, or reproduction of this transmission by unintended
>> recipients is not authorized and may be unlawful.<ATT00001..c>
>

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

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Cc: Flemming Andreasen <fandreas@cisco.com>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] Multipath RTP (interface advertisement in SDP)
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As a chair.

We encourage everyone to read this draft and provide comments to the 
protocols that deal with MMUSIC, namely, SDP and ICE.

Nevertheless, Christer Holmberg has volunteered to provide a review of 
the document from the point of view of MMUSIC. It should be ready within 
the next two weeks. Thank you Christer!

/Miguel

On 25/07/2011 14:26, Varun Singh wrote:
> Greetings,
>
> The draft is part of AVTCore but we have recently introduced an "interface" attribute in SDP, as an alternative to in-band interface advertisement. The draft also contains some O/A examples and these changes can be found in the SDP Considerations section. http://tools.ietf.org/rfcdiff?url2=draft-singh-avtcore-mprtp-02.txt
>
> Comments and suggestions are welcome from MMUSIC.
>
> Regards,
> Varun
>
> Begin forwarded message:
>>
>> A new version of I-D, draft-singh-avtcore-mprtp-02.txt has been successfully submitted by Varun Singh and posted to the IETF repository.
>>
>> Filename: draft-singh-avtcore-mprtp
>> Revision: 02
>> Title: Multipath RTP (MPRTP)
>> Creation date: 2011-07-08
>> WG ID: Individual Submission
>> Number of pages: 34
>>
>> Abstract:
>> The Real-time Transport Protocol (RTP) is used to deliver real-time
>> content and, along with the RTP Control Protocol (RTCP), forms the
>> control channel between the sender and receiver. However, RTP and
>> RTCP assume a single delivery path between the sender and receiver
>> and make decisions based on the measured characteristics of this
>> single path. Increasingly, endpoints are becoming multi-homed, which
>> means that they are connected via multiple Internet paths. Network
>> utilization can be improved when endpoints use multiple parallel
>> paths for communication. The resulting increase in reliability and
>> throughput can also enhance the user experience. This document
>> extends the Real-time Transport Protocol (RTP) so that a single
>> session can take advantage of the availability of multiple paths
>> between two endpoints.
>>
>>
>>
>>
>> The IETF Secretariat

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

From HKaplan@acmepacket.com  Wed Jul 27 16:10:27 2011
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To: "Miguel A. Garcia" <Miguel.A.Garcia@ericsson.com>
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Thread-Topic: [MMUSIC] resurecting the CCAP attribute
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On Jul 27, 2011, at 2:17 PM, Miguel A. Garcia wrote:

> As an individual....
> As far as I understand, the requirement from 3GPP is to be able to offer=
=20
> an audio stream of circuit-switched or over regular IP. As such, the=20
> authors of the misc-cap draft do not have a strong requirement for=20
> letting a mixture of different network types (3GPP requirement) or the=20
> same (your requirement, the altc draft).

Would a requirement from 3GPP help?  If so we can submit a CR and start the=
 process in 3gpp to do so.  It's not like I *want* to offer both ipv4+ipv6,=
 but the service providers are demanding it, both for transitioning their I=
MS-based core, and for access and peering.


> I would say that we need to clarify whether we can come up with a text=20
> that indicates what is the intended usage and the allowed usage.

Yeah some text to point at would help. :)

-hadriel


From paulbeaumont.ietf@gmail.com  Thu Jul 28 12:18:59 2011
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All

Apologies, being first time to IETF and its mailing lists, if this is not th=
e right mailing list to post this request to.

On initial inspection it looks like it is not possible to specify in SDP the=
 specific packetisation period associated with a codec that is available thr=
ough negotiation. Specific case I am concerned with is to support (1) G.711 @=
 PP=3D10ms and (2) G.711 @ PP=3D20ms; prioritised in that order. The ptime p=
arameter would appear to apply to both.

Please can you advise where I should submit this question or if this is the r=
ight forum then whether the above observation is correct and any supplementa=
l information/comments you may have?

Thanks
Paul Beaumont


--Apple-Mail-46-713619380
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<html><body bgcolor=3D"#FFFFFF"><div><div></div><div><span class=3D"Apple-st=
yle-span" style=3D"-webkit-tap-highlight-color: rgba(26, 26, 26, 0.296875); -=
webkit-composition-fill-color: rgba(175, 192, 227, 0.230469); -webkit-compos=
ition-frame-color: rgba(77, 128, 180, 0.230469); font-size: medium; color: r=
gb(0, 35, 163); ">All<br><br>Apologies, being first time to IETF and its mai=
ling lists, if this is not the right mailing list to post this request to.<b=
r><br>On initial inspection it looks like it is not possible to specify in S=
DP the specific packetisation period associated with a codec that is availab=
le through negotiation. Specific case I am concerned with is to support (1) G=
.711 @ PP=3D10ms and (2) G.711 @ PP=3D20ms; prioritised in that order. The p=
time parameter would appear to apply to both.<br><br>Please can you advise w=
here I should submit this question or if this is the right forum then whethe=
r the above observation is correct and any supplemental information/comments=
 you may have?<br><br>Thanks<br>Paul Beaumont</span><br><br></div></div><div=
></div></body></html>=

--Apple-Mail-46-713619380--

From randell-ietf@jesup.org  Thu Jul 28 12:38:34 2011
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On 7/28/2011 3:19 PM, Paul Beaumont wrote:
> On initial inspection it looks like it is not possible to specify in 
> SDP the specific packetisation period associated with a codec that is 
> available through negotiation. Specific case I am concerned with is to 
> support (1) G.711 @ PP=10ms and (2) G.711 @ PP=20ms; prioritised in 
> that order. The ptime parameter would appear to apply to both.

You can't specify the packetization; you can merely specify a preferred 
ptime and a maxptime.  The
sender can use any packetization time they want up to maxptime.  The 
codec may limit it further (see iLBC), and the granularity will vary 
with codec.

-- 
Randell Jesup
randell-ietf@jesup.org


From stephen.botzko@gmail.com  Thu Jul 28 22:13:03 2011
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Subject: Re: [MMUSIC] New rev of the Traffic Class attribute draft
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Generally speaking, I like the new structure, and think the parent classes
and application components are pretty good.

I am thinking there is room for some more refinement on the adjective
substrings - particularly Immersive and Desktop-Video.  For one thing, there
is a space in between Immersive and Desktop-Video (room videoconferencing).
This usually has higher resolution and video quality than PC desktop video,
but less than Immersive.

Perhaps more importantly, I am not clear from the draft what the adjective
substrings are for.  In particular I do not understand if the adjective
strings are intended to nuance the traffic characteristics for the parent
class+application component in some way (and if so, how).

For video, the main distinction would appear to be bandwidth.  If so, we
should consider how to account for future codec improvements.  An HEVC video
codec (expected to be standardized in 2013) will reduce the bandwidth
requirement over today's AVC/H.264 by 50%.  So tomorrow's "Immersive" class
of system might have similar bandwidth requirements to an older traditional
videoconferencing system.

Stephen Botzko

On Wed, Jul 13, 2011 at 5:36 PM, James M. Polk <jmpolk@cisco.com> wrote:

> MMUSIC
>
> We have recently submitted the next version of the traffic class attribute,
> here
>
> http://www.ietf.org/internet-**drafts/draft-polk-mmusic-**
> traffic-class-for-sdp-02.txt<http://www.ietf.org/internet-drafts/draft-polk-mmusic-traffic-class-for-sdp-02.txt>
>
> Most of the work in this version has to do with a newly inserted Section 2
> (i.e., the old section 2 is now section 3).
>
> This new section 2 creates and defines the framework for this attribute
> (something Christer mentioned he would like to see.
>
> Also, several mentioned they wanted to see term or string definitions we
> were proposing that the last version lacked. This is in the new version.
>
> With the addition to this new section, the biggest changes are as follows:
>
> - we changed the attribute string have this format
>
> parent.application(.adjective)**(.adjective)(.admitted/non-**admitted)
>
> (this isn't the exact ABNF, which is in Section 3 now)
>
> - We added that the parent and application components are a pairing.
>
> - we added that there can be zero or more adjective components describing
> the traffic class
>
> - we provide a list of each component for we review, i.e., separate lists
> of parent, application, adjective and cac-class components. Some of the
> application components are settle (i.e., we have one for Netflix type of
> traffic, but don't have an actual string picked yet).
>
> Most of the rest of the changes were minor nits or changes to the new
> attribute structure.
>
> Comments are really appreciated.
>
> James, Paul & Subha
>
> ______________________________**_________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/**listinfo/mmusic<https://www.ietf.org/mailman/listinfo/mmusic>
>

--bcaec548aa59eba49004a92e5571
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Generally speaking, I like the new structure, and think the parent classes =
and application components are pretty good.<br><br>I am thinking there is r=
oom for some more refinement on the adjective substrings - particularly Imm=
ersive and Desktop-Video.=A0 For one thing, there is a space in between Imm=
ersive and Desktop-Video (room videoconferencing).=A0 This usually has high=
er resolution and video quality than PC desktop video, but less than Immers=
ive.<br>
<br>Perhaps more importantly, I am not clear from the draft what the adject=
ive substrings are for.=A0 In particular I do not understand if the adjecti=
ve strings are intended to nuance the traffic characteristics for the paren=
t class+application component in some way (and if so, how).=A0 <br>
<br>For video, the main distinction would appear to be bandwidth.=A0 If so,=
 we should consider how to account for future codec improvements.=A0 An HEV=
C video codec (expected to be standardized in 2013) will reduce the bandwid=
th requirement over today&#39;s AVC/H.264 by 50%.=A0 So tomorrow&#39;s &quo=
t;Immersive&quot; class of system might have similar bandwidth requirements=
 to an older traditional videoconferencing system.<br>
<br>Stephen Botzko<br><br><div class=3D"gmail_quote">On Wed, Jul 13, 2011 a=
t 5:36 PM, James M. Polk <span dir=3D"ltr">&lt;<a href=3D"mailto:jmpolk@cis=
co.com">jmpolk@cisco.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmai=
l_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left=
:1ex;">
MMUSIC<br>
<br>
We have recently submitted the next version of the traffic class attribute,=
 here<br>
<br>
<a href=3D"http://www.ietf.org/internet-drafts/draft-polk-mmusic-traffic-cl=
ass-for-sdp-02.txt" target=3D"_blank">http://www.ietf.org/internet-<u></u>d=
rafts/draft-polk-mmusic-<u></u>traffic-class-for-sdp-02.txt</a><br>
<br>
Most of the work in this version has to do with a newly inserted Section 2 =
(i.e., the old section 2 is now section 3).<br>
<br>
This new section 2 creates and defines the framework for this attribute (so=
mething Christer mentioned he would like to see.<br>
<br>
Also, several mentioned they wanted to see term or string definitions we we=
re proposing that the last version lacked. This is in the new version.<br>
<br>
With the addition to this new section, the biggest changes are as follows:<=
br>
<br>
- we changed the attribute string have this format<br>
<br>
parent.application(.adjective)<u></u>(.adjective)(.admitted/non-<u></u>admi=
tted)<br>
<br>
(this isn&#39;t the exact ABNF, which is in Section 3 now)<br>
<br>
- We added that the parent and application components are a pairing.<br>
<br>
- we added that there can be zero or more adjective components describing t=
he traffic class<br>
<br>
- we provide a list of each component for we review, i.e., separate lists o=
f parent, application, adjective and cac-class components. Some of the appl=
ication components are settle (i.e., we have one for Netflix type of traffi=
c, but don&#39;t have an actual string picked yet).<br>

<br>
Most of the rest of the changes were minor nits or changes to the new attri=
bute structure.<br>
<br>
Comments are really appreciated.<br>
<br>
James, Paul &amp; Subha<br>
<br>
______________________________<u></u>_________________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/mmusic</a><br>
</blockquote></div><br>

--bcaec548aa59eba49004a92e5571--

From albrecht.schwarz@alcatel-lucent.com  Fri Jul 29 02:31:47 2011
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From: "Schwarz, Albrecht (Albrecht)" <albrecht.schwarz@alcatel-lucent.com>
To: Paul Beaumont <paulbeaumont.ietf@gmail.com>, "codec@ietf.org" <codec@ietf.org>
Date: Fri, 29 Jul 2011 11:31:08 +0200
Thread-Topic: [codec] SDP Support for Multiple Packetisation Periods
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Subject: Re: [MMUSIC] [codec] SDP Support for Multiple Packetisation Periods
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Paul,

1) SDP =3D> IETF MMUSIC

2) ptime problem statement ("well known"), see

Multiple Packetization Times in the Session Description Protocol (SDP):
Problem Statement, Requirements & Solution
draft-garcia-mmusic-multiple-ptimes-problem-03.txt
http://tools.ietf.org/html/draft-garcia-mmusic-multiple-ptimes-problem-03

3) Solution:
=3D revised SDP Offer/Answer
see
a) SDP) Capability Negotiation RFC 5939
   http://tools.ietf.org/html/rfc5939
b) SDP Media Mapabilities Negotiation
   http://tools.ietf.org/wg/mmusic/draft-ietf-mmusic-sdp-media-capabilities=
/

Thus, revised SDP Offer/Answer must be used (instead of legacy SDP Offer/An=
swer RFC 3264).

-Albrecht

=20

> -----Original Message-----
> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org]=20
> On Behalf Of Paul Beaumont
> Sent: Donnerstag, 28. Juli 2011 19:31
> To: codec@ietf.org
> Subject: [codec] SDP Support for Multiple Packetisation Periods
>=20
> All
>=20
> Apologies, being first time to IETF and its mailing lists, if=20
> this is not the right mailing list to post this request to.
>=20
> On initial inspection it looks like it is not possible to=20
> specify in SDP the specific packetisation period associated=20
> with a codec that is available through negotiation. Specific=20
> case I am concerned with is to support (1) G.711 @ PP=3D10ms=20
> and (2) G.711 @ PP=3D20ms; prioritised in that order. The ptime=20
> parameter would appear to apply to both.
>=20
> Please can you advise where I should submit this question or=20
> if this is the right frum then whether the above observation=20
> is correct and any supplemental information/comments you may have?=20
>=20
> Thanks
> Paul Beaumont
>=20
>=20
>=20
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec
> =

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Cc: mmusic@ietf.org, "'Subha Dhesikan \(sdhesika\)'" <sdhesika@cisco.com>
Subject: Re: [MMUSIC] New rev of the Traffic Class attribute draft
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Stephen,

 

We do want the adjectives to help serve as a means of differentiating
traffic.  Conversational.audio is quite different from Conversational.video
in many respects.  Likewise, Conversational.video and
Conversational.video.immersive would suggest to me that we need to give
better treatment of media flows to the latter.

 

You suggestion to introduce Conversational.video.room-conferencing (or
similar) is quite reasonable.  It is because we might want to give different
treatment to different kinds of conversational video that we propose this
structure.  Exactly how that treatment might differ is beyond the scope of
this draft, though.

 

What we should do is have an IANA registry for the standard adjectives (as
suggested in the draft).  Whatever document proposes inclusion of a new
adjective should probably describe the type of traffic and why the adjective
is introduced, much as we did for Conversational in section 2.2.

 

I'm not sure how we would relate the bandwidth requirements with the traffic
class labels, though.  I personally view the labels as useful for logging,
QoS, etc.  What is important is understanding the traffic characteristics.
An immersive video system ("Telepresence"), regardless of how much bandwidth
is consumed, needs to have minimal delay in the network as compared to
things like Multimedia-Streaming or even Conversational.video.desktop-video.

 

In any case, we should definitely consider any proposed additional
adjectives at the outset.  We did not attempt to define adjectives for some
classes, but the framework is there: we can introduce them now or in the
future.  Some would most certainly be introduced in the future.

 

Paul

 

From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of
Stephen Botzko
Sent: Friday, July 29, 2011 1:13 AM
To: James M. Polk
Cc: mmusic@ietf.org; Subha Dhesikan (sdhesika); Paul Jones
Subject: Re: [MMUSIC] New rev of the Traffic Class attribute draft

 

Generally speaking, I like the new structure, and think the parent classes
and application components are pretty good.

I am thinking there is room for some more refinement on the adjective
substrings - particularly Immersive and Desktop-Video.  For one thing, there
is a space in between Immersive and Desktop-Video (room videoconferencing).
This usually has higher resolution and video quality than PC desktop video,
but less than Immersive.

Perhaps more importantly, I am not clear from the draft what the adjective
substrings are for.  In particular I do not understand if the adjective
strings are intended to nuance the traffic characteristics for the parent
class+application component in some way (and if so, how).  

For video, the main distinction would appear to be bandwidth.  If so, we
should consider how to account for future codec improvements.  An HEVC video
codec (expected to be standardized in 2013) will reduce the bandwidth
requirement over today's AVC/H.264 by 50%.  So tomorrow's "Immersive" class
of system might have similar bandwidth requirements to an older traditional
videoconferencing system.

Stephen Botzko

On Wed, Jul 13, 2011 at 5:36 PM, James M. Polk <jmpolk@cisco.com> wrote:

MMUSIC

We have recently submitted the next version of the traffic class attribute,
here

http://www.ietf.org/internet-drafts/draft-polk-mmusic-traffic-class-for-sdp-
02.txt

Most of the work in this version has to do with a newly inserted Section 2
(i.e., the old section 2 is now section 3).

This new section 2 creates and defines the framework for this attribute
(something Christer mentioned he would like to see.

Also, several mentioned they wanted to see term or string definitions we
were proposing that the last version lacked. This is in the new version.

With the addition to this new section, the biggest changes are as follows:

- we changed the attribute string have this format

parent.application(.adjective)(.adjective)(.admitted/non-admitted)

(this isn't the exact ABNF, which is in Section 3 now)

- We added that the parent and application components are a pairing.

- we added that there can be zero or more adjective components describing
the traffic class

- we provide a list of each component for we review, i.e., separate lists of
parent, application, adjective and cac-class components. Some of the
application components are settle (i.e., we have one for Netflix type of
traffic, but don't have an actual string picked yet).

Most of the rest of the changes were minor nits or changes to the new
attribute structure.

Comments are really appreciated.

James, Paul & Subha

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

 


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D'>Stephen,<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>We do want the adjectives to help serve as a means of differentiating =
traffic. &nbsp;Conversational.audio is quite different from =
Conversational.video in many respects.&nbsp; Likewise, =
Conversational.video and Conversational.video.immersive would suggest to =
me that we need to give better treatment of media flows to the =
latter.<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>You suggestion to introduce Conversational.video.room-conferencing =
(or similar) is quite reasonable.&nbsp; It is because we might want to =
give different treatment to different kinds of conversational video that =
we propose this structure.&nbsp; Exactly how that treatment might differ =
is beyond the scope of this draft, though.<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>What we should do is have an IANA registry for the standard =
adjectives (as suggested in the draft).&nbsp; Whatever document proposes =
inclusion of a new adjective should probably describe the type of =
traffic and why the adjective is introduced, much as we did for =
Conversational in section 2.2.<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>I&#8217;m not sure how we would relate the bandwidth requirements =
with the traffic class labels, though.&nbsp; I personally view the =
labels as useful for logging, QoS, etc.&nbsp; What is important is =
understanding the traffic characteristics.&nbsp; An immersive video =
system (&#8220;Telepresence&#8221;), regardless of how much bandwidth is =
consumed, needs to have minimal delay in the network as compared to =
things like Multimedia-Streaming or even =
Conversational.video.desktop-video.<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>In any case, we should definitely consider any proposed additional =
adjectives at the outset.&nbsp; We did not attempt to define adjectives =
for some classes, but the framework is there: we can introduce them now =
or in the future.&nbsp; Some would most certainly be introduced in the =
future.<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Paul<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><div =
style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt'><div><div style=3D'border:none;border-top:solid #B5C4DF =
1.0pt;padding:3.0pt 0in 0in 0in'><p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> =
mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] <b>On Behalf Of =
</b>Stephen Botzko<br><b>Sent:</b> Friday, July 29, 2011 1:13 =
AM<br><b>To:</b> James M. Polk<br><b>Cc:</b> mmusic@ietf.org; Subha =
Dhesikan (sdhesika); Paul Jones<br><b>Subject:</b> Re: [MMUSIC] New rev =
of the Traffic Class attribute draft<o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'>Generally speaking, I like the new =
structure, and think the parent classes and application components are =
pretty good.<br><br>I am thinking there is room for some more refinement =
on the adjective substrings - particularly Immersive and =
Desktop-Video.&nbsp; For one thing, there is a space in between =
Immersive and Desktop-Video (room videoconferencing).&nbsp; This usually =
has higher resolution and video quality than PC desktop video, but less =
than Immersive.<br><br>Perhaps more importantly, I am not clear from the =
draft what the adjective substrings are for.&nbsp; In particular I do =
not understand if the adjective strings are intended to nuance the =
traffic characteristics for the parent class+application component in =
some way (and if so, how).&nbsp; <br><br>For video, the main distinction =
would appear to be bandwidth.&nbsp; If so, we should consider how to =
account for future codec improvements.&nbsp; An HEVC video codec =
(expected to be standardized in 2013) will reduce the bandwidth =
requirement over today's AVC/H.264 by 50%.&nbsp; So tomorrow's =
&quot;Immersive&quot; class of system might have similar bandwidth =
requirements to an older traditional videoconferencing =
system.<br><br>Stephen Botzko<o:p></o:p></p><div><p class=3DMsoNormal>On =
Wed, Jul 13, 2011 at 5:36 PM, James M. Polk &lt;<a =
href=3D"mailto:jmpolk@cisco.com">jmpolk@cisco.com</a>&gt; =
wrote:<o:p></o:p></p><p class=3DMsoNormal>MMUSIC<br><br>We have recently =
submitted the next version of the traffic class attribute, =
here<br><br><a =
href=3D"http://www.ietf.org/internet-drafts/draft-polk-mmusic-traffic-cla=
ss-for-sdp-02.txt" =
target=3D"_blank">http://www.ietf.org/internet-drafts/draft-polk-mmusic-t=
raffic-class-for-sdp-02.txt</a><br><br>Most of the work in this version =
has to do with a newly inserted Section 2 (i.e., the old section 2 is =
now section 3).<br><br>This new section 2 creates and defines the =
framework for this attribute (something Christer mentioned he would like =
to see.<br><br>Also, several mentioned they wanted to see term or string =
definitions we were proposing that the last version lacked. This is in =
the new version.<br><br>With the addition to this new section, the =
biggest changes are as follows:<br><br>- we changed the attribute string =
have this =
format<br><br>parent.application(.adjective)(.adjective)(.admitted/non-ad=
mitted)<br><br>(this isn't the exact ABNF, which is in Section 3 =
now)<br><br>- We added that the parent and application components are a =
pairing.<br><br>- we added that there can be zero or more adjective =
components describing the traffic class<br><br>- we provide a list of =
each component for we review, i.e., separate lists of parent, =
application, adjective and cac-class components. Some of the application =
components are settle (i.e., we have one for Netflix type of traffic, =
but don't have an actual string picked yet).<br><br>Most of the rest of =
the changes were minor nits or changes to the new attribute =
structure.<br><br>Comments are really appreciated.<br><br>James, Paul =
&amp; =
Subha<br><br>_______________________________________________<br>mmusic =
mailing list<br><a href=3D"mailto:mmusic@ietf.org" =
target=3D"_blank">mmusic@ietf.org</a><br><a =
href=3D"https://www.ietf.org/mailman/listinfo/mmusic" =
target=3D"_blank">https://www.ietf.org/mailman/listinfo/mmusic</a><o:p></=
o:p></p></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></div></body></html>
------=_NextPart_000_0590_01CC4DD4.6DD346B0--


From stephen.botzko@gmail.com  Fri Jul 29 07:07:17 2011
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References: <201107132136.p6DLamPn002152@mtv-core-3.cisco.com> <CAMC7SJ6QNXn5T-=D-49pgpZ=jM5wCEX39p+53ahJgd=WE2jznQ@mail.gmail.com> <058f01cc4df5$f4e38720$deaa9560$@packetizer.com>
Date: Fri, 29 Jul 2011 10:07:15 -0400
Message-ID: <CAMC7SJ7X0Tf6gT6PZ+drSbNS_hO42Ri=ochuXcBrQGSjT2rA=g@mail.gmail.com>
From: Stephen Botzko <stephen.botzko@gmail.com>
To: "Paul E. Jones" <paulej@packetizer.com>
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Cc: "James M. Polk" <jmpolk@cisco.com>, mmusic@ietf.org, "Subha Dhesikan \(sdhesika\)" <sdhesika@cisco.com>
Subject: Re: [MMUSIC] New rev of the Traffic Class attribute draft
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Hi Paul

I am thinking there are at least two ways to approach the differentiation.

One way is to further refine the traffic characteristics of the base
component.  Presently "conversational" only specifies very low tolerance to
loss, delay and jitter.  So I could envision using the other components to
characterize other attributes about the traffic.  Personally I think that
has merit, but I think there needs to be some care in ensuring that the
characterization that is appropriate is clear.  That way, as new
applications are registered, there would hopefully be useful traffic
information provided.

Another way is to use the adjectives to somehow describe the priority or
"importance" of the flow.  Is that what you mean when you say Immersive
needs "better treatment"?  Personally I would find that problematic.
Certainly my own company's QOS classes are not organized that way.
Generally within the conversational class, the purpose of the meeting is a
more important criteria to drive the priority of the traffic.

All this is to say that the framework itself seems reasonable, but there
needs to be more discussion, particularly on the adjectives.

Stephen Botzko

On Fri, Jul 29, 2011 at 9:46 AM, Paul E. Jones <paulej@packetizer.com>wrote=
:

> Stephen,****
>
> ** **
>
> We do want the adjectives to help serve as a means of differentiating
> traffic.  Conversational.audio is quite different from Conversational.vid=
eo
> in many respects.  Likewise, Conversational.video and
> Conversational.video.immersive would suggest to me that we need to give
> better treatment of media flows to the latter.****
>
> ** **
>
> You suggestion to introduce Conversational.video.room-conferencing (or
> similar) is quite reasonable.  It is because we might want to give differ=
ent
> treatment to different kinds of conversational video that we propose this
> structure.  Exactly how that treatment might differ is beyond the scope o=
f
> this draft, though.****
>
> ** **
>
> What we should do is have an IANA registry for the standard adjectives (a=
s
> suggested in the draft).  Whatever document proposes inclusion of a new
> adjective should probably describe the type of traffic and why the adject=
ive
> is introduced, much as we did for Conversational in section 2.2.****
>
> ** **
>
> I=92m not sure how we would relate the bandwidth requirements with the
> traffic class labels, though.  I personally view the labels as useful for
> logging, QoS, etc.  What is important is understanding the traffic
> characteristics.  An immersive video system (=93Telepresence=94), regardl=
ess of
> how much bandwidth is consumed, needs to have minimal delay in the networ=
k
> as compared to things like Multimedia-Streaming or even
> Conversational.video.desktop-video.****
>
> ** **
>
> In any case, we should definitely consider any proposed additional
> adjectives at the outset.  We did not attempt to define adjectives for so=
me
> classes, but the framework is there: we can introduce them now or in the
> future.  Some would most certainly be introduced in the future.****
>
> ** **
>
> Paul****
>
> ** **
>
> *From:* mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] *On
> Behalf Of *Stephen Botzko
> *Sent:* Friday, July 29, 2011 1:13 AM
> *To:* James M. Polk
> *Cc:* mmusic@ietf.org; Subha Dhesikan (sdhesika); Paul Jones
> *Subject:* Re: [MMUSIC] New rev of the Traffic Class attribute draft****
>
> ** **
>
> Generally speaking, I like the new structure, and think the parent classe=
s
> and application components are pretty good.
>
> I am thinking there is room for some more refinement on the adjective
> substrings - particularly Immersive and Desktop-Video.  For one thing, th=
ere
> is a space in between Immersive and Desktop-Video (room videoconferencing=
).
> This usually has higher resolution and video quality than PC desktop vide=
o,
> but less than Immersive.
>
> Perhaps more importantly, I am not clear from the draft what the adjectiv=
e
> substrings are for.  In particular I do not understand if the adjective
> strings are intended to nuance the traffic characteristics for the parent
> class+application component in some way (and if so, how).
>
> For video, the main distinction would appear to be bandwidth.  If so, we
> should consider how to account for future codec improvements.  An HEVC vi=
deo
> codec (expected to be standardized in 2013) will reduce the bandwidth
> requirement over today's AVC/H.264 by 50%.  So tomorrow's "Immersive" cla=
ss
> of system might have similar bandwidth requirements to an older tradition=
al
> videoconferencing system.
>
> Stephen Botzko****
>
> On Wed, Jul 13, 2011 at 5:36 PM, James M. Polk <jmpolk@cisco.com> wrote:*=
*
> **
>
> MMUSIC
>
> We have recently submitted the next version of the traffic class attribut=
e,
> here
>
>
> http://www.ietf.org/internet-drafts/draft-polk-mmusic-traffic-class-for-s=
dp-02.txt
>
> Most of the work in this version has to do with a newly inserted Section =
2
> (i.e., the old section 2 is now section 3).
>
> This new section 2 creates and defines the framework for this attribute
> (something Christer mentioned he would like to see.
>
> Also, several mentioned they wanted to see term or string definitions we
> were proposing that the last version lacked. This is in the new version.
>
> With the addition to this new section, the biggest changes are as follows=
:
>
> - we changed the attribute string have this format
>
> parent.application(.adjective)(.adjective)(.admitted/non-admitted)
>
> (this isn't the exact ABNF, which is in Section 3 now)
>
> - We added that the parent and application components are a pairing.
>
> - we added that there can be zero or more adjective components describing
> the traffic class
>
> - we provide a list of each component for we review, i.e., separate lists
> of parent, application, adjective and cac-class components. Some of the
> application components are settle (i.e., we have one for Netflix type of
> traffic, but don't have an actual string picked yet).
>
> Most of the rest of the changes were minor nits or changes to the new
> attribute structure.
>
> Comments are really appreciated.
>
> James, Paul & Subha
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic****
>
> ** **
>

--bcaec548aa59750f7b04a935ccd5
Content-Type: text/html; charset=windows-1252
Content-Transfer-Encoding: quoted-printable

Hi Paul<br><br>I am thinking there are at least two ways to approach the di=
fferentiation.<br><br>One way is to further refine the traffic characterist=
ics of the base component.=A0 Presently &quot;conversational&quot; only spe=
cifies very low tolerance to loss, delay and jitter.=A0 So I could envision=
 using the other components to characterize other attributes about the traf=
fic.=A0 Personally I think that has merit, but I think there needs to be so=
me care in ensuring that the characterization that is appropriate is clear.=
=A0 That way, as new applications are registered, there would hopefully be =
useful traffic information provided.<br>
<br>Another way is to use the adjectives to somehow describe the priority o=
r &quot;importance&quot; of the flow.=A0 Is that what you mean when you say=
 Immersive needs &quot;better treatment&quot;?=A0 Personally I would find t=
hat problematic.=A0 Certainly my own company&#39;s QOS classes are not orga=
nized that way.=A0 Generally within the conversational class, the purpose o=
f the meeting is a more important criteria to drive the priority of the tra=
ffic. <br>
<br>All this is to say that the framework itself seems reasonable, but ther=
e needs to be more discussion, particularly on the adjectives.<br><br>Steph=
en Botzko<br><br><div class=3D"gmail_quote">On Fri, Jul 29, 2011 at 9:46 AM=
, Paul E. Jones <span dir=3D"ltr">&lt;<a href=3D"mailto:paulej@packetizer.c=
om">paulej@packetizer.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;"><div link=3D"blue" vlink=3D"purple" lang=3D=
"EN-US"><div><p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#=
1F497D">Stephen,<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#1F497D"><u></=
u>=A0<u></u></span></p><p class=3D"MsoNormal"><span style=3D"font-size:11.0=
pt;color:#1F497D">We do want the adjectives to help serve as a means of dif=
ferentiating traffic. =A0Conversational.audio is quite different from Conve=
rsational.video in many respects.=A0 Likewise, Conversational.video and Con=
versational.video.immersive would suggest to me that we need to give better=
 treatment of media flows to the latter.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#1F497D"><u></=
u>=A0<u></u></span></p><p class=3D"MsoNormal"><span style=3D"font-size:11.0=
pt;color:#1F497D">You suggestion to introduce Conversational.video.room-con=
ferencing (or similar) is quite reasonable.=A0 It is because we might want =
to give different treatment to different kinds of conversational video that=
 we propose this structure.=A0 Exactly how that treatment might differ is b=
eyond the scope of this draft, though.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#1F497D"><u></=
u>=A0<u></u></span></p><p class=3D"MsoNormal"><span style=3D"font-size:11.0=
pt;color:#1F497D">What we should do is have an IANA registry for the standa=
rd adjectives (as suggested in the draft).=A0 Whatever document proposes in=
clusion of a new adjective should probably describe the type of traffic and=
 why the adjective is introduced, much as we did for Conversational in sect=
ion 2.2.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#1F497D"><u></=
u>=A0<u></u></span></p><p class=3D"MsoNormal"><span style=3D"font-size:11.0=
pt;color:#1F497D">I=92m not sure how we would relate the bandwidth requirem=
ents with the traffic class labels, though.=A0 I personally view the labels=
 as useful for logging, QoS, etc.=A0 What is important is understanding the=
 traffic characteristics.=A0 An immersive video system (=93Telepresence=94)=
, regardless of how much bandwidth is consumed, needs to have minimal delay=
 in the network as compared to things like Multimedia-Streaming or even Con=
versational.video.desktop-video.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#1F497D"><u></=
u>=A0<u></u></span></p><p class=3D"MsoNormal"><span style=3D"font-size:11.0=
pt;color:#1F497D">In any case, we should definitely consider any proposed a=
dditional adjectives at the outset.=A0 We did not attempt to define adjecti=
ves for some classes, but the framework is there: we can introduce them now=
 or in the future.=A0 Some would most certainly be introduced in the future=
.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#1F497D"><u></=
u>=A0<u></u></span></p><p class=3D"MsoNormal"><span style=3D"font-size:11.0=
pt;color:#1F497D">Paul<u></u><u></u></span></p><p class=3D"MsoNormal"><span=
 style=3D"font-size:11.0pt;color:#1F497D"><u></u>=A0<u></u></span></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt"><div><div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;paddin=
g:3.0pt 0in 0in 0in"><p class=3D"MsoNormal"><b><span style=3D"font-size:10.=
0pt">From:</span></b><span style=3D"font-size:10.0pt"> <a href=3D"mailto:mm=
usic-bounces@ietf.org" target=3D"_blank">mmusic-bounces@ietf.org</a> [mailt=
o:<a href=3D"mailto:mmusic-bounces@ietf.org" target=3D"_blank">mmusic-bounc=
es@ietf.org</a>] <b>On Behalf Of </b>Stephen Botzko<br>
<b>Sent:</b> Friday, July 29, 2011 1:13 AM<br><b>To:</b> James M. Polk<br><=
b>Cc:</b> <a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.=
org</a>; Subha Dhesikan (sdhesika); Paul Jones<br><b>Subject:</b> Re: [MMUS=
IC] New rev of the Traffic Class attribute draft<u></u><u></u></span></p>
</div></div><div><div></div><div class=3D"h5"><p class=3D"MsoNormal"><u></u=
>=A0<u></u></p><p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">Genera=
lly speaking, I like the new structure, and think the parent classes and ap=
plication components are pretty good.<br>
<br>I am thinking there is room for some more refinement on the adjective s=
ubstrings - particularly Immersive and Desktop-Video.=A0 For one thing, the=
re is a space in between Immersive and Desktop-Video (room videoconferencin=
g).=A0 This usually has higher resolution and video quality than PC desktop=
 video, but less than Immersive.<br>
<br>Perhaps more importantly, I am not clear from the draft what the adject=
ive substrings are for.=A0 In particular I do not understand if the adjecti=
ve strings are intended to nuance the traffic characteristics for the paren=
t class+application component in some way (and if so, how).=A0 <br>
<br>For video, the main distinction would appear to be bandwidth.=A0 If so,=
 we should consider how to account for future codec improvements.=A0 An HEV=
C video codec (expected to be standardized in 2013) will reduce the bandwid=
th requirement over today&#39;s AVC/H.264 by 50%.=A0 So tomorrow&#39;s &quo=
t;Immersive&quot; class of system might have similar bandwidth requirements=
 to an older traditional videoconferencing system.<br>
<br>Stephen Botzko<u></u><u></u></p><div><p class=3D"MsoNormal">On Wed, Jul=
 13, 2011 at 5:36 PM, James M. Polk &lt;<a href=3D"mailto:jmpolk@cisco.com"=
 target=3D"_blank">jmpolk@cisco.com</a>&gt; wrote:<u></u><u></u></p><p clas=
s=3D"MsoNormal">
MMUSIC<br><br>We have recently submitted the next version of the traffic cl=
ass attribute, here<br><br><a href=3D"http://www.ietf.org/internet-drafts/d=
raft-polk-mmusic-traffic-class-for-sdp-02.txt" target=3D"_blank">http://www=
.ietf.org/internet-drafts/draft-polk-mmusic-traffic-class-for-sdp-02.txt</a=
><br>
<br>Most of the work in this version has to do with a newly inserted Sectio=
n 2 (i.e., the old section 2 is now section 3).<br><br>This new section 2 c=
reates and defines the framework for this attribute (something Christer men=
tioned he would like to see.<br>
<br>Also, several mentioned they wanted to see term or string definitions w=
e were proposing that the last version lacked. This is in the new version.<=
br><br>With the addition to this new section, the biggest changes are as fo=
llows:<br>
<br>- we changed the attribute string have this format<br><br>parent.applic=
ation(.adjective)(.adjective)(.admitted/non-admitted)<br><br>(this isn&#39;=
t the exact ABNF, which is in Section 3 now)<br><br>- We added that the par=
ent and application components are a pairing.<br>
<br>- we added that there can be zero or more adjective components describi=
ng the traffic class<br><br>- we provide a list of each component for we re=
view, i.e., separate lists of parent, application, adjective and cac-class =
components. Some of the application components are settle (i.e., we have on=
e for Netflix type of traffic, but don&#39;t have an actual string picked y=
et).<br>
<br>Most of the rest of the changes were minor nits or changes to the new a=
ttribute structure.<br><br>Comments are really appreciated.<br><br>James, P=
aul &amp; Subha<br><br>_______________________________________________<br>
mmusic mailing list<br><a href=3D"mailto:mmusic@ietf.org" target=3D"_blank"=
>mmusic@ietf.org</a><br><a href=3D"https://www.ietf.org/mailman/listinfo/mm=
usic" target=3D"_blank">https://www.ietf.org/mailman/listinfo/mmusic</a><u>=
</u><u></u></p>
</div><p class=3D"MsoNormal"><u></u>=A0<u></u></p></div></div></div></div><=
/div></blockquote></div><br>

--bcaec548aa59750f7b04a935ccd5--

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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Stephen Botzko'" <stephen.botzko@gmail.com>
References: <201107132136.p6DLamPn002152@mtv-core-3.cisco.com>	<CAMC7SJ6QNXn5T-=D-49pgpZ=jM5wCEX39p+53ahJgd=WE2jznQ@mail.gmail.com>	<058f01cc4df5$f4e38720$deaa9560$@packetizer.com> <CAMC7SJ7X0Tf6gT6PZ+drSbNS_hO42Ri=ochuXcBrQGSjT2rA=g@mail.gmail.com>
In-Reply-To: <CAMC7SJ7X0Tf6gT6PZ+drSbNS_hO42Ri=ochuXcBrQGSjT2rA=g@mail.gmail.com>
Date: Sat, 30 Jul 2011 01:44:12 -0400
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Cc: "'James M. Polk'" <jmpolk@cisco.com>, mmusic@ietf.org, "'Subha Dhesikan \(sdhesika\)'" <sdhesika@cisco.com>
Subject: Re: [MMUSIC] New rev of the Traffic Class attribute draft
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Stephen,

 

What we tried to do was limit the number of base classes to some finite
number that was largely, if not completely, representative of the classes of
traffic for which we might want to provide QoS.  We spent a fair amount of
time trying to figure out how to make just the base class sufficient, but
decided we needed a better approach that included adjectives.

 

The adjectives are just adjectives.  How one relates that to QoS policies is
another matter.  I am suggesting that we might want to consider
Conversational.immersive more important than Conversational.desktop-video.
However, I can accept an argument that Conversational.desktop-video._ceo or
Conversational.desktop-video._executive_summit might be given better
treatment than the routine Telepresence call.

 

The intent was that we define a structure such that the adjectives can help
dictate QoS policies, provide useful information to the endpoints as to the
nature of the media flow at the remote end, etc.  The old class names that
existed were insufficient for describing the various types of traffic
flowing on the network.

 

Paul

 

From: Stephen Botzko [mailto:stephen.botzko@gmail.com] 
Sent: Friday, July 29, 2011 10:07 AM
To: Paul E. Jones
Cc: James M. Polk; mmusic@ietf.org; Subha Dhesikan (sdhesika)
Subject: Re: [MMUSIC] New rev of the Traffic Class attribute draft

 

Hi Paul

I am thinking there are at least two ways to approach the differentiation.

One way is to further refine the traffic characteristics of the base
component.  Presently "conversational" only specifies very low tolerance to
loss, delay and jitter.  So I could envision using the other components to
characterize other attributes about the traffic.  Personally I think that
has merit, but I think there needs to be some care in ensuring that the
characterization that is appropriate is clear.  That way, as new
applications are registered, there would hopefully be useful traffic
information provided.

Another way is to use the adjectives to somehow describe the priority or
"importance" of the flow.  Is that what you mean when you say Immersive
needs "better treatment"?  Personally I would find that problematic.
Certainly my own company's QOS classes are not organized that way.
Generally within the conversational class, the purpose of the meeting is a
more important criteria to drive the priority of the traffic. 

All this is to say that the framework itself seems reasonable, but there
needs to be more discussion, particularly on the adjectives.

Stephen Botzko

On Fri, Jul 29, 2011 at 9:46 AM, Paul E. Jones <paulej@packetizer.com>
wrote:

Stephen,

 

We do want the adjectives to help serve as a means of differentiating
traffic.  Conversational.audio is quite different from Conversational.video
in many respects.  Likewise, Conversational.video and
Conversational.video.immersive would suggest to me that we need to give
better treatment of media flows to the latter.

 

You suggestion to introduce Conversational.video.room-conferencing (or
similar) is quite reasonable.  It is because we might want to give different
treatment to different kinds of conversational video that we propose this
structure.  Exactly how that treatment might differ is beyond the scope of
this draft, though.

 

What we should do is have an IANA registry for the standard adjectives (as
suggested in the draft).  Whatever document proposes inclusion of a new
adjective should probably describe the type of traffic and why the adjective
is introduced, much as we did for Conversational in section 2.2.

 

I'm not sure how we would relate the bandwidth requirements with the traffic
class labels, though.  I personally view the labels as useful for logging,
QoS, etc.  What is important is understanding the traffic characteristics.
An immersive video system ("Telepresence"), regardless of how much bandwidth
is consumed, needs to have minimal delay in the network as compared to
things like Multimedia-Streaming or even Conversational.video.desktop-video.

 

In any case, we should definitely consider any proposed additional
adjectives at the outset.  We did not attempt to define adjectives for some
classes, but the framework is there: we can introduce them now or in the
future.  Some would most certainly be introduced in the future.

 

Paul

 

From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of
Stephen Botzko
Sent: Friday, July 29, 2011 1:13 AM
To: James M. Polk
Cc: mmusic@ietf.org; Subha Dhesikan (sdhesika); Paul Jones
Subject: Re: [MMUSIC] New rev of the Traffic Class attribute draft

 

Generally speaking, I like the new structure, and think the parent classes
and application components are pretty good.

I am thinking there is room for some more refinement on the adjective
substrings - particularly Immersive and Desktop-Video.  For one thing, there
is a space in between Immersive and Desktop-Video (room videoconferencing).
This usually has higher resolution and video quality than PC desktop video,
but less than Immersive.

Perhaps more importantly, I am not clear from the draft what the adjective
substrings are for.  In particular I do not understand if the adjective
strings are intended to nuance the traffic characteristics for the parent
class+application component in some way (and if so, how).  

For video, the main distinction would appear to be bandwidth.  If so, we
should consider how to account for future codec improvements.  An HEVC video
codec (expected to be standardized in 2013) will reduce the bandwidth
requirement over today's AVC/H.264 by 50%.  So tomorrow's "Immersive" class
of system might have similar bandwidth requirements to an older traditional
videoconferencing system.

Stephen Botzko

On Wed, Jul 13, 2011 at 5:36 PM, James M. Polk <jmpolk@cisco.com> wrote:

MMUSIC

We have recently submitted the next version of the traffic class attribute,
here

http://www.ietf.org/internet-drafts/draft-polk-mmusic-traffic-class-for-sdp-
02.txt

Most of the work in this version has to do with a newly inserted Section 2
(i.e., the old section 2 is now section 3).

This new section 2 creates and defines the framework for this attribute
(something Christer mentioned he would like to see.

Also, several mentioned they wanted to see term or string definitions we
were proposing that the last version lacked. This is in the new version.

With the addition to this new section, the biggest changes are as follows:

- we changed the attribute string have this format

parent.application(.adjective)(.adjective)(.admitted/non-admitted)

(this isn't the exact ABNF, which is in Section 3 now)

- We added that the parent and application components are a pairing.

- we added that there can be zero or more adjective components describing
the traffic class

- we provide a list of each component for we review, i.e., separate lists of
parent, application, adjective and cac-class components. Some of the
application components are settle (i.e., we have one for Netflix type of
traffic, but don't have an actual string picked yet).

Most of the rest of the changes were minor nits or changes to the new
attribute structure.

Comments are really appreciated.

James, Paul & Subha

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

 

 


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<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]--></head><body lang=3DEN-US link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Stephen,<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>What we tried to do was limit the number of base classes to some =
finite number that was largely, if not completely, representative of the =
classes of traffic for which we might want to provide QoS. &nbsp;We =
spent a fair amount of time trying to figure out how to make just the =
base class sufficient, but decided we needed a better approach that =
included adjectives.<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>The adjectives are just adjectives.&nbsp; How one relates that to QoS =
policies is another matter.&nbsp; I am suggesting that we <i>might</i> =
want to consider Conversational.immersive more important than =
Conversational.desktop-video.&nbsp; However, I can accept an argument =
that Conversational.desktop-video._ceo or =
Conversational.desktop-video._executive_summit might be given better =
treatment than the routine Telepresence call.<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>The intent was that we define a structure such that the adjectives =
can help dictate QoS policies, provide useful information to the =
endpoints as to the nature of the media flow at the remote end, =
etc.&nbsp; The old class names that existed were insufficient for =
describing the various types of traffic flowing on the =
network.<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Paul<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><div =
style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt'><div><div style=3D'border:none;border-top:solid #B5C4DF =
1.0pt;padding:3.0pt 0in 0in 0in'><p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> =
Stephen Botzko [mailto:stephen.botzko@gmail.com] <br><b>Sent:</b> =
Friday, July 29, 2011 10:07 AM<br><b>To:</b> Paul E. Jones<br><b>Cc:</b> =
James M. Polk; mmusic@ietf.org; Subha Dhesikan =
(sdhesika)<br><b>Subject:</b> Re: [MMUSIC] New rev of the Traffic Class =
attribute draft<o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'>Hi Paul<br><br>I am thinking there are at =
least two ways to approach the differentiation.<br><br>One way is to =
further refine the traffic characteristics of the base component.&nbsp; =
Presently &quot;conversational&quot; only specifies very low tolerance =
to loss, delay and jitter.&nbsp; So I could envision using the other =
components to characterize other attributes about the traffic.&nbsp; =
Personally I think that has merit, but I think there needs to be some =
care in ensuring that the characterization that is appropriate is =
clear.&nbsp; That way, as new applications are registered, there would =
hopefully be useful traffic information provided.<br><br>Another way is =
to use the adjectives to somehow describe the priority or =
&quot;importance&quot; of the flow.&nbsp; Is that what you mean when you =
say Immersive needs &quot;better treatment&quot;?&nbsp; Personally I =
would find that problematic.&nbsp; Certainly my own company's QOS =
classes are not organized that way.&nbsp; Generally within the =
conversational class, the purpose of the meeting is a more important =
criteria to drive the priority of the traffic. <br><br>All this is to =
say that the framework itself seems reasonable, but there needs to be =
more discussion, particularly on the adjectives.<br><br>Stephen =
Botzko<o:p></o:p></p><div><p class=3DMsoNormal>On Fri, Jul 29, 2011 at =
9:46 AM, Paul E. Jones &lt;<a =
href=3D"mailto:paulej@packetizer.com">paulej@packetizer.com</a>&gt; =
wrote:<o:p></o:p></p><div><div><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span =
style=3D'font-size:11.0pt;color:#1F497D'>Stephen,</span><o:p></o:p></p><p=
 class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span =
style=3D'font-size:11.0pt;color:#1F497D'>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span =
style=3D'font-size:11.0pt;color:#1F497D'>We do want the adjectives to =
help serve as a means of differentiating traffic. =
&nbsp;Conversational.audio is quite different from Conversational.video =
in many respects.&nbsp; Likewise, Conversational.video and =
Conversational.video.immersive would suggest to me that we need to give =
better treatment of media flows to the latter.</span><o:p></o:p></p><p =
class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span =
style=3D'font-size:11.0pt;color:#1F497D'>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span =
style=3D'font-size:11.0pt;color:#1F497D'>You suggestion to introduce =
Conversational.video.room-conferencing (or similar) is quite =
reasonable.&nbsp; It is because we might want to give different =
treatment to different kinds of conversational video that we propose =
this structure.&nbsp; Exactly how that treatment might differ is beyond =
the scope of this draft, though.</span><o:p></o:p></p><p =
class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span =
style=3D'font-size:11.0pt;color:#1F497D'>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span =
style=3D'font-size:11.0pt;color:#1F497D'>What we should do is have an =
IANA registry for the standard adjectives (as suggested in the =
draft).&nbsp; Whatever document proposes inclusion of a new adjective =
should probably describe the type of traffic and why the adjective is =
introduced, much as we did for Conversational in section =
2.2.</span><o:p></o:p></p><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span =
style=3D'font-size:11.0pt;color:#1F497D'>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span =
style=3D'font-size:11.0pt;color:#1F497D'>I&#8217;m not sure how we would =
relate the bandwidth requirements with the traffic class labels, =
though.&nbsp; I personally view the labels as useful for logging, QoS, =
etc.&nbsp; What is important is understanding the traffic =
characteristics.&nbsp; An immersive video system =
(&#8220;Telepresence&#8221;), regardless of how much bandwidth is =
consumed, needs to have minimal delay in the network as compared to =
things like Multimedia-Streaming or even =
Conversational.video.desktop-video.</span><o:p></o:p></p><p =
class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span =
style=3D'font-size:11.0pt;color:#1F497D'>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span =
style=3D'font-size:11.0pt;color:#1F497D'>In any case, we should =
definitely consider any proposed additional adjectives at the =
outset.&nbsp; We did not attempt to define adjectives for some classes, =
but the framework is there: we can introduce them now or in the =
future.&nbsp; Some would most certainly be introduced in the =
future.</span><o:p></o:p></p><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span =
style=3D'font-size:11.0pt;color:#1F497D'>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span =
style=3D'font-size:11.0pt;color:#1F497D'>Paul</span><o:p></o:p></p><p =
class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span =
style=3D'font-size:11.0pt;color:#1F497D'>&nbsp;</span><o:p></o:p></p><div=
 style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt'><div><div style=3D'border:none;border-top:solid #B5C4DF =
1.0pt;padding:3.0pt 0in 0in 0in'><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b><span =
style=3D'font-size:10.0pt'>From:</span></b><span =
style=3D'font-size:10.0pt'> <a href=3D"mailto:mmusic-bounces@ietf.org" =
target=3D"_blank">mmusic-bounces@ietf.org</a> [mailto:<a =
href=3D"mailto:mmusic-bounces@ietf.org" =
target=3D"_blank">mmusic-bounces@ietf.org</a>] <b>On Behalf Of =
</b>Stephen Botzko<br><b>Sent:</b> Friday, July 29, 2011 1:13 =
AM<br><b>To:</b> James M. Polk<br><b>Cc:</b> <a =
href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.org</a>; =
Subha Dhesikan (sdhesika); Paul Jones<br><b>Subject:</b> Re: [MMUSIC] =
New rev of the Traffic Class attribute =
draft</span><o:p></o:p></p></div></div><div><div><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;margin-bottom:12.0pt'>Generally =
speaking, I like the new structure, and think the parent classes and =
application components are pretty good.<br><br>I am thinking there is =
room for some more refinement on the adjective substrings - particularly =
Immersive and Desktop-Video.&nbsp; For one thing, there is a space in =
between Immersive and Desktop-Video (room videoconferencing).&nbsp; This =
usually has higher resolution and video quality than PC desktop video, =
but less than Immersive.<br><br>Perhaps more importantly, I am not clear =
from the draft what the adjective substrings are for.&nbsp; In =
particular I do not understand if the adjective strings are intended to =
nuance the traffic characteristics for the parent class+application =
component in some way (and if so, how).&nbsp; <br><br>For video, the =
main distinction would appear to be bandwidth.&nbsp; If so, we should =
consider how to account for future codec improvements.&nbsp; An HEVC =
video codec (expected to be standardized in 2013) will reduce the =
bandwidth requirement over today's AVC/H.264 by 50%.&nbsp; So tomorrow's =
&quot;Immersive&quot; class of system might have similar bandwidth =
requirements to an older traditional videoconferencing =
system.<br><br>Stephen Botzko<o:p></o:p></p><div><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>On Wed, Jul =
13, 2011 at 5:36 PM, James M. Polk &lt;<a =
href=3D"mailto:jmpolk@cisco.com" =
target=3D"_blank">jmpolk@cisco.com</a>&gt; wrote:<o:p></o:p></p><p =
class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>MMUSIC<br><b=
r>We have recently submitted the next version of the traffic class =
attribute, here<br><br><a =
href=3D"http://www.ietf.org/internet-drafts/draft-polk-mmusic-traffic-cla=
ss-for-sdp-02.txt" =
target=3D"_blank">http://www.ietf.org/internet-drafts/draft-polk-mmusic-t=
raffic-class-for-sdp-02.txt</a><br><br>Most of the work in this version =
has to do with a newly inserted Section 2 (i.e., the old section 2 is =
now section 3).<br><br>This new section 2 creates and defines the =
framework for this attribute (something Christer mentioned he would like =
to see.<br><br>Also, several mentioned they wanted to see term or string =
definitions we were proposing that the last version lacked. This is in =
the new version.<br><br>With the addition to this new section, the =
biggest changes are as follows:<br><br>- we changed the attribute string =
have this =
format<br><br>parent.application(.adjective)(.adjective)(.admitted/non-ad=
mitted)<br><br>(this isn't the exact ABNF, which is in Section 3 =
now)<br><br>- We added that the parent and application components are a =
pairing.<br><br>- we added that there can be zero or more adjective =
components describing the traffic class<br><br>- we provide a list of =
each component for we review, i.e., separate lists of parent, =
application, adjective and cac-class components. Some of the application =
components are settle (i.e., we have one for Netflix type of traffic, =
but don't have an actual string picked yet).<br><br>Most of the rest of =
the changes were minor nits or changes to the new attribute =
structure.<br><br>Comments are really appreciated.<br><br>James, Paul =
&amp; =
Subha<br><br>_______________________________________________<br>mmusic =
mailing list<br><a href=3D"mailto:mmusic@ietf.org" =
target=3D"_blank">mmusic@ietf.org</a><br><a =
href=3D"https://www.ietf.org/mailman/listinfo/mmusic" =
target=3D"_blank">https://www.ietf.org/mailman/listinfo/mmusic</a><o:p></=
o:p></p></div><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p></div></div></div></div></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></div></body></html>
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From miguel.a.garcia@ericsson.com  Sun Jul 31 01:33:36 2011
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Date: Sun, 31 Jul 2011 10:33:32 +0200
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Cc: "codec@ietf.org" <codec@ietf.org>, mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] [codec] SDP Support for Multiple Packetisation Periods
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Yeap, I was also going to point to the (already expired) I-D on multiple 
packetization problem draft,

http://tools.ietf.org/id/draft-garcia-mmusic-multiple-ptimes-problem-03.txt

The authors didn't see much interest in solving this problem, so the 
draft was abandoned. If the community realizes that the problem still 
persists, we can easily resurrect it.

/Miguel

On 29/07/2011 11:31, Schwarz, Albrecht (Albrecht) wrote:
> Paul,
>
> 1) SDP =>  IETF MMUSIC
>
> 2) ptime problem statement ("well known"), see
>
> Multiple Packetization Times in the Session Description Protocol (SDP):
> Problem Statement, Requirements&  Solution
> draft-garcia-mmusic-multiple-ptimes-problem-03.txt
> http://tools.ietf.org/html/draft-garcia-mmusic-multiple-ptimes-problem-03
>
> 3) Solution:
> = revised SDP Offer/Answer
> see
> a) SDP) Capability Negotiation RFC 5939
>     http://tools.ietf.org/html/rfc5939
> b) SDP Media Mapabilities Negotiation
>     http://tools.ietf.org/wg/mmusic/draft-ietf-mmusic-sdp-media-capabilities/
>
> Thus, revised SDP Offer/Answer must be used (instead of legacy SDP Offer/Answer RFC 3264).
>
> -Albrecht
>
>
>
>> -----Original Message-----
>> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org]
>> On Behalf Of Paul Beaumont
>> Sent: Donnerstag, 28. Juli 2011 19:31
>> To: codec@ietf.org
>> Subject: [codec] SDP Support for Multiple Packetisation Periods
>>
>> All
>>
>> Apologies, being first time to IETF and its mailing lists, if
>> this is not the right mailing list to post this request to.
>>
>> On initial inspection it looks like it is not possible to
>> specify in SDP the specific packetisation period associated
>> with a codec that is available through negotiation. Specific
>> case I am concerned with is to support (1) G.711 @ PP=10ms
>> and (2) G.711 @ PP=20ms; prioritised in that order. The ptime
>> parameter would appear to apply to both.
>>
>> Please can you advise where I should submit this question or
>> if this is the right frum then whether the above observation
>> is correct and any supplemental information/comments you may have?
>>
>> Thanks
>> Paul Beaumont
>>
>>
>>
>> _______________________________________________
>> codec mailing list
>> codec@ietf.org
>> https://www.ietf.org/mailman/listinfo/codec
>>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>

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