
From miguel.a.garcia@ericsson.com  Mon Apr  2 06:18:16 2012
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Date: Mon, 2 Apr 2012 15:18:14 +0200
From: "Miguel A. Garcia" <Miguel.A.Garcia@ericsson.com>
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Cc: Flemming Andreasen <fandreas@cisco.com>
Subject: [MMUSIC] WGLC of draft-ietf-mmusic-media-loopback-18
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This is to start a 2-week Working Group Last Call for

        draft-ietf-mmusic-media-loopback-18.txt

The WGLC ends on April 16th, 2012.

Please reply to this e-mail to send comments, so that the authors and the 
mailing list are all copied.

/Miguel

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

From miguel.a.garcia@ericsson.com  Tue Apr  3 05:35:17 2012
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Cc: Flemming Andreasen <fandreas@cisco.com>
Subject: [MMUSIC] Feedback on draft-boucadair-mmusic-altc-04 as Independent Submission
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This mail is related to draft-boucadair-mmusic-altc-04.txt, available at:

http://datatracker.ietf.org/doc/draft-boucadair-mmusic-altc/

This draft has been submitted to the RFC Editor for publication in the 
independent submission stream. Because of this, it can be published as 
either Informational or Experimental RFC.

The RFC Editor, in these cases, gets feedback from the community about 
the possible publication, including consultation to the IESG (see RFC 5742).

We would like to get your input, basically, indicating whether you are OK 
with the publication of this Informational/Experimental RFC or whether 
you have concerns. We will try to have a consolidated set of comments to 
the RFC Editor.

The draft proposes a mechanism to include several IP addresses related to 
the same media stream in an SDP offer, including alternative IPv4 and 
IPv6 addresses.

The draft presents a problem: it defines a SIP option tag. SIP option 
tags can only be defined in standards track RFCs (See RFC 3261 Section 
19.2), but this document can only be Informational or Experimental.

Please let the mailing list know your thoughts.

BR,

           Flemming and Miguel (chairs)
-- 
Miguel A. Garcia
+34-91-339-3608
Ericsson Spain

From brett@broadsoft.com  Tue Apr  3 06:30:25 2012
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From: Brett Tate <brett@broadsoft.com>
To: mmusic <mmusic@ietf.org>
Date: Tue, 3 Apr 2012 06:30:22 -0700
Thread-Topic: [MMUSIC] Feedback on draft-boucadair-mmusic-altc-04 as Independent Submission
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The functionality provided by draft-boucadair-mmusic-altc is useful.  I pre=
fer that this draft progress to be an RFC.  It should progress as standard =
track; however publishing as Informational is adequate (especially if can r=
eserve the "altc" option-tag) if no IETF working group is willing to work o=
n the draft.

If the option-tag issue causes concern, it currently appears to provide lit=
tle value other than a hint.  Thus, it could be removed or uselessly commun=
icated by other means (such as defining a "ns-supported" header to mean the=
 same thing as "supported" when non standard track).

Thanks,
Brett


From ietf@meetecho.com  Tue Apr  3 07:34:08 2012
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Dear all,=0A=0Athe full recording (synchronized video, audio, slides and =
jabber room)=0Aof MMUSIC session at IETF-83 is available.=0A=0AYou can wa=
tch it by either clicking the proper link on the remote =0Aparticipation =
page =0A(http://www.ietf.org/meeting/83/remote-participation.html#Meetech=
o), or =0Aby directly accessing the following URL:=0Ahttp://ietf83.conf.m=
eetecho.com/index.php/Recorded_Sessions#MMUSIC_IETF83=0A=0AFor the chair(=
s): please feel free to put the link to the recording in =0Athe minutes, =
if you think this might be useful.=0A=0AIn case of problems with the play=
out, just drop an e-mail to=0Ateam@meetecho.com.=0A=0ACheers,=0Athe Meete=
cho team=C2=A0=0A=0A=C2=A0=0AMeetecho s.r.l.=0AWeb Conferencing and Colla=
boration Tools=0Awww.meetecho.com=0A=C2=A0

--_=__=_XaM3_.1333463634.2A.468187.42.24781.52.42.007.1312405235
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=0A<div class=3D"xam_msg_class">=0A<div style=3D"font: normal 13px Arial;=
 color:rgb(0, 0, 0);"><div>Dear all,=0A<br>=0A<br>the full recording (syn=
chronized video, audio, slides and jabber room)=0A<br>of MMUSIC session a=
t IETF-83 is available.=0A<br>=0A<br>You can watch it by either clicking =
the proper link on the remote =0Aparticipation page =0A(<a class=3D"moz-t=
xt-link-freetext" href=3D"http://www.ietf.org/meeting/83/remote-participa=
tion.html#Meetecho">http://www.ietf.org/meeting/83/remote-participation.h=
tml#Meetecho</a>), or =0Aby directly accessing the following URL:=0A<br><=
a class=3D"moz-txt-link-freetext" href=3D"http://ietf83.conf.meetecho.com=
/index.php/Recorded_Sessions#XRBLOCK_IETF83">http://ietf83.conf.meetecho.=
com/index.php/Recorded_Sessions#MMUSIC_IETF83</a>=0A<br>=0A<br>For the ch=
air(s): please feel free to put the link to the recording in =0Athe minut=
es, if you think this might be useful.=0A<br>=0A<br>In case of problems w=
ith the playout, just drop an e-mail to=0A<br><a target=3D"_self" class=3D=
"moz-txt-link-abbreviated">team@meetecho.com</a>.=0A<br>=0A<br>Cheers,=0A=
<br>the Meetecho team&nbsp;=0A</div><div>&nbsp;</div><div>Meetecho s.r.l.=
</div><div>Web Conferencing and Collaboration Tools</div><div>www.meetech=
o.com</div><div>&nbsp;</div></div>=0A</div>=0A

--_=__=_XaM3_.1333463634.2A.468187.42.24781.52.42.007.1312405235--


From vsingh.ietf@gmail.com  Tue Apr  3 08:35:22 2012
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References: <454A16E4DA86094EB66BB2C1DBBBC0180765E609@XMB-BGL-41B.cisco.com> <CAEbPqrw-TsLXut5iufxEJ7gy0KoD-8fuHuVQ=6Cic5UMsqaGzg@mail.gmail.com> <454A16E4DA86094EB66BB2C1DBBBC0180765E90B@XMB-BGL-41B.cisco.com>
From: Varun Singh <vsingh.ietf@gmail.com>
Date: Tue, 3 Apr 2012 18:35:00 +0300
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To: "Tirumaleswar Reddy (tireddy)" <tireddy@cisco.com>
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Cc: =?ISO-8859-1?Q?J=F6rg_Ott?= <jo@comnet.tkk.fi>, mmusic@ietf.org, "Dan Wing \(dwing\)" <dwing@cisco.com>
Subject: Re: [MMUSIC] Comments on draft-singh-avtcore-mprtp-04
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Hi Tiru,

Clarifications inline.

On Fri, Mar 23, 2012 at 13:35, Tirumaleswar Reddy (tireddy)
<tireddy@cisco.com> wrote:
>
>>
>> Hi Tiru,
>>
>> Response inline.
>>
>> On Wed, Mar 21, 2012 at 20:07, Tirumaleswar Reddy (tireddy)
>> <tireddy@cisco.com> wrote:
>> > Hi Varun -
>> >
>> > I have few comments related to the draft
>> >
>> > a)How is the situation handled when new network paths become
>> > available/disabled after the call is setup ?
>> >
>> > =A0=A0=A0 (3G enabled/disabled after the call is setup)
>> >
>>
>> This is a local decision and MPRTP will use all available paths.
>> Either the application or ICE or something else needs to tell MPRTP
>> when an interface is available or unavailable.
>>
>> At the media level (in the worst case) the sending endpoint will
>> discover a path is unavailable when a subflow reports 100% loss and it
>> will stop using it. Alternatively, if the endpoint discovers that one
>> of its interface is disabled it can advertise a new set of interfaces
>> (excluding the disabled interface from the set) to the other endpoint.
>
> Please refer RFC 6419 on the problems and behavior of various OS related =
to MIF. For example Section 3.1.2 Connection Manager only selects the best =
possible connection for the app based on the destination. The default conne=
ction manager always prefers wired over wireless.
>

You are right that the connection manager chooses the interface but
this is probably true for any protocol that tries to do multipath. The
application probably needs to either override this behavior (e.g.,
sudo/ raw sockets/ kernel configuration) or the scheduling algorithm
needs to make sure that it does not cause congestion by assuming
multipath and sending too much data.

>>
>> > b)There is little information related to IPv6. How SIP Client would
>> pick
>> > among list of ULA, link-local addresses, IPv6 global addresses with
>> > different precedence using ICE (refer to
>> > draft-keranen-mmusic-ice-adress-selection) ?
>> >
>>
>> It is also a local decision. This should not be different from the
>> choices made by an endpoint using a single path. if an endpoint is
>> using ICE then it just inherits the ICE priorities.
>
> For example consider a IPv6 Mobile Node making SIP call in PMIPv6. This n=
ode could be assigned both MAG and LMA prefixes. Let's say the destination =
is reachable through both the prefixes. But the issue is only LMA prefix wo=
uld offer mobility if Mobile Node moves out of the hotspot MAG prefix will =
not be available. In a different region he may get a different MAG prefix. =
But if we consider a branch with multiple ISP - such complexities may not a=
rise. The problem in this case would be the SIP client being provided the l=
ist of host candidate addresses with the preference provided by RFC 3484 (a=
nd using ICE). You will need support from OS to achieve this.
>

Yes, support from the OS or some rules (described in the
aforementioned RFC/drafts) are needed to restrict the list of
interface candidates. We will make this clear in the upcoming drafts.

Cheers,
Varun


--=20
http://www.netlab.tkk.fi/~varun/

From miguel.a.garcia@ericsson.com  Wed Apr  4 03:20:13 2012
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Hi Brett.

Notice that this draft has been sent directly to the RFC Editor for 
publication in the Independent Stream. This means that it can only be 
published as Informational or Experimental.

Additionally, MMUSIC has been requested for comments, but bear in mind, 
that this document is not the produce of the IETF.

/Miguel

On 03/04/2012 15:30, Brett Tate wrote:
> The functionality provided by draft-boucadair-mmusic-altc is useful.  I prefer that this draft progress to be an RFC.  It should progress as standard track; however publishing as Informational is adequate (especially if can reserve the "altc" option-tag) if no IETF working group is willing to work on the draft.
>
> If the option-tag issue causes concern, it currently appears to provide little value other than a hint.  Thus, it could be removed or uselessly communicated by other means (such as defining a "ns-supported" header to mean the same thing as "supported" when non standard track).
>
> Thanks,
> Brett
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>

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

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To: "Miguel A. Garcia" <Miguel.A.Garcia@ericsson.com>, mmusic <mmusic@ietf.org>
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> Notice that this draft has been sent directly to the RFC Editor for
> publication in the Independent Stream. This means that it can only be
> published as Informational or Experimental.
>=20
> Additionally, MMUSIC has been requested for comments, but bear in mind,
> that this document is not the produce of the IETF.

I agree.  My mentioning of "standard track" and "no IETF working group is w=
illing to work on the draft" was mostly just an attempt to reconcile why an=
 Informational draft defines a new option-tag even though IANA (per RFC 326=
1 and RFC 5727) may disallow registering/reserving the new option-tag.  As =
highlighted within IETF 77 minutes (and thread http://www.ietf.org/mail-arc=
hive/web/mmusic/current/msg09016.html), the attempts to get MMUSIC to adopt=
 altc or other ICE alternatives (for IPv4/IPv6 transition) have failed.

For what it's worth, I'd actually prefer that the new option-tag be removed=
 from the draft since it is mostly only a hint and complicates things for s=
ome B2BUA/3PCC situations.  However as indicated, the draft is an independe=
nt submission.  Thus, my preference is likely of little value unless the ne=
w option-tag prevents the draft from becoming an RFC (and becoming an RFC c=
ontinues to be preferred by the authors).


> On 03/04/2012 15:30, Brett Tate wrote:
> > The functionality provided by draft-boucadair-mmusic-altc is useful.
> I prefer that this draft progress to be an RFC.  It should progress as
> standard track; however publishing as Informational is adequate
> (especially if can reserve the "altc" option-tag) if no IETF working
> group is willing to work on the draft.
> >
> > If the option-tag issue causes concern, it currently appears to
> provide little value other than a hint.  Thus, it could be removed or
> uselessly communicated by other means (such as defining a "ns-
> supported" header to mean the same thing as "supported" when non
> standard track).



From miguel.a.garcia@ericsson.com  Tue Apr 10 00:18:31 2012
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Cc: "draft-ietf-mmusic-media-loopback.authors@tools.ietf.org" <draft-ietf-mmusic-media-loopback.authors@tools.ietf.org>, Flemming Andreasen <fandreas@cisco.com>
Subject: Re: [MMUSIC] WGLC of draft-ietf-mmusic-media-loopback-18
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A short note:

A reviewer of Media MIME types team sent some minor/editorial comments to 
Section 13 of the draft.

The review is available here:

http://www.ietf.org/mail-archive/web/ietf-types/current/msg01649.html

/Miguel

On 02/04/2012 15:18, Miguel A. Garcia wrote:
> This is to start a 2-week Working Group Last Call for
>
>          draft-ietf-mmusic-media-loopback-18.txt
>
> The WGLC ends on April 16th, 2012.
>
> Please reply to this e-mail to send comments, so that the authors and the
> mailing list are all copied.
>
> /Miguel
>

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

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Cc: draft-ietf-mmusic-media-loopback.authors@tools.ietf.org, mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] WGLC of draft-ietf-mmusic-media-loopback-18
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I have reviewed loopback-18 and have a couple of comments:


General:
- I'm missing considerations around SRTP. It needs to be described in 
here how that works (e.g. what do you copy for RTP packet loopback and 
just be clear that each side does its normaly SRTP processing on both 
send and receive).


Specific sections:
- Abstract and Introduction: Says it defines new SDP media attributes. 
Should also mention it defines new MIME types

- 3.2, first paragraph:
Change "specific media types" to "specific media descriptions"
I believe the "MAY" should be a "MUST"

- 4.1, first paragraph
Change "property media attribute" to "value media attribute" and add 
reference to [RFC4566]

- 4.2, first paragraph
Change "value media attribute" to "property media attribute" and add 
reference to [RFC4566]

- 5.1, third paragraph
Instead of "terminate the session", we should rather just talk about 
"reject the offer".

- 5.2, first paragraph
add "and hence MUST NOT be used" to the end.

- 5.2
The first two examples use RFC 2119 language ("MUST") whereas the last 
one does not. Given these are examples, I think they should all use 
lower-case "must"

-6, third paragraph
rtp-media-loopback gets congestion considerations, whereas 
rtp-pkt-loopback in the the preceding paragraph does not. I believe it 
applies equally well to both.

- 7, second paragraph
Overall, I find this paragraph difficult to follow - it would be helpful 
to rephrase and/or elaborate.
<quote>
To keep the implementation of loopback-mirrors simple it is
     mandated that no payload format other than encapsulated or direct
     loopback formats can be used in the packets generated by a
     loopback-mirror. As described in RFC 3550 [RFC3550], sequence
     numbers and timestamps in the RTP header are generated with initial
     random values for security reasons. If this were not mandated and
     the source payload is sequence number aware, the loopback-mirror
     will be required to understand that payload format to generate
     looped back packets that do not violate RFC 3550 [RFC3550].
     Requiring looped back packets to be in one of the two formats means
     loopback-mirror does not have to look into the actual payload
     received before generating the loopback packets.
</quote>
Some specific questions:
a) If this is only an issue for the loopback-mirror, then why do we need 
to mandate it ? If it affects the source, then we need to elaborate on 
this in the O/A section, since it presumably imposes restrictions on the 
answerer operation (that were not called out there). Also, if we keep it 
as "mandated", then we need to use RFC 2119 language instead.

b) What does "this" refer to in the third sentence. The immediately 
preceding sentence, or the one before it ?

c) The paragraph does not seem to account for the "media loopback" case. 
This is probably because it is the payload format section where media 
loopback is not defined and hence it's not intended to. It's confusing 
as it reads though.


- 11, second paragraph
s/recommended/RECOMMENDED

- 11, third paragraph
The "infinite looping" attack seems to suggest there should be some form 
of rate limiting at the mirror (would be helpful at the source as well 
of course)

- 13.2
Change from MIME to Media Type registration of RTP payload formats per 
RFC 4855.  Text in registrations need to be updated accordingly 
(s/MIME/media type/).


I have a few additional nit fixes which I will forward directly to the 
authors.

Thanks

-- Flemming






On 4/2/12 9:18 AM, Miguel A. Garcia wrote:
> This is to start a 2-week Working Group Last Call for
>
>        draft-ietf-mmusic-media-loopback-18.txt
>
> The WGLC ends on April 16th, 2012.
>
> Please reply to this e-mail to send comments, so that the authors and 
> the mailing list are all copied.
>
> /Miguel
>

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I have reviewed loopback-18 and have a couple of comments: <br>
<br>
<br>
General:<br>
- I'm missing considerations around SRTP. It needs to be described in
here how that works (e.g. what do you copy for RTP packet loopback and
just be clear that each side does its normaly SRTP processing on both
send and receive). <br>
<br>
<br>
Specific sections:<br>
- Abstract and Introduction: Says it defines new SDP media attributes.
Should also mention it defines new MIME types<br>
<br>
- 3.2, first paragraph: <br>
Change "specific media types" to "specific media descriptions" <br>
I believe the "MAY" should be a "MUST"<br>
<br>
- 4.1, first paragraph<br>
Change "property media attribute" to "value media attribute" and add
reference to [RFC4566]<br>
<br>
- 4.2, first paragraph<br>
Change "value media attribute" to "property media attribute" and add
reference to [RFC4566]<br>
<br>
- 5.1, third paragraph<br>
Instead of "terminate the session", we should rather just talk about
"reject the offer". <br>
<br>
- 5.2, first paragraph<br>
add "and hence MUST NOT be used" to the end. <br>
<br>
- 5.2<br>
The first two examples use RFC 2119 language ("MUST") whereas the last
one does not. Given these are examples, I think they should all use
lower-case "must"<br>
<br>
-6, third paragraph<br>
rtp-media-loopback gets congestion considerations, whereas
rtp-pkt-loopback in the the preceding paragraph does not. I believe it
applies equally well to both. <br>
<br>
- 7, second paragraph<br>
Overall, I find this paragraph difficult to follow - it would be
helpful to rephrase and/or elaborate. <br>
&lt;quote&gt;<br>
To keep the implementation of loopback-mirrors simple it is <br>
&nbsp;&nbsp;&nbsp; mandated that no payload format other than encapsulated or direct <br>
&nbsp;&nbsp;&nbsp; loopback formats can be used in the packets generated by a <br>
&nbsp;&nbsp;&nbsp; loopback-mirror. As described in RFC 3550 [RFC3550], sequence <br>
&nbsp;&nbsp;&nbsp; numbers and timestamps in the RTP header are generated with initial
<br>
&nbsp;&nbsp;&nbsp; random values for security reasons. If this were not mandated and <br>
&nbsp;&nbsp;&nbsp; the source payload is sequence number aware, the loopback-mirror <br>
&nbsp;&nbsp;&nbsp; will be required to understand that payload format to generate <br>
&nbsp;&nbsp;&nbsp; looped back packets that do not violate RFC 3550 [RFC3550]. <br>
&nbsp;&nbsp;&nbsp; Requiring looped back packets to be in one of the two formats means
<br>
&nbsp;&nbsp;&nbsp; loopback-mirror does not have to look into the actual payload <br>
&nbsp;&nbsp;&nbsp; received before generating the loopback packets. <br>
&lt;/quote&gt;<br>
Some specific questions:<br>
a) <span style="font-size: 12pt; font-family: Cambria;">If this is
only an issue for the loopback-mirror, then why do we need to
mandate it ? If it affects the source, then we need to elaborate on
this in the
O/A section, since it presumably imposes restrictions on the answerer
operation
(that were not called out there). Also, if we keep it as "mandated",
then we need to use RFC 2119 language instead. </span>
<br>
<br>
b) <span style="font-size: 12pt; font-family: Cambria;">What does
"this" refer to in the third sentence. The immediately preceding
sentence,
or the one before it ? </span>
<br>
<br>
c) The paragraph does not seem to account for the "media loopback"
case. This is probably because it is the payload format section where
media loopback is not defined and hence it's not intended to. It's
confusing as it reads though. <br>
<br>
<br>
- 11, second paragraph<br>
s/recommended/RECOMMENDED<br>
<br>
- 11, third paragraph<br>
The "infinite looping" attack seems to suggest there should be some
form of rate limiting at the mirror (would be helpful at the source as
well of course)<br>
<br>
- 13.2<br>
<style>@font-face {
  font-family: "Cambria";
}p.MsoNormal, li.MsoNormal, div.MsoNormal { margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: "Times New Roman"; }div.Section1 { page: Section1; }</style>
Change from MIME to Media Type registration of RTP payload formats per
RFC 4855.&nbsp; Text in registrations need to be updated accordingly
(s/MIME/media type/).<br>
<br>
<br>
I have a few additional nit fixes which I will forward directly to the
authors. <br>
<br>
Thanks <br>
<br>
-- Flemming<br>
<br>
<br>
<br>
<br>
<br>
<br>
On 4/2/12 9:18 AM, Miguel A. Garcia wrote:
<blockquote cite="mid:4F79A716.3080804@ericsson.com" type="cite">This
is to start a 2-week Working Group Last Call for
  <br>
  <br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; draft-ietf-mmusic-media-loopback-18.txt
  <br>
  <br>
The WGLC ends on April 16th, 2012.
  <br>
  <br>
Please reply to this e-mail to send comments, so that the authors and
the mailing list are all copied.
  <br>
  <br>
/Miguel
  <br>
  <br>
</blockquote>
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From: Andrew Allen <aallen@rim.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: Issue in draft-ietf-mmusic-sdp-cs
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Subject: [MMUSIC] Issue in draft-ietf-mmusic-sdp-cs
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In 5.2.1 of the draft  it shows:

=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 c=3D PSTN - -
=A0
as a valid syntax however in section 5.6.1 it states:

The endpoint MUST set the <nettype> in the "c=3D" line to "PSTN", and the <a=
ddrtype> to "E164".=A0

Offline discussions with the authors indicates that they believe this is a 
bug in 5.6.1, and that the text should say that if the endpoint is not aware=
 of its E.164 number, both the <addrtype> and <connectionaddress> should be=
 set to "-".

It is proposed that a  change is made to  paragraph in 5.6.1 to 
clearly separate the cases when the endpoint is aware of its E.164 number an=
d when it is not.

Andrew
 
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From miguel.a.garcia@ericsson.com  Tue Apr 10 23:21:20 2012
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As an author,

I acknowledge that the draft should be clear, and somehow, contradicts 
itself. What the draft tries to say is what Andrew just pointed out: if 
the endpoint is not aware of its E.164 address, but just that there is a 
Circuit-Switched connection, then the c line should be:

  c=PSTN - -

/Miguel


On 10/04/2012 21:15, Andrew Allen wrote:
>
> In 5.2.1 of the draft  it shows:
>
>                  c= PSTN - -
>
> as a valid syntax however in section 5.6.1 it states:
>
> The endpoint MUST set the<nettype>  in the "c=" line to "PSTN", and the<addrtype>  to "E164".
>
> Offline discussions with the authors indicates that they believe this is a
> bug in 5.6.1, and that the text should say that if the endpoint is not aware of its E.164 number, both the<addrtype>  and<connectionaddress>  should be set to "-".
>
> It is proposed that a  change is made to  paragraph in 5.6.1 to
> clearly separate the cases when the endpoint is aware of its E.164 number and when it is not.
>
> Andrew
>
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> This transmission (including any attachments) may contain confidential information, privileged material (including material protected by the solicitor-client or other applicable privileges), or constitute non-public information. Any use of this information by anyone other than the intended recipient is prohibited. If you have received this 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.
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-- 
Miguel A. Garcia
+34-91-339-3608
Ericsson Spain

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Subject: [MMUSIC] [Technical Errata Reported] RFC5547 (3190)
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The following errata report has been submitted for RFC5547,
"A Session Description Protocol (SDP) Offer/Answer Mechanism to Enable File Transfer".

--------------------------------------
You may review the report below and at:
http://www.rfc-editor.org/errata_search.php?rfc=5547&eid=3190

--------------------------------------
Type: Technical
Reported by: Bruno CHATRAS <bruno.chatras@orange.com>

Section: GLOBAL

Original Text
-------------
Section 9.1., paragraph 7:
OLD:

    --boundary71
    Content-Type: application/sdp
    Content-Length: [length of SDP]

NEW:

    --boundary71
    Content-Type: application/sdp


Section 9.1., paragraph 9:
OLD:

    --boundary71
    Content-Type: image/jpeg
    Content-Transfer-Encoding: binary
    Content-ID: <id2@alicepc.example.com>
    Content-Length: [length of image]
    Content-Disposition: icon

NEW:

    --boundary71
    Content-Type: image/jpeg
    Content-Transfer-Encoding: binary
    Content-ID: <id2@alicepc.example.com>
    Content-Disposition: icon


Section 9.2., paragraph 24:
OLD:

    --boundary71
    Content-Type: application/sdp
    Content-Length: [length of SDP]

NEW:

    --boundary71
    Content-Type: application/sdp


Section 9.2., paragraph 26:
OLD:

    --boundary71
    Content-Type: image/jpeg
    Content-Transfer-Encoding: binary
    Content-ID: <id3@alicepc.example.com>
    Content-Length: [length of image]
    Content-Disposition: icon

NEW:

    --boundary71
    Content-Type: image/jpeg
    Content-Transfer-Encoding: binary
    Content-ID: <id3@alicepc.example.com>
    Content-Disposition: icon


Corrected Text
--------------


Notes
-----
A Content-Length header is shown for a body-part within a multipart body. But Content-Length is an HTTP/SIP header, not a IANA-registered MIME header and should therefore not appear at that location in valid examples. The length of a body part within a multipart body is determined by MIME framing. A Content-Length header found for a body-part within a multipart body is meaningless and should be ignored.

This was discussed on both the SIP Implementors and SIP Core mailing lists.

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

--------------------------------------
RFC5547 (draft-ietf-mmusic-file-transfer-mech-11)
--------------------------------------
Title               : A Session Description Protocol (SDP) Offer/Answer Mechanism to Enable File Transfer
Publication Date    : May 2009
Author(s)           : M. Garcia-Martin, M. Isomaki, G. Camarillo, S. Loreto, P. Kyzivat
Category            : PROPOSED STANDARD
Source              : Multiparty Multimedia Session Control
Area                : Real-time Applications and Infrastructure
Stream              : IETF
Verifying Party     : IESG

From magnus.westerlund@ericsson.com  Fri Apr 13 05:22:07 2012
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Cc: "draft-ietf-mmusic-media-loopback.authors@tools.ietf.org" <draft-ietf-mmusic-media-loopback.authors@tools.ietf.org>, Flemming Andreasen <fandreas@cisco.com>, mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] WGLC of draft-ietf-mmusic-media-loopback-18
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Hi,

I have reviewed the version in WGLC. I have some few comments:

1. Section 5.5:
"Note that for any form of NAT traversal to function, symmetric
    RTP/RTCP MUST be used."

Any reason to no reference RFC 4961 here?

2. Section 7.1.1 to 7.1.3 and 7.2.1 to 7.2.3 all are missing a space
between the section number and the section title.

3. Section 6:
 A loopback mirror that is compliant to this specification and
    accepts media with rtp-pkt-loopback loopback-type MUST loopback the
    incoming RTP packets using either the encapsulated RTP payload
    format or the direct loopback RTP payload format as defined in
    section 7 of this specification.

I don't think this has addressed my previous WG last call comment about
that the MUST needs an escape clause or at least a pointer to the issue
of congestion control. Because if you have no escape clause then the
loopback source MUST perform congestion control not only on the path
from the source to the mirror, but also on the mirror to the source.

4. Section 7.1.2:
    The outer RTP header of the encapsulating packet MUST be followed
    by the payload header defined in this section.  If the received RTP
    packet has to be looped back in multiple encapsulating packets due
    to fragmentation, the encapsulating RTP header in each packet MUST
    be followed by the payload header defined in this section.

I think this is are unfortunate formulations as it actually defines that
RTP header extensions and CSRC lists are not allowed. Especially header
extensions should not be disallowed. I would suggest a formulation saying:

The RTP payload of the encapsulating RTP packet starts with the payload
header defined in this section.

And go on and define the payload structure in relation to the start of
the payload.

5. Section 7.1.2:
    The
    Receive timestamp MUST be based on the same clock used by the
    loopback-source.

I still think this formulation is extremely wrong. One alternative is
the same clocks source, like if they where using the same NTP server. Or
is it actually "clock rate" that is intended here?

Se my previous comment 28 for more input about this:
http://www.ietf.org/mail-archive/web/mmusic/current/msg08971.html


6. I can't locate any answer to my issue number 30:

30. Section 7.2.1:
Sequence Number: The RTP sequence number SHOULD be generated by the
    loopback-mirror in the usual manner with a constant random offset.

What is the acceptable cases for when to break it? I assume the
intention with not saying MUST is that one can copy it from the
incomming packet. Thus when source -> mirror packet losses occur causing
wholes in the return path which isn't real loss on the mirror->source
path. It also causes third party monitor to report losses that aren't
real on this stream.

Can you please summarize any discussion that has happened?

7. I also don't see any changes related to my comment 34:

"34. Section 9.
This section is thoroughly inadequate. The reason is that you have a
forward path that is coupled to the reverse path. If the loopback is to
work correctly the mirror needs to be able to send back what the source
sent to it. Thus it is the loopback source that needs to adapt the media
bit-rate so that both forward and reverse path can handle the flows. If
doesn't the mirror will need to reduce the bit-rate it transmits and
that can only happen in two ways, dropping packets or transcoding them
to a lower rate fulfilling the congestion control requirement. Neither
will fulfill the purpose of the loopback as I see it. This issue needs
discussion, including some considerations in how to actually accomplish
it with the tools we have available."

This is coupled to issue 3 above. If that above MUST stands you
absolutely have to have more detailed discussion about dealing with the
whole loop, rather than a single leg that is the norm.

8. Section 11.

The amplification or reflection this allows has at least been
sufficiently discussed. I am far from certain that the mitigation is
clear enough.

I do support Flemming's comment about the SRTP implications of using
loopback are.

9. Another not addressed issue:

41. The media type registrations "rate" parameter I think needs to be
redefined. It needs to reference the rate of the payload types it
intends to mirror back.

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 vnagarjuna@saperix.com  Fri Apr 13 05:39:31 2012
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--20cf3056397d82769e04bd8ec33c
Content-Type: text/plain; charset=ISO-8859-1

On Fri, Apr 13, 2012 at 8:22 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

>
> 5. Section 7.1.2:
>    The
>    Receive timestamp MUST be based on the same clock used by the
>    loopback-source.
>
> I still think this formulation is extremely wrong. One alternative is
> the same clocks source, like if they where using the same NTP server. Or
> is it actually "clock rate" that is intended here?
>
> Se my previous comment 28 for more input about this:
> http://www.ietf.org/mail-archive/web/mmusic/current/msg08971.html
>
>
Magnus,

This was intended to be "clock rate" not clock.

Regards,
nagarjuna

--20cf3056397d82769e04bd8ec33c
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<br><br><div class=3D"gmail_quote">On Fri, Apr 13, 2012 at 8:22 AM, Magnus =
Westerlund <span dir=3D"ltr">&lt;<a href=3D"mailto:magnus.westerlund@ericss=
on.com">magnus.westerlund@ericsson.com</a>&gt;</span> wrote:<br><blockquote=
 class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc soli=
d;padding-left:1ex">
<br>
5. Section 7.1.2:<br>
 =A0 =A0The<br>
 =A0 =A0Receive timestamp MUST be based on the same clock used by the<br>
 =A0 =A0loopback-source.<br>
<br>
I still think this formulation is extremely wrong. One alternative is<br>
the same clocks source, like if they where using the same NTP server. Or<br=
>
is it actually &quot;clock rate&quot; that is intended here?<br>
<br>
Se my previous comment 28 for more input about this:<br>
<a href=3D"http://www.ietf.org/mail-archive/web/mmusic/current/msg08971.htm=
l" target=3D"_blank">http://www.ietf.org/mail-archive/web/mmusic/current/ms=
g08971.html</a><br>
<br></blockquote><div><br></div><div>Magnus,</div><div><br></div><div>This =
was intended to be &quot;clock rate&quot; not clock.</div><div><br></div><d=
iv>Regards,</div><div>nagarjuna=A0</div></div>

--20cf3056397d82769e04bd8ec33c--

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From: "Belling, Thomas (NSN - DE/Munich)" <thomas.belling@nsn.com>
To: "ext Miguel A. Garcia" <Miguel.A.Garcia@ericsson.com>, "Andrew Allen" <aallen@rim.com>
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Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] Issue in draft-ietf-mmusic-sdp-cs
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Dear Miguel,

Might be a matter of taste and can certainly be overcome with clear
descriptive text, but:

For me a dash instead of E.164 somehow sounds like the host not even
knowing its addresstype.

Thomas

-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf
Of ext Miguel A. Garcia
Sent: Wednesday, April 11, 2012 8:21 AM
To: Andrew Allen
Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] Issue in draft-ietf-mmusic-sdp-cs

As an author,

I acknowledge that the draft should be clear, and somehow, contradicts
itself. What the draft tries to say is what Andrew just pointed out: if
the endpoint is not aware of its E.164 address, but just that there is a
Circuit-Switched connection, then the c line should be:

  c=3DPSTN - -

/Miguel


On 10/04/2012 21:15, Andrew Allen wrote:
>
> In 5.2.1 of the draft  it shows:
>
>                  c=3D PSTN - -
>
> as a valid syntax however in section 5.6.1 it states:
>
> The endpoint MUST set the<nettype>  in the "c=3D" line to "PSTN", and
the<addrtype>  to "E164".
>
> Offline discussions with the authors indicates that they believe this=20
> is a bug in 5.6.1, and that the text should say that if the endpoint
is not aware of its E.164 number, both the<addrtype>
and<connectionaddress>  should be set to "-".
>
> It is proposed that a  change is made to  paragraph in 5.6.1 to=20
> clearly separate the cases when the endpoint is aware of its E.164
number and when it is not.
>
> Andrew
>
> ---------------------------------------------------------------------
> This transmission (including any attachments) may contain confidential
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>

--
Miguel A. Garcia
+34-91-339-3608
Ericsson Spain
_______________________________________________
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From: Andrew Allen <aallen@rim.com>
To: "Belling, Thomas (NSN - DE/Munich)" <thomas.belling@nsn.com>, "ext Miguel A. Garcia" <Miguel.A.Garcia@ericsson.com>
Thread-Topic: [MMUSIC] Issue in draft-ietf-mmusic-sdp-cs
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Subject: Re: [MMUSIC] Issue in draft-ietf-mmusic-sdp-cs
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Thomas

That's certainly one way you can look at it.

However I think the approach the authors have taken is that the address type=
 is the type of the connectionaddress value that follows. Since the connecti=
onaddress has no value ("-") putting the value E164 in the address type is p=
ointless.

Having "PSTN - -" syntax is consistent with the current 3GPP documentation t=
oo so I would prefer to keep it that way.

Andrew

> -----Original Message-----
> From: Belling, Thomas (NSN - DE/Munich) [mailto:thomas.belling@nsn.com]
> Sent: Friday, April 13, 2012 10:05 AM
> To: ext Miguel A. Garcia; Andrew Allen
> Cc: mmusic@ietf.org
> Subject: RE: [MMUSIC] Issue in draft-ietf-mmusic-sdp-cs
> 
> Dear Miguel,
> 
> Might be a matter of taste and can certainly be overcome with clear
> descriptive text, but:
> 
> For me a dash instead of E.164 somehow sounds like the host not even
> knowing its addresstype.
> 
> Thomas
> 
> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf
> Of ext Miguel A. Garcia
> Sent: Wednesday, April 11, 2012 8:21 AM
> To: Andrew Allen
> Cc: mmusic@ietf.org
> Subject: Re: [MMUSIC] Issue in draft-ietf-mmusic-sdp-cs
> 
> As an author,
> 
> I acknowledge that the draft should be clear, and somehow, contradicts
> itself. What the draft tries to say is what Andrew just pointed out: if
> the endpoint is not aware of its E.164 address, but just that there is a
> Circuit-Switched connection, then the c line should be:
> 
>   c=3DPSTN - -
> 
> /Miguel
> 
> 
> On 10/04/2012 21:15, Andrew Allen wrote:
> >
> > In 5.2.1 of the draft  it shows:
> >
> >                  c=3D PSTN - -
> >
> > as a valid syntax however in section 5.6.1 it states:
> >
> > The endpoint MUST set the<nettype>  in the "c=3D" line to "PSTN", and
> the<addrtype>  to "E164".
> >
> > Offline discussions with the authors indicates that they believe this
> > is a bug in 5.6.1, and that the text should say that if the endpoint
> is not aware of its E.164 number, both the<addrtype>
> and<connectionaddress>  should be set to "-".
> >
> > It is proposed that a  change is made to  paragraph in 5.6.1 to
> > clearly separate the cases when the endpoint is aware of its E.164
> number and when it is not.
> >
> > Andrew
> >
> > ---------------------------------------------------------------------
> > This transmission (including any attachments) may contain confidential
> information, privileged material (including material protected by the
> solicitor-client or other applicable privileges), or constitute
> non-public information. Any use of this information by anyone other than
> the intended recipient is prohibited. If you have received this
> 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.
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www.ietf.org/mailman/listinfo/mmusic
> >
> 
> --
> Miguel A. Garcia
> +34-91-339-3608
> Ericsson Spain
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

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From HKaplan@acmepacket.com  Sat Apr 14 04:52:41 2012
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: [MMUSIC] WGLC of draft-ietf-mmusic-media-loopback-18
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Cc: "draft-ietf-mmusic-media-loopback.authors@tools.ietf.org" <draft-ietf-mmusic-media-loopback.authors@tools.ietf.org>, Flemming Andreasen <fandreas@cisco.com>, mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] WGLC of draft-ietf-mmusic-media-loopback-18
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Hey Magnus,
inline...

On Apr 13, 2012, at 8:22 AM, Magnus Westerlund wrote:

> Hi,
>=20
> I have reviewed the version in WGLC. I have some few comments:
>=20
> 1. Section 5.5:
> "Note that for any form of NAT traversal to function, symmetric
>    RTP/RTCP MUST be used."
>=20
> Any reason to no reference RFC 4961 here?

Will do.

>=20
> 2. Section 7.1.1 to 7.1.3 and 7.2.1 to 7.2.3 all are missing a space
> between the section number and the section title.

Ya the formatting got screwed up by the generic text printing of the MS Wor=
d doc.

>=20
> 3. Section 6:
> A loopback mirror that is compliant to this specification and
>    accepts media with rtp-pkt-loopback loopback-type MUST loopback the
>    incoming RTP packets using either the encapsulated RTP payload
>    format or the direct loopback RTP payload format as defined in
>    section 7 of this specification.
>=20
> I don't think this has addressed my previous WG last call comment about
> that the MUST needs an escape clause or at least a pointer to the issue
> of congestion control. Because if you have no escape clause then the
> loopback source MUST perform congestion control not only on the path
> from the source to the mirror, but also on the mirror to the source.

OK.

>=20
> 4. Section 7.1.2:
>    The outer RTP header of the encapsulating packet MUST be followed
>    by the payload header defined in this section.  If the received RTP
>    packet has to be looped back in multiple encapsulating packets due
>    to fragmentation, the encapsulating RTP header in each packet MUST
>    be followed by the payload header defined in this section.
>=20
> I think this is are unfortunate formulations as it actually defines that
> RTP header extensions and CSRC lists are not allowed. Especially header
> extensions should not be disallowed. I would suggest a formulation saying=
:
>=20
> The RTP payload of the encapsulating RTP packet starts with the payload
> header defined in this section.
>=20
> And go on and define the payload structure in relation to the start of
> the payload.

Ah sorry I thought I cleaned up that wording to make it clear.  I'll fix th=
at.


>=20
> 5. Section 7.1.2:
>    The
>    Receive timestamp MUST be based on the same clock used by the
>    loopback-source.
>=20
> I still think this formulation is extremely wrong. One alternative is
> the same clocks source, like if they where using the same NTP server. Or
> is it actually "clock rate" that is intended here?
>=20
> Se my previous comment 28 for more input about this:
> http://www.ietf.org/mail-archive/web/mmusic/current/msg08971.html

Sorry I meant clock rate - will change.


>=20
>=20
> 6. I can't locate any answer to my issue number 30:
>=20
> 30. Section 7.2.1:
> Sequence Number: The RTP sequence number SHOULD be generated by the
>    loopback-mirror in the usual manner with a constant random offset.
>=20
> What is the acceptable cases for when to break it? I assume the
> intention with not saying MUST is that one can copy it from the
> incomming packet. Thus when source -> mirror packet losses occur causing
> wholes in the return path which isn't real loss on the mirror->source
> path. It also causes third party monitor to report losses that aren't
> real on this stream.
>=20
> Can you please summarize any discussion that has happened?

Actually there has been no discussion on this topic as far as I know - it w=
as not a SHOULD to allow copying, it's a SHOULD because it's a SHOULD in RF=
C 3550.
If you'd prefer, I can just say MUST follow RFC 3550.

>=20
> 7. I also don't see any changes related to my comment 34:
>=20
> "34. Section 9.
> This section is thoroughly inadequate. The reason is that you have a
> forward path that is coupled to the reverse path. If the loopback is to
> work correctly the mirror needs to be able to send back what the source
> sent to it. Thus it is the loopback source that needs to adapt the media
> bit-rate so that both forward and reverse path can handle the flows. If
> doesn't the mirror will need to reduce the bit-rate it transmits and
> that can only happen in two ways, dropping packets or transcoding them
> to a lower rate fulfilling the congestion control requirement. Neither
> will fulfill the purpose of the loopback as I see it. This issue needs
> discussion, including some considerations in how to actually accomplish
> it with the tools we have available."
>=20
> This is coupled to issue 3 above. If that above MUST stands you
> absolutely have to have more detailed discussion about dealing with the
> whole loop, rather than a single leg that is the norm.

OK we should start a thread on this.


>=20
> 8. Section 11.
>=20
> The amplification or reflection this allows has at least been
> sufficiently discussed. I am far from certain that the mitigation is
> clear enough.
>=20
> I do support Flemming's comment about the SRTP implications of using
> loopback are.

Yup.

>=20
> 9. Another not addressed issue:
>=20
> 41. The media type registrations "rate" parameter I think needs to be
> redefined. It needs to reference the rate of the payload types it
> intends to mirror back.

OK.


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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "mmusic (E-mail)" <mmusic@ietf.org>
Thread-Topic: media-loopback congestion
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Cc: "draft-ietf-mmusic-media-loopback.authors@tools.ietf.org" <draft-ietf-mmusic-media-loopback.authors@tools.ietf.org>
Subject: [MMUSIC] media-loopback congestion
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Magnus has raised a comment/concern as follows:

Currently section 6 says this:
   A loopback mirror that is compliant to this specification and
   accepts media with rtp-pkt-loopback loopback-type MUST loopback the
   incoming RTP packets using either the encapsulated RTP payload
   format or the direct loopback RTP payload format as defined in
   section 7 of this specification.

And section 9 about Congestion Control says this:
    All the participants in a loopback session SHOULD implement
    congestion control mechanisms as defined by the RTP profile under
    which the loopback mechanism is implemented. For audio video
    profiles, implementations SHOULD conform to the mechanism defined
    in Section 2 of RFC 3551.

Magnus' comment is this:
This section [9] is thoroughly inadequate. The reason is that you have a
forward path that is coupled to the reverse path. If the loopback is to
work correctly the mirror needs to be able to send back what the source
sent to it. Thus it is the loopback source that needs to adapt the media
bit-rate so that both forward and reverse path can handle the flows. If
doesn't the mirror will need to reduce the bit-rate it transmits and
that can only happen in two ways, dropping packets or transcoding them
to a lower rate fulfilling the congestion control requirement. Neither
will fulfill the purpose of the loopback as I see it. This issue needs
discussion, including some considerations in how to actually accomplish
it with the tools we have available.


From HKaplan@acmepacket.com  Sat Apr 14 05:09:47 2012
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "mmusic (E-mail)" <mmusic@ietf.org>, Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: [MMUSIC] media-loopback congestion
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Date: Sat, 14 Apr 2012 12:09:42 +0000
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Cc: "draft-ietf-mmusic-media-loopback.authors@tools.ietf.org" <draft-ietf-mmusic-media-loopback.authors@tools.ietf.org>
Subject: Re: [MMUSIC] media-loopback congestion
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One of Magnus concerns for below is that section 6's MUST statement does no=
t have an escape clause allowing the mirror to not mirror a packet back if =
it perceives congestion.  I propose to put in the following at the end of t=
he section 6 shown below:
..., when it can send packets back.  Since congestion might be detected by =
the mirror, for example, the loopback mirror might not necessarily loopback=
 all incoming RTP packets.

Does this work for everyone for this section?

There still needs to be discussion on section 9.

-hadriel


On Apr 14, 2012, at 7:59 AM, Hadriel Kaplan wrote:

> Magnus has raised a comment/concern as follows:
>=20
> Currently section 6 says this:
>   A loopback mirror that is compliant to this specification and
>   accepts media with rtp-pkt-loopback loopback-type MUST loopback the
>   incoming RTP packets using either the encapsulated RTP payload
>   format or the direct loopback RTP payload format as defined in
>   section 7 of this specification.


From gunnar.hellstrom@omnitor.se  Sat Apr 14 07:07:47 2012
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Cc: draft-ietf-mmusic-media-loopback.authors@tools.ietf.org, Flemming Andreasen <fandreas@cisco.com>, mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] WGLC of draft-ietf-mmusic-media-loopback-18
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I had concerns with the NAT traversal and keep-alive in earlier versions.

I am happy with the current version when the new comments and answers 
during WGLC are taken into account.

Good to see  progress.

/Gunnar


Miguel A. Garcia skrev 2012-04-02 15:18:
> This is to start a 2-week Working Group Last Call for
>
>        draft-ietf-mmusic-media-loopback-18.txt
>
> The WGLC ends on April 16th, 2012.
>
> Please reply to this e-mail to send comments, so that the authors and 
> the mailing list are all copied.
>
> /Miguel
>

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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: "Hadriel Kaplan" <HKaplan@acmepacket.com>, "mmusic (E-mail)" <mmusic@ietf.org>, "Magnus Westerlund" <magnus.westerlund@ericsson.com>
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Cc: draft-ietf-mmusic-media-loopback.authors@tools.ietf.org
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Hadriel,

|..., when it can send packets back.

This looks open for interpretation since there can be other reasons
apart from congestions where the mirror can't send packets back. My
suggestion is to state explicitly that congestion in the only exception:

   A loopback mirror that is compliant to this specification and
   accepts media with rtp-pkt-loopback loopback-type MUST loopback the
   incoming RTP packets using either the encapsulated RTP payload
   format or the direct loopback RTP payload format as defined in
   section 7 of this specification, except when the loopback mirror=20
   perceives congestion. When the loopback mirror perceives congestion
   it might not necessarily loopback all incoming RTP packets.

Muthu

|-----Original Message-----
|From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
Behalf Of Hadriel Kaplan
|Sent: Saturday, April 14, 2012 5:40 PM
|To: mmusic (E-mail); Magnus Westerlund
|Cc: draft-ietf-mmusic-media-loopback.authors@tools.ietf.org
|Subject: Re: [MMUSIC] media-loopback congestion
|
|
|One of Magnus concerns for below is that section 6's MUST statement
does not have an escape clause
|allowing the mirror to not mirror a packet back if it perceives
congestion.  I propose to put in the
|following at the end of the section 6 shown below:
|..., when it can send packets back.  Since congestion might be detected
by the mirror, for example,
|the loopback mirror might not necessarily loopback all incoming RTP
packets.
|
|Does this work for everyone for this section?
|
|There still needs to be discussion on section 9.
|
|-hadriel
|
|
|On Apr 14, 2012, at 7:59 AM, Hadriel Kaplan wrote:
|
|> Magnus has raised a comment/concern as follows:
|>
|> Currently section 6 says this:
|>   A loopback mirror that is compliant to this specification and
|>   accepts media with rtp-pkt-loopback loopback-type MUST loopback the
|>   incoming RTP packets using either the encapsulated RTP payload
|>   format or the direct loopback RTP payload format as defined in
|>   section 7 of this specification.
|
|_______________________________________________
|mmusic mailing list
|mmusic@ietf.org
|https://www.ietf.org/mailman/listinfo/mmusic

From miguel.a.garcia@ericsson.com  Mon Apr 16 00:44:57 2012
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Hi Thomas:

Please see below.

On 13/04/2012 17:05, Belling, Thomas (NSN - DE/Munich) wrote:
> Dear Miguel,
>
> Might be a matter of taste and can certainly be overcome with clear
> descriptive text, but:
>
> For me a dash instead of E.164 somehow sounds like the host not even
> knowing its addresstype.

And certainly that is the case. The device knows it is behind a circuit 
switched connection. It doesn't know any address, not an E.164, or not 
any other addressing scheme (say, E.212). So, if it doesn't know it, 
let's signal it with a dash. I don't see the point in indicating that the 
devices knows it has an E.164 address allocated to it, when it does not 
really know it.

/Miguel

> Thomas
>
> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf
> Of ext Miguel A. Garcia
> Sent: Wednesday, April 11, 2012 8:21 AM
> To: Andrew Allen
> Cc: mmusic@ietf.org
> Subject: Re: [MMUSIC] Issue in draft-ietf-mmusic-sdp-cs
>
> As an author,
>
> I acknowledge that the draft should be clear, and somehow, contradicts
> itself. What the draft tries to say is what Andrew just pointed out: if
> the endpoint is not aware of its E.164 address, but just that there is a
> Circuit-Switched connection, then the c line should be:
>
>    c=PSTN - -
>
> /Miguel
>
>
> On 10/04/2012 21:15, Andrew Allen wrote:
>>
>> In 5.2.1 of the draft  it shows:
>>
>>                   c= PSTN - -
>>
>> as a valid syntax however in section 5.6.1 it states:
>>
>> The endpoint MUST set the<nettype>   in the "c=" line to "PSTN", and
> the<addrtype>   to "E164".
>>
>> Offline discussions with the authors indicates that they believe this
>> is a bug in 5.6.1, and that the text should say that if the endpoint
> is not aware of its E.164 number, both the<addrtype>
> and<connectionaddress>   should be set to "-".
>>
>> It is proposed that a  change is made to  paragraph in 5.6.1 to
>> clearly separate the cases when the endpoint is aware of its E.164
> number and when it is not.
>>
>> Andrew
>>
>> ---------------------------------------------------------------------
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>> _______________________________________________
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>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>>
>
> --
> Miguel A. Garcia
> +34-91-339-3608
> Ericsson Spain
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

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

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Subject: Re: [MMUSIC] [Technical Errata Reported] RFC5547 (3190)
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Hi:

According to the discussion in SIPCORE, I agree with the reported errata. 
It should be moved to status "verified".

Thanks to Bruno and Paul for reporting it.

/Miguel

On 12/04/2012 18:06, RFC Errata System wrote:
>
> The following errata report has been submitted for RFC5547,
> "A Session Description Protocol (SDP) Offer/Answer Mechanism to Enable File Transfer".
>
> --------------------------------------
> You may review the report below and at:
> http://www.rfc-editor.org/errata_search.php?rfc=5547&eid=3190
>
> --------------------------------------
> Type: Technical
> Reported by: Bruno CHATRAS<bruno.chatras@orange.com>
>
> Section: GLOBAL
>
> Original Text
> -------------
> Section 9.1., paragraph 7:
>
> OLD:
>
>
>
>      --boundary71
>
>      Content-Type: application/sdp
>
>      Content-Length: [length of SDP]
>
>
>
> NEW:
>
>
>
>      --boundary71
>
>      Content-Type: application/sdp
>
>
>
>
>
> Section 9.1., paragraph 9:
>
> OLD:
>
>
>
>      --boundary71
>
>      Content-Type: image/jpeg
>
>      Content-Transfer-Encoding: binary
>
>      Content-ID:<id2@alicepc.example.com>
>
>      Content-Length: [length of image]
>
>      Content-Disposition: icon
>
>
>
> NEW:
>
>
>
>      --boundary71
>
>      Content-Type: image/jpeg
>
>      Content-Transfer-Encoding: binary
>
>      Content-ID:<id2@alicepc.example.com>
>
>      Content-Disposition: icon
>
>
>
>
>
> Section 9.2., paragraph 24:
>
> OLD:
>
>
>
>      --boundary71
>
>      Content-Type: application/sdp
>
>      Content-Length: [length of SDP]
>
>
>
> NEW:
>
>
>
>      --boundary71
>
>      Content-Type: application/sdp
>
>
>
>
>
> Section 9.2., paragraph 26:
>
> OLD:
>
>
>
>      --boundary71
>
>      Content-Type: image/jpeg
>
>      Content-Transfer-Encoding: binary
>
>      Content-ID:<id3@alicepc.example.com>
>
>      Content-Length: [length of image]
>
>      Content-Disposition: icon
>
>
>
> NEW:
>
>
>
>      --boundary71
>
>      Content-Type: image/jpeg
>
>      Content-Transfer-Encoding: binary
>
>      Content-ID:<id3@alicepc.example.com>
>
>      Content-Disposition: icon
>
>
>
> Corrected Text
> --------------
>
>
> Notes
> -----
> A Content-Length header is shown for a body-part within a multipart body. But Content-Length is an HTTP/SIP header, not a IANA-registered MIME header and should therefore not appear at that location in valid examples. The length of a body part within a multipart body is determined by MIME framing. A Content-Length header found for a body-part within a multipart body is meaningless and should be ignored.
>
>
>
> This was discussed on both the SIP Implementors and SIP Core mailing lists.
>
> Instructions:
> -------------
> This errata is currently posted as "Reported". If necessary, please
> use "Reply All" to discuss whether it should be verified or
> rejected. When a decision is reached, the verifying party (IESG)
> can log in to change the status and edit the report, if necessary.
>
> --------------------------------------
> RFC5547 (draft-ietf-mmusic-file-transfer-mech-11)
> --------------------------------------
> Title               : A Session Description Protocol (SDP) Offer/Answer Mechanism to Enable File Transfer
> Publication Date    : May 2009
> Author(s)           : M. Garcia-Martin, M. Isomaki, G. Camarillo, S. Loreto, P. Kyzivat
> Category            : PROPOSED STANDARD
> Source              : Multiparty Multimedia Session Control
> Area                : Real-time Applications and Infrastructure
> Stream              : IETF
> Verifying Party     : IESG

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

From miguel.a.garcia@ericsson.com  Mon Apr 16 02:15:32 2012
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>, "markus.isomaki@nokia.com" <markus.isomaki@nokia.com>, "fandreas@cisco.com" <fandreas@cisco.com>, "bruno.chatras@orange.com" <bruno.chatras@orange.com>, RFC Errata System <rfc-editor@rfc-editor.org>
Subject: Re: [MMUSIC] [Technical Errata Reported] RFC5547 (3190)
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Resending with Paul's correct e-mail address.

On 16/04/2012 11:10, Miguel A. Garcia wrote:
> Hi:
>
> According to the discussion in SIPCORE, I agree with the reported errata.
> It should be moved to status "verified".
>
> Thanks to Bruno and Paul for reporting it.
>
> /Miguel
>
> On 12/04/2012 18:06, RFC Errata System wrote:
>>
>> The following errata report has been submitted for RFC5547,
>> "A Session Description Protocol (SDP) Offer/Answer Mechanism to Enable File Transfer".
>>
>> --------------------------------------
>> You may review the report below and at:
>> http://www.rfc-editor.org/errata_search.php?rfc=5547&eid=3190
>>
>> --------------------------------------
>> Type: Technical
>> Reported by: Bruno CHATRAS<bruno.chatras@orange.com>
>>
>> Section: GLOBAL
>>
>> Original Text
>> -------------
>> Section 9.1., paragraph 7:
>>
>> OLD:
>>
>>
>>
>>       --boundary71
>>
>>       Content-Type: application/sdp
>>
>>       Content-Length: [length of SDP]
>>
>>
>>
>> NEW:
>>
>>
>>
>>       --boundary71
>>
>>       Content-Type: application/sdp
>>
>>
>>
>>
>>
>> Section 9.1., paragraph 9:
>>
>> OLD:
>>
>>
>>
>>       --boundary71
>>
>>       Content-Type: image/jpeg
>>
>>       Content-Transfer-Encoding: binary
>>
>>       Content-ID:<id2@alicepc.example.com>
>>
>>       Content-Length: [length of image]
>>
>>       Content-Disposition: icon
>>
>>
>>
>> NEW:
>>
>>
>>
>>       --boundary71
>>
>>       Content-Type: image/jpeg
>>
>>       Content-Transfer-Encoding: binary
>>
>>       Content-ID:<id2@alicepc.example.com>
>>
>>       Content-Disposition: icon
>>
>>
>>
>>
>>
>> Section 9.2., paragraph 24:
>>
>> OLD:
>>
>>
>>
>>       --boundary71
>>
>>       Content-Type: application/sdp
>>
>>       Content-Length: [length of SDP]
>>
>>
>>
>> NEW:
>>
>>
>>
>>       --boundary71
>>
>>       Content-Type: application/sdp
>>
>>
>>
>>
>>
>> Section 9.2., paragraph 26:
>>
>> OLD:
>>
>>
>>
>>       --boundary71
>>
>>       Content-Type: image/jpeg
>>
>>       Content-Transfer-Encoding: binary
>>
>>       Content-ID:<id3@alicepc.example.com>
>>
>>       Content-Length: [length of image]
>>
>>       Content-Disposition: icon
>>
>>
>>
>> NEW:
>>
>>
>>
>>       --boundary71
>>
>>       Content-Type: image/jpeg
>>
>>       Content-Transfer-Encoding: binary
>>
>>       Content-ID:<id3@alicepc.example.com>
>>
>>       Content-Disposition: icon
>>
>>
>>
>> Corrected Text
>> --------------
>>
>>
>> Notes
>> -----
>> A Content-Length header is shown for a body-part within a multipart body. But Content-Length is an HTTP/SIP header, not a IANA-registered MIME header and should therefore not appear at that location in valid examples. The length of a body part within a multipart body is determined by MIME framing. A Content-Length header found for a body-part within a multipart body is meaningless and should be ignored.
>>
>>
>>
>> This was discussed on both the SIP Implementors and SIP Core mailing lists.
>>
>> Instructions:
>> -------------
>> This errata is currently posted as "Reported". If necessary, please
>> use "Reply All" to discuss whether it should be verified or
>> rejected. When a decision is reached, the verifying party (IESG)
>> can log in to change the status and edit the report, if necessary.
>>
>> --------------------------------------
>> RFC5547 (draft-ietf-mmusic-file-transfer-mech-11)
>> --------------------------------------
>> Title               : A Session Description Protocol (SDP) Offer/Answer Mechanism to Enable File Transfer
>> Publication Date    : May 2009
>> Author(s)           : M. Garcia-Martin, M. Isomaki, G. Camarillo, S. Loreto, P. Kyzivat
>> Category            : PROPOSED STANDARD
>> Source              : Multiparty Multimedia Session Control
>> Area                : Real-time Applications and Infrastructure
>> Stream              : IETF
>> Verifying Party     : IESG
>

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

From magnus.westerlund@ericsson.com  Mon Apr 16 02:47:19 2012
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Subject: Re: [MMUSIC] media-loopback congestion
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On 2012-04-16 08:04, Muthu Arul Mozhi Perumal (mperumal) wrote:
> Hadriel,
> 
> |..., when it can send packets back.
> 
> This looks open for interpretation since there can be other reasons
> apart from congestions where the mirror can't send packets back. My
> suggestion is to state explicitly that congestion in the only exception:
> 
>    A loopback mirror that is compliant to this specification and
>    accepts media with rtp-pkt-loopback loopback-type MUST loopback the
>    incoming RTP packets using either the encapsulated RTP payload
>    format or the direct loopback RTP payload format as defined in
>    section 7 of this specification, except when the loopback mirror 
>    perceives congestion. When the loopback mirror perceives congestion
>    it might not necessarily loopback all incoming RTP packets.


To start with Hadriel's question if this is sufficient. I don't think
so. The reasons is that unless you have a media level loopback capable
of doing transcoding to a format with variable bit-rate the loopback
case is such that you don't want to modify the packets in the return
path. From the perspective of reaching the goals of loopback it would
actually be best to put all the responsibility of the congestion control
on the media source. And it needs to take both the forward and return
path into account when making congestion control decisions.

Actually this is in many ways simpler than having one way control. This
as the packet sender and receiver are the same. The source will know
what it sent and when and can make accurate full RTT measurements to
determine delay variations. It is also certain of all the packets it
sent. If the mirror is truly deterministic in its behavior then you can
accurately model for these conditions. In fact having the mirror detect
and stop sending when it sees congestion actually makes the source work
more difficult.

So I see several approaches here.

1) Specify that the source is sole responsible for both paths.

2) Specify that the source is sole responsible and have the mirror
verify compliance by running a circuit breaker algorithm to verify that
at least the return path isn't seriously messing things up.

3) Specify that each side runs independent congestion control and then
discuss the impact on the return path when there is congestion in that
direction and not the forward one.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
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From mperumal@cisco.com  Mon Apr 16 22:53:00 2012
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Date: Tue, 17 Apr 2012 11:22:51 +0530
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Thread-Topic: [MMUSIC] media-loopback congestion
Thread-Index: Ac0btfZZvTzR6TCkQrm1AZZfU8vyywAnkabQ
References: <1DB4397F-D929-49FD-9462-095DF9196172@acmepacket.com> <581DCD23-A142-4414-8662-05B84FD8AA50@acmepacket.com> <1D062974A4845E4D8A343C65380492020827EB48@XMB-BGL-414.cisco.com> <4F8BEAA4.4090906@ericsson.com>
From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>
X-OriginalArrivalTime: 17 Apr 2012 05:52:53.0312 (UTC) FILETIME=[5475C400:01CD1C5E]
Cc: draft-ietf-mmusic-media-loopback.authors@tools.ietf.org, "mmusic \(E-mail\)" <mmusic@ietf.org>
Subject: Re: [MMUSIC] media-loopback congestion
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|From the perspective of reaching the goals of loopback it would
|actually be best to put all the responsibility of the congestion=20
|control on the media source. And it needs to take both the forward
|and return path into account when making congestion control decisions.

I agree. The purpose of the packet loopback mode is to provide a simple =
mechanism to test how good the media path is and not to test how good =
the mirror can respond to network congestion, IMHO. If there is packet =
loss due to congestion, so it be -- the loopback source will know that. =
It might then want to repeat the test to see if this is a temporary =
issue. If subsequent tests produce packet loss, it might conclude that =
the media path isn't good for making an audio/video call using the coded =
it used.

Requiring the mirror to implement congestion control to conceal packet =
loss would be of little value for this case. Worse, if were a mechanism =
different from what the mirror endpoint might use when it actually =
renders and generates media -- it might give a false impression about =
the quality of the media path.

|2) Specify that the source is sole responsible and have the mirror
|verify compliance by running a circuit breaker algorithm to verify
|that at least the return path isn't seriously messing things up.

I agree the source should be made responsible for congestion control for =
both the paths. In addition, a mirror running circuit breaker could be =
useful if it does the same when it renders and generates media -- might =
help the user to determine using packet loopback at point media stops =
working. On the other hand, if the purpose of the test is to see how =
seriously the return path is messed up, then a circuit breaker at the =
mirror may not be needed. So, the draft could specify that the mirror =
MAY run a circuit breaker, I think.

On similar lines, having the mirror report congestion to the source =
might be useful as well.

Muthu

|-----Original Message-----
|From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
|Sent: Monday, April 16, 2012 3:17 PM
|To: Muthu Arul Mozhi Perumal (mperumal)
|Cc: Hadriel Kaplan; mmusic (E-mail); =
draft-ietf-mmusic-media-loopback.authors@tools.ietf.org
|Subject: Re: [MMUSIC] media-loopback congestion
|
|On 2012-04-16 08:04, Muthu Arul Mozhi Perumal (mperumal) wrote:
|> Hadriel,
|>
|> |..., when it can send packets back.
|>
|> This looks open for interpretation since there can be other reasons
|> apart from congestions where the mirror can't send packets back. My
|> suggestion is to state explicitly that congestion in the only =
exception:
|>
|>    A loopback mirror that is compliant to this specification and
|>    accepts media with rtp-pkt-loopback loopback-type MUST loopback =
the
|>    incoming RTP packets using either the encapsulated RTP payload
|>    format or the direct loopback RTP payload format as defined in
|>    section 7 of this specification, except when the loopback mirror
|>    perceives congestion. When the loopback mirror perceives =
congestion
|>    it might not necessarily loopback all incoming RTP packets.
|
|
|To start with Hadriel's question if this is sufficient. I don't think
|so. The reasons is that unless you have a media level loopback capable
|of doing transcoding to a format with variable bit-rate the loopback
|case is such that you don't want to modify the packets in the return
|path. From the perspective of reaching the goals of loopback it would
|actually be best to put all the responsibility of the congestion =
control
|on the media source. And it needs to take both the forward and return
|path into account when making congestion control decisions.
|
|Actually this is in many ways simpler than having one way control. This
|as the packet sender and receiver are the same. The source will know
|what it sent and when and can make accurate full RTT measurements to
|determine delay variations. It is also certain of all the packets it
|sent. If the mirror is truly deterministic in its behavior then you can
|accurately model for these conditions. In fact having the mirror detect
|and stop sending when it sees congestion actually makes the source work
|more difficult.
|
|So I see several approaches here.
|
|1) Specify that the source is sole responsible for both paths.
|
|2) Specify that the source is sole responsible and have the mirror
|verify compliance by running a circuit breaker algorithm to verify that
|at least the return path isn't seriously messing things up.
|
|3) Specify that each side runs independent congestion control and then
|discuss the impact on the return path when there is congestion in that
|direction and not the forward one.
|
|Cheers
|
|Magnus Westerlund
|
|----------------------------------------------------------------------
|Multimedia Technologies, Ericsson Research EAB/TVM
|----------------------------------------------------------------------
|Ericsson AB                | Phone  +46 10 7148287
|F=E4r=F6gatan 6                | Mobile +46 73 0949079
|SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
|----------------------------------------------------------------------


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From miguel.a.garcia@ericsson.com  Wed Apr 18 05:46:28 2012
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Cc: "draft-ietf-mmusic-media-loopback.authors@tools.ietf.org" <draft-ietf-mmusic-media-loopback.authors@tools.ietf.org>, Flemming Andreasen <fandreas@cisco.com>
Subject: Re: [MMUSIC] WGLC of draft-ietf-mmusic-media-loopback-18
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The WGLC of draft-ietf-mmusic-media-loopback-18.txt has concluded. Thanks 
to all who provided comments.

Please authors, address the issues discussed in the last call and submit 
a new version of the document as early as you can.

/Miguel

On 02/04/2012 15:18, Miguel A. Garcia wrote:
> This is to start a 2-week Working Group Last Call for
>
>          draft-ietf-mmusic-media-loopback-18.txt
>
> The WGLC ends on April 16th, 2012.
>
> Please reply to this e-mail to send comments, so that the authors and the
> mailing list are all copied.
>
> /Miguel
>

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

From miguel.a.garcia@ericsson.com  Wed Apr 18 06:02:09 2012
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This is to start a 4-week Working Group Last Call for

      draft-ietf-mmusic-rfc2326bis-29.txt

The WGLC ends on May 16th, 2012. Notice that this is a special 4-week 
WGLC due to the length of the document.

We would especially like reviewers to focus on the changes since the last 
WGLC, which was for version -23.

Please reply to this e-mail that includes the authors and the mailing list.

/Miguel

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

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From: "Dan Wing" <dwing@cisco.com>
To: <zhou.sujing@zte.com.cn>
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Subject: Re: [MMUSIC] Hi, May I ask for your opinion on draft-zhou-mmusic-sdes-keymod-01?
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> -----Original Message-----
> From: zhou.sujing@zte.com.cn [mailto:zhou.sujing@zte.com.cn]
> Sent: Tuesday, April 17, 2012 7:49 PM
> To: Dan Wing
> Cc: mmusic@ietf.org
> Subject: RE: RE: RE: Hi, May I ask for your opinion on draft-zhou-
> mmusic-sdes-keymod-01?
>=20
>=20
> Regards~~~
>=20
> -Sujing Zhou
>=20
> "Dan Wing" <dwing@cisco.com> =D0=B4=D3=DA 2012-04-17 23:42:36:
>=20
> > > -----Original Message-----
> > > From: zhou.sujing@zte.com.cn [mailto:zhou.sujing@zte.com.cn]
> > > Sent: Monday, April 16, 2012 10:10 PM
> > > To: Dan Wing
> > > Subject: =B4=F0=B8=B4: RE: Hi, May I ask for your opinion on =
draft-zhou-
> mmusic-
> > > sdes-keymod-01?
> > >
> > >
> > > Yes=A3=ACthe offer cannot determin if re-trageting or forking are
> occuring.
> > > I added a new parameter "keymod" with NULL=A1=A1value in the offer
> message
> > > (in 01 version) no matter wich scenarios,
> > > to inform the answerer that
> > >
> > >  "Hi, I am kind of concerned about my outgoing
> > > message be intercept by some intermediates, so I am ready to =
accept
> any
> > > answer message with
> > > keymod parameter, if you send values in keymod parameter, I can
> update
> > > my key accordingly"
> > >
> > > It is up to the answerer to decide if it will send keymod =
parameter
> > > with valid value.
> > >
> > > If the answer does not care, it can send keymod with NULL value,
> and
> > > this case is the same as the current SDES;
> > > If the answer cares, it can send keymod with non-NULL value, and
> this
> > > case is what our document is solving.
> > >
> > > The answerer may be able to detect it is in a re-targeting/forking
> > > scenario, and be triggered to send keymod;
> > > Or, the answerer could not detect the situation, and be encouraged
> to
> > > send keymod, because offerer asked for.
> >
> > The offerer should just always issue a re-Invite with a new =
a=3Dcrypto
> key.
> >
> > I guess you are trying to optimize that so that the offerer and
> answerer
> > somehow determine if a re-Invite might be necessary?  Why not
> generalize
> > that optimization, so that the answerer indicates "re-direct
> occurred" and
> > indicates "forking occurred"?  That could be useful for this SDESC
> re-Invite
> > optimization, and could be useful elsewhere.  However, I believe it
> is
> > impossible for the Answerer to reliably determine if a re-direct
> occurred or
> > if forking occurred, which is why I have doubt that the proposal in
> your I-D
> > can work as you hope.
> >
> > -d
>=20
> Let the offerer send a re-Invite or an UPDATE with a new a=3Dcrypto =
key
> is indeed an solution.
> But it require more roundtrips.
>=20
> Our motivations are
> -to optimize the procedure
> -to enhance the security of SDES
>=20
> your question is around the applicability of the optimized procedure.
> In the case of re-direct occurs, the anwser always knows it is a re-
> directed call,should no problem, our 00 draft works.
> In the case  of forking, the answer does not know it is a forked call,
> our 00 draft does not work in this case.
>=20
> Our 01 verion does not let answer alone to determin if a re-direct or =
a
> forking occur.
> Instead, our 01 version is a general upgrade to current SDES protocol,
> that will come to our
> another motivation :enhance the security of SDES
>=20
> Generaly it is preferable the session key  between two peers  be
> established with contribution from both peers,otherwise we will get
> into trouble
> as  SDES now in the scenarios of re-targetting and forking.
> Our 01 version actually suggests to change the unidirectional key
> transport in SDES into a key agreement(indicated by "keymod"):
> offerer provides: k1
> answer provides: keymod value
> the outgoing key from offerer to answerer is derived from k1 and =
keymod
> value no matter in which situation.
> Re-targeting and forking  happen to be the scenarios that especially
> benefit from the change.

Which involves the same number of (signaling) round-trips, right?

-d


>=20
>=20
>=20
> >
> >
> > >
> > >
> > > Regards~~~
> > >
> > > -Sujing Zhou
> > >
> > > "Dan Wing" <dwing@cisco.com> =D0=B4=D3=DA 2012-04-17 05:14:58:
> > >
> > > > I don't see how sending additional parameters solves the =
problem.
> > > >
> > > > The problem is the offerer cannot determine if re-targeting or
> > > forking
> > > > occurred, and thus cannot determine if it needs to send a new
> > > (updated)
> > > > offer.  The only solution is to always send a new offer.
> > > >
> > > > -d
> > > >
> > > >
> > > > > -----Original Message-----
> > > > > From: zhou.sujing@zte.com.cn [mailto:zhou.sujing@zte.com.cn]
> > > > > Sent: Monday, April 16, 2012 12:29 AM
> > > > > To: Dan Wing
> > > > > Subject: Hi, May I ask for your opinion on draft-zhou-mmusic-
> sdes-
> > > > > keymod-01?
> > > > >
> > > > >
> > > > > Hi, Wing,
> > > > >
> > > > >   I=A1=A1have updated my draft incorporating your question.
> > > > > http://tools.ietf.org/html/draft-zhou-mmusic-sdes-keymod-01
> > > > >   May I ask for your opinion and interest in it?
> > > > >
> > > > > Thank you!
> > > > >
> > > > > Regards~~~
> > > > >
> > > > > -Sujing Zhou
> > > > >
> > > > > mmusic-bounces@ietf.org =D0=B4=D3=DA 2012-03-21 09:32:40:
> > > > >
> > > > > >
> > > > > > Hi=A3=AC Wing
> > > > > >
> > > > > >     Your point is good, I may update our draft (in 01
> version)
> > > like
> > > > > > the following paragraph:
> > > > > > " 3.1.  Generating the Initial Offer - Unicast Streams
> > > > > >
> > > > > >    The generation of the initial offer for a unicast stream
> MUST
> > > > > follow
> > > > > >    that of the crypto attribute RFC4568 [RFC4568], and MAY
> > > > > >
> > > > > >    also include an additional "keymod" parameter with =
keymod-
> val
> > > > > being
> > > > > >    NULL.  It indicates to the ultimate answerer that the
> offerer
> > > > > wants
> > > > > >    to employ the mechanism specified in
> > > > > >
> > > > > >    this document, a key agreement mechanism with a higher
> > > security
> > > > > level
> > > > > >    than the original SDES.
> > > > > > "
> > > > > > So that answerer does not need to know it is in a re-
> targeting or
> > > > > > forking scenario,
> > > > > > it is offerer suggesting use keymod mechanism.
> > > > > >
> > > > > >
> > > > > >
> > > > > > Regards~~~
> > > > > >
> > > > > > -Sujing Zhou
> > > > > >
> > > > > > "Dan Wing" <dwing@cisco.com> =D0=B4=D3=DA 2012-03-18 =
10:43:42:
> > > > > >
> > > > > > > To be effective, draft-zhou-mmusic-sdes-keymod needs the
> > > Answerer
> > > > > to send
> > > > > > > the "keymod" extension if it knows, or believes, the =
INVITE
> was
> > > re-
> > > > > targeted
> > > > > > > or forked.  I have two questions:
> > > > > > >
> > > > > > > I see for the described re-targeting case (Section 5.1 of
> > > > > > > draft-zhou-mmusic-sdes-keymod-00), a "cause" parameter
> > > [RFC4458]
> > > > > can be a
> > > > > > > useful indicator that a call was re-targeted.  Is the
> "cause"
> > > > > parameter
> > > > > > > always present when a call is re-targeted?
> > > > > > >
> > > > > > > How does the Answerer know if a call was forked?
> > > > > > >
> > > > > > > -d
> > > > > > >
> > > > > > >
> > > > > > >
> > > > > > >
> > > > > > > _______________________________________________
> > > > > > mmusic mailing list
> > > > > > mmusic@ietf.org
> > > > > > https://www.ietf.org/mailman/listinfo/mmusic
> > > >
> > > >
> > > >
> >
> >
> >



From HKaplan@acmepacket.com  Wed Apr 18 08:01:48 2012
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Thread-Topic: [MMUSIC] WGLC of draft-ietf-mmusic-media-loopback-18
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Roger, will do.

We still have on open issue on the congestion stuff, however.

-hadriel


On Apr 18, 2012, at 8:46 AM, Miguel A. Garcia wrote:

> The WGLC of draft-ietf-mmusic-media-loopback-18.txt has concluded. Thanks=
 to all who provided comments.
>=20
> Please authors, address the issues discussed in the last call and submit =
a new version of the document as early as you can.
>=20
> /Miguel
>=20
> On 02/04/2012 15:18, Miguel A. Garcia wrote:
>> This is to start a 2-week Working Group Last Call for
>>=20
>>         draft-ietf-mmusic-media-loopback-18.txt
>>=20
>> The WGLC ends on April 16th, 2012.
>>=20
>> Please reply to this e-mail to send comments, so that the authors and th=
e
>> mailing list are all copied.
>>=20
>> /Miguel
>>=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


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> > 
> > Generaly it is preferable the session key  between two peers  be
> > established with contribution from both peers,otherwise we will get
> > into trouble
> > as  SDES now in the scenarios of re-targetting and forking.
> > Our 01 version actually suggests to change the unidirectional key
> > transport in SDES into a key agreement(indicated by "keymod"):
> > offerer provides: k1
> > answer provides: keymod value
> > the outgoing key from offerer to answerer is derived from k1 and 
keymod
> > value no matter in which situation.
> > Re-targeting and forking  happen to be the scenarios that especially
> > benefit from the change.
> 
> Which involves the same number of (signaling) round-trips, right?

In my opinion, the new method does not add extra round trips, it has the 
same round trips with 
the current SDES without re-INVITE or UPDATE.

offerer-->answerer:INVITE
       a=crypto:1 AES_CM_128_HMAC_SHA1_80
        inline:d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj|2^20|1:32 --->k1
        keymod:rand|xor|
offerer<--answerer:Response
       a=crypto:1 AES_CM_128_HMAC_SHA1_32
       inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj|2^20|1:32; --->k2
       keymod:rand|xor|WVNfX19zZW1jdGwgKCkgew==         ->keymod value

after the single round,
    k1 and keymod value-->k1' to protect session from offerer to answerer
   k2 -->  to protect session from answerer   to offerer

> 
> -d

--=_alternative 000465DE482579E5_=
Content-Type: text/html; charset="US-ASCII"


<br><tt><font size=2><br>
&gt; &gt; <br>
&gt; &gt; Generaly it is preferable the session key &nbsp;between two peers
&nbsp;be<br>
&gt; &gt; established with contribution from both peers,otherwise we will
get<br>
&gt; &gt; into trouble<br>
&gt; &gt; as &nbsp;SDES now in the scenarios of re-targetting and forking.<br>
&gt; &gt; Our 01 version actually suggests to change the unidirectional
key<br>
&gt; &gt; transport in SDES into a key agreement(indicated by &quot;keymod&quot;):<br>
&gt; &gt; offerer provides: k1<br>
&gt; &gt; answer provides: keymod value<br>
&gt; &gt; the outgoing key from offerer to answerer is derived from k1
and keymod<br>
&gt; &gt; value no matter in which situation.<br>
&gt; &gt; Re-targeting and forking &nbsp;happen to be the scenarios that
especially<br>
&gt; &gt; benefit from the change.<br>
&gt; <br>
&gt; Which involves the same number of (signaling) round-trips, right?<br>
</font></tt>
<br><tt><font size=2>In my opinion, the new method does not add extra round
trips, it has the same round trips with </font></tt>
<br><tt><font size=2>the current SDES without re-INVITE or UPDATE.</font></tt>
<br>
<br><tt><font size=2>offerer--&gt;answerer:INVITE</font></tt>
<br><tt><font size=2>&nbsp; &nbsp; &nbsp; &nbsp;a=crypto:1 AES_CM_128_HMAC_SHA1_80</font></tt>
<br><tt><font size=2>&nbsp; &nbsp; &nbsp; &nbsp; inline:d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj|2^20|1:32
---&gt;k1</font></tt>
<br><tt><font size=2>&nbsp; &nbsp; &nbsp; &nbsp; keymod:rand|xor|</font></tt>
<br><font size=2>offerer&lt;--answerer:Response</font>
<br><tt><font size=2>&nbsp; &nbsp; &nbsp; &nbsp;a=crypto:1 AES_CM_128_HMAC_SHA1_32</font></tt>
<br><tt><font size=2>&nbsp; &nbsp; &nbsp; &nbsp;inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj|2^20|1:32;
---&gt;k2</font></tt>
<br><tt><font size=2>&nbsp; &nbsp; &nbsp; &nbsp;keymod:rand|xor|WVNfX19zZW1jdGwgKCkgew==
&nbsp; &nbsp; &nbsp; &nbsp; -&gt;keymod value</font></tt>
<br>
<br><tt><font size=2>after the single round,</font></tt>
<br><tt><font size=2>&nbsp; &nbsp; k1 and keymod value--&gt;k1' to protect
session from offerer to answerer</font></tt>
<br><tt><font size=2>&nbsp; &nbsp;k2 --&gt; &nbsp;to protect session from
answerer &nbsp; to offerer</font></tt>
<br>
<br><tt><font size=2>&gt; <br>
&gt; -d<br>
</font></tt>
--=_alternative 000465DE482579E5_=--


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From: "Dan Wing" <dwing@cisco.com>
To: <zhou.sujing@zte.com.cn>
References: <079c01cd1d68$05710670$10531350$@com> <OF24015959.EB99E9AE-ON482579E5.0003B462-482579E5.000465E0@zte.com.cn>
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Subject: Re: [MMUSIC] Hi, May I ask for your opinion on draft-zhou-mmusic-sdes-keymod-01?
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> -----Original Message-----
> From: zhou.sujing@zte.com.cn [mailto:zhou.sujing@zte.com.cn]
> Sent: Wednesday, April 18, 2012 5:48 PM
> To: Dan Wing
> Cc: mmusic@ietf.org
> Subject: Re: RE: RE: RE: Hi, May I ask for your opinion on draft-zhou-
> mmusic-sdes-keymod-01?
> 
> 
> 
> > >
> > > Generaly it is preferable the session key  between two peers  be
> > > established with contribution from both peers,otherwise we will get
> > > into trouble
> > > as  SDES now in the scenarios of re-targetting and forking.
> > > Our 01 version actually suggests to change the unidirectional key
> > > transport in SDES into a key agreement(indicated by "keymod"):
> > > offerer provides: k1
> > > answer provides: keymod value
> > > the outgoing key from offerer to answerer is derived from k1 and
> keymod
> > > value no matter in which situation.
> > > Re-targeting and forking  happen to be the scenarios that
> especially
> > > benefit from the change.
> >
> > Which involves the same number of (signaling) round-trips, right?
> 
> In my opinion, the new method does not add extra round trips, it has
> the same round trips with
> the current SDES without re-INVITE or UPDATE.
> 
> offerer-->answerer:INVITE
>        a=crypto:1 AES_CM_128_HMAC_SHA1_80
>         inline:d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj|2^20|1:32 ---
> >k1
>         keymod:rand|xor|
> offerer<--answerer:Response
>        a=crypto:1 AES_CM_128_HMAC_SHA1_32
>        inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj|2^20|1:32; ---
> >k2
>        keymod:rand|xor|WVNfX19zZW1jdGwgKCkgew==         ->keymod value
> 
> after the single round,
>     k1 and keymod value-->k1' to protect session from offerer to
> answerer
>    k2 -->  to protect session from answerer   to offerer

I now understand what you're proposing, thanks for explaining it this way.

That avoids a signaling round trip, but does require the Offerer and
Answerer support keymod.  If either of them don't, the Offerer needs to
always do a re-Invite.  So this appears a reasonable optimization to avoid
always doing a re-Invite.

-d



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--=_alternative 0011C4E5482579E5_=--


From magnus.westerlund@ericsson.com  Thu Apr 19 02:10:24 2012
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Date: Thu, 19 Apr 2012 11:10:17 +0200
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [MMUSIC] WGLC of draft-ietf-mmusic-rfc2326bis-29.txt
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On 2012-04-18 15:02, Miguel A. Garcia wrote:
> This is to start a 4-week Working Group Last Call for
> 
>       draft-ietf-mmusic-rfc2326bis-29.txt
> 
> The WGLC ends on May 16th, 2012. Notice that this is a special 4-week 
> WGLC due to the length of the document.
> 
> We would especially like reviewers to focus on the changes since the last 
> WGLC, which was for version -23.
> 
> Please reply to this e-mail that includes the authors and the mailing list.

Hi,

As an author I can post the link of the Diff between 23 and now:

https://tools.ietf.org/rfcdiff?url1=draft-ietf-mmusic-rfc2326bis-23&difftype=--html&submit=Go!&url2=draft-ietf-mmusic-rfc2326bis-29

This is quite a substantiall amount of changes. Unfortunately the
section reordering makes it difficult to spot changes in some sub-sections.

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 dwing@cisco.com  Thu Apr 19 08:26:09 2012
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From: "Dan Wing" <dwing@cisco.com>
To: <zhou.sujing@zte.com.cn>
References: <0b9d01cd1dcc$4f46db30$edd49190$@com> <OFA855A751.2A28087D-ON482579E5.0011A9EC-482579E5.0011C4E5@zte.com.cn>
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Subject: Re: [MMUSIC] Hi, May I ask for your opinion on draft-zhou-mmusic-sdes-keymod-01?
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> -----Original Message-----
> From: zhou.sujing@zte.com.cn [mailto:zhou.sujing@zte.com.cn]
> Sent: Wednesday, April 18, 2012 8:14 PM
> To: Dan Wing
> Cc: mmusic@ietf.org
> Subject: =B4=F0=B8=B4: RE: RE: RE: RE: Hi, May I ask for your opinion =
on draft-
> zhou-mmusic-sdes-keymod-01?
>=20
>=20
> Then do your support our draft being considered into a =
WG=A3=A8mmusic=A3=A9
> work item=A3=BF

It should be considered, yes.

-d


>=20
> Regards~~~
>=20
> -Sujing Zhou
>=20
> "Dan Wing" <dwing@cisco.com> =D0=B4=D3=DA 2012-04-19 09:32:40:
>=20
> > > -----Original Message-----
> > > From: zhou.sujing@zte.com.cn [mailto:zhou.sujing@zte.com.cn]
> > > Sent: Wednesday, April 18, 2012 5:48 PM
> > > To: Dan Wing
> > > Cc: mmusic@ietf.org
> > > Subject: Re: RE: RE: RE: Hi, May I ask for your opinion on draft-
> zhou-
> > > mmusic-sdes-keymod-01?
> > >
> > >
> > >
> > > > >
> > > > > Generaly it is preferable the session key  between two peers
> be
> > > > > established with contribution from both peers,otherwise we =
will
> get
> > > > > into trouble
> > > > > as  SDES now in the scenarios of re-targetting and forking.
> > > > > Our 01 version actually suggests to change the unidirectional
> key
> > > > > transport in SDES into a key agreement(indicated by "keymod"):
> > > > > offerer provides: k1
> > > > > answer provides: keymod value
> > > > > the outgoing key from offerer to answerer is derived from k1
> and
> > > keymod
> > > > > value no matter in which situation.
> > > > > Re-targeting and forking  happen to be the scenarios that
> > > especially
> > > > > benefit from the change.
> > > >
> > > > Which involves the same number of (signaling) round-trips, =
right?
> > >
> > > In my opinion, the new method does not add extra round trips, it
> has
> > > the same round trips with
> > > the current SDES without re-INVITE or UPDATE.
> > >
> > > offerer-->answerer:INVITE
> > >        a=3Dcrypto:1 AES_CM_128_HMAC_SHA1_80
> > >         inline:d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj|2^20|1:32 =
-
> --
> > > >k1
> > >         keymod:rand|xor|
> > > offerer<--answerer:Response
> > >        a=3Dcrypto:1 AES_CM_128_HMAC_SHA1_32
> > >        inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj|2^20|1:32; =
-
> --
> > > >k2
> > >        keymod:rand|xor|WVNfX19zZW1jdGwgKCkgew=3D=3D         =
->keymod
> value
> > >
> > > after the single round,
> > >     k1 and keymod value-->k1' to protect session from offerer to
> > > answerer
> > >    k2 -->  to protect session from answerer   to offerer
> >
> > I now understand what you're proposing, thanks for explaining it =
this
> way.
> >
> > That avoids a signaling round trip, but does require the Offerer and
> > Answerer support keymod.  If either of them don't, the Offerer needs
> to
> > always do a re-Invite.  So this appears a reasonable optimization to
> avoid
> > always doing a re-Invite.
> >
> > -d
> >
> >
> >



From harald@alvestrand.no  Thu Apr 19 08:34:04 2012
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Date: Thu, 19 Apr 2012 17:34:01 +0200
From: Harald Alvestrand <harald@alvestrand.no>
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Subject: [MMUSIC] RFC 6263 - interpretation validation: "a=imageattr" is advisory, not mandatory
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Hello MMUSICers,

Roni Even and I just had a discussion on the rtcweb list, which led to 
an interesting conclusion with regard to the meaning of a=imageattr from 
RFC 6236.

My question (part of a thread) was:

>  I interpret the example in 4.2.1 (second alternative) as consisting of
>  4 messages; in an offer/answer exchange, this would look like:
>  
>  1) Alice ->  Bob: OFFER
>        a=imageattr:97 send [x=800,y=640,sar=1.1,q=0.6] [x=480,y=320] \
>                       recv [x=330,y=250]
>  2) Bob ->  Alice: ANSWER
>        a=imageattr:97 recv [x=800,y=640,sar=1.1] \
>                       send [x=[320:16:640],y=[240:16:480],par=[1.2-1.3]]
>  3) Alice ->  Bob: OFFER (re-invite, probably)
>        a=imageattr:97 send [x=800,y=640,sar=1.1] \
>                       recv [x=336,y=256]
>  4) Bob ->  Alice: ANSWER
>        a=imageattr:97 recv [x=800,y=640,sar=1.1] \
>                       send [x=336,y=256]
>  
>  So after message 2, Bob cannot send anything, since he can't reproduce
>  what Alice asked for, right?
>  
>  And after message 2, Alice can only send 800x640 with a sar of 1.1,
>  right? It's forbidden for Alice to send a noncompensated image, or a
>  640x480 image?
>  
>  Again, it's the use of the words "wishes" and "desires" that I want to
>  be sure I understand correctly - as used in RFC 6236, they both mean
>  "demand" / "require" / "MUST". Right?
>  

Roni's response was:
> This is a good question, my view will be that since Alice gave a preference
> like profile and level (and maybe bandwidth) in the original offer than Bob
> can send any video that will be within this level. The image attribute says
> that Alice prefers to get something specific to help with processing (note
> that Alice can limit the received video by using max-mbps and max-fs for
> H.264). It may be good to discuss it with others from MMUSIC.
> The usage of the image attribute in my view is more relevant to allow the
> receiver to ask for a specific image size since this is not defined in H.264
> (in H.263 you can specify the common specific resolutions and frame rate
> using the optional parameters in RFC 4629). This was a requirement to allow
> a receiver to ask for a specific size.
> The usage of "wish" was because H.264 encoder is not required to support
> scale to a specific resolution and a compliant decoder must support
> everything within the negotiated profile and level. So when a receiver asks
> for a size it can happen only if the encoder can do it but if the encoder
> cannot it is still compliant.
If this interpretation is correct, it means that the imageattr attribute is useful only for carrying "desired" (non-mandatory) constraints on the incoming video stream, and cannot be used to require a specific resolution or range of resolutions from the sender (mandatory constraints).

If RTCWEB wishes to have such mandatory constraints, it must propose a followup or extension to RFC 6236; it cannot simply refer to RFC 6236 for this.

Is this a correct interpretation?

                         Harald Alvestrand



From miguel.a.garcia@ericsson.com  Mon Apr 23 02:57:07 2012
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Subject: [MMUSIC] Minutes of MMUSIC WG meeting at IETF 83, Paris
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Hi,

The minutes of the MMUSIC WG meeting at IETF 83 (Paris), have been posted:

http://www.ietf.org/proceedings/83/minutes/minutes-83-mmusic.htm

Please review them and send comments and corrections.

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

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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
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Subject: [MMUSIC] FW: I-D Action: draft-muthu-mmusic-offer-answer-g723-g729-00.txt
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We have resubmitted the I-D clarifying the offer/answer considerations
for the Annex A flavor of G.723 and the Annex B flavors of G.729, G.729D
and G.729E, to the MMUSIC WG.=20

This clarification is missing in RFC4856, leading to interop issues,
like:
http://sipforum.org/pipermail/discussion/2008-January/004026.html

The draft was earlier submitted to the Payload WG and discussed there:
http://www.ietf.org/mail-archive/web/payload/current/msg00362.html

The draft was presented at IETF-83 in the AVTEXT session (as a Payload
item). Following instructions from the AD and chairs we are resubmitting
it to MMUSIC.

Comments welcome..

Muthu

-----Original Message-----
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[mailto:i-d-announce-bounces@ietf.org] On Behalf Of
internet-drafts@ietf.org
Sent: Tuesday, April 24, 2012 11:33 AM
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Subject: I-D Action: draft-muthu-mmusic-offer-answer-g723-g729-00.txt


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

	Title           : Offer/Answer Considerations for G.723 Annex A
and G.729 Annex B
	Author(s)       : Muthu Arul Mozhi Perumal
                          Parthasarathi Ravindran
	Filename        :
draft-muthu-mmusic-offer-answer-g723-g729-00.txt
	Pages           : 8
	Date            : 2012-04-23

   [RFC4856] describes the annexa parameter for G723 and the annexb
   parameter for G729, G729D and G729E. However, the specification does
   not describe the offerer and answerer behavior when the value of the
   annexa or annexb parameter does not match in the SDP offer and
   answer.  This document provides the offer/answer considerations for
   these parameters.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-muthu-mmusic-offer-answer-g723
-g729-00.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-muthu-mmusic-offer-answer-g723-
g729-00.txt

The IETF datatracker page for this Internet-Draft is:
https://datatracker.ietf.org/doc/draft-muthu-mmusic-offer-answer-g723-g7
29/

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From: "Yuepeiyu (Roy)" <yuepeiyu@huawei.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] WGLC of draft-ietf-mmusic-rfc2326bis-29.txt
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WG,


We have updated our individual draft for the maximum SSRC SDP signaling.
The main change is that the fallback policy has been changed from a
single SSRC to being application specific.

Feedback much appreciated.

Cheers

Magnus Westerlund


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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.

	Title           : Multiple Synchronization sources (SSRC) in RTP Session S=
ignaling
	Author(s)       : Magnus Westerlund
                          Bo Burman
                          Fredrik Jansson
	Filename        : draft-westerlund-avtcore-max-ssrc-01.txt
	Pages           : 13
	Date            : 2012-04-26

   RTP has always been a protocol that supports multiple participants
   each sending their own media streams in an RTP session.
   Unfortunately many implementations are designed only for point to
   point voice over IP with a single source in each end-point.  Even
   client implementations aimed at video conferences have often been
   built with the assumption around central mixers that only deliver a
   single media stream per media type.  Thus any application that wants
   to allow for more advance usage where multiple media streams are sent
   and received by an end-point has an issue with legacy
   implementations.  This document describes the problem and proposes a
   signalling solution for how to use multiple SSRCs within one RTP
   session and at the same time handle the legacy issues.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-westerlund-avtcore-max-ssrc-01.tx=
t

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-westerlund-avtcore-max-ssrc-01.txt

The IETF datatracker page for this Internet-Draft is:
https://datatracker.ietf.org/doc/draft-westerlund-avtcore-max-ssrc/

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From magnus.westerlund@ericsson.com  Thu Apr 26 08:56:39 2012
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Cc: Flemming Andreasen <fandreas@cisco.com>, "draft-ietf-mmusic-rfc2326bis.authors@tools.ietf.org" <draft-ietf-mmusic-rfc2326bis.authors@tools.ietf.org>, mmusic <mmusic@ietf.org>
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Hi,

I have to add a WG last call comment on my own document. This comes from
a discussion on the AVTCORE ECN for RTP document with Pete Resnick
regarding ABNF extension definitions and the qdtext definition of UTF-8.

Fist of all it doesn't provide any escaping of strings including " and
secondly its definition of UTF-8 and additional characters are in error.


>     quoted-string   =  ( DQ *qdtext DQ )
>     qdtext          =  %x20-21 / %x23-7E / %x80-FF / UTF8-NONASCII
>                        ; any UTF-8 TEXT except<">
>     DQ              =  %x22  ; US-ASCII double-quote mark (34)
>     UTF8-NONASCII    =  %xC0-DF 1UTF8-CONT
>                      /  %xE0-EF 2UTF8-CONT
>                      /  %xF0-F7 3UTF8-CONT
>                      /  %xF8-FB 4UTF8-CONT
>                      /  %xFC-FD 5UTF8-CONT
>     UTF8-CONT        =  %x80-BF
>
First of all, I don't think you should be including %x80-FF not in the
context of UTF-8. As you said, if you want binary content (or random
octets), you can always BASE64 the thing. This should be for strings,
and that means either US-ASCII or UTF-8 and nothing else.

I think you should incorporate definitions from 5234 (DQUOTE) and 3629
(UTF8-1, UTF8-2, UTF8-3, and UTF8-4) and simply make this:

quoted-string = ( DQUOTE *qdtext DQUOTE )
qdtext = %x20-21 / %x23-7E / UTF8-NONASCII
UTF8-NONASCII = UTF8-1 / UTF8-2 / UTF8-3 / UTF8-4

If you want to add the quoted-pair escaping mechanism, the simplest
would be:

quoted-string = ( DQUOTE *qdtext DQUOTE )
qdtext = %x20-21 / %x23-5B / %x5D-7E / quoted-pair / UTF8-NONASCII
    ; No DQUOTE and no "\"
quoted-pair = "\\" / ( "\" DQUOTE )
UTF8-NONASCII = UTF8-1 / UTF8-2 / UTF8-3 / UTF8-4

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 fluffy@iii.ca  Sun Apr 29 08:49:24 2012
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I went thought this pretty carefully from the point of view of could I =
implement it, does it work, does it meet the use cases I care about, and =
are the corner cases where it won't work. As bizarre as this seems, I =
have could not find any problems - I think this is ready to publish.=20

I did not do a careful check of grammar or stuff like that because 1) I =
suck at that at 2) the RFC Ed will get that. I also did not check the =
ABNF - someone should do that.=20

Look good,

Cullen



From christer.holmberg@ericsson.com  Mon Apr 30 05:27:30 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>
Date: Mon, 30 Apr 2012 14:27:25 +0200
Thread-Topic: [MMUSIC] Review of draft-ietf-mmusic-sdp-media-capabilities-13
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Hi,

Eventhough a very detailed mechanism, in the same way as Cullen I could not=
 come up with cases where the mechanism would not work.

I also took a detailed look at the ABNF, where I have some comments.

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


General:

SDP normally has a rather strict syntax. For example, elements are normally=
 separated explicitly with "SP". However, you use "1*WSP" in most places. I=
s there a reason for having a more relaxed syntax in media cap?


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




Section 3.3.2.1:



The section contains examples with a number of rmcap attributes:



       a=3Drmcap:1,2 audio G729/8000

       a=3Drmcap:1 video H263-1998/90000

       a=3Drmcap:2 video H263-2000/90000



However, I don't think these are valid according to the syntax in section 3=
.3.1, which does not define the media type ("audio", "video", etc):



               a=3Drmcap:<media-cap-num-list> <encoding-name>/<clock-rate> =
[/<encoding-parms>]





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





Section 3.3.8:



The example contains the following line:



               a=3Drmcap:51 *



However, I don't think that is valid according to the syntax in section 3.3=
.1, which mandates both encoding name and clock-rate:



               a=3Drmcap:<media-cap-num-list> <encoding-name>/<clock-rate> =
[/<encoding-parms>]





In addition, I find no definition of using "*" for the encoding-name (I ass=
ume it is some kind of wildcard).





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





Section 4.1:



See issue on 3.3.8.





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





a=3Domcap:



There is no example showing the a=3Domcap attribute. I think it would be go=
od to have one.





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





a=3Dsescap:



There is no example showing the usage of optional-configs. I think it would=
 be good to have one.





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


Regards,

Christer

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<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]--></head><body lang=3DEN-US link=3Dblue vli=
nk=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span lang=3DFI =
style=3D'font-size:12.0pt'>Hi,<o:p></o:p></span></p><p class=3DMsoNormal><s=
pan lang=3DFI style=3D'font-size:12.0pt'><o:p>&nbsp;</o:p></span></p><p cla=
ss=3DMsoNormal><span style=3D'font-size:12.0pt'>Eventhough a very detailed =
mechanism, in the same way as Cullen I could not come up with cases where t=
he mechanism would not work.<o:p></o:p></span></p><p class=3DMsoNormal><spa=
n style=3D'font-size:12.0pt'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNorm=
al><span style=3D'font-size:12.0pt'>I also took a detailed look at the ABNF=
, where I have some comments.<o:p></o:p></span></p><p class=3DMsoNormal><sp=
an style=3D'font-size:12.0pt'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNor=
mal><span style=3D'font-size:12.0pt'>-----------------<o:p></o:p></span></p=
><p class=3DMsoNormal><span style=3D'font-size:12.0pt'><o:p>&nbsp;</o:p></s=
pan></p><p class=3DMsoNormal><span style=3D'font-size:12.0pt'><o:p>&nbsp;</=
o:p></span></p><p class=3DMsoNormal><span style=3D'font-size:12.0pt'>Genera=
l:<o:p></o:p></span></p><p class=3DMsoNormal><span style=3D'font-size:12.0p=
t'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span style=3D'font-siz=
e:12.0pt'>SDP normally has a rather strict syntax. For example, elements ar=
e normally separated explicitly with &#8220;SP&#8221;. However, you use &#8=
220;1*WSP&#8221; in most places. Is there a reason for having a more relaxe=
d syntax in media cap?<o:p></o:p></span></p><p class=3DMsoNormal><span styl=
e=3D'font-size:12.0pt'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><sp=
an style=3D'font-size:12.0pt'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNor=
mal><span style=3D'font-size:12.0pt'>----------------<o:p></o:p></span></p>=
<p class=3DMsoNormal><span style=3D'font-size:12.0pt'><o:p>&nbsp;</o:p></sp=
an></p><pre><span style=3D'font-size:12.0pt;font-family:"Calibri","sans-ser=
if"'><o:p>&nbsp;</o:p></span></pre><pre><span style=3D'font-size:12.0pt;fon=
t-family:"Calibri","sans-serif"'>Section 3.3.2.1:<o:p></o:p></span></pre><p=
re><span style=3D'font-size:12.0pt;font-family:"Calibri","sans-serif"'><o:p=
>&nbsp;</o:p></span></pre><pre><span style=3D'font-size:12.0pt;font-family:=
"Calibri","sans-serif"'>The section contains examples with a number of rmca=
p attributes:<o:p></o:p></span></pre><pre><span style=3D'font-size:12.0pt;f=
ont-family:"Calibri","sans-serif"'><o:p>&nbsp;</o:p></span></pre><pre><span=
 style=3D'font-size:12.0pt;font-family:"Calibri","sans-serif"'>&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp; a=3Drmcap:1,2 audio G729/8000<o:p></o:p></span></p=
re><pre><span style=3D'font-size:12.0pt;font-family:"Calibri","sans-serif"'=
>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=3Drmcap:1 video H263-1998/90000<o:p=
></o:p></span></pre><pre><span style=3D'font-size:12.0pt;font-family:"Calib=
ri","sans-serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=3Drmcap:2 video H2=
63-2000/90000<o:p></o:p></span></pre><pre><span style=3D'font-size:12.0pt;f=
ont-family:"Calibri","sans-serif"'><o:p>&nbsp;</o:p></span></pre><pre><span=
 style=3D'font-size:12.0pt;font-family:"Calibri","sans-serif"'>However, I d=
on&#8217;t think these are valid according to the syntax in section 3.3.1, =
which does not define the media type (&#8220;audio&#8221;, &#8220;video&#82=
21;, etc):<o:p></o:p></span></pre><pre><span style=3D'font-size:12.0pt;font=
-family:"Calibri","sans-serif"'><o:p>&nbsp;</o:p></span></pre><pre><span st=
yle=3D'font-size:12.0pt;font-family:"Calibri","sans-serif"'>&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=3Dr=
mcap:&lt;media-cap-num-list&gt; &lt;encoding-name&gt;/&lt;clock-rate&gt; [/=
&lt;encoding-parms&gt;]<o:p></o:p></span></pre><pre><span style=3D'font-siz=
e:12.0pt;font-family:"Calibri","sans-serif"'><o:p>&nbsp;</o:p></span></pre>=
<pre><span style=3D'font-size:12.0pt;font-family:"Calibri","sans-serif"'><o=
:p>&nbsp;</o:p></span></pre><pre><span style=3D'font-size:12.0pt;font-famil=
y:"Calibri","sans-serif"'>---------------<o:p></o:p></span></pre><pre><span=
 style=3D'font-size:12.0pt;font-family:"Calibri","sans-serif"'><o:p>&nbsp;<=
/o:p></span></pre><pre><span style=3D'font-size:12.0pt;font-family:"Calibri=
","sans-serif"'><o:p>&nbsp;</o:p></span></pre><pre><span style=3D'font-size=
:12.0pt;font-family:"Calibri","sans-serif"'>Section 3.3.8:<o:p></o:p></span=
></pre><pre><span style=3D'font-size:12.0pt;font-family:"Calibri","sans-ser=
if"'><o:p>&nbsp;</o:p></span></pre><pre><span style=3D'font-size:12.0pt;fon=
t-family:"Calibri","sans-serif"'>The example contains the following line:<o=
:p></o:p></span></pre><pre><span style=3D'font-size:12.0pt;font-family:"Cal=
ibri","sans-serif"'><o:p>&nbsp;</o:p></span></pre><pre><span style=3D'font-=
size:12.0pt;font-family:"Calibri","sans-serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=3Drmcap:51 *<o:=
p></o:p></span></pre><pre><span style=3D'font-size:12.0pt;font-family:"Cali=
bri","sans-serif"'><o:p>&nbsp;</o:p></span></pre><pre><span style=3D'font-s=
ize:12.0pt;font-family:"Calibri","sans-serif"'>However, I don&#8217;t think=
 that is valid according to the syntax in section 3.3.1, which mandates bot=
h encoding name and clock-rate:<o:p></o:p></span></pre><pre><span style=3D'=
font-size:12.0pt;font-family:"Calibri","sans-serif"'><o:p>&nbsp;</o:p></spa=
n></pre><pre><span style=3D'font-size:12.0pt;font-family:"Calibri","sans-se=
rif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp; a=3Drmcap:&lt;media-cap-num-list&gt; &lt;encoding-name&gt;/=
&lt;clock-rate&gt; [/&lt;encoding-parms&gt;]<o:p></o:p></span></pre><pre><s=
pan style=3D'font-size:12.0pt;font-family:"Calibri","sans-serif"'><o:p>&nbs=
p;</o:p></span></pre><pre><span style=3D'font-size:12.0pt;font-family:"Cali=
bri","sans-serif"'><o:p>&nbsp;</o:p></span></pre><pre><span style=3D'font-s=
ize:12.0pt;font-family:"Calibri","sans-serif"'>In addition, I find no defin=
ition of using &#8220;*&#8221; for the encoding-name (I assume it is some k=
ind of wildcard).<o:p></o:p></span></pre><pre><span style=3D'font-size:12.0=
pt;font-family:"Calibri","sans-serif"'><o:p>&nbsp;</o:p></span></pre><pre><=
span style=3D'font-size:12.0pt;font-family:"Calibri","sans-serif"'><o:p>&nb=
sp;</o:p></span></pre><pre><span style=3D'font-size:12.0pt;font-family:"Cal=
ibri","sans-serif"'>---------------<o:p></o:p></span></pre><pre><span style=
=3D'font-size:12.0pt;font-family:"Calibri","sans-serif"'><o:p>&nbsp;</o:p><=
/span></pre><pre><span style=3D'font-size:12.0pt;font-family:"Calibri","san=
s-serif"'><o:p>&nbsp;</o:p></span></pre><pre><span style=3D'font-size:12.0p=
t;font-family:"Calibri","sans-serif"'>Section 4.1:<o:p></o:p></span></pre><=
pre><span style=3D'font-size:12.0pt;font-family:"Calibri","sans-serif"'><o:=
p>&nbsp;</o:p></span></pre><pre><span style=3D'font-size:12.0pt;font-family=
:"Calibri","sans-serif"'>See issue on 3.3.8.<o:p></o:p></span></pre><pre><s=
pan style=3D'font-size:12.0pt;font-family:"Calibri","sans-serif"'><o:p>&nbs=
p;</o:p></span></pre><pre><span style=3D'font-size:12.0pt;font-family:"Cali=
bri","sans-serif"'><o:p>&nbsp;</o:p></span></pre><pre><span style=3D'font-s=
ize:12.0pt;font-family:"Calibri","sans-serif"'>---------------<o:p></o:p></=
span></pre><pre><span style=3D'font-size:12.0pt;font-family:"Calibri","sans=
-serif"'><o:p>&nbsp;</o:p></span></pre><pre><span style=3D'font-size:12.0pt=
;font-family:"Calibri","sans-serif"'><o:p>&nbsp;</o:p></span></pre><pre><sp=
an style=3D'font-size:12.0pt;font-family:"Calibri","sans-serif"'>a=3Domcap:=
<o:p></o:p></span></pre><pre><span style=3D'font-size:12.0pt;font-family:"C=
alibri","sans-serif"'><o:p>&nbsp;</o:p></span></pre><pre><span style=3D'fon=
t-size:12.0pt;font-family:"Calibri","sans-serif"'>There is no example showi=
ng the a=3Domcap attribute. I think it would be good to have one.<o:p></o:p=
></span></pre><pre><span style=3D'font-size:12.0pt;font-family:"Calibri","s=
ans-serif"'><o:p>&nbsp;</o:p></span></pre><pre><span style=3D'font-size:12.=
0pt;font-family:"Calibri","sans-serif"'><o:p>&nbsp;</o:p></span></pre><pre>=
<span style=3D'font-size:12.0pt;font-family:"Calibri","sans-serif"'>-------=
--------<o:p></o:p></span></pre><pre><span style=3D'font-size:12.0pt;font-f=
amily:"Calibri","sans-serif"'><o:p>&nbsp;</o:p></span></pre><pre><span styl=
e=3D'font-size:12.0pt;font-family:"Calibri","sans-serif"'><o:p>&nbsp;</o:p>=
</span></pre><pre><span style=3D'font-size:12.0pt;font-family:"Calibri","sa=
ns-serif"'>a=3Dsescap:<o:p></o:p></span></pre><pre><span style=3D'font-size=
:12.0pt;font-family:"Calibri","sans-serif"'><o:p>&nbsp;</o:p></span></pre><=
pre><span style=3D'font-size:12.0pt;font-family:"Calibri","sans-serif"'>The=
re is no example showing the usage of optional-configs. I think it would be=
 good to have one.<o:p></o:p></span></pre><pre><span style=3D'font-size:12.=
0pt;font-family:"Calibri","sans-serif"'><o:p>&nbsp;</o:p></span></pre><pre>=
<span style=3D'font-size:12.0pt;font-family:"Calibri","sans-serif"'><o:p>&n=
bsp;</o:p></span></pre><pre><span style=3D'font-size:12.0pt;font-family:"Ca=
libri","sans-serif"'>---------------<o:p></o:p></span></pre><pre><span styl=
e=3D'font-size:12.0pt;font-family:"Calibri","sans-serif"'><o:p>&nbsp;</o:p>=
</span></pre><p class=3DMsoNormal><span style=3D'font-size:12.0pt'>Regards,=
<o:p></o:p></span></p><p class=3DMsoNormal><span style=3D'font-size:12.0pt'=
><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span style=3D'font-size:=
12.0pt'>Christer</span><span style=3D'font-size:12.0pt'><o:p></o:p></span><=
/p></div></body></html>=

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From: "Pedro Capelastegui" <capelastegui@dit.upm.es>
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Date: Mon, 30 Apr 2012 18:57:17 +0200
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Subject: [MMUSIC] Submission of draft-capelastegui-mmusic-3dv-sdp-00
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Hi all,

 

I have submitted an individual draft to address the description of 3D video
streams in SDP. The document is intended as an alternative to
draft-ietf-mmusic-signal-3d-format-00, using a different mechanism based on
decoding dependencies to indicate the relationship between different streams
in a 3D video stream.

 

The document can be found at 

https://datatracker.ietf.org/doc/draft-capelastegui-mmusic-3dv-sdp/

 

Comments are welcome.

 

Regards,

Pedro


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</o:shapelayout></xml><![endif]--></head><body lang=3DEN-US link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal>Hi =
all,<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>I have submitted an individual draft to address the =
description of 3D video streams in SDP. The document is intended as an =
alternative to draft-ietf-mmusic-signal-3d-format-00, using a different =
mechanism based on decoding dependencies to indicate the relationship =
between different streams in a 3D video stream.<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>The document =
can be found at <o:p></o:p></p><p class=3DMsoNormal><a =
href=3D"https://datatracker.ietf.org/doc/draft-capelastegui-mmusic-3dv-sd=
p/">https://datatracker.ietf.org/doc/draft-capelastegui-mmusic-3dv-sdp/</=
a><o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>Comments are welcome.<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>Regards,<o:p></o:p></p><p =
class=3DMsoNormal>Pedro<o:p></o:p></p></div></body></html>
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