
From pkyzivat@alum.mit.edu  Thu Nov  1 14:16:44 2012
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Date: Thu, 01 Nov 2012 17:16:39 -0400
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: [MMUSIC] Comments on draft-holmberg-mmusic-sdp-mmt-negotiation-00
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Here are a number of comments on 
draft-holmberg-mmusic-sdp-mmt-negotiation-00, written as I prepare for 
the meeting.

Section 5.2.2.1:

I understand why an offerer would typically *want* to group one or more 
media descriptions for specific media types with an "anymedia" media 
description. But why MUST that be done? Why can't an offerer just offer 
the anymedia description if he isn't interested in backward compatibility?

Section 6.1

It is unclear to me whether the <media> value "anymedia" that is defined 
he is legitimate according to 4566. Section 8.2.1 of 4566 says:

    The set of media types is intended to be small and SHOULD NOT be
    extended except under rare circumstances.  The same rules should
    apply for media names as for top-level media content types, and where
    possible the same name should be registered for SDP as for MIME.  For
    media other than existing top-level media content types, a Standards
    Track RFC MUST be produced for a new top-level content type to be
    registered, and the registration MUST provide good justification why
    no existing media name is appropriate (the "Standards Action" policy
    of RFC 2434 [8].

The above suggests that the name needs to be a top level media content 
type, except for the "where possible". (And it's unclear what sort of 
exceptions "where possible" was intended to cover. The above uses the 
iana registry for "Media Types", which is shared with MIME usage. And 
from my quick scan it is not evident how to register a new top level type.

So, I think it might actually require a revision to 4566 to make this 
change - breaking this case out and exempting it from the registration 
process. If so then I think it would be helpful to choose something more 
distinctive than "anymedia" to denote this quite different case. I 
suggest "*". (This is syntactically valid in the <media> field according 
to the existing 4566 syntax, so it *ought* to be backward compatible.)

Section 7 - mmtype attribute:

Note that the use of payload types in m-lines is only for those with one 
of the RTP proto values. For UDP the media subtype is placed in the 
<fmt> field of the m-line rather than in rtpmap.

But with some rewording and explanation what you propose should work in 
most practical cases using the <fmt> value directly. E.g.

   a=mmtype: PCMU/8000 audio
   a=mmtype: H261/90000 video

The only time it wouldn't work is if you wanted to use a common subtype 
for two top level (e.g. both audio/foo and video/foo). That isn't likely.

(I'm having a separate discussion about how the new SCTP... <proto> 
values might describe the assignment of SCTP streams for particular 
purposes. At some point we *might* end up having a way to specify use of 
RTP over an SCTP stream. That then might start to interact with this if 
we wanted to mix audio/video in one such stream. But maybe it is 
premature to deal with that.)

Section 7.3.2:

I think you need to say *something* about O/A. I guess what makes sense 
is that 7.3.1 applies independently to the offer and the answer.

Section 10:

*Something* needs to be done about "anymedia" or whatever replaces it. 
As currently written you would need to register a new media type value. 
With my suggestions you might avoid that but have to identify this draft 
as an extension to 4566.

Unfortunately it is a bad time to revise 4566, since there is a 4566bis 
pending. Conceivably this whole thing could be added in to 4566bis, if 
the schedule works for that.

	Thanks,
	Paul

From pkyzivat@alum.mit.edu  Thu Nov  1 16:04:04 2012
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Date: Thu, 01 Nov 2012 19:04:00 -0400
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: [MMUSIC] Comments on draft-westerlund-mmusic-max-ssrc-00
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I just reviewed this and have some comments:

In Section 4.4 on o/a usage, *an* offerer must use this mechanism in the 
*initial* offer, while *the* answerer must use the mechanism in *each* 
subsequent *offer* and *answer*.

IIUC this means the initial offerer on the initial o/a of a session 
might use the mechanism on the initial offer and never again, while the 
answerer of the initial o/a has to use it on every offer and answer.

This is very asymmetric, and I suspect not what you intend.

I *think* what you intend is that a supporter generating an initial 
offer in a session must use the mechanism. Then, if the answer doesn't 
use the mechanism the offerer should cease using it in subsequent o/a 
exchanges in the session.

OTOH, if the initial answer uses the mechanism then it should be used by 
both ends for all subsequent o/a exchanges.

Is that what you intended?

That makes more sense. But this won't necessarily work right if there is 
a B2BUA that passes SDP through unchanged, and then at some point 
replaces a one session with another on one side of the B2BUA. (I.e. a 
3pcc transfer.) If the other end of the new session differs in support 
for this draft from the replaced endpoint it won't be properly 
negotiated for the new session.

This can be solved by the B2BUA editing the SDP, but that is a high 
burden for some B2BUAs.

ISTM that there is no real harm in an end that supports this *always* 
including it in offers.

A smaller point:

Is there any reason not to also allow this as a session level attribute, 
applying to all the media descriptions that don't have their own?

It could be convenient to just put the following at session level:

a=max-send-ssrc:{*:1}
a=max-recv-ssrc:{*:1}

And finally, section 5 has an example with the following which must be 
wrong because it is missing a ":":

a=max-send-ssrc:{* 1}

	Thanks,
	Paul

From pkyzivat@alum.mit.edu  Thu Nov  1 17:04:36 2012
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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: [MMUSIC] Comments on draft-ietf-mmusic-traffic-class-for-sdp-02
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I have some comments on this new version of the document. (Sorry!)

Section 3:

I know we have been through this many times, and its much better now. 
But I still see one issue:

The ABNF says an adjective is a tcl-token, which allows "-" and 
alphanumerics. But then the text says that an adjective and be preceded 
by underscore ("_") to indicate that it is non-standard. This means you 
are *extending* the syntax in text beyond what the ABNF allows. IMO that 
is a very bad idea.

Instead, I recommend altering the syntax to permit the underscore in 
adjectives in the way you intend it to be used. I would suggest some 
syntax for this but I don't know if you want to permit non-standard 
qualified adjectives.

Also, as we have previously discussed, this way of using non-standard 
adjectives is quite analogous to the use "P-" or "X-" headers that has 
been found troublesome and generally banned. So I expect this might have 
trouble getting through IESG review. I thought we had discussed having 
the non-standard adjectives be registered using a 
first-come-first-served registration policy.

ISTM that the portion of this section that discusses "aq" ought to be in 
its own section, and then referenced from the iana considerations section.

Section 6.4:

I don't find any explanation of how or if to register qualified 
adjectives. The section title says "Unqualified Adjective Registration", 
but the inital set of registered values includes "aq" (the qualifier 
part of qualified adjectives), and some values intended to be qualfied 
by "aq:" (such as "admitted"). This certainly implies to me that I can 
use each of these as an unqualified adjective.

If the intent is to register such things then I would expect to see the 
full qualified adjectives listed for registration. E.g. "aq:admitted"

	Thanks,
	Paul

From zhengyli@cisco.com  Fri Nov  2 02:31:28 2012
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From: "Rockson Li (zhengyli)" <zhengyli@cisco.com>
To: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>, "Miguel A. Garcia" <Miguel.A.Garcia@ericsson.com>, mmusic <mmusic@ietf.org>
Thread-Topic: [MMUSIC] WG Poll for consensus on WG item: latching draft
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Subject: Re: [MMUSIC] WG Poll for consensus on WG item: latching draft
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+1

On 10/18/12 8:09 PM, "Muthu Arul Mozhi Perumal (mperumal)"
<mperumal@cisco.com> wrote:

>I support it. I think it is quite informative.
>
>Muthu
>
>|-----Original Message-----
>|From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf
>Of Miguel A. Garcia
>|Sent: Thursday, October 18, 2012 2:13 PM
>|To: mmusic
>|Cc: Flemming Andreasen (fandreas); Dan Wing (dwing)
>|Subject: [MMUSIC] WG Poll for consensus on WG item: latching draft
>|
>|At the last IETF meeting we had a presentation of the following draft:
>|
>|http://datatracker.ietf.org/doc/draft-ivov-mmusic-latching/
>|
>|At that time, there was interest in adopting this document as WG item. We
>|currently have a milestone to submit an informational document describing
>|current practices in hosted NAT traversal, where this document fits.
>|
>|We would like to poll the working group to reconfirm the consensus on
>|adopting the above mentioned draft as a working group item.
>|
>|If you support this work or have problems with the adoption of this
>|document as WG item, please indicate it by replying to this e-mail within
>|the next 5 days.
>|
>|     Miguel and Flemming (co-chairs)
>|--
>|Miguel A. Garcia
>|+34-91-339-3608
>|Ericsson Spain
>|_______________________________________________
>|mmusic mailing list
>|mmusic@ietf.org
>|https://www.ietf.org/mailman/listinfo/mmusic
>_______________________________________________
>mmusic mailing list
>mmusic@ietf.org
>https://www.ietf.org/mailman/listinfo/mmusic


From harald@alvestrand.no  Fri Nov  2 03:35:19 2012
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Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-02.txt
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On 10/24/2012 08:05 AM, Salvatore Loreto wrote:
> Hi there,
>
> I have updated the draft trying to capture the result of the 
> discussion we had in this mailing list
> writing down for SCTP a <fmt> syntax similar to that used for RTP,
> i.e. with "channle numbers" being used in a way analogous to payload 
> type numbers.
FWIW, I think locking ourselves into this scheme is a Very Bad Idea. It 
will make it impossible to do many of the novel and interesting 
applications envisaged by WebRTC participants.

If you want to have the ability to declare some scoping for SCTP usage, 
I think we should at least:

- Extend the "a=streams" parameter to take a range, including "*" for 
"unlimited".
a=streams:1-* would mean that one expects to use at least 1 stream, but 
no upper limit.
a=streams:0-255 would mean that one may or may not use streams, but 
can't support more than 255.

- Extend the "a=datachannel" syntax to take a default notation, and not 
limit other fields.
a=datachannel:* RTCWeb would then mean that all datachannels not 
explicitly specified will use the RTCWeb protocol, and label and other 
options will be passed using the RTCWeb initialization protocol.

I would like to just hardcode in my browser that it always sends

m=application 1234 DTLS/SCTP 0
a=streams:0-*
a=datachannel:* RTCWeb

and really have SDP O/A stay out of the way of datachannel setup. I 
don't see the need to do more.
But others may have other needs.

>
> The proposal is a very first attempt to see if this is the direction 
> we want to go
> and it is obviously incomplete lacking the possibility to define
> all the properties of the specific channel...
>
> feedback and comments are very welcome and necessary at this point
>
> best regards
> Salvatore
>
>
>
>
> On 10/22/12 10:07 PM, internet-drafts@ietf.org wrote:
>> A New Internet-Draft is available from the on-line Internet-Drafts 
>> directories.
>>   This draft is a work item of the Multiparty Multimedia Session 
>> Control Working Group of the IETF.
>>
>>     Title           : Stream Control Transmission Protocol 
>> (SCTP)-Based Media Transport in the Session Description Protocol (SDP)
>>     Author(s)       : Salvatore Loreto
>>                            Gonzalo Camarillo
>>     Filename        : draft-ietf-mmusic-sctp-sdp-02.txt
>>     Pages           : 13
>>     Date            : 2012-10-22
>>
>> Abstract:
>>     SCTP (Stream Control Transmission Protocol) is a transport protocol
>>     used to establish associations between two endpoints.  This document
>>     describes how to express media transport over SCTP in SDP (Session
>>     Description Protocol).  This document defines the 'SCTP', 
>> 'SCTP/DTLS'
>>     and 'DTLS/SCTP' protocol identifiers for SDP.
>>
>>
>> The IETF datatracker status page for this draft is:
>> https://datatracker.ietf.org/doc/draft-ietf-mmusic-sctp-sdp
>>
>> There's also a htmlized version available at:
>> http://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-02
>>
>> A diff from the previous version is available at:
>> http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-sctp-sdp-02
>>
>>
>> Internet-Drafts are also available by anonymous FTP at:
>> ftp://ftp.ietf.org/internet-drafts/
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From harald@alvestrand.no  Fri Nov  2 03:39:11 2012
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Subject: [MMUSIC] SCTP question: Where does it multiplex?
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As currently specified, the DTLS/SCTP protocol is carefully specified in 
such a way that it can ride on the same 5-tuple as an SRTP session. I'd 
like to keep that property when we SDP-negotiate it.

The proposed syntax for DTLS/SCTP seems to be

m=<something, likely application> <port number> DTLS/SCTP <fmt data, 
mostly meaningless>

I see two ways to define this as a multiplex:

- Give the same port number as another m= line in the same SDP session
- Give a meaningless port number, and use the a=group:BUNDLE (of 
whatever flavor) to give which port number it's actually multiplexed 
together with.

The best way to do this may be obvious after the BUNDLE discussion, but 
in case there are other considerations .... are there thoughts?

                Harald


From pkyzivat@alum.mit.edu  Fri Nov  2 07:42:28 2012
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Harald,

(Speaking as co-chair of CLUE)

In clue we have some goals that overlap with this work and with rtcweb.

For one thing, we want to ensure that an webrtc client browser can act 
as a clue client. That means that the things that one must do to act as 
a clue client should be things that can be done from a browser using 
webrtc APIs and rtcweb protocols.

But of course we also expect clue clients that are *not* webrtc clients, 
that are built on a native sip infrastructure. And so we require that 
all combinations of native and webrtc clients to interoperate.

Also, in clue we need a single reliable bi-directional data channel for 
clue-specific signaling. We have mostly been considering the same 
SDP/DTLS/UDP transport that rtcweb is, partly because you guys have 
already done most of the work of defining it, and partly because we hope 
that means the APIs you define for it will mean that a browser based 
clue client will be able to implement that part of clue without needing 
any new clue-specific APIs. (Note that we only need *one* channel over 
the transport for this purpose. We don't care which one, but of course 
we do care that the two ends agree on what one it is.)

(The following point is more speculative - we haven't really considered 
it formally in clue, though there have been a few passing discussions 
here and there. And so this part is personal, not as chair.)

It is also the case that a clue client (either flavor) is likely to have 
need to simultaneously implement BFCP. This isn't really special to 
clue. When webrtc is used to hook a browser into any sip conference it 
may be necessary to use BFCP to access the advanced features of the 
conference.

It would be *nice* if the clue protocol and the bfcp protocol could 
coexist over a single SCTP transport, each taking its own channel. That 
would also make it possible for a webrtc cient to implement bfcp using 
the webrtc datachannel APIs. (If bfcp is run over TCP or UDP then AFAIK 
a webrtc client would not be able to support it.)

Of course there is currently no definition for bfcp over SCPT, and the 
work is still wrapping up to define it over UDP. So this is more wishful 
thinking on my part than anything else.

*If* we had both a clue protocol binding and a bfcp binding to SCTP data 
channels, then there would be a need for a way to set up a session that 
contains an SCTP transport m-line, and the channels to be used for clue 
and/or bfcp. This would be needed both natively, and with webrtc/rtcweb. 
The stuff that Salvatore is proposing in the sctp-sdp draft will 
hopefully support that.

(End of personal speculative part.)

*** Note: I am *not* arguing that channels may only be used if they have 
been negotiated in SDP. I think we should be able to work it out so that 
some channels are reserved in SDP and others can be used based on 
implicit agreement and/or by dynamic run-time negotiation. Obviously 
some work is required to make those things coexist, but that isn't 
rocket science.

What I want to avoid is the need for clue to support dynamic assignment 
of the channels it needs using some rtcweb-specific channel assignment 
protocol. Nor would I like to see some standardized static assignment of 
channels to protocols. (E.g. BFCP always uses channel 10 and CLUE always 
uses channel 11.)

AFAICT, the alternative to the above is that a webrtc clue client would 
be forced to run "its end" of the clue and bfcp protocols in the web 
server rather than the browser, and then exchange that information with 
the browser via translation to html or to a separate rtcweb data 
channel. While this is *possible*, it seems less than ideal.

Also, even ignoring rtcweb compatibility, when running native sip, I 
think we will want a way, in SDP, to negotiate one SCTP channel for the 
clue protocol and maybe another for bfcp (if that becomes supported).

	Thanks,
	Paul

On 11/2/12 6:35 AM, Harald Alvestrand wrote:
> On 10/24/2012 08:05 AM, Salvatore Loreto wrote:
>> Hi there,
>>
>> I have updated the draft trying to capture the result of the
>> discussion we had in this mailing list
>> writing down for SCTP a <fmt> syntax similar to that used for RTP,
>> i.e. with "channle numbers" being used in a way analogous to payload
>> type numbers.
> FWIW, I think locking ourselves into this scheme is a Very Bad Idea. It
> will make it impossible to do many of the novel and interesting
> applications envisaged by WebRTC participants.
>
> If you want to have the ability to declare some scoping for SCTP usage,
> I think we should at least:
>
> - Extend the "a=streams" parameter to take a range, including "*" for
> "unlimited".
> a=streams:1-* would mean that one expects to use at least 1 stream, but
> no upper limit.
> a=streams:0-255 would mean that one may or may not use streams, but
> can't support more than 255.
>
> - Extend the "a=datachannel" syntax to take a default notation, and not
> limit other fields.
> a=datachannel:* RTCWeb would then mean that all datachannels not
> explicitly specified will use the RTCWeb protocol, and label and other
> options will be passed using the RTCWeb initialization protocol.
>
> I would like to just hardcode in my browser that it always sends
>
> m=application 1234 DTLS/SCTP 0
> a=streams:0-*
> a=datachannel:* RTCWeb
>
> and really have SDP O/A stay out of the way of datachannel setup. I
> don't see the need to do more.
> But others may have other needs.
>
>>
>> The proposal is a very first attempt to see if this is the direction
>> we want to go
>> and it is obviously incomplete lacking the possibility to define
>> all the properties of the specific channel...
>>
>> feedback and comments are very welcome and necessary at this point
>>
>> best regards
>> Salvatore
>>
>>
>>
>>
>> On 10/22/12 10:07 PM, internet-drafts@ietf.org wrote:
>>> A New Internet-Draft is available from the on-line Internet-Drafts
>>> directories.
>>>   This draft is a work item of the Multiparty Multimedia Session
>>> Control Working Group of the IETF.
>>>
>>>     Title           : Stream Control Transmission Protocol
>>> (SCTP)-Based Media Transport in the Session Description Protocol (SDP)
>>>     Author(s)       : Salvatore Loreto
>>>                            Gonzalo Camarillo
>>>     Filename        : draft-ietf-mmusic-sctp-sdp-02.txt
>>>     Pages           : 13
>>>     Date            : 2012-10-22
>>>
>>> Abstract:
>>>     SCTP (Stream Control Transmission Protocol) is a transport protocol
>>>     used to establish associations between two endpoints.  This document
>>>     describes how to express media transport over SCTP in SDP (Session
>>>     Description Protocol).  This document defines the 'SCTP',
>>> 'SCTP/DTLS'
>>>     and 'DTLS/SCTP' protocol identifiers for SDP.
>>>
>>>
>>> The IETF datatracker status page for this draft is:
>>> https://datatracker.ietf.org/doc/draft-ietf-mmusic-sctp-sdp
>>>
>>> There's also a htmlized version available at:
>>> http://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-02
>>>
>>> A diff from the previous version is available at:
>>> http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-sctp-sdp-02
>>>
>>>
>>> Internet-Drafts are also available by anonymous FTP at:
>>> ftp://ftp.ietf.org/internet-drafts/
>>>
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


From pkyzivat@alum.mit.edu  Fri Nov  2 07:53:32 2012
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On 11/2/12 6:39 AM, Harald Alvestrand wrote:
> As currently specified, the DTLS/SCTP protocol is carefully specified in
> such a way that it can ride on the same 5-tuple as an SRTP session. I'd
> like to keep that property when we SDP-negotiate it.

It is? How? Where is that specified?

	Thanks,
	Paul

> The proposed syntax for DTLS/SCTP seems to be
>
> m=<something, likely application> <port number> DTLS/SCTP <fmt data,
> mostly meaningless>
>
> I see two ways to define this as a multiplex:
>
> - Give the same port number as another m= line in the same SDP session
> - Give a meaningless port number, and use the a=group:BUNDLE (of
> whatever flavor) to give which port number it's actually multiplexed
> together with.
>
> The best way to do this may be obvious after the BUNDLE discussion, but
> in case there are other considerations .... are there thoughts?
>
>                 Harald
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


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On Oct 28, 2012, at 4:19 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> I support this - I think it is desirable to have this behavior =
documented, since behavior similar to this is fairly widespread. =
Hopefully having it documented will help prevent gratuitous variation.

+1, with a bit of trepidation. But Paul has the right of it.



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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, IETF MMUSIC WG <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Comments on draft-ietf-mmusic-traffic-class-for-sdp-02
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Please see inline.

> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Paul Kyzivat
> Sent: Thursday, November 01, 2012 8:05 PM
> To: IETF MMUSIC WG
> Subject: [MMUSIC] Comments on draft-ietf-mmusic-traffic-class-for-sdp-02
>=20
> I have some comments on this new version of the document. (Sorry!)
>=20
> Section 3:
>=20
> I know we have been through this many times, and its much better now.
> But I still see one issue:
>=20
> The ABNF says an adjective is a tcl-token, which allows "-" and
> alphanumerics. But then the text says that an adjective and be preceded
> by underscore ("_") to indicate that it is non-standard. This means you
> are *extending* the syntax in text beyond what the ABNF allows. IMO that
> is a very bad idea.
>=20
> Instead, I recommend altering the syntax to permit the underscore in
> adjectives in the way you intend it to be used. I would suggest some
> syntax for this but I don't know if you want to permit non-standard
> qualified adjectives.

I'd like to recommend removing the text regarding the '_' and leaving the A=
BNF as is. At IETF 84 I thought we agreed to address this by lowering the b=
ar to register a new adjective.

Cheers,
Charles

> Also, as we have previously discussed, this way of using non-standard
> adjectives is quite analogous to the use "P-" or "X-" headers that has
> been found troublesome and generally banned. So I expect this might have
> trouble getting through IESG review. I thought we had discussed having
> the non-standard adjectives be registered using a
> first-come-first-served registration policy.
>=20
> ISTM that the portion of this section that discusses "aq" ought to be in
> its own section, and then referenced from the iana considerations section=
.
>=20
> Section 6.4:
>=20
> I don't find any explanation of how or if to register qualified
> adjectives. The section title says "Unqualified Adjective Registration",
> but the inital set of registered values includes "aq" (the qualifier
> part of qualified adjectives), and some values intended to be qualfied
> by "aq:" (such as "admitted"). This certainly implies to me that I can
> use each of these as an unqualified adjective.
>=20
> If the intent is to register such things then I would expect to see the
> full qualified adjectives listed for registration. E.g. "aq:admitted"
>=20
> 	Thanks,
> 	Paul
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

From miguel.a.garcia@ericsson.com  Sat Nov  3 02:37:35 2012
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Subject: Re: [MMUSIC] WG Poll for consensus on WG item: latching draft
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This is to close the WG poll for consensus on the latching draft, as a WG 
item.

We have received a few indications of this document being of interest for 
the working group to document this current practice. We have received no 
objections.

With this, we can accept this document as WG item. Please authors, submit 
a new version as WG item (draft-ietf-mmusic-...) as early as possible.

/Miguel

On 18/10/2012 10:42, Miguel A. Garcia wrote:
> At the last IETF meeting we had a presentation of the following draft:
>
> http://datatracker.ietf.org/doc/draft-ivov-mmusic-latching/
>
> At that time, there was interest in adopting this document as WG item. We
> currently have a milestone to submit an informational document describing
> current practices in hosted NAT traversal, where this document fits.
>
> We would like to poll the working group to reconfirm the consensus on
> adopting the above mentioned draft as a working group item.
>
> If you support this work or have problems with the adoption of this
> document as WG item, please indicate it by replying to this e-mail within
> the next 5 days.
>
>      Miguel and Flemming (co-chairs)

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

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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: comments on draft-ietf-mmusic-sdp-bundle-negotiation-01
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I have a few of comments on this draft, mostly in regard to the description=
 of the offer/answer procedures.

1) Section 6.2 end as follows:

   If an SDP Offer, offering a "BUNDLE" group, and the SDP Offerer has
   reasons to believe that the rejection is due to the usage of a single
   port number value for multiple "m=3D" lines, the SDP Offerer SHOULD
   send a new SDP Offer, without a "BUNDLE" group, where a separate port
   number value is provide for each "m=3D" line of the SDP offer.

I think this was meant to be as follows:

   If an SDP Offer, offering a "BUNDLE" group, is rejected, and the SDP Off=
erer has
   reasons to believe that the rejection is due to the usage of a single
   port number value for multiple "m=3D" lines, the SDP Offerer SHOULD
   send a new SDP Offer, without a "BUNDLE" group, where a separate port
   number value is provide for each "m=3D" line of the SDP offer.

Additionally, on what basis would the offerer have "reasons to believe the =
rejection is due to the usage of a single port number value"? Perhaps if it=
 is the initial INVITE, this assumption is made, whereas if this is a re-IN=
VITE following a previously successful negotiation, it is assumed to be som=
e other case? This seems strange to me. I think it is better to remove the =
"reasons to believe" qualifier.

2) Section 6.4.2, reads:

   The total proposed bandwidth is the sum of the proposed bandwidth for
   each "m=3D" line associated with a negotiated BUNDLE group.

I'm afraid this is an overly simplistic approach. For example, a typical vi=
deo call with content sharing includes one audio m-line and 2 video m-lines=
. The bandwidth is declared at the session level as well as at the media le=
vel, but the sum of the media levels is typically greater than that of the =
session level. When content sharing occurs, it steals some bandwidth from t=
he main video. In the case of BUNDLED m-lines, the proposed bandwidth shoul=
d therefore be the lesser of the sum of the BUNDLE m-lines and the session =
bandwidth.

3) Section 7.2 reads,

   o  - A given SSRC SHOULD NOT transmit RTP packets using payload types
      that originates from different "m=3D" lines.

   NOTE: The last bullet above is to avoid sending multiple media types
   from the same SSRC.  If transmission of multiple media types are done
   with time overlap RTP and RTCP fails to function.  Even if done in
   proper sequence this causes RTP Timestamp rate switching issues [ref
   to draft-ietf-avtext-multiple-clock-rates].

Why not replace the text in the bullet point with the text from the note, i=
.e.

   o  - A given SSRC SHOULD NOT be used for sending multiple media types .
=20
   NOTE: If transmission of multiple media types is done
   with time overlap, RTP and RTCP fail to function.  Even if done in
   proper sequence this causes RTP Timestamp rate switching issues [ref
   to draft-ietf-avtext-multiple-clock-rates].

Cheers,
Charles

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Subject: Re: [MMUSIC] Comments on draft-ietf-mmusic-traffic-class-for-sdp-02
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At 09:06 PM 11/2/2012, Charles Eckel (eckelcu) wrote:
>Please see inline.
>
> > -----Original Message-----
> > From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> > Behalf Of Paul Kyzivat
> > Sent: Thursday, November 01, 2012 8:05 PM
> > To: IETF MMUSIC WG
> > Subject: [MMUSIC] Comments on draft-ietf-mmusic-traffic-class-for-sdp-02
> >
> > I have some comments on this new version of the document. (Sorry!)
> >
> > Section 3:
> >
> > I know we have been through this many times, and its much better now.
> > But I still see one issue:
> >
> > The ABNF says an adjective is a tcl-token, which allows "-" and
> > alphanumerics. But then the text says that an adjective and be preceded
> > by underscore ("_") to indicate that it is non-standard. This means you
> > are *extending* the syntax in text beyond what the ABNF allows. IMO that
> > is a very bad idea.
> >
> > Instead, I recommend altering the syntax to permit the underscore in
> > adjectives in the way you intend it to be used. I would suggest some
> > syntax for this but I don't know if you want to permit non-standard
> > qualified adjectives.
>
>I'd like to recommend removing the text regarding the '_' and 
>leaving the ABNF as is. At IETF 84 I thought we agreed to address 
>this by lowering the bar to register a new adjective.

The -03 version at 
http://hive.packetizer.com/users/paulej/internet-drafts/draft-ietf-mmusic-traffic-class-for-sdp-03.txt 
has removed the '_' from all the text.


>Cheers,
>Charles
>
> > Also, as we have previously discussed, this way of using non-standard
> > adjectives is quite analogous to the use "P-" or "X-" headers that has
> > been found troublesome and generally banned. So I expect this might have
> > trouble getting through IESG review. I thought we had discussed having
> > the non-standard adjectives be registered using a
> > first-come-first-served registration policy.
> >
> > ISTM that the portion of this section that discusses "aq" ought to be in
> > its own section, and then referenced from the iana considerations section.
> >
> > Section 6.4:
> >
> > I don't find any explanation of how or if to register qualified
> > adjectives. The section title says "Unqualified Adjective Registration",
> > but the inital set of registered values includes "aq" (the qualifier
> > part of qualified adjectives), and some values intended to be qualfied
> > by "aq:" (such as "admitted"). This certainly implies to me that I can
> > use each of these as an unqualified adjective.
> >
> > If the intent is to register such things then I would expect to see the
> > full qualified adjectives listed for registration. E.g. "aq:admitted"
> >
> >       Thanks,
> >       Paul
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www.ietf.org/mailman/listinfo/mmusic
>_______________________________________________
>mmusic mailing list
>mmusic@ietf.org
>https://www.ietf.org/mailman/listinfo/mmusic


From eckelcu@cisco.com  Sun Nov  4 16:42:56 2012
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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: comments on draft-holmberg-mmusic-sdp-mmt-negotiation-00
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Subject: [MMUSIC] comments on draft-holmberg-mmusic-sdp-mmt-negotiation-00
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I read the draft, and I have some concerns it with. It seems that in genera=
l one needs to duplicate much of the information from each individual media=
 description in the anymedia media description. I think I could live with t=
hat; however, the bigger problem is in regard to figuring out what needs to=
 be specified in the anymedia media description. In the good cases, you mer=
ely duplicate the SDP attributes from each individual media description in =
the anymedia media description. In the bad case, you need to define new SDP=
 attributes.
Here are some related comments in regard to specific section of the existin=
g draft.

1) Section 5.2.2.2 seems to imply that accepting/rejecting an anymedia medi=
a description is always an all of nothing matter. If it contains 2 audio li=
nes and 2 video lines, I need to accept all of them or none of them. It tha=
t is the intent, it seems pretty limiting to me.

2) Section 6.3 says directionality of anymedia applies to all media lines. =
This seems limiting as well. I may want to temporarily stop a video stream =
(i.e. make inactive or recvonly), but continue to have the audio be sendrec=
v.

3) Section 6.4, I think you need to be able to specify the bandwidth for ea=
ch media stream in the anymedia group, so new syntax will be needed to deal=
 with the bandwidth attribute.

Cheers,
Charles


From christer.holmberg@ericsson.com  Sun Nov  4 17:00:16 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [MMUSIC] comments on draft-holmberg-mmusic-sdp-mmt-negotiation-00
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Hi Charles,
=20
>I read the draft, and I have some concerns it with. It seems that in gener=
al one needs to duplicate much of the information from each individual medi=
a description in the anymedia media description. I think I could live with =
that;=20
>however, the bigger problem is in regard to figuring out what needs to be =
specified in the anymedia media description. In the good cases, you merely =
duplicate the SDP attributes from each individual media description in the =
anymedia=20
>media description. In the bad case, you need to define new SDP attributes.
>Here are some related comments in regard to specific section of the existi=
ng draft.
>
>1) Section 5.2.2.2 seems to imply that accepting/rejecting an anymedia med=
ia description is always an all of nothing matter. If it contains 2 audio l=
ines and 2 video lines, I need to accept all of them or none of them. It th=
at is the=20
>intent, it seems pretty limiting to me.

That is not the intent.

However, it is related to a more general (not specific to muliplexing) ques=
tion that needs to be clarified - how individual sources/streams within an =
SDP m- line are rejected.

>2) Section 6.3 says directionality of anymedia applies to all media lines.=
 This seems limiting as well. I may want to temporarily stop a video stream=
 (i.e. make inactive or recvonly), but continue to have the audio be sendre=
cv.

You can use the ssrc attribute to set the directionality for individual sou=
rces/streams.

>3) Section 6.4, I think you need to be able to specify the bandwidth for e=
ach media stream in the anymedia group, so new syntax will be needed to dea=
l with the bandwidth attribute.

Yes.

But, again, it is a generic thing for when you have multiple sources/stream=
s per m- line.


Regards,

Christer

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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: comments on draft-holmberg-mmusic-sdp-mmt-negotiation-00
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Subject: Re: [MMUSIC] comments on draft-holmberg-mmusic-sdp-mmt-negotiation-00
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> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> Sent: Sunday, November 04, 2012 8:00 PM
> To: Charles Eckel (eckelcu); mmusic@ietf.org
> Subject: RE: comments on draft-holmberg-mmusic-sdp-mmt-negotiation-00
>=20
>=20
> Hi Charles,
>=20
> >I read the draft, and I have some concerns it with. It seems that in gen=
eral
> one needs to duplicate much of the information from each individual media
> description in the anymedia media description. I think I could live with =
that;
> >however, the bigger problem is in regard to figuring out what needs to b=
e
> specified in the anymedia media description. In the good cases, you merel=
y
> duplicate the SDP attributes from each individual media description in th=
e
> anymedia
> >media description. In the bad case, you need to define new SDP attribute=
s.
> >Here are some related comments in regard to specific section of the
> existing draft.
> >
> >1) Section 5.2.2.2 seems to imply that accepting/rejecting an anymedia
> media description is always an all of nothing matter. If it contains 2 au=
dio
> lines and 2 video lines, I need to accept all of them or none of them. It=
 that
> is the
> >intent, it seems pretty limiting to me.
>=20
> That is not the intent.
>=20
> However, it is related to a more general (not specific to muliplexing)
> question that needs to be clarified - how individual sources/streams with=
in
> an SDP m- line are rejected.

Okay, that is true; but the need for such a mechanism is more common needed=
 if all media is in a single description rather than in multiple media desc=
riptions that are then bundled. The anymedia syntax really forces the issue=
 for even relatively simple use cases.

> >2) Section 6.3 says directionality of anymedia applies to all media line=
s.
> This seems limiting as well. I may want to temporarily stop a video strea=
m
> (i.e. make inactive or recvonly), but continue to have the audio be sendr=
ecv.
>=20
> You can use the ssrc attribute to set the directionality for individual
> sources/streams.
>=20
> >3) Section 6.4, I think you need to be able to specify the bandwidth for
> each media stream in the anymedia group, so new syntax will be needed to
> deal with the bandwidth attribute.
>=20
> Yes.
>=20
> But, again, it is a generic thing for when you have multiple sources/stre=
ams
> per m- line.

In the case of multiple m-lines, you can assign a bandwidth per m-line and =
have the session bandwidth cap the total bandwidth for the session. With an=
ymedia and a single m-line, you lose the granularity you previously had at =
the m-line level and are left with specifying the bandwidth for the anymedi=
a line as a whole.
For example, today I might offer the following for a video call with conten=
t sharing:

v=3D0 =09
o=3Danonymous 1240218157 1240218157 IN IP4 10.193.128.35=20
s=3D-=20
i=3DmyUserAgent=20
c=3DIN IP4 10.193.128.35=20
b=3DTIAS:512000
t=3D0 0=20
m=3Daudio 6000 RTP/AVP 0 18 101
a=3Drtpmap:0 PCMU/8000=20
a=3Drtpmap:18 G729/8000=20
a=3Drtpmap:101 telephone-event/8000=20
a=3Dfmtp:101 0-15=20
m=3Dvideo 6002 RTP/AVP 96
b=3DTIAS:512000=20
a=3Drtpmap:96 H264/90000=20
a=3Dfmtp:96 profile-level-id=3D428014=20
a=3Dcontent:main
m=3Dvideo 6002 RTP/AVP 96
b=3DTIAS:256000=20
a=3Drtpmap:96 H264/90000=20
a=3Dfmtp:96 profile-level-id=3D428014
a=3Dcontent:slides

In this case, depending on the answer, the audio will take between 8k and 6=
4k, the main video can take the remainder of the 512k until content sharing=
 starts, and once content sharing starts, the main video could drop down a =
bit to free some bandwidth for the content .
With mmt, all I can specify is 512k for the total. I cannot subdivide it pe=
r media stream as above. To do so, I would need to specify some new SDP and=
/or encode the SSRC values I plan to use in SDP.=20

Cheers,
Charles


>=20
> Regards,
>=20
> Christer

From internet-drafts@ietf.org  Mon Nov  5 05:25:49 2012
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-latching-00.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Multiparty Multimedia Session Control Wor=
king Group of the IETF.

	Title           : Latching: Hosted NAT Traversal (HNT) for Media in Real-T=
ime Communication
	Author(s)       : Emil Ivov
                          Hadriel Kaplan
                          Dan Wing
	Filename        : draft-ietf-mmusic-latching-00.txt
	Pages           : 15
	Date            : 2012-11-05

Abstract:
   This document describes behavior of signalling intermediaries in RTC
   deployments, sometimes referred to as Session Border Controllers
   (SBCs), when performing Hosted NAT Traversal (HNT).  HNT is a set of
   mechanisms, such as media relaying and latching, that such
   intermediaries use to enable other RTC devices behind NATs to
   communicate with each other.  This document is non-normative, and is
   only written to explain HNT in order to provide a reference to the
   IETF community, as well as an informative description to
   manufacturers, and users.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-latching-00


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


From harald@alvestrand.no  Mon Nov  5 06:55:39 2012
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Subject: Re: [MMUSIC] SCTP question: Where does it multiplex?
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On 11/02/2012 03:53 PM, Paul Kyzivat wrote:
> On 11/2/12 6:39 AM, Harald Alvestrand wrote:
>> As currently specified, the DTLS/SCTP protocol is carefully specified in
>> such a way that it can ride on the same 5-tuple as an SRTP session. I'd
>> like to keep that property when we SDP-negotiate it.
>
> It is? How? Where is that specified?
RFC 5764 section 5.1.2 (DTLS/SCTP packets are DTLS packets).


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Subject: [MMUSIC] BUNDLE: Session bandwidth (Re: comments on draft-ietf-mmusic-sdp-bundle-negotiation-01)
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On 11/04/2012 07:23 AM, Charles Eckel (eckelcu) wrote:
> 2) Section 6.4.2, reads:
>
>     The total proposed bandwidth is the sum of the proposed bandwidth for
>     each "m=" line associated with a negotiated BUNDLE group.
>
> I'm afraid this is an overly simplistic approach. For example, a typical video call with content sharing includes one audio m-line and 2 video m-lines. The bandwidth is declared at the session level as well as at the media level, but the sum of the media levels is typically greater than that of the session level. When content sharing occurs, it steals some bandwidth from the main video. In the case of BUNDLED m-lines, the proposed bandwidth should therefore be the lesser of the sum of the BUNDLE m-lines and the session bandwidth.
>
I'm not sure this really matters; what we're trying to compute is the b= 
parameter for the "implied RTP session" in the BUNDLE group. What 
matters is that it's computed consistently, which argues for the 
simplest possible computation.

Last time I looked b=CT was scoped to a whole conference ("supplied for 
the session"), while b=AS was scoped to a single RTP session. But I may 
be misinterpreting RFC 4566 section 5.8.

But yes, that's one of the advantages of the MMT approach - the sender 
gets to specify instead of the receiver having to compute.




From fandreas@cisco.com  Mon Nov  5 11:00:00 2012
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Date: Mon, 05 Nov 2012 13:59:37 -0500
From: Flemming Andreasen <fandreas@cisco.com>
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] comments on draft-holmberg-mmusic-sdp-mmt-negotiation-00
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Taking a step back, the big issue I have with the various "bundle"-like 
proposals is how we deal with parameters/attributes that have to be 
reconciled across multiple media descriptions. I have yet to see 
solutions for how to deal with that in a generic manner.

What I like about the "mmt"-style proposal is that we avoid that issue 
by treating the multiplexed media stream as a single media description 
and hence we explicitly provide the parameters/attributes we want to use 
for that multiplexed media description. As you note, we lose some 
expressiveness in the process of doing that. Use of source-specific 
media attributes (RFC 5576) brings some of that back, but not 
everything. We have other parameters and, as you note, a potential need 
to specify things at the "stream"-level (which could consist of more 
than one SSRC), however you could probably come up with a solution 
similar to RFC 5576 for those (level of indirection for the actual 
parameter/attribute). It's a more verbose solution than the 
"bundle"-like approaches, but it also provides a clearer and cleaner 
solution IMO (of course the middlebox guys will probably tell us shortly 
that this will never make it through their boxes)

Thanks

-- Flemming (as Individual)


On 11/4/12 10:29 PM, Charles Eckel (eckelcu) wrote:
>> -----Original Message-----
>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
>> Sent: Sunday, November 04, 2012 8:00 PM
>> To: Charles Eckel (eckelcu); mmusic@ietf.org
>> Subject: RE: comments on draft-holmberg-mmusic-sdp-mmt-negotiation-00
>>
>>
>> Hi Charles,
>>
>>> I read the draft, and I have some concerns it with. It seems that in general
>> one needs to duplicate much of the information from each individual media
>> description in the anymedia media description. I think I could live with that;
>>> however, the bigger problem is in regard to figuring out what needs to be
>> specified in the anymedia media description. In the good cases, you merely
>> duplicate the SDP attributes from each individual media description in the
>> anymedia
>>> media description. In the bad case, you need to define new SDP attributes.
>>> Here are some related comments in regard to specific section of the
>> existing draft.
>>> 1) Section 5.2.2.2 seems to imply that accepting/rejecting an anymedia
>> media description is always an all of nothing matter. If it contains 2 audio
>> lines and 2 video lines, I need to accept all of them or none of them. It that
>> is the
>>> intent, it seems pretty limiting to me.
>> That is not the intent.
>>
>> However, it is related to a more general (not specific to muliplexing)
>> question that needs to be clarified - how individual sources/streams within
>> an SDP m- line are rejected.
> Okay, that is true; but the need for such a mechanism is more common needed if all media is in a single description rather than in multiple media descriptions that are then bundled. The anymedia syntax really forces the issue for even relatively simple use cases.
>
>>> 2) Section 6.3 says directionality of anymedia applies to all media lines.
>> This seems limiting as well. I may want to temporarily stop a video stream
>> (i.e. make inactive or recvonly), but continue to have the audio be sendrecv.
>>
>> You can use the ssrc attribute to set the directionality for individual
>> sources/streams.
>>
>>> 3) Section 6.4, I think you need to be able to specify the bandwidth for
>> each media stream in the anymedia group, so new syntax will be needed to
>> deal with the bandwidth attribute.
>>
>> Yes.
>>
>> But, again, it is a generic thing for when you have multiple sources/streams
>> per m- line.
> In the case of multiple m-lines, you can assign a bandwidth per m-line and have the session bandwidth cap the total bandwidth for the session. With anymedia and a single m-line, you lose the granularity you previously had at the m-line level and are left with specifying the bandwidth for the anymedia line as a whole.
> For example, today I might offer the following for a video call with content sharing:
>
> v=0 	
> o=anonymous 1240218157 1240218157 IN IP4 10.193.128.35
> s=-
> i=myUserAgent
> c=IN IP4 10.193.128.35
> b=TIAS:512000
> t=0 0
> m=audio 6000 RTP/AVP 0 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> m=video 6002 RTP/AVP 96
> b=TIAS:512000
> a=rtpmap:96 H264/90000
> a=fmtp:96 profile-level-id=428014
> a=content:main
> m=video 6002 RTP/AVP 96
> b=TIAS:256000
> a=rtpmap:96 H264/90000
> a=fmtp:96 profile-level-id=428014
> a=content:slides
>
> In this case, depending on the answer, the audio will take between 8k and 64k, the main video can take the remainder of the 512k until content sharing starts, and once content sharing starts, the main video could drop down a bit to free some bandwidth for the content .
> With mmt, all I can specify is 512k for the total. I cannot subdivide it per media stream as above. To do so, I would need to specify some new SDP and/or encode the SSRC values I plan to use in SDP.
>
> Cheers,
> Charles
>
>
>> Regards,
>>
>> Christer
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
> .
>


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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: Harald Alvestrand <harald@alvestrand.no>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] BUNDLE: Session bandwidth (Re: comments on draft-ietf-mmusic-sdp-bundle-negotiation-01)
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Subject: Re: [MMUSIC] BUNDLE: Session bandwidth (Re: comments on	draft-ietf-mmusic-sdp-bundle-negotiation-01)
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> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Harald Alvestrand
> Sent: Monday, November 05, 2012 10:45 AM
> To: mmusic@ietf.org
> Subject: [MMUSIC] BUNDLE: Session bandwidth (Re: comments on draft-
> ietf-mmusic-sdp-bundle-negotiation-01)
>=20
> On 11/04/2012 07:23 AM, Charles Eckel (eckelcu) wrote:
> > 2) Section 6.4.2, reads:
> >
> >     The total proposed bandwidth is the sum of the proposed bandwidth f=
or
> >     each "m=3D" line associated with a negotiated BUNDLE group.
> >
> > I'm afraid this is an overly simplistic approach. For example, a typica=
l video
> call with content sharing includes one audio m-line and 2 video m-lines. =
The
> bandwidth is declared at the session level as well as at the media level,=
 but
> the sum of the media levels is typically greater than that of the session
> level. When content sharing occurs, it steals some bandwidth from the mai=
n
> video. In the case of BUNDLED m-lines, the proposed bandwidth should
> therefore be the lesser of the sum of the BUNDLE m-lines and the session
> bandwidth.
> >
> I'm not sure this really matters; what we're trying to compute is the b=
=3D
> parameter for the "implied RTP session" in the BUNDLE group. What
> matters is that it's computed consistently, which argues for the
> simplest possible computation.
>=20
> Last time I looked b=3DCT was scoped to a whole conference ("supplied for
> the session"), while b=3DAS was scoped to a single RTP session. But I may
> be misinterpreting RFC 4566 section 5.8.

I recommend using b=3DTIAS, as defined in RFC 3890. This parameter may be u=
sed at either the session or the media level, and it scope accordingly.=20
What needs to be avoided is summing up the b=3DTIAS/AS values from the bund=
led media lines and using that as that bandwidth for the bundle even if it =
exceeds the b=3DTIAS/AS/CT value from the session level.

> But yes, that's one of the advantages of the MMT approach - the sender
> gets to specify instead of the receiver having to compute.

The MMT approach has different challenges when it comes to bandwidth. That =
is being tracked in another thread (http://www.ietf.org/mail-archive/web/mm=
usic/current/msg09817.html).

Cheers,
Charles

>=20
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

From harald@alvestrand.no  Mon Nov  5 13:18:49 2012
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Subject: Re: [MMUSIC] BUNDLE: Session bandwidth (Re: comments on draft-ietf-mmusic-sdp-bundle-negotiation-01)
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On 11/05/2012 09:04 PM, Charles Eckel (eckelcu) wrote:
>> -----Original Message-----
>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
>> Behalf Of Harald Alvestrand
>> Sent: Monday, November 05, 2012 10:45 AM
>> To: mmusic@ietf.org
>> Subject: [MMUSIC] BUNDLE: Session bandwidth (Re: comments on draft-
>> ietf-mmusic-sdp-bundle-negotiation-01)
>>
>> On 11/04/2012 07:23 AM, Charles Eckel (eckelcu) wrote:
>>> 2) Section 6.4.2, reads:
>>>
>>>      The total proposed bandwidth is the sum of the proposed bandwidth for
>>>      each "m=" line associated with a negotiated BUNDLE group.
>>>
>>> I'm afraid this is an overly simplistic approach. For example, a typical video
>> call with content sharing includes one audio m-line and 2 video m-lines. The
>> bandwidth is declared at the session level as well as at the media level, but
>> the sum of the media levels is typically greater than that of the session
>> level. When content sharing occurs, it steals some bandwidth from the main
>> video. In the case of BUNDLED m-lines, the proposed bandwidth should
>> therefore be the lesser of the sum of the BUNDLE m-lines and the session
>> bandwidth.
>> I'm not sure this really matters; what we're trying to compute is the b=
>> parameter for the "implied RTP session" in the BUNDLE group. What
>> matters is that it's computed consistently, which argues for the
>> simplest possible computation.
>>
>> Last time I looked b=CT was scoped to a whole conference ("supplied for
>> the session"), while b=AS was scoped to a single RTP session. But I may
>> be misinterpreting RFC 4566 section 5.8.
> I recommend using b=TIAS, as defined in RFC 3890. This parameter may be used at either the session or the media level, and it scope accordingly.
> What needs to be avoided is summing up the b=TIAS/AS values from the bundled media lines and using that as that bandwidth for the bundle even if it exceeds the b=TIAS/AS/CT value from the session level.
If one of the bundling approaches is chosen, would you recommend setting 
the RTP session (=media level) bandwidth as min(SDP session b= line, 
sum(media b= line))?
>
>> But yes, that's one of the advantages of the MMT approach - the sender
>> gets to specify instead of the receiver having to compute.
> The MMT approach has different challenges when it comes to bandwidth. That is being tracked in another thread (http://www.ietf.org/mail-archive/web/mmusic/current/msg09817.html).
>
> Cheers,
> Charles
>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic


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Subject: [MMUSIC] MMT bandwidth limits (Re: comments on draft-holmberg-mmusic-sdp-mmt-negotiation-00)
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On 11/05/2012 04:29 AM, Charles Eckel (eckelcu) wrote:
> In the case of multiple m-lines, you can assign a bandwidth per m-line and have the session bandwidth cap the total bandwidth for the session. With anymedia and a single m-line, you lose the granularity you previously had at the m-line level and are left with specifying the bandwidth for the anymedia line as a whole.
> For example, today I might offer the following for a video call with content sharing:
>
> v=0 	
> o=anonymous 1240218157 1240218157 IN IP4 10.193.128.35
> s=-
> i=myUserAgent
> c=IN IP4 10.193.128.35
> b=TIAS:512000
> t=0 0
> m=audio 6000 RTP/AVP 0 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> m=video 6002 RTP/AVP 96
> b=TIAS:512000
> a=rtpmap:96 H264/90000
> a=fmtp:96 profile-level-id=428014
> a=content:main
> m=video 6002 RTP/AVP 96
> b=TIAS:256000
> a=rtpmap:96 H264/90000
> a=fmtp:96 profile-level-id=428014
> a=content:slides
>
> In this case, depending on the answer, the audio will take between 8k and 64k, the main video can take the remainder of the 512k until content sharing starts, and once content sharing starts, the main video could drop down a bit to free some bandwidth for the content .
> With mmt, all I can specify is 512k for the total. I cannot subdivide it per media stream as above. To do so, I would need to specify some new SDP and/or encode the SSRC values I plan to use in SDP.
I'm curious - if I send this in an offer, is there any part of the SDP 
description that says that the respondent, when sending, should cut down 
on his video bandwidth if video + audio would otherwise be > 512K?

I don't see any "priority" flags in there.

(another thing that confuses me a bit is how b= represents bandwidth in 
the 2 directions for a point-to-point session - if A sends an offer with 
session b=TIAS:512K, and B sends A an answer with session b=TIAS:512K, 
and they both start sending video at 300K, are they below the cap or 
above the cap?

Again, it's probably clear to those who've been around here for a while)

                 Harald



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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: [MMUSIC] BUNDLE: Session bandwidth (Re: comments on draft-ietf-mmusic-sdp-bundle-negotiation-01)
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Subject: Re: [MMUSIC] BUNDLE: Session bandwidth (Re: comments on draft-ietf-mmusic-sdp-bundle-negotiation-01)
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> -----Original Message-----
> From: Harald Alvestrand [mailto:harald@alvestrand.no]
> Sent: Monday, November 05, 2012 4:19 PM
> To: Charles Eckel (eckelcu)
> Cc: mmusic@ietf.org
> Subject: Re: [MMUSIC] BUNDLE: Session bandwidth (Re: comments on draft-
> ietf-mmusic-sdp-bundle-negotiation-01)
>=20
> On 11/05/2012 09:04 PM, Charles Eckel (eckelcu) wrote:
> >> -----Original Message-----
> >> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> >> Behalf Of Harald Alvestrand
> >> Sent: Monday, November 05, 2012 10:45 AM
> >> To: mmusic@ietf.org
> >> Subject: [MMUSIC] BUNDLE: Session bandwidth (Re: comments on draft-
> >> ietf-mmusic-sdp-bundle-negotiation-01)
> >>
> >> On 11/04/2012 07:23 AM, Charles Eckel (eckelcu) wrote:
> >>> 2) Section 6.4.2, reads:
> >>>
> >>>      The total proposed bandwidth is the sum of the proposed bandwidt=
h
> for
> >>>      each "m=3D" line associated with a negotiated BUNDLE group.
> >>>
> >>> I'm afraid this is an overly simplistic approach. For example, a typi=
cal
> video
> >> call with content sharing includes one audio m-line and 2 video m-line=
s.
> The
> >> bandwidth is declared at the session level as well as at the media lev=
el,
> but
> >> the sum of the media levels is typically greater than that of the sess=
ion
> >> level. When content sharing occurs, it steals some bandwidth from the
> main
> >> video. In the case of BUNDLED m-lines, the proposed bandwidth should
> >> therefore be the lesser of the sum of the BUNDLE m-lines and the sessi=
on
> >> bandwidth.
> >> I'm not sure this really matters; what we're trying to compute is the =
b=3D
> >> parameter for the "implied RTP session" in the BUNDLE group. What
> >> matters is that it's computed consistently, which argues for the
> >> simplest possible computation.
> >>
> >> Last time I looked b=3DCT was scoped to a whole conference ("supplied =
for
> >> the session"), while b=3DAS was scoped to a single RTP session. But I =
may
> >> be misinterpreting RFC 4566 section 5.8.
> > I recommend using b=3DTIAS, as defined in RFC 3890. This parameter may =
be
> used at either the session or the media level, and it scope accordingly.
> > What needs to be avoided is summing up the b=3DTIAS/AS values from the
> bundled media lines and using that as that bandwidth for the bundle even =
if
> it exceeds the b=3DTIAS/AS/CT value from the session level.
> If one of the bundling approaches is chosen, would you recommend setting
> the RTP session (=3Dmedia level) bandwidth as min(SDP session b=3D line,
> sum(media b=3D line))?

Yes.

Cheers,
Charles

> >> But yes, that's one of the advantages of the MMT approach - the sender
> >> gets to specify instead of the receiver having to compute.
> > The MMT approach has different challenges when it comes to bandwidth.
> That is being tracked in another thread (http://www.ietf.org/mail-
> archive/web/mmusic/current/msg09817.html).
> >
> > Cheers,
> > Charles
> >
> >>
> >> _______________________________________________
> >> mmusic mailing list
> >> mmusic@ietf.org
> >> https://www.ietf.org/mailman/listinfo/mmusic


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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: Harald Alvestrand <harald@alvestrand.no>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] MMT bandwidth limits (Re: comments on draft-holmberg-mmusic-sdp-mmt-negotiation-00)
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Date: Mon, 5 Nov 2012 23:58:29 +0000
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References: <92B7E61ADAC1BB4F941F943788C088281074BD@xmb-aln-x08.cisco.com> <7594FB04B1934943A5C02806D1A2204B024E1B@ESESSMB209.ericsson.se> <92B7E61ADAC1BB4F941F943788C0882810758B@xmb-aln-x08.cisco.com> <50982E6B.1010904@alvestrand.no>
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Subject: Re: [MMUSIC] MMT bandwidth limits (Re: comments on	draft-holmberg-mmusic-sdp-mmt-negotiation-00)
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> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Harald Alvestrand
> Sent: Monday, November 05, 2012 4:24 PM
> To: mmusic@ietf.org
> Subject: [MMUSIC] MMT bandwidth limits (Re: comments on draft-
> holmberg-mmusic-sdp-mmt-negotiation-00)
>=20
> On 11/05/2012 04:29 AM, Charles Eckel (eckelcu) wrote:
> > In the case of multiple m-lines, you can assign a bandwidth per m-line =
and
> have the session bandwidth cap the total bandwidth for the session. With
> anymedia and a single m-line, you lose the granularity you previously had=
 at
> the m-line level and are left with specifying the bandwidth for the
> anymedia line as a whole.
> > For example, today I might offer the following for a video call with co=
ntent
> sharing:
> >
> > v=3D0
> > o=3Danonymous 1240218157 1240218157 IN IP4 10.193.128.35
> > s=3D-
> > i=3DmyUserAgent
> > c=3DIN IP4 10.193.128.35
> > b=3DTIAS:512000
> > t=3D0 0
> > m=3Daudio 6000 RTP/AVP 0 18 101
> > a=3Drtpmap:0 PCMU/8000
> > a=3Drtpmap:18 G729/8000
> > a=3Drtpmap:101 telephone-event/8000
> > a=3Dfmtp:101 0-15
> > m=3Dvideo 6002 RTP/AVP 96
> > b=3DTIAS:512000
> > a=3Drtpmap:96 H264/90000
> > a=3Dfmtp:96 profile-level-id=3D428014
> > a=3Dcontent:main
> > m=3Dvideo 6002 RTP/AVP 96
> > b=3DTIAS:256000
> > a=3Drtpmap:96 H264/90000
> > a=3Dfmtp:96 profile-level-id=3D428014
> > a=3Dcontent:slides
> >
> > In this case, depending on the answer, the audio will take between 8k a=
nd
> 64k, the main video can take the remainder of the 512k until content shar=
ing
> starts, and once content sharing starts, the main video could drop down a=
 bit
> to free some bandwidth for the content .
> > With mmt, all I can specify is 512k for the total. I cannot subdivide i=
t per
> media stream as above. To do so, I would need to specify some new SDP
> and/or encode the SSRC values I plan to use in SDP.
> I'm curious - if I send this in an offer, is there any part of the SDP
> description that says that the respondent, when sending, should cut down
> on his video bandwidth if video + audio would otherwise be > 512K?
>=20
> I don't see any "priority" flags in there.

There is no explicit priority. It is up to the application to decide. RFC 3=
890 does say:

   At the SDP session level, the TIAS value is the maximal amount of
   bandwidth needed when all declared media streams are used.  This MAY
   be less than the sum of all the individual media streams values.
   This is due to the possibility that not all streams have their
   maximum at the same point in time.=20

> (another thing that confuses me a bit is how b=3D represents bandwidth in
> the 2 directions for a point-to-point session - if A sends an offer with
> session b=3DTIAS:512K, and B sends A an answer with session b=3DTIAS:512K=
,
> and they both start sending video at 300K, are they below the cap or
> above the cap?

That is fine. This is just a statement of the maximum bandwidth the impleme=
ntation is willing to receive. The values can be asymmetric, and this is ju=
st a maximum. It does not indicate anything about how much the implementati=
on will send.

Cheers,
Charles

> Again, it's probably clear to those who've been around here for a while)
=20
>                  Harald
>=20
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

From eckelcu@cisco.com  Mon Nov  5 16:06:32 2012
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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: "Flemming Andreasen (fandreas)" <fandreas@cisco.com>
Thread-Topic: [MMUSIC] comments on draft-holmberg-mmusic-sdp-mmt-negotiation-00
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] comments on draft-holmberg-mmusic-sdp-mmt-negotiation-00
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Hi Flemming,

You may very well be right, and I think Christer agrees with your assessmen=
t as well. I do not necessarily disagree, but I am not sure that I agree ei=
ther.
The bandwidth attribute is one example for which I think BUNDLE works bette=
r than MMT. However, bandwidth may be the exception and it may be that the =
majority of SDP attributes are more straightforward with MMT.
An example or two would probably help convince me, or maybe a beer :)

Cheers,
Charles

> -----Original Message-----
> From: Flemming Andreasen (fandreas)
> Sent: Monday, November 05, 2012 2:00 PM
> To: Charles Eckel (eckelcu)
> Cc: Christer Holmberg; mmusic@ietf.org
> Subject: Re: [MMUSIC] comments on draft-holmberg-mmusic-sdp-mmt-
> negotiation-00
>=20
> Taking a step back, the big issue I have with the various "bundle"-like
> proposals is how we deal with parameters/attributes that have to be
> reconciled across multiple media descriptions. I have yet to see
> solutions for how to deal with that in a generic manner.
>=20
> What I like about the "mmt"-style proposal is that we avoid that issue
> by treating the multiplexed media stream as a single media description
> and hence we explicitly provide the parameters/attributes we want to use
> for that multiplexed media description. As you note, we lose some
> expressiveness in the process of doing that. Use of source-specific
> media attributes (RFC 5576) brings some of that back, but not
> everything. We have other parameters and, as you note, a potential need
> to specify things at the "stream"-level (which could consist of more
> than one SSRC), however you could probably come up with a solution
> similar to RFC 5576 for those (level of indirection for the actual
> parameter/attribute). It's a more verbose solution than the
> "bundle"-like approaches, but it also provides a clearer and cleaner
> solution IMO (of course the middlebox guys will probably tell us shortly
> that this will never make it through their boxes)
>=20
> Thanks
>=20
> -- Flemming (as Individual)
>=20
>=20
> On 11/4/12 10:29 PM, Charles Eckel (eckelcu) wrote:
> >> -----Original Message-----
> >> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> >> Sent: Sunday, November 04, 2012 8:00 PM
> >> To: Charles Eckel (eckelcu); mmusic@ietf.org
> >> Subject: RE: comments on draft-holmberg-mmusic-sdp-mmt-negotiation-
> 00
> >>
> >>
> >> Hi Charles,
> >>
> >>> I read the draft, and I have some concerns it with. It seems that in
> general
> >> one needs to duplicate much of the information from each individual
> media
> >> description in the anymedia media description. I think I could live wi=
th
> that;
> >>> however, the bigger problem is in regard to figuring out what needs t=
o
> be
> >> specified in the anymedia media description. In the good cases, you
> merely
> >> duplicate the SDP attributes from each individual media description in
> the
> >> anymedia
> >>> media description. In the bad case, you need to define new SDP
> attributes.
> >>> Here are some related comments in regard to specific section of the
> >> existing draft.
> >>> 1) Section 5.2.2.2 seems to imply that accepting/rejecting an anymedi=
a
> >> media description is always an all of nothing matter. If it contains 2=
 audio
> >> lines and 2 video lines, I need to accept all of them or none of them.=
 It
> that
> >> is the
> >>> intent, it seems pretty limiting to me.
> >> That is not the intent.
> >>
> >> However, it is related to a more general (not specific to muliplexing)
> >> question that needs to be clarified - how individual sources/streams
> within
> >> an SDP m- line are rejected.
> > Okay, that is true; but the need for such a mechanism is more common
> needed if all media is in a single description rather than in multiple me=
dia
> descriptions that are then bundled. The anymedia syntax really forces the
> issue for even relatively simple use cases.
> >
> >>> 2) Section 6.3 says directionality of anymedia applies to all media l=
ines.
> >> This seems limiting as well. I may want to temporarily stop a video
> stream
> >> (i.e. make inactive or recvonly), but continue to have the audio be
> sendrecv.
> >>
> >> You can use the ssrc attribute to set the directionality for individua=
l
> >> sources/streams.
> >>
> >>> 3) Section 6.4, I think you need to be able to specify the bandwidth =
for
> >> each media stream in the anymedia group, so new syntax will be needed
> to
> >> deal with the bandwidth attribute.
> >>
> >> Yes.
> >>
> >> But, again, it is a generic thing for when you have multiple
> sources/streams
> >> per m- line.
> > In the case of multiple m-lines, you can assign a bandwidth per m-line =
and
> have the session bandwidth cap the total bandwidth for the session. With
> anymedia and a single m-line, you lose the granularity you previously had=
 at
> the m-line level and are left with specifying the bandwidth for the
> anymedia line as a whole.
> > For example, today I might offer the following for a video call with co=
ntent
> sharing:
> >
> > v=3D0
> > o=3Danonymous 1240218157 1240218157 IN IP4 10.193.128.35
> > s=3D-
> > i=3DmyUserAgent
> > c=3DIN IP4 10.193.128.35
> > b=3DTIAS:512000
> > t=3D0 0
> > m=3Daudio 6000 RTP/AVP 0 18 101
> > a=3Drtpmap:0 PCMU/8000
> > a=3Drtpmap:18 G729/8000
> > a=3Drtpmap:101 telephone-event/8000
> > a=3Dfmtp:101 0-15
> > m=3Dvideo 6002 RTP/AVP 96
> > b=3DTIAS:512000
> > a=3Drtpmap:96 H264/90000
> > a=3Dfmtp:96 profile-level-id=3D428014
> > a=3Dcontent:main
> > m=3Dvideo 6002 RTP/AVP 96
> > b=3DTIAS:256000
> > a=3Drtpmap:96 H264/90000
> > a=3Dfmtp:96 profile-level-id=3D428014
> > a=3Dcontent:slides
> >
> > In this case, depending on the answer, the audio will take between 8k a=
nd
> 64k, the main video can take the remainder of the 512k until content shar=
ing
> starts, and once content sharing starts, the main video could drop down a=
 bit
> to free some bandwidth for the content .
> > With mmt, all I can specify is 512k for the total. I cannot subdivide i=
t per
> media stream as above. To do so, I would need to specify some new SDP
> and/or encode the SSRC values I plan to use in SDP.
> >
> > Cheers,
> > Charles
> >
> >
> >> Regards,
> >>
> >> Christer
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www.ietf.org/mailman/listinfo/mmusic
> > .
> >


From fandreas@cisco.com  Mon Nov  5 19:07:36 2012
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Subject: [MMUSIC] Comments on draft-westerlund-mmusic-max-ssrc-00
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Hi

A few high-level questions/comments on this draft:

1) The draft concerns itself with resource constraints within a single 
media description, however in practice, limitations will often span 
media description boundaries due to use of shared resources. Why the 
limited scope ?

2) The draft only considers resource limitations in terms of codecs, 
however there are many other potential limiting factors to be considered 
(endpoint-wise and network-wise). For example, use of security 
mechanisms may impact what can be supported, the frequency with which 
packets are sent, etc. Why is the scope limited to only the codecs 
themselves ?

3) SDP Media Capabilities provide a general framework and partial 
solution to indicate resource constraints that span media descriptions. 
While it does not cover the specific issue being handled here (multiple 
instances of a given codec), it provides a more general framework that 
lends itself to 1) and 2) above. Any particular reason to not use this 
framework ?

Lastly, given the royalty-bearing IPR disclosure associated with the 
draft (please correct me if I misread this), I think it would be 
desirable to make a clear distinction between the problem to solve and 
the solution proposed to see if we can find a royalty-free approach.

Thanks

-- Flemming (as Individual)

From magnus.westerlund@ericsson.com  Tue Nov  6 05:01:09 2012
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Date: Tue, 6 Nov 2012 08:01:01 -0500
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Subject: [MMUSIC] Relation between SRCNAME and MSID
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Hi,

I got to reading the MSID this morning and as it was some questions how
SRCNMANE (draft-westerlund-avtext-rtcp-sdes-srcname-02) relates to MSID
(draft-alvestrand-mmusic-msid-01) I will give my view on this.

So SRCNAME is a hierarchical structure who's root is a media stream
source, i.e. a camera or a microphone. Its primary usage is to bind
SSRCs that are all related to that media source together. A simple
example would be if one produced simulcast, i.e. alternative encodings,
of the same media source V1, you can have a source name that is V1.enc1
for the first encoding, V2.enc2 for the second one, and so on. When you
apply RTP retransmission or FEC you do that to a particular SSRC, which
then is one of the encodings. Or even more complex if an encoding is an
SVC MST stream, then you have several streams in different RTP sessions
for the different layers, and your FEC or retransmissions only applies
to one of these substreams. Thus you can have a SSRC that carries a
SRCNAME of V1.enc1.baselayer and use that label for both the media
stream and on the FEC SSRC to tie them together. So SRCNAME shares a
property with MSID that is that is ties SSRCs both within and across RTP
Sessions together.

So MSID, especially in its usage with WebRTC to create the WebRTC
MediaStream actually have an interesting relation to SRCNAMES. As MSID
are application level groupings of particular mediaStreamTracks, a MSID
can be used to tie in RTP media stream and its related SSRCs (the ones
carrying the FEC or the layers) into a MediaStream object. Lets try an
simple example to illustrate.

A WebRTC end-point has one video camera (V1) and one microphone (A1).
The video is to be provided at two different encodings V1.enc1 and
V1.enc2 in SRCNAME terms. But for a WebRTC application it can actually
select to deal with these two encodings as two different WebRTC
MediaStreams which I will give MSID values Var1 and Var2, where the
variants contain one video encoding and the audio

In SDP using a=ssrc to provide the SRCNAME it would look like this


m=video
a=ssrc: 1111 SRCNAME:V1.enc1
a=ssrc: 1111 msid:Var1 v1
a=ssrc: 2222 SRCNAME:V1.enc2
a=ssrc: 2222 msid:Var2 v1

a=audio
a=ssrc: 3333 SRCNAME:A1
a=ssrc: 3333 msid:Var1 a1
a=ssrc: 3333 msid:Var2 a1

In other words a given SSRC might actually be a MediaStreamTrack in
multiple MediaStreams and thus have multiple MSIDs. While a SSRC will
only have one SRCNAME, but that one may specify to a higher detail.

If we would have RTP retransmission in the above, a question around MSID
arises. Should all SSRCs that are part of representing be given a MSID
or would it be simpler to just label the media entry point to the
hierarchical relations SRCNAME is capable of expressing? And I need to
point out that MSID can't replace the SRCNAME as currently defined as it
doesn't make clear the hierarchical relation of the SSRCs.

If you are interested in discussion SRCNAME please show up in AVTEXT on
Wednesday at 1300-1430.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
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From ron.even.tlv@gmail.com  Tue Nov  6 06:04:55 2012
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>, "'mmusic \(E-mail\)'" <mmusic@ietf.org>
References: <50990A0D.305@ericsson.com>
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Date: Tue, 6 Nov 2012 09:02:18 -0500
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Subject: Re: [MMUSIC] Relation between SRCNAME and MSID
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Hi,
I had the same question when looking at mapping RTP streams to CLUE =
media
capture and after trying to address simulcast in that environment it =
looked
to me that srcname can be used for mapping since if we look at hierarchy =
of
CNAME.srcname we can say that if we have an endpoint that have multiple
media sources (e.g. 3 cameras) this can be reflected using the same =
CNAME
with srcnames V1,V2 and V3.=20
If using srcname for the simulcast case it make sense not to have a
different identifier for the non simulcast case.

Roni Even =20

-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf =
Of
Magnus Westerlund
Sent: 06 November, 2012 8:01 AM
To: mmusic (E-mail)
Subject: [MMUSIC] Relation between SRCNAME and MSID

Hi,

I got to reading the MSID this morning and as it was some questions how
SRCNMANE (draft-westerlund-avtext-rtcp-sdes-srcname-02) relates to MSID
(draft-alvestrand-mmusic-msid-01) I will give my view on this.

So SRCNAME is a hierarchical structure who's root is a media stream =
source,
i.e. a camera or a microphone. Its primary usage is to bind SSRCs that =
are
all related to that media source together. A simple example would be if =
one
produced simulcast, i.e. alternative encodings, of the same media source =
V1,
you can have a source name that is V1.enc1 for the first encoding, =
V2.enc2
for the second one, and so on. When you apply RTP retransmission or FEC =
you
do that to a particular SSRC, which then is one of the encodings. Or =
even
more complex if an encoding is an SVC MST stream, then you have several
streams in different RTP sessions for the different layers, and your FEC =
or
retransmissions only applies to one of these substreams. Thus you can =
have a
SSRC that carries a SRCNAME of V1.enc1.baselayer and use that label for =
both
the media stream and on the FEC SSRC to tie them together. So SRCNAME =
shares
a property with MSID that is that is ties SSRCs both within and across =
RTP
Sessions together.

So MSID, especially in its usage with WebRTC to create the WebRTC
MediaStream actually have an interesting relation to SRCNAMES. As MSID =
are
application level groupings of particular mediaStreamTracks, a MSID can =
be
used to tie in RTP media stream and its related SSRCs (the ones carrying =
the
FEC or the layers) into a MediaStream object. Lets try an simple example =
to
illustrate.

A WebRTC end-point has one video camera (V1) and one microphone (A1).
The video is to be provided at two different encodings V1.enc1 and
V1.enc2 in SRCNAME terms. But for a WebRTC application it can actually
select to deal with these two encodings as two different WebRTC =
MediaStreams
which I will give MSID values Var1 and Var2, where the variants contain =
one
video encoding and the audio

In SDP using a=3Dssrc to provide the SRCNAME it would look like this


m=3Dvideo
a=3Dssrc: 1111 SRCNAME:V1.enc1
a=3Dssrc: 1111 msid:Var1 v1
a=3Dssrc: 2222 SRCNAME:V1.enc2
a=3Dssrc: 2222 msid:Var2 v1

a=3Daudio
a=3Dssrc: 3333 SRCNAME:A1
a=3Dssrc: 3333 msid:Var1 a1
a=3Dssrc: 3333 msid:Var2 a1

In other words a given SSRC might actually be a MediaStreamTrack in =
multiple
MediaStreams and thus have multiple MSIDs. While a SSRC will only have =
one
SRCNAME, but that one may specify to a higher detail.

If we would have RTP retransmission in the above, a question around MSID
arises. Should all SSRCs that are part of representing be given a MSID =
or
would it be simpler to just label the media entry point to the =
hierarchical
relations SRCNAME is capable of expressing? And I need to point out that
MSID can't replace the SRCNAME as currently defined as it doesn't make =
clear
the hierarchical relation of the SSRCs.

If you are interested in discussion SRCNAME please show up in AVTEXT on
Wednesday at 1300-1430.

Cheers

Magnus Westerlund

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From jonathan@vidyo.com  Tue Nov  6 08:12:25 2012
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From: Jonathan Lennox <jonathan@vidyo.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>
Date: Tue, 6 Nov 2012 11:12:21 -0500
Thread-Topic: Another minor update for ICE: rel-addr privacy in relay candidates
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Subject: [MMUSIC] Another minor update for ICE: rel-addr privacy in relay candidates
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Hi, All --

I wanted to mention another minor issue with ICE.  This was an issue Justin=
 Uberti raised back in February.

Right now, relayed candidates must include the corresponding server-reflexi=
ve address in their rel-addr field.  I believe the motivation was so middle=
boxes could give any special network treatment they needed between the NAT =
and the TURN server.

In privacy cases, however, where an endpoint is deliberately sending only r=
elayed candidates to conceal its location, this is clearly terrible.

The conclusion on the list at that time was to send 0.0.0.0:0, or equivalen=
t, for the rel-addr in this case.  This should presumably be recommended in=
 the ICE-bis draft.

An alternative might be to omit rel-addr entirely in these cases, but stric=
t 5245 parsers might fail if you did that.

--
Jonathan Lennox
jonathan@vidyo.com



From internet-drafts@ietf.org  Tue Nov  6 15:02:10 2012
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Date: Tue, 06 Nov 2012 15:02:09 -0800
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-media-loopback-24.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Multiparty Multimedia Session Control Wor=
king Group of the IETF.

	Title           : An Extension to the Session Description Protocol (SDP) a=
nd Real-time Transport Protocol (RTP) for Media Loopback
	Author(s)       : Hadriel Kaplan
                          Kaynam Hedayat
                          Nagarjuna Venna
                          Paul E. Jones
                          Nathan Stratton
	Filename        : draft-ietf-mmusic-media-loopback-24.txt
	Pages           : 33
	Date            : 2012-11-06

Abstract:
    The wide deployment of Voice over IP (VoIP), Text and Video over IP
    services has introduced new challenges in managing and maintaining
    real-time voice/text/video quality, reliability, and overall
    performance.  In particular, media delivery is an area that needs
    attention.  One method of meeting these challenges is monitoring
    the media delivery performance by looping media back to the
    transmitter.  This is typically referred to as "active monitoring"
    of services.   Media loopback is especially popular in ensuring the
    quality of transport to the edge of a given VoIP, Real-time Text or
    Video over IP service.  Today in networks that deliver real-time
    media, short of running 'ping' and 'traceroute' to the edge,
    administrators are left without the necessary tools to actively
    monitor, manage, and diagnose quality issues with their service.
    The extension defined herein adds new SDP media types and
    attributes, which enable establishment of media sessions where the
    media is looped back to the transmitter. Such media sessions will
    serve as monitoring and troubleshooting tools by providing the
    means for measurement of more advanced VoIP, Real-time Text and
    Video over IP performance metrics.



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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-media-loopback-24

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-media-loopback-24


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


From martin.thomson@gmail.com  Wed Nov  7 05:59:43 2012
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From: Martin Thomson <martin.thomson@gmail.com>
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Subject: [MMUSIC] Signaling trickle support
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Matthew, Justin and I were discussing the problem of signaling trickle
ICE support and the following proposal was made:

The 24th bit in SDP is currently set to zero.  This bit is
specifically reserved for forward compatibility purposes.  We can use
this bit.  Set this bit to one to indicate support for trickle ICE.

This meets the "works if supported, fails if it doesn't criteria".

This solution may also be appropriate for bundle/mmt/whatever.

From emil@sip-communicator.org  Wed Nov  7 07:18:50 2012
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Subject: Re: [MMUSIC] Signaling trickle support
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Hey Martin,

On 07.11.12, 08:59, Martin Thomson wrote:
> Matthew, Justin and I were discussing the problem of signaling trickle
> ICE support and the following proposal was made:
> 
> The 24th bit in SDP is currently set to zero.  This bit is
> specifically reserved for forward compatibility purposes.  We can use
> this bit.  Set this bit to one to indicate support for trickle ICE.

bit 24? So, you mean setting the version to 1? Is that it?

Could you please point us to the spec you have in mind?

Emil

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Subject: Re: [MMUSIC] Signaling trickle support
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On 07.11.12, 10:18, Emil Ivov wrote:
> Hey Martin,
> 
> On 07.11.12, 08:59, Martin Thomson wrote:
>> Matthew, Justin and I were discussing the problem of signaling trickle
>> ICE support and the following proposal was made:
>>
>> The 24th bit in SDP is currently set to zero.  This bit is
>> specifically reserved for forward compatibility purposes.  We can use
>> this bit.  Set this bit to one to indicate support for trickle ICE.
> 
> bit 24? So, you mean setting the version to 1? Is that it?

OK, it appears (from our offline chat) that this is indeed what you meant.

So, I couldn't find anything in 4566 or 3264 that says an offer MUST be
rejected in the case where the version number is different from 0, but I
might be missing some text (could you please point me to the right place
if you have it?)

That said I think it's a great idea and we should even take it further!
For example we could give v=2 a meaning of bundle support. v=3 would
then mean cundle. v=4 would of course be MMT.

Then, v=5 would be trickle+bundle, v=6: trickle+cundle, v=7:
trickle+mmt, v=8: bundle+mmt, v=9: cundle+mmt, v=10:trickle+bundle+mmt.

And of course if we get a rejection for v=10, we would then be able to
reissue three separate offers for v=1, v=2, and v=4 and this way we'll
know exactly why the offer failed.

Now all that's left is to put this into text and go for WGLC because it
is crystal clear no one will have a problem with it ;)

Emil


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From pkyzivat@alum.mit.edu  Wed Nov  7 11:27:36 2012
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Subject: Re: [MMUSIC] Signaling trickle support
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On 11/7/12 8:59 AM, Martin Thomson wrote:
> Matthew, Justin and I were discussing the problem of signaling trickle
> ICE support and the following proposal was made:
>
> The 24th bit in SDP is currently set to zero.  This bit is
> specifically reserved for forward compatibility purposes.  We can use
> this bit.  Set this bit to one to indicate support for trickle ICE.
>
> This meets the "works if supported, fails if it doesn't criteria".
>
> This solution may also be appropriate for bundle/mmt/whatever.

After some side discussion, I guess you are talking about bits in the 
value of the v= field ???

If so, I still have no idea what you are talking about. The syntax for 
the value of v= is 1*DIGIT. You can choose to represent that in binary 
and then choose bit 24 if you want, but that has no significance in 4566.

Assigning a new SDP version to signal Trickle ICE support seems like an 
incredibly bad idea.

	Thanks,
	Paul


From miguel.a.garcia@ericsson.com  Wed Nov  7 23:30:21 2012
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Subject: Re: [MMUSIC] Signaling trickle support
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As an individual...

I concur with Paul, changing the v= version in the SDP is a very bad idea.

Let met quote RFC 4566:


    The "v=" field gives the version of the Session Description Protocol.
    This memo defines version 0.  There is no minor version number.


This means that if you define another version different than zero, then, 
as a side effect, you are creating a different version of SDP than the 
one specified in RFC 4566. So, you will need to create an RFC specifying 
how such version of SDP works.

/Miguel

On 07/11/2012 20:27, Paul Kyzivat wrote:
> On 11/7/12 8:59 AM, Martin Thomson wrote:
>> Matthew, Justin and I were discussing the problem of signaling trickle
>> ICE support and the following proposal was made:
>>
>> The 24th bit in SDP is currently set to zero.  This bit is
>> specifically reserved for forward compatibility purposes.  We can use
>> this bit.  Set this bit to one to indicate support for trickle ICE.
>>
>> This meets the "works if supported, fails if it doesn't criteria".
>>
>> This solution may also be appropriate for bundle/mmt/whatever.
>
> After some side discussion, I guess you are talking about bits in the
> value of the v= field ???
>
> If so, I still have no idea what you are talking about. The syntax for
> the value of v= is 1*DIGIT. You can choose to represent that in binary
> and then choose bit 24 if you want, but that has no significance in 4566.
>
> Assigning a new SDP version to signal Trickle ICE support seems like an
> incredibly bad idea.
>
>      Thanks,
>      Paul
>
> _______________________________________________
> 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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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Miguel Garcia A <miguel.a.garcia@ericsson.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>, Martin Thomson <martin.thomson@gmail.com>, Emil Ivov <emcho@jitsi.org>
Thread-Topic: [MMUSIC] Signaling trickle support
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Date: Thu, 8 Nov 2012 14:23:19 +0000
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Subject: Re: [MMUSIC] Signaling trickle support
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+1

Regards,

Christer

-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 Miguel A. Garcia
Sent: 8. marraskuuta 2012 9:30
To: Paul Kyzivat; Martin Thomson; Emil Ivov
Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] Signaling trickle support

As an individual...

I concur with Paul, changing the v=3D version in the SDP is a very bad idea=
.

Let met quote RFC 4566:


    The "v=3D" field gives the version of the Session Description Protocol.
    This memo defines version 0.  There is no minor version number.


This means that if you define another version different than zero, then, as=
 a side effect, you are creating a different version of SDP than the one sp=
ecified in RFC 4566. So, you will need to create an RFC specifying how such=
 version of SDP works.

/Miguel

On 07/11/2012 20:27, Paul Kyzivat wrote:
> On 11/7/12 8:59 AM, Martin Thomson wrote:
>> Matthew, Justin and I were discussing the problem of signaling=20
>> trickle ICE support and the following proposal was made:
>>
>> The 24th bit in SDP is currently set to zero.  This bit is=20
>> specifically reserved for forward compatibility purposes.  We can use=20
>> this bit.  Set this bit to one to indicate support for trickle ICE.
>>
>> This meets the "works if supported, fails if it doesn't criteria".
>>
>> This solution may also be appropriate for bundle/mmt/whatever.
>
> After some side discussion, I guess you are talking about bits in the=20
> value of the v=3D field ???
>
> If so, I still have no idea what you are talking about. The syntax for=20
> the value of v=3D is 1*DIGIT. You can choose to represent that in binary=
=20
> and then choose bit 24 if you want, but that has no significance in 4566.
>
> Assigning a new SDP version to signal Trickle ICE support seems like=20
> an incredibly bad idea.
>
>      Thanks,
>      Paul
>
> _______________________________________________
> 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

From keith.drage@alcatel-lucent.com  Thu Nov  8 06:31:10 2012
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>
Date: Thu, 8 Nov 2012 15:30:37 +0100
Thread-Topic: Comments on draft-ietf-mmusic-sdp-cs-13
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Subject: [MMUSIC] Comments on draft-ietf-mmusic-sdp-cs-13
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Neither of these comments are much beyond editorial. Just discovered while =
working through the document.

1) 5.6.3 of draft-ietf-mmusic-sdp-cs-13 contains the text:

	o Note that it if delivery of the Answer is delayed for some reason,	the c=
ircuit-switched call attempt may arrive at the Offerer before	the Answer ha=
s been processed. In this case, since the=09
	correlation mechanisms are negotiated as part of the Offer/Answer=09
	exchange, the Answerer cannot know whether or not the incoming=09
	circuit-switched call attempt is correlated with the session being	negotia=
ted, the Offerer SHOULD answer the circuit-switched call=09
	attempt only after it has received and processed the Answer.=09

Can we please not start a sentence that contains the condition for implemen=
ting a normative requirement with the words "Note that".

2) Section 5.6 is the only part that indicates a dependence on SDP offer/an=
swer as defined in RFC 3264. There are other normative parts of this docume=
nt that are at least based on the prerequisite that RFC 3264 is being used.=
 The normative reference would appear to need to be more general. Does any =
of this document have a life outside RFC 3264?

Regards

Keith
		=09

From fandreas@cisco.com  Thu Nov  8 07:01:55 2012
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] Signaling trickle support
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This is very different from the way we normally handle extensions in SDP 
both from a pure signaling and negotiation point of view. I share the 
sentiment of others that this does not look like a good idea but would 
like to understand if there were other considerations for suggesting 
this radically different approach.

Thanks

-- Flemming

On 11/8/12 9:23 AM, Christer Holmberg wrote:
> +1
>
> Regards,
>
> Christer
>
> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of Miguel A. Garcia
> Sent: 8. marraskuuta 2012 9:30
> To: Paul Kyzivat; Martin Thomson; Emil Ivov
> Cc: mmusic@ietf.org
> Subject: Re: [MMUSIC] Signaling trickle support
>
> As an individual...
>
> I concur with Paul, changing the v= version in the SDP is a very bad idea.
>
> Let met quote RFC 4566:
>
>
>      The "v=" field gives the version of the Session Description Protocol.
>      This memo defines version 0.  There is no minor version number.
>
>
> This means that if you define another version different than zero, then, as a side effect, you are creating a different version of SDP than the one specified in RFC 4566. So, you will need to create an RFC specifying how such version of SDP works.
>
> /Miguel
>
> On 07/11/2012 20:27, Paul Kyzivat wrote:
>> On 11/7/12 8:59 AM, Martin Thomson wrote:
>>> Matthew, Justin and I were discussing the problem of signaling
>>> trickle ICE support and the following proposal was made:
>>>
>>> The 24th bit in SDP is currently set to zero.  This bit is
>>> specifically reserved for forward compatibility purposes.  We can use
>>> this bit.  Set this bit to one to indicate support for trickle ICE.
>>>
>>> This meets the "works if supported, fails if it doesn't criteria".
>>>
>>> This solution may also be appropriate for bundle/mmt/whatever.
>> After some side discussion, I guess you are talking about bits in the
>> value of the v= field ???
>>
>> If so, I still have no idea what you are talking about. The syntax for
>> the value of v= is 1*DIGIT. You can choose to represent that in binary
>> and then choose bit 24 if you want, but that has no significance in 4566.
>>
>> Assigning a new SDP version to signal Trickle ICE support seems like
>> an incredibly bad idea.
>>
>>       Thanks,
>>       Paul
>>
>> _______________________________________________
>> 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
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
> .
>


From emil@sip-communicator.org  Thu Nov  8 07:13:04 2012
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References: <CABkgnnVR-prV8xgzPx9H-MYRM_vTui-8GGfTD0yy1JPvJJzrrQ@mail.gmail.com> <509AB627.1090507@alum.mit.edu> <509B5F87.1010807@ericsson.com> <7594FB04B1934943A5C02806D1A2204B027800@ESESSMB209.ericsson.se> <509BC947.709@cisco.com>
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] Signaling trickle support
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I actually think this was supposed to be a joke. At least that's how I
got it.

Emil

On 08.11.12, 10:01, Flemming Andreasen wrote:
> This is very different from the way we normally handle extensions in SDP 
> both from a pure signaling and negotiation point of view. I share the 
> sentiment of others that this does not look like a good idea but would 
> like to understand if there were other considerations for suggesting 
> this radically different approach.
> 
> Thanks
> 
> -- Flemming
> 
> On 11/8/12 9:23 AM, Christer Holmberg wrote:
>> +1
>>
>> Regards,
>>
>> Christer
>>
>> -----Original Message-----
>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of Miguel A. Garcia
>> Sent: 8. marraskuuta 2012 9:30
>> To: Paul Kyzivat; Martin Thomson; Emil Ivov
>> Cc: mmusic@ietf.org
>> Subject: Re: [MMUSIC] Signaling trickle support
>>
>> As an individual...
>>
>> I concur with Paul, changing the v= version in the SDP is a very bad idea.
>>
>> Let met quote RFC 4566:
>>
>>
>>      The "v=" field gives the version of the Session Description Protocol.
>>      This memo defines version 0.  There is no minor version number.
>>
>>
>> This means that if you define another version different than zero, then, as a side effect, you are creating a different version of SDP than the one specified in RFC 4566. So, you will need to create an RFC specifying how such version of SDP works.
>>
>> /Miguel
>>
>> On 07/11/2012 20:27, Paul Kyzivat wrote:
>>> On 11/7/12 8:59 AM, Martin Thomson wrote:
>>>> Matthew, Justin and I were discussing the problem of signaling
>>>> trickle ICE support and the following proposal was made:
>>>>
>>>> The 24th bit in SDP is currently set to zero.  This bit is
>>>> specifically reserved for forward compatibility purposes.  We can use
>>>> this bit.  Set this bit to one to indicate support for trickle ICE.
>>>>
>>>> This meets the "works if supported, fails if it doesn't criteria".
>>>>
>>>> This solution may also be appropriate for bundle/mmt/whatever.
>>> After some side discussion, I guess you are talking about bits in the
>>> value of the v= field ???
>>>
>>> If so, I still have no idea what you are talking about. The syntax for
>>> the value of v= is 1*DIGIT. You can choose to represent that in binary
>>> and then choose bit 24 if you want, but that has no significance in 4566.
>>>
>>> Assigning a new SDP version to signal Trickle ICE support seems like
>>> an incredibly bad idea.
>>>
>>>       Thanks,
>>>       Paul
>>>
>>> _______________________________________________
>>> 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
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>> .
>>
> 
> 

-- 
https://jitsi.org

From martin.thomson@gmail.com  Thu Nov  8 07:46:52 2012
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Date: Thu, 8 Nov 2012 10:46:49 -0500
Message-ID: <CABkgnnVMf53Vk_Rd-0erZN4ZLDKM4ePOvMfLJnfOUNVxdWYLqQ@mail.gmail.com>
From: Martin Thomson <martin.thomson@gmail.com>
To: Emil Ivov <emcho@jitsi.org>
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Cc: Flemming Andreasen <fandreas@cisco.com>, "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] Signaling trickle support
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Trolled.

Though to be honest, I don't see a way around this particular problem.
 The idea of using a Require: trickle header is fraught with problems;
the idea that you might use an a= line doesn't give you the
characteristics you desire.  I just don't know how to cause the
failure that is needed when this isn't supported.

Besides, the failure modes all lead to an expenditure of time that is
worse than the simple candidate gathering scenarios (reflexive,
relay), though not "other" gathering methods (UPnP, etc...).

half-trickle is a reasonable solution.

On 8 November 2012 10:12, Emil Ivov <emcho@jitsi.org> wrote:
> I actually think this was supposed to be a joke. At least that's how I
> got it.
>
> Emil
>
> On 08.11.12, 10:01, Flemming Andreasen wrote:
>> This is very different from the way we normally handle extensions in SDP
>> both from a pure signaling and negotiation point of view. I share the
>> sentiment of others that this does not look like a good idea but would
>> like to understand if there were other considerations for suggesting
>> this radically different approach.
>>
>> Thanks
>>
>> -- Flemming
>>
>> On 11/8/12 9:23 AM, Christer Holmberg wrote:
>>> +1
>>>
>>> Regards,
>>>
>>> Christer
>>>
>>> -----Original Message-----
>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of Miguel A. Garcia
>>> Sent: 8. marraskuuta 2012 9:30
>>> To: Paul Kyzivat; Martin Thomson; Emil Ivov
>>> Cc: mmusic@ietf.org
>>> Subject: Re: [MMUSIC] Signaling trickle support
>>>
>>> As an individual...
>>>
>>> I concur with Paul, changing the v= version in the SDP is a very bad idea.
>>>
>>> Let met quote RFC 4566:
>>>
>>>
>>>      The "v=" field gives the version of the Session Description Protocol.
>>>      This memo defines version 0.  There is no minor version number.
>>>
>>>
>>> This means that if you define another version different than zero, then, as a side effect, you are creating a different version of SDP than the one specified in RFC 4566. So, you will need to create an RFC specifying how such version of SDP works.
>>>
>>> /Miguel
>>>
>>> On 07/11/2012 20:27, Paul Kyzivat wrote:
>>>> On 11/7/12 8:59 AM, Martin Thomson wrote:
>>>>> Matthew, Justin and I were discussing the problem of signaling
>>>>> trickle ICE support and the following proposal was made:
>>>>>
>>>>> The 24th bit in SDP is currently set to zero.  This bit is
>>>>> specifically reserved for forward compatibility purposes.  We can use
>>>>> this bit.  Set this bit to one to indicate support for trickle ICE.
>>>>>
>>>>> This meets the "works if supported, fails if it doesn't criteria".
>>>>>
>>>>> This solution may also be appropriate for bundle/mmt/whatever.
>>>> After some side discussion, I guess you are talking about bits in the
>>>> value of the v= field ???
>>>>
>>>> If so, I still have no idea what you are talking about. The syntax for
>>>> the value of v= is 1*DIGIT. You can choose to represent that in binary
>>>> and then choose bit 24 if you want, but that has no significance in 4566.
>>>>
>>>> Assigning a new SDP version to signal Trickle ICE support seems like
>>>> an incredibly bad idea.
>>>>
>>>>       Thanks,
>>>>       Paul
>>>>
>>>> _______________________________________________
>>>> 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
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>> .
>>>
>>
>>
>
> --
> https://jitsi.org

From fandreas@cisco.com  Thu Nov  8 07:53:10 2012
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Subject: Re: [MMUSIC] Signaling trickle support
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On 11/8/12 10:46 AM, Martin Thomson wrote:
> Trolled.
>
> Though to be honest, I don't see a way around this particular problem.
>   The idea of using a Require: trickle header is fraught with problems;
> the idea that you might use an a= line doesn't give you the
> characteristics you desire.  I just don't know how to cause the
> failure that is needed when this isn't supported.
Can we capture the desired characteristics and failure mode for starters ?

> Besides, the failure modes all lead to an expenditure of time that is
> worse than the simple candidate gathering scenarios (reflexive,
> relay), though not "other" gathering methods (UPnP, etc...).
Is there a requirement here that needs to be written down ?

> half-trickle is a reasonable solution.
Sorry - what's half-trickle ?

-- Flemming


> On 8 November 2012 10:12, Emil Ivov <emcho@jitsi.org> wrote:
>> I actually think this was supposed to be a joke. At least that's how I
>> got it.
>>
>> Emil
>>
>> On 08.11.12, 10:01, Flemming Andreasen wrote:
>>> This is very different from the way we normally handle extensions in SDP
>>> both from a pure signaling and negotiation point of view. I share the
>>> sentiment of others that this does not look like a good idea but would
>>> like to understand if there were other considerations for suggesting
>>> this radically different approach.
>>>
>>> Thanks
>>>
>>> -- Flemming
>>>
>>> On 11/8/12 9:23 AM, Christer Holmberg wrote:
>>>> +1
>>>>
>>>> Regards,
>>>>
>>>> Christer
>>>>
>>>> -----Original Message-----
>>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of Miguel A. Garcia
>>>> Sent: 8. marraskuuta 2012 9:30
>>>> To: Paul Kyzivat; Martin Thomson; Emil Ivov
>>>> Cc: mmusic@ietf.org
>>>> Subject: Re: [MMUSIC] Signaling trickle support
>>>>
>>>> As an individual...
>>>>
>>>> I concur with Paul, changing the v= version in the SDP is a very bad idea.
>>>>
>>>> Let met quote RFC 4566:
>>>>
>>>>
>>>>       The "v=" field gives the version of the Session Description Protocol.
>>>>       This memo defines version 0.  There is no minor version number.
>>>>
>>>>
>>>> This means that if you define another version different than zero, then, as a side effect, you are creating a different version of SDP than the one specified in RFC 4566. So, you will need to create an RFC specifying how such version of SDP works.
>>>>
>>>> /Miguel
>>>>
>>>> On 07/11/2012 20:27, Paul Kyzivat wrote:
>>>>> On 11/7/12 8:59 AM, Martin Thomson wrote:
>>>>>> Matthew, Justin and I were discussing the problem of signaling
>>>>>> trickle ICE support and the following proposal was made:
>>>>>>
>>>>>> The 24th bit in SDP is currently set to zero.  This bit is
>>>>>> specifically reserved for forward compatibility purposes.  We can use
>>>>>> this bit.  Set this bit to one to indicate support for trickle ICE.
>>>>>>
>>>>>> This meets the "works if supported, fails if it doesn't criteria".
>>>>>>
>>>>>> This solution may also be appropriate for bundle/mmt/whatever.
>>>>> After some side discussion, I guess you are talking about bits in the
>>>>> value of the v= field ???
>>>>>
>>>>> If so, I still have no idea what you are talking about. The syntax for
>>>>> the value of v= is 1*DIGIT. You can choose to represent that in binary
>>>>> and then choose bit 24 if you want, but that has no significance in 4566.
>>>>>
>>>>> Assigning a new SDP version to signal Trickle ICE support seems like
>>>>> an incredibly bad idea.
>>>>>
>>>>>        Thanks,
>>>>>        Paul
>>>>>
>>>>> _______________________________________________
>>>>> 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
>>>> _______________________________________________
>>>> mmusic mailing list
>>>> mmusic@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>> .
>>>>
>>>
>> --
>> https://jitsi.org
> .
>


From emil@sip-communicator.org  Thu Nov  8 08:29:59 2012
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References: <CABkgnnVR-prV8xgzPx9H-MYRM_vTui-8GGfTD0yy1JPvJJzrrQ@mail.gmail.com> <509AB627.1090507@alum.mit.edu> <509B5F87.1010807@ericsson.com> <7594FB04B1934943A5C02806D1A2204B027800@ESESSMB209.ericsson.se> <509BC947.709@cisco.com> <509BCBFB.2020305@jitsi.org> <CABkgnnVMf53Vk_Rd-0erZN4ZLDKM4ePOvMfLJnfOUNVxdWYLqQ@mail.gmail.com> <509BD563.9080808@cisco.com>
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] Signaling trickle support
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Hey Flemming,

On 08.11.12, 10:53, Flemming Andreasen wrote:
> 
> On 11/8/12 10:46 AM, Martin Thomson wrote:
>> Trolled.
>>
>> Though to be honest, I don't see a way around this particular problem.
>>   The idea of using a Require: trickle header is fraught with problems;
>> the idea that you might use an a= line doesn't give you the
>> characteristics you desire.  I just don't know how to cause the
>> failure that is needed when this isn't supported.
> Can we capture the desired characteristics and failure mode for starters ?

The "really" desired characteristic is knowing in advance whether or not
the remote party can do trickle so that you know whether you can afford
to send an empty offer without causing it to freak out.

Since protocols like SIP don't really have things like XMPP disco and
Cap Neg that would have allowed it, we were considering other options.
One such option would be to send a trickle offer and format it in a way
that would cause a non-trickle answerer to reject it in a _predictable_
way. This would allow the offerer to retry with a regular ICE offer.

>> Besides, the failure modes all lead to an expenditure of time that is
>> worse than the simple candidate gathering scenarios (reflexive,
>> relay), though not "other" gathering methods (UPnP, etc...).
> Is there a requirement here that needs to be written down ?

The point that Martin is making is that sending an offer and then
waiting for the failure may potentially take longer than just gathering
your candidates. This is true especially in cases where the offerer is
only planning on using a single port, on a single interface, with a
single STUN/TURN server and no other candidate gathering mechanisms.

But then, if this really is the case, the offerer knows it and can
choose not to do the probing.

>> half-trickle is a reasonable solution.
> Sorry - what's half-trickle ?

We had that in the slides on Tuesday but it's not yet in the draft. The
idea is that if you can't confirm support for trickle in the answerer,
you just send a vanilla ICE offer and you also indicate you do support
trickle. A trickle ICE answerer can then trickle candidates and in
general you get half the optimizations of Trickle ICE. You even get more
than that in case the offerer is able to predictively gather candidates
before the call is initiated.

We'll describe this in the next version.

Cheers,
Emil

> 
> -- Flemming
> 
> 
>> On 8 November 2012 10:12, Emil Ivov <emcho@jitsi.org> wrote:
>>> I actually think this was supposed to be a joke. At least that's how I
>>> got it.
>>>
>>> Emil
>>>
>>> On 08.11.12, 10:01, Flemming Andreasen wrote:
>>>> This is very different from the way we normally handle extensions in SDP
>>>> both from a pure signaling and negotiation point of view. I share the
>>>> sentiment of others that this does not look like a good idea but would
>>>> like to understand if there were other considerations for suggesting
>>>> this radically different approach.
>>>>
>>>> Thanks
>>>>
>>>> -- Flemming
>>>>
>>>> On 11/8/12 9:23 AM, Christer Holmberg wrote:
>>>>> +1
>>>>>
>>>>> Regards,
>>>>>
>>>>> Christer
>>>>>
>>>>> -----Original Message-----
>>>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of Miguel A. Garcia
>>>>> Sent: 8. marraskuuta 2012 9:30
>>>>> To: Paul Kyzivat; Martin Thomson; Emil Ivov
>>>>> Cc: mmusic@ietf.org
>>>>> Subject: Re: [MMUSIC] Signaling trickle support
>>>>>
>>>>> As an individual...
>>>>>
>>>>> I concur with Paul, changing the v= version in the SDP is a very bad idea.
>>>>>
>>>>> Let met quote RFC 4566:
>>>>>
>>>>>
>>>>>       The "v=" field gives the version of the Session Description Protocol.
>>>>>       This memo defines version 0.  There is no minor version number.
>>>>>
>>>>>
>>>>> This means that if you define another version different than zero, then, as a side effect, you are creating a different version of SDP than the one specified in RFC 4566. So, you will need to create an RFC specifying how such version of SDP works.
>>>>>
>>>>> /Miguel
>>>>>
>>>>> On 07/11/2012 20:27, Paul Kyzivat wrote:
>>>>>> On 11/7/12 8:59 AM, Martin Thomson wrote:
>>>>>>> Matthew, Justin and I were discussing the problem of signaling
>>>>>>> trickle ICE support and the following proposal was made:
>>>>>>>
>>>>>>> The 24th bit in SDP is currently set to zero.  This bit is
>>>>>>> specifically reserved for forward compatibility purposes.  We can use
>>>>>>> this bit.  Set this bit to one to indicate support for trickle ICE.
>>>>>>>
>>>>>>> This meets the "works if supported, fails if it doesn't criteria".
>>>>>>>
>>>>>>> This solution may also be appropriate for bundle/mmt/whatever.
>>>>>> After some side discussion, I guess you are talking about bits in the
>>>>>> value of the v= field ???
>>>>>>
>>>>>> If so, I still have no idea what you are talking about. The syntax for
>>>>>> the value of v= is 1*DIGIT. You can choose to represent that in binary
>>>>>> and then choose bit 24 if you want, but that has no significance in 4566.
>>>>>>
>>>>>> Assigning a new SDP version to signal Trickle ICE support seems like
>>>>>> an incredibly bad idea.
>>>>>>
>>>>>>        Thanks,
>>>>>>        Paul
>>>>>>
>>>>>> _______________________________________________
>>>>>> 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
>>>>> _______________________________________________
>>>>> mmusic mailing list
>>>>> mmusic@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>>> .
>>>>>
>>>>
>>> --
>>> https://jitsi.org
>> .
>>
> 
> 

-- 
https://jitsi.org

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References: <CABkgnnVR-prV8xgzPx9H-MYRM_vTui-8GGfTD0yy1JPvJJzrrQ@mail.gmail.com> <509AB627.1090507@alum.mit.edu> <509B5F87.1010807@ericsson.com> <7594FB04B1934943A5C02806D1A2204B027800@ESESSMB209.ericsson.se> <509BC947.709@cisco.com> <509BCBFB.2020305@jitsi.org> <CABkgnnVMf53Vk_Rd-0erZN4ZLDKM4ePOvMfLJnfOUNVxdWYLqQ@mail.gmail.com> <509BD563.9080808@cisco.com> <509BDE01.9060705@jitsi.org>
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] Signaling trickle support
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On 11/8/12 11:29 AM, Emil Ivov wrote:
> Hey Flemming,
>
> On 08.11.12, 10:53, Flemming Andreasen wrote:
>> On 11/8/12 10:46 AM, Martin Thomson wrote:
>>> Trolled.
>>>
>>> Though to be honest, I don't see a way around this particular problem.
>>>    The idea of using a Require: trickle header is fraught with problems;
>>> the idea that you might use an a= line doesn't give you the
>>> characteristics you desire.  I just don't know how to cause the
>>> failure that is needed when this isn't supported.
>> Can we capture the desired characteristics and failure mode for starters ?
> The "really" desired characteristic is knowing in advance whether or not
> the remote party can do trickle so that you know whether you can afford
> to send an empty offer without causing it to freak out.
Why would you send an empty offer ?

> Since protocols like SIP don't really have things like XMPP disco and
> Cap Neg that would have allowed it, we were considering other options.
> One such option would be to send a trickle offer and format it in a way
> that would cause a non-trickle answerer to reject it in a _predictable_
> way. This would allow the offerer to retry with a regular ICE offer.
I thought the point with trickle-ICE was to reduce call setup latency 
and hence you would want to make sure that things complete in as few 
round-trips as possible. This doesn't seem to line up with that (?).

>
>>> Besides, the failure modes all lead to an expenditure of time that is
>>> worse than the simple candidate gathering scenarios (reflexive,
>>> relay), though not "other" gathering methods (UPnP, etc...).
>> Is there a requirement here that needs to be written down ?
> The point that Martin is making is that sending an offer and then
> waiting for the failure may potentially take longer than just gathering
> your candidates. This is true especially in cases where the offerer is
> only planning on using a single port, on a single interface, with a
> single STUN/TURN server and no other candidate gathering mechanisms.
I would agree with that.
>
> But then, if this really is the case, the offerer knows it and can
> choose not to do the probing.
Agreed.
>>> half-trickle is a reasonable solution.
>> Sorry - what's half-trickle ?
> We had that in the slides on Tuesday but it's not yet in the draft. The
> idea is that if you can't confirm support for trickle in the answerer,
Do you mean: if you don't know if the answerer supports it before 
sending an offer to him [since confirming support in the answer should 
be easy]  ?
> you just send a vanilla ICE offer and you also indicate you do support
> trickle. A trickle ICE answerer can then trickle candidates and in
> general you get half the optimizations of Trickle ICE. You even get more
> than that in case the offerer is able to predictively gather candidates
> before the call is initiated.
Agreed.
>
> We'll describe this in the next version.
Ok - thanks for the explanation.

-- Flemming
>
> Cheers,
> Emil
>
>> -- Flemming
>>
>>
>>> On 8 November 2012 10:12, Emil Ivov <emcho@jitsi.org> wrote:
>>>> I actually think this was supposed to be a joke. At least that's how I
>>>> got it.
>>>>
>>>> Emil
>>>>
>>>> On 08.11.12, 10:01, Flemming Andreasen wrote:
>>>>> This is very different from the way we normally handle extensions in SDP
>>>>> both from a pure signaling and negotiation point of view. I share the
>>>>> sentiment of others that this does not look like a good idea but would
>>>>> like to understand if there were other considerations for suggesting
>>>>> this radically different approach.
>>>>>
>>>>> Thanks
>>>>>
>>>>> -- Flemming
>>>>>
>>>>> On 11/8/12 9:23 AM, Christer Holmberg wrote:
>>>>>> +1
>>>>>>
>>>>>> Regards,
>>>>>>
>>>>>> Christer
>>>>>>
>>>>>> -----Original Message-----
>>>>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of Miguel A. Garcia
>>>>>> Sent: 8. marraskuuta 2012 9:30
>>>>>> To: Paul Kyzivat; Martin Thomson; Emil Ivov
>>>>>> Cc: mmusic@ietf.org
>>>>>> Subject: Re: [MMUSIC] Signaling trickle support
>>>>>>
>>>>>> As an individual...
>>>>>>
>>>>>> I concur with Paul, changing the v= version in the SDP is a very bad idea.
>>>>>>
>>>>>> Let met quote RFC 4566:
>>>>>>
>>>>>>
>>>>>>        The "v=" field gives the version of the Session Description Protocol.
>>>>>>        This memo defines version 0.  There is no minor version number.
>>>>>>
>>>>>>
>>>>>> This means that if you define another version different than zero, then, as a side effect, you are creating a different version of SDP than the one specified in RFC 4566. So, you will need to create an RFC specifying how such version of SDP works.
>>>>>>
>>>>>> /Miguel
>>>>>>
>>>>>> On 07/11/2012 20:27, Paul Kyzivat wrote:
>>>>>>> On 11/7/12 8:59 AM, Martin Thomson wrote:
>>>>>>>> Matthew, Justin and I were discussing the problem of signaling
>>>>>>>> trickle ICE support and the following proposal was made:
>>>>>>>>
>>>>>>>> The 24th bit in SDP is currently set to zero.  This bit is
>>>>>>>> specifically reserved for forward compatibility purposes.  We can use
>>>>>>>> this bit.  Set this bit to one to indicate support for trickle ICE.
>>>>>>>>
>>>>>>>> This meets the "works if supported, fails if it doesn't criteria".
>>>>>>>>
>>>>>>>> This solution may also be appropriate for bundle/mmt/whatever.
>>>>>>> After some side discussion, I guess you are talking about bits in the
>>>>>>> value of the v= field ???
>>>>>>>
>>>>>>> If so, I still have no idea what you are talking about. The syntax for
>>>>>>> the value of v= is 1*DIGIT. You can choose to represent that in binary
>>>>>>> and then choose bit 24 if you want, but that has no significance in 4566.
>>>>>>>
>>>>>>> Assigning a new SDP version to signal Trickle ICE support seems like
>>>>>>> an incredibly bad idea.
>>>>>>>
>>>>>>>         Thanks,
>>>>>>>         Paul
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> 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
>>>>>> _______________________________________________
>>>>>> mmusic mailing list
>>>>>> mmusic@ietf.org
>>>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>>>> .
>>>>>>
>>>> --
>>>> https://jitsi.org
>>> .
>>>
>>


From emil@sip-communicator.org  Thu Nov  8 11:25:26 2012
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References: <CABkgnnVR-prV8xgzPx9H-MYRM_vTui-8GGfTD0yy1JPvJJzrrQ@mail.gmail.com> <509AB627.1090507@alum.mit.edu> <509B5F87.1010807@ericsson.com> <7594FB04B1934943A5C02806D1A2204B027800@ESESSMB209.ericsson.se> <509BC947.709@cisco.com> <509BCBFB.2020305@jitsi.org> <CABkgnnVMf53Vk_Rd-0erZN4ZLDKM4ePOvMfLJnfOUNVxdWYLqQ@mail.gmail.com> <509BD563.9080808@cisco.com> <509BDE01.9060705@jitsi.org> <509C02F2.10303@cisco.com>
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] Signaling trickle support
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Hey Flemming,

On 08.11.12, 14:07, Flemming Andreasen wrote:
> 
> On 11/8/12 11:29 AM, Emil Ivov wrote:
>> Hey Flemming,
>>
>> On 08.11.12, 10:53, Flemming Andreasen wrote:
>>> On 11/8/12 10:46 AM, Martin Thomson wrote:
>>>> Trolled.
>>>>
>>>> Though to be honest, I don't see a way around this particular problem.
>>>>    The idea of using a Require: trickle header is fraught with problems;
>>>> the idea that you might use an a= line doesn't give you the
>>>> characteristics you desire.  I just don't know how to cause the
>>>> failure that is needed when this isn't supported.
>>> Can we capture the desired characteristics and failure mode for starters ?
>> The "really" desired characteristic is knowing in advance whether or not
>> the remote party can do trickle so that you know whether you can afford
>> to send an empty offer without causing it to freak out.
> Why would you send an empty offer ?

Normally the offer should contain only candidates that are readily
available (such as host candidates). However if the offerer does not
wish to disclose those for privacy reasons, then it would just send an
offer with no candidates. This is for example how Chrome currently behaves.

It also places 0.0.0.0:1 in the c= and m= lines.

That's currently explain in section 4 of the draft:

http://tools.ietf.org/html/draft-rescorla-mmusic-ice-trickle-00#section-4


>> Since protocols like SIP don't really have things like XMPP disco and
>> Cap Neg that would have allowed it, we were considering other options.
>> One such option would be to send a trickle offer and format it in a way
>> that would cause a non-trickle answerer to reject it in a _predictable_
>> way. This would allow the offerer to retry with a regular ICE offer.
> I thought the point with trickle-ICE was to reduce call setup latency

It is.

> and hence you would want to make sure that things complete in as few 
> round-trips as possible. This doesn't seem to line up with that (?).

The thing is that in cases where trickle ICE support cannot be
pre-confirmed you have to fall back to half trickle. The probing
requests would hence allow for an optimisation compared to that.

With half-trickle the offerer has to gather all candidates anyway so,
while doing so, they might as well try and check if they really need to
complete it before sending their first offer.

>>>> Besides, the failure modes all lead to an expenditure of time that is
>>>> worse than the simple candidate gathering scenarios (reflexive,
>>>> relay), though not "other" gathering methods (UPnP, etc...).
>>> Is there a requirement here that needs to be written down ?
>> The point that Martin is making is that sending an offer and then
>> waiting for the failure may potentially take longer than just gathering
>> your candidates. This is true especially in cases where the offerer is
>> only planning on using a single port, on a single interface, with a
>> single STUN/TURN server and no other candidate gathering mechanisms.
> I would agree with that.
>>
>> But then, if this really is the case, the offerer knows it and can
>> choose not to do the probing.
> Agreed.
>>>> half-trickle is a reasonable solution.
>>> Sorry - what's half-trickle ?
>> We had that in the slides on Tuesday but it's not yet in the draft. The
>> idea is that if you can't confirm support for trickle in the answerer,
> Do you mean: if you don't know if the answerer supports it before 
> sending an offer to him 

Yes. That's what I meant. The offerer needs to know that in advance in
order to decide how to form their offer.

Cheers,
Emil

> [since confirming support in the answer should 
> be easy]  ?
>> you just send a vanilla ICE offer and you also indicate you do support
>> trickle. A trickle ICE answerer can then trickle candidates and in
>> general you get half the optimizations of Trickle ICE. You even get more
>> than that in case the offerer is able to predictively gather candidates
>> before the call is initiated.
> Agreed.
>>
>> We'll describe this in the next version.
> Ok - thanks for the explanation.
> 
> -- Flemming
>>
>> Cheers,
>> Emil
>>
>>> -- Flemming
>>>
>>>
>>>> On 8 November 2012 10:12, Emil Ivov <emcho@jitsi.org> wrote:
>>>>> I actually think this was supposed to be a joke. At least that's how I
>>>>> got it.
>>>>>
>>>>> Emil
>>>>>
>>>>> On 08.11.12, 10:01, Flemming Andreasen wrote:
>>>>>> This is very different from the way we normally handle extensions in SDP
>>>>>> both from a pure signaling and negotiation point of view. I share the
>>>>>> sentiment of others that this does not look like a good idea but would
>>>>>> like to understand if there were other considerations for suggesting
>>>>>> this radically different approach.
>>>>>>
>>>>>> Thanks
>>>>>>
>>>>>> -- Flemming
>>>>>>
>>>>>> On 11/8/12 9:23 AM, Christer Holmberg wrote:
>>>>>>> +1
>>>>>>>
>>>>>>> Regards,
>>>>>>>
>>>>>>> Christer
>>>>>>>
>>>>>>> -----Original Message-----
>>>>>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of Miguel A. Garcia
>>>>>>> Sent: 8. marraskuuta 2012 9:30
>>>>>>> To: Paul Kyzivat; Martin Thomson; Emil Ivov
>>>>>>> Cc: mmusic@ietf.org
>>>>>>> Subject: Re: [MMUSIC] Signaling trickle support
>>>>>>>
>>>>>>> As an individual...
>>>>>>>
>>>>>>> I concur with Paul, changing the v= version in the SDP is a very bad idea.
>>>>>>>
>>>>>>> Let met quote RFC 4566:
>>>>>>>
>>>>>>>
>>>>>>>        The "v=" field gives the version of the Session Description Protocol.
>>>>>>>        This memo defines version 0.  There is no minor version number.
>>>>>>>
>>>>>>>
>>>>>>> This means that if you define another version different than zero, then, as a side effect, you are creating a different version of SDP than the one specified in RFC 4566. So, you will need to create an RFC specifying how such version of SDP works.
>>>>>>>
>>>>>>> /Miguel
>>>>>>>
>>>>>>> On 07/11/2012 20:27, Paul Kyzivat wrote:
>>>>>>>> On 11/7/12 8:59 AM, Martin Thomson wrote:
>>>>>>>>> Matthew, Justin and I were discussing the problem of signaling
>>>>>>>>> trickle ICE support and the following proposal was made:
>>>>>>>>>
>>>>>>>>> The 24th bit in SDP is currently set to zero.  This bit is
>>>>>>>>> specifically reserved for forward compatibility purposes.  We can use
>>>>>>>>> this bit.  Set this bit to one to indicate support for trickle ICE.
>>>>>>>>>
>>>>>>>>> This meets the "works if supported, fails if it doesn't criteria".
>>>>>>>>>
>>>>>>>>> This solution may also be appropriate for bundle/mmt/whatever.
>>>>>>>> After some side discussion, I guess you are talking about bits in the
>>>>>>>> value of the v= field ???
>>>>>>>>
>>>>>>>> If so, I still have no idea what you are talking about. The syntax for
>>>>>>>> the value of v= is 1*DIGIT. You can choose to represent that in binary
>>>>>>>> and then choose bit 24 if you want, but that has no significance in 4566.
>>>>>>>>
>>>>>>>> Assigning a new SDP version to signal Trickle ICE support seems like
>>>>>>>> an incredibly bad idea.
>>>>>>>>
>>>>>>>>         Thanks,
>>>>>>>>         Paul
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> 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
>>>>>>> _______________________________________________
>>>>>>> mmusic mailing list
>>>>>>> mmusic@ietf.org
>>>>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>>>>> .
>>>>>>>
>>>>> --
>>>>> https://jitsi.org
>>>> .
>>>>
>>>
> 
> 

-- 
https://jitsi.org

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Subject: Re: [MMUSIC] Signaling trickle support
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On Nov 8, 2012, at 2:25 PM, Emil Ivov <emcho@jitsi.org> wrote:

> Hey Flemming,
>=20
> On 08.11.12, 14:07, Flemming Andreasen wrote:
>>>=20
>> Why would you send an empty offer ?
>=20
> Normally the offer should contain only candidates that are readily
> available (such as host candidates). However if the offerer does not
> wish to disclose those for privacy reasons, then it would just send an
> offer with no candidates. This is for example how Chrome currently behave=
s.
>=20
> It also places 0.0.0.0:1 in the c=3D and m=3D lines.
>=20
> That's currently explain in section 4 of the draft:
>=20
> http://tools.ietf.org/html/draft-rescorla-mmusic-ice-trickle-00#section-4
>=20

For SIP, why not just send an offerless INVITE? (it might even work sometim=
es ;)

Likewise for RTCWeb, send a request for a session without SDP whatsoever, a=
nd make the other side send an offer first (and write your JS to support th=
at case).

-hadriel


From emil@sip-communicator.org  Thu Nov  8 11:48:19 2012
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Subject: Re: [MMUSIC] Signaling trickle support
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Hey Hadriel,

On 08.11.12, 14:34, Hadriel Kaplan wrote:
> 
> 
> On Nov 8, 2012, at 2:25 PM, Emil Ivov <emcho@jitsi.org> wrote:
> 
>> Hey Flemming,
>>
>> On 08.11.12, 14:07, Flemming Andreasen wrote:
>>>>
>>> Why would you send an empty offer ?
>>
>> Normally the offer should contain only candidates that are readily
>> available (such as host candidates). However if the offerer does not
>> wish to disclose those for privacy reasons, then it would just send an
>> offer with no candidates. This is for example how Chrome currently behaves.
>>
>> It also places 0.0.0.0:1 in the c= and m= lines.
>>
>> That's currently explain in section 4 of the draft:
>>
>> http://tools.ietf.org/html/draft-rescorla-mmusic-ice-trickle-00#section-4
>>
> 
> For SIP, why not just send an offerless INVITE? (it might even work sometimes ;)
> 
> Likewise for RTCWeb, send a request for a session without SDP whatsoever, and make the other side send an offer first (and write your JS to support that case).

I was thinking along those lines at one point too:

http://www.ietf.org/mail-archive/web/mmusic/current/msg09681.html

Christer made a comment that this would work badly with legacy:

http://www.ietf.org/mail-archive/web/mmusic/current/msg09698.html

And besides, that would also prevent you from letting the offerer (and
the calling user) choose what kind of call/content they'd like to have
(i.e. audio or video or a conference or a slide show, etc).

Cheers,
Emil


-- 
https://jitsi.org

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Subject: Re: [MMUSIC] Signaling trickle support
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On 11/8/12 2:25 PM, Emil Ivov wrote:
> Hey Flemming,
>
> On 08.11.12, 14:07, Flemming Andreasen wrote:
>> On 11/8/12 11:29 AM, Emil Ivov wrote:
>>> Hey Flemming,
>>>
>>> On 08.11.12, 10:53, Flemming Andreasen wrote:
>>>> On 11/8/12 10:46 AM, Martin Thomson wrote:
>>>>> Trolled.
>>>>>
>>>>> Though to be honest, I don't see a way around this particular problem.
>>>>>     The idea of using a Require: trickle header is fraught with problems;
>>>>> the idea that you might use an a= line doesn't give you the
>>>>> characteristics you desire.  I just don't know how to cause the
>>>>> failure that is needed when this isn't supported.
>>>> Can we capture the desired characteristics and failure mode for starters ?
>>> The "really" desired characteristic is knowing in advance whether or not
>>> the remote party can do trickle so that you know whether you can afford
>>> to send an empty offer without causing it to freak out.
>> Why would you send an empty offer ?
> Normally the offer should contain only candidates that are readily
> available (such as host candidates). However if the offerer does not
> wish to disclose those for privacy reasons, then it would just send an
> offer with no candidates. This is for example how Chrome currently behaves.
>
> It also places 0.0.0.0:1 in the c= and m= lines.
>
> That's currently explain in section 4 of the draft:
>
> http://tools.ietf.org/html/draft-rescorla-mmusic-ice-trickle-00#section-4
Ok (that's not an empty offer though).

>
>>> Since protocols like SIP don't really have things like XMPP disco and
>>> Cap Neg that would have allowed it, we were considering other options.
>>> One such option would be to send a trickle offer and format it in a way
>>> that would cause a non-trickle answerer to reject it in a _predictable_
>>> way. This would allow the offerer to retry with a regular ICE offer.
>> I thought the point with trickle-ICE was to reduce call setup latency
> It is.
>
>> and hence you would want to make sure that things complete in as few
>> round-trips as possible. This doesn't seem to line up with that (?).
> The thing is that in cases where trickle ICE support cannot be
> pre-confirmed you have to fall back to half trickle. The probing
> requests would hence allow for an optimisation compared to that.
Somewhat unclear to me as it builds upon an assumption that it is 
somehow "more efficient"to probe than to collect candidates. Clarifying 
the criteria for something being "more efficient" would also be helpful 
here (what is it truly that we are after; setup delay reduction, fewer 
NAT bindings, ...)

> With half-trickle the offerer has to gather all candidates anyway so,
> while doing so, they might as well try and check if they really need to
> complete it before sending their first offer.
Unclear to me since you can collect candidates before you know who to 
send the offer to.

Thanks

-- Flemming
>>>>> Besides, the failure modes all lead to an expenditure of time that is
>>>>> worse than the simple candidate gathering scenarios (reflexive,
>>>>> relay), though not "other" gathering methods (UPnP, etc...).
>>>> Is there a requirement here that needs to be written down ?
>>> The point that Martin is making is that sending an offer and then
>>> waiting for the failure may potentially take longer than just gathering
>>> your candidates. This is true especially in cases where the offerer is
>>> only planning on using a single port, on a single interface, with a
>>> single STUN/TURN server and no other candidate gathering mechanisms.
>> I would agree with that.
>>> But then, if this really is the case, the offerer knows it and can
>>> choose not to do the probing.
>> Agreed.
>>>>> half-trickle is a reasonable solution.
>>>> Sorry - what's half-trickle ?
>>> We had that in the slides on Tuesday but it's not yet in the draft. The
>>> idea is that if you can't confirm support for trickle in the answerer,
>> Do you mean: if you don't know if the answerer supports it before
>> sending an offer to him
> Yes. That's what I meant. The offerer needs to know that in advance in
> order to decide how to form their offer.
>
> Cheers,
> Emil
>
>> [since confirming support in the answer should
>> be easy]  ?
>>> you just send a vanilla ICE offer and you also indicate you do support
>>> trickle. A trickle ICE answerer can then trickle candidates and in
>>> general you get half the optimizations of Trickle ICE. You even get more
>>> than that in case the offerer is able to predictively gather candidates
>>> before the call is initiated.
>> Agreed.
>>> We'll describe this in the next version.
>> Ok - thanks for the explanation.
>>
>> -- Flemming
>>> Cheers,
>>> Emil
>>>
>>>> -- Flemming
>>>>
>>>>
>>>>> On 8 November 2012 10:12, Emil Ivov <emcho@jitsi.org> wrote:
>>>>>> I actually think this was supposed to be a joke. At least that's how I
>>>>>> got it.
>>>>>>
>>>>>> Emil
>>>>>>
>>>>>> On 08.11.12, 10:01, Flemming Andreasen wrote:
>>>>>>> This is very different from the way we normally handle extensions in SDP
>>>>>>> both from a pure signaling and negotiation point of view. I share the
>>>>>>> sentiment of others that this does not look like a good idea but would
>>>>>>> like to understand if there were other considerations for suggesting
>>>>>>> this radically different approach.
>>>>>>>
>>>>>>> Thanks
>>>>>>>
>>>>>>> -- Flemming
>>>>>>>
>>>>>>> On 11/8/12 9:23 AM, Christer Holmberg wrote:
>>>>>>>> +1
>>>>>>>>
>>>>>>>> Regards,
>>>>>>>>
>>>>>>>> Christer
>>>>>>>>
>>>>>>>> -----Original Message-----
>>>>>>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of Miguel A. Garcia
>>>>>>>> Sent: 8. marraskuuta 2012 9:30
>>>>>>>> To: Paul Kyzivat; Martin Thomson; Emil Ivov
>>>>>>>> Cc: mmusic@ietf.org
>>>>>>>> Subject: Re: [MMUSIC] Signaling trickle support
>>>>>>>>
>>>>>>>> As an individual...
>>>>>>>>
>>>>>>>> I concur with Paul, changing the v= version in the SDP is a very bad idea.
>>>>>>>>
>>>>>>>> Let met quote RFC 4566:
>>>>>>>>
>>>>>>>>
>>>>>>>>         The "v=" field gives the version of the Session Description Protocol.
>>>>>>>>         This memo defines version 0.  There is no minor version number.
>>>>>>>>
>>>>>>>>
>>>>>>>> This means that if you define another version different than zero, then, as a side effect, you are creating a different version of SDP than the one specified in RFC 4566. So, you will need to create an RFC specifying how such version of SDP works.
>>>>>>>>
>>>>>>>> /Miguel
>>>>>>>>
>>>>>>>> On 07/11/2012 20:27, Paul Kyzivat wrote:
>>>>>>>>> On 11/7/12 8:59 AM, Martin Thomson wrote:
>>>>>>>>>> Matthew, Justin and I were discussing the problem of signaling
>>>>>>>>>> trickle ICE support and the following proposal was made:
>>>>>>>>>>
>>>>>>>>>> The 24th bit in SDP is currently set to zero.  This bit is
>>>>>>>>>> specifically reserved for forward compatibility purposes.  We can use
>>>>>>>>>> this bit.  Set this bit to one to indicate support for trickle ICE.
>>>>>>>>>>
>>>>>>>>>> This meets the "works if supported, fails if it doesn't criteria".
>>>>>>>>>>
>>>>>>>>>> This solution may also be appropriate for bundle/mmt/whatever.
>>>>>>>>> After some side discussion, I guess you are talking about bits in the
>>>>>>>>> value of the v= field ???
>>>>>>>>>
>>>>>>>>> If so, I still have no idea what you are talking about. The syntax for
>>>>>>>>> the value of v= is 1*DIGIT. You can choose to represent that in binary
>>>>>>>>> and then choose bit 24 if you want, but that has no significance in 4566.
>>>>>>>>>
>>>>>>>>> Assigning a new SDP version to signal Trickle ICE support seems like
>>>>>>>>> an incredibly bad idea.
>>>>>>>>>
>>>>>>>>>          Thanks,
>>>>>>>>>          Paul
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> 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
>>>>>>>> _______________________________________________
>>>>>>>> mmusic mailing list
>>>>>>>> mmusic@ietf.org
>>>>>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>>>>>> .
>>>>>>>>
>>>>>> --
>>>>>> https://jitsi.org
>>>>> .
>>>>>
>>


From emil@sip-communicator.org  Thu Nov  8 15:24:43 2012
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Message-ID: <509C3F33.8060809@jitsi.org>
Date: Thu, 08 Nov 2012 18:24:35 -0500
From: Emil Ivov <emcho@jitsi.org>
Organization: Jitsi
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To: Flemming Andreasen <fandreas@cisco.com>
References: <CABkgnnVR-prV8xgzPx9H-MYRM_vTui-8GGfTD0yy1JPvJJzrrQ@mail.gmail.com> <509AB627.1090507@alum.mit.edu> <509B5F87.1010807@ericsson.com> <7594FB04B1934943A5C02806D1A2204B027800@ESESSMB209.ericsson.se> <509BC947.709@cisco.com> <509BCBFB.2020305@jitsi.org> <CABkgnnVMf53Vk_Rd-0erZN4ZLDKM4ePOvMfLJnfOUNVxdWYLqQ@mail.gmail.com> <509BD563.9080808@cisco.com> <509BDE01.9060705@jitsi.org> <509C02F2.10303@cisco.com> <509C0724.3050300@jitsi.org> <509C3066.6020209@cisco.com>
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] Signaling trickle support
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Hey Flemming,

On 08.11.12, 17:21, Flemming Andreasen wrote:
>>>> Since protocols like SIP don't really have things like XMPP disco and
>>>> Cap Neg that would have allowed it, we were considering other options.
>>>> One such option would be to send a trickle offer and format it in a way
>>>> that would cause a non-trickle answerer to reject it in a _predictable_
>>>> way. This would allow the offerer to retry with a regular ICE offer.
>>> I thought the point with trickle-ICE was to reduce call setup latency
>> It is.
>>
>>> and hence you would want to make sure that things complete in as few
>>> round-trips as possible. This doesn't seem to line up with that (?).
>> The thing is that in cases where trickle ICE support cannot be
>> pre-confirmed you have to fall back to half trickle. The probing
>> requests would hence allow for an optimisation compared to that.
> Somewhat unclear to me as it builds upon an assumption that it is 
> somehow "more efficient"to probe than to collect candidates.

Indeed, this would not always be the case and it would basically come
down to comparing one signalling RTT to the duration of candidate
gathering.

As you agreed in one of your previous mails, the relation may vary:

Executing a single STUN transaction is likely to be less time consuming
than a signalling RTT. Candidate gathering however may also last a lot
longer than a single offer/answer exchange.

However, an agent that has several interfaces and uses them to gather
candidates for three streams that don't have bundle nor RTCP muxing is
going to need quite some time to complete a harvest. Add to this UPnP
allocations and a second STUN server and you could easily find the
signalling RTT tens of times shorter than the candidate gathering.

In all cases the UA would have a fairly good idea of what to expect so
it could easily determine whether or not a probe would make sense.

> Clarifying 
> the criteria for something being "more efficient" would also be helpful 
> here (what is it truly that we are after; setup delay reduction, fewer 
> NAT bindings, ...)

I believe we can agree that in the context of trickle the only thing
that matters is call setup duration.

>> With half-trickle the offerer has to gather all candidates anyway so,
>> while doing so, they might as well try and check if they really need to
>> complete it before sending their first offer.
> Unclear to me since you can collect candidates before you know who to 
> send the offer to.

You could ... but not always. A hard phone can easily do this when
someone picks it up. A softphone however doesn't have a clear early cue
about a pending call initiation so the only way would be to always
maintain a set of active candidates and to keep them alive. Not
necessarily the most desirable option, especially for large deployments.

A web page could be even worse, especially in cases where the calling
functionality is only a side feature (e.g. a support or a customer
service call option).

Cheers,
Emil

> Thanks
> 
> -- Flemming
>>>>>> Besides, the failure modes all lead to an expenditure of time that is
>>>>>> worse than the simple candidate gathering scenarios (reflexive,
>>>>>> relay), though not "other" gathering methods (UPnP, etc...).
>>>>> Is there a requirement here that needs to be written down ?
>>>> The point that Martin is making is that sending an offer and then
>>>> waiting for the failure may potentially take longer than just gathering
>>>> your candidates. This is true especially in cases where the offerer is
>>>> only planning on using a single port, on a single interface, with a
>>>> single STUN/TURN server and no other candidate gathering mechanisms.
>>> I would agree with that.
>>>> But then, if this really is the case, the offerer knows it and can
>>>> choose not to do the probing.
>>> Agreed.
>>>>>> half-trickle is a reasonable solution.
>>>>> Sorry - what's half-trickle ?
>>>> We had that in the slides on Tuesday but it's not yet in the draft. The
>>>> idea is that if you can't confirm support for trickle in the answerer,
>>> Do you mean: if you don't know if the answerer supports it before
>>> sending an offer to him
>> Yes. That's what I meant. The offerer needs to know that in advance in
>> order to decide how to form their offer.
>>
>> Cheers,
>> Emil
>>
>>> [since confirming support in the answer should
>>> be easy]  ?
>>>> you just send a vanilla ICE offer and you also indicate you do support
>>>> trickle. A trickle ICE answerer can then trickle candidates and in
>>>> general you get half the optimizations of Trickle ICE. You even get more
>>>> than that in case the offerer is able to predictively gather candidates
>>>> before the call is initiated.
>>> Agreed.
>>>> We'll describe this in the next version.
>>> Ok - thanks for the explanation.
>>>
>>> -- Flemming
>>>> Cheers,
>>>> Emil
>>>>
>>>>> -- Flemming
>>>>>
>>>>>
>>>>>> On 8 November 2012 10:12, Emil Ivov <emcho@jitsi.org> wrote:
>>>>>>> I actually think this was supposed to be a joke. At least that's how I
>>>>>>> got it.
>>>>>>>
>>>>>>> Emil
>>>>>>>
>>>>>>> On 08.11.12, 10:01, Flemming Andreasen wrote:
>>>>>>>> This is very different from the way we normally handle extensions in SDP
>>>>>>>> both from a pure signaling and negotiation point of view. I share the
>>>>>>>> sentiment of others that this does not look like a good idea but would
>>>>>>>> like to understand if there were other considerations for suggesting
>>>>>>>> this radically different approach.
>>>>>>>>
>>>>>>>> Thanks
>>>>>>>>
>>>>>>>> -- Flemming
>>>>>>>>
>>>>>>>> On 11/8/12 9:23 AM, Christer Holmberg wrote:
>>>>>>>>> +1
>>>>>>>>>
>>>>>>>>> Regards,
>>>>>>>>>
>>>>>>>>> Christer
>>>>>>>>>
>>>>>>>>> -----Original Message-----
>>>>>>>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of Miguel A. Garcia
>>>>>>>>> Sent: 8. marraskuuta 2012 9:30
>>>>>>>>> To: Paul Kyzivat; Martin Thomson; Emil Ivov
>>>>>>>>> Cc: mmusic@ietf.org
>>>>>>>>> Subject: Re: [MMUSIC] Signaling trickle support
>>>>>>>>>
>>>>>>>>> As an individual...
>>>>>>>>>
>>>>>>>>> I concur with Paul, changing the v= version in the SDP is a very bad idea.
>>>>>>>>>
>>>>>>>>> Let met quote RFC 4566:
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>         The "v=" field gives the version of the Session Description Protocol.
>>>>>>>>>         This memo defines version 0.  There is no minor version number.
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> This means that if you define another version different than zero, then, as a side effect, you are creating a different version of SDP than the one specified in RFC 4566. So, you will need to create an RFC specifying how such version of SDP works.
>>>>>>>>>
>>>>>>>>> /Miguel
>>>>>>>>>
>>>>>>>>> On 07/11/2012 20:27, Paul Kyzivat wrote:
>>>>>>>>>> On 11/7/12 8:59 AM, Martin Thomson wrote:
>>>>>>>>>>> Matthew, Justin and I were discussing the problem of signaling
>>>>>>>>>>> trickle ICE support and the following proposal was made:
>>>>>>>>>>>
>>>>>>>>>>> The 24th bit in SDP is currently set to zero.  This bit is
>>>>>>>>>>> specifically reserved for forward compatibility purposes.  We can use
>>>>>>>>>>> this bit.  Set this bit to one to indicate support for trickle ICE.
>>>>>>>>>>>
>>>>>>>>>>> This meets the "works if supported, fails if it doesn't criteria".
>>>>>>>>>>>
>>>>>>>>>>> This solution may also be appropriate for bundle/mmt/whatever.
>>>>>>>>>> After some side discussion, I guess you are talking about bits in the
>>>>>>>>>> value of the v= field ???
>>>>>>>>>>
>>>>>>>>>> If so, I still have no idea what you are talking about. The syntax for
>>>>>>>>>> the value of v= is 1*DIGIT. You can choose to represent that in binary
>>>>>>>>>> and then choose bit 24 if you want, but that has no significance in 4566.
>>>>>>>>>>
>>>>>>>>>> Assigning a new SDP version to signal Trickle ICE support seems like
>>>>>>>>>> an incredibly bad idea.
>>>>>>>>>>
>>>>>>>>>>          Thanks,
>>>>>>>>>>          Paul
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> 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
>>>>>>>>> _______________________________________________
>>>>>>>>> mmusic mailing list
>>>>>>>>> mmusic@ietf.org
>>>>>>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>>>>>>> .
>>>>>>>>>
>>>>>>> --
>>>>>>> https://jitsi.org
>>>>>> .
>>>>>>
>>>
> 
> 

-- 
https://jitsi.org

From palmarti@cisco.com  Fri Nov  9 04:45:54 2012
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From: "Pal Martinsen (palmarti)" <palmarti@cisco.com>
To: Emil Ivov <emcho@jitsi.org>
Thread-Topic: [MMUSIC] Signaling trickle support
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References: <CABkgnnVR-prV8xgzPx9H-MYRM_vTui-8GGfTD0yy1JPvJJzrrQ@mail.gmail.com> <509AB627.1090507@alum.mit.edu> <509B5F87.1010807@ericsson.com> <7594FB04B1934943A5C02806D1A2204B027800@ESESSMB209.ericsson.se> <509BC947.709@cisco.com> <509BCBFB.2020305@jitsi.org> <CABkgnnVMf53Vk_Rd-0erZN4ZLDKM4ePOvMfLJnfOUNVxdWYLqQ@mail.gmail.com> <509BD563.9080808@cisco.com> <509BDE01.9060705@jitsi.org> <509C02F2.10303@cisco.com> <509C0724.3050300@jitsi.org> <509C3066.6020209@cisco.com> <509C3F33.8060809@jitsi.org>
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Cc: "Flemming Andreasen \(fandreas\)" <fandreas@cisco.com>, "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] Signaling trickle support
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On Nov 9, 2012, at 24:24 AM, Emil Ivov <emcho@jitsi.org> wrote:

> Hey Flemming,
>=20
> On 08.11.12, 17:21, Flemming Andreasen wrote:
>>>>> Since protocols like SIP don't really have things like XMPP disco and
>>>>> Cap Neg that would have allowed it, we were considering other options=
.
>>>>> One such option would be to send a trickle offer and format it in a w=
ay
>>>>> that would cause a non-trickle answerer to reject it in a _predictabl=
e_
>>>>> way. This would allow the offerer to retry with a regular ICE offer.
>>>> I thought the point with trickle-ICE was to reduce call setup latency
>>> It is.
>>>=20
>>>> and hence you would want to make sure that things complete in as few
>>>> round-trips as possible. This doesn't seem to line up with that (?).
>>> The thing is that in cases where trickle ICE support cannot be
>>> pre-confirmed you have to fall back to half trickle. The probing
>>> requests would hence allow for an optimisation compared to that.
>> Somewhat unclear to me as it builds upon an assumption that it is=20
>> somehow "more efficient"to probe than to collect candidates.
>=20
> Indeed, this would not always be the case and it would basically come
> down to comparing one signalling RTT to the duration of candidate
> gathering.
>=20
> As you agreed in one of your previous mails, the relation may vary:
>=20
> Executing a single STUN transaction is likely to be less time consuming
> than a signalling RTT. Candidate gathering however may also last a lot
> longer than a single offer/answer exchange.
>=20
> However, an agent that has several interfaces and uses them to gather
> candidates for three streams that don't have bundle nor RTCP muxing is
> going to need quite some time to complete a harvest.

>From our experience single allocation takes aprox 30ms. This includes one r=
edirect due to alternate server and one extra roundtrip because of authenti=
cation. The time the TURN server is using to process the request is neglica=
ble and so  it is RTT that is important. If the RTT is to large media would=
 be troublesome anyway due to excessive jitter and delay.

Number of media lines and interfaces does not matter as this is done in par=
allel (or pretty close to). Our implementation have 5 media lines (Audio, 2=
x Video, BFCP, FECC), resulting in 9 TURN allocations.=20

What we have been struggling with is delay in the signaling path. We have s=
een delays in the range of 3-5 seconds for the 200OK to traverse the signal=
ing path. That caused some failing ICE calls due to to large skew and we di=
d not complete the transactions in what we have set as a reasonable time.=20

> Add to this UPnP
> allocations and a second STUN server and you could easily find the
> signalling RTT tens of times shorter than the candidate gathering.
>=20
My understanding is that UPnP or PCP can be used to open pinholes without t=
he need for keepalives.
So in theory this can be done when the endpoints registers. At least if we =
are not overly concerned about the extra port allocations on the NAT.

But I agree with the point. Additional steps to gather "better" candidates =
can take time.


> In all cases the UA would have a fairly good idea of what to expect so
> it could easily determine whether or not a probe would make sense.
>=20
>> Clarifying=20
>> the criteria for something being "more efficient" would also be helpful=
=20
>> here (what is it truly that we are after; setup delay reduction, fewer=20
>> NAT bindings, ...)
>=20
> I believe we can agree that in the context of trickle the only thing
> that matters is call setup duration.
>=20
Personally I see great benefit in having the ability to add candidates (and=
 remove them) without doing a update or re-invite. This would make mobility=
 scenarios easier, and developing a smarter and more optimized media experi=
ence can be a lot easier. =20

.-.
P=E5l-Erik


>>> With half-trickle the offerer has to gather all candidates anyway so,
>>> while doing so, they might as well try and check if they really need to
>>> complete it before sending their first offer.
>> Unclear to me since you can collect candidates before you know who to=20
>> send the offer to.
>=20
> You could ... but not always. A hard phone can easily do this when
> someone picks it up. A softphone however doesn't have a clear early cue
> about a pending call initiation so the only way would be to always
> maintain a set of active candidates and to keep them alive. Not
> necessarily the most desirable option, especially for large deployments.
>=20
> A web page could be even worse, especially in cases where the calling
> functionality is only a side feature (e.g. a support or a customer
> service call option).
>=20
> Cheers,
> Emil
>=20
>> Thanks
>>=20
>> -- Flemming
>>>>>>> Besides, the failure modes all lead to an expenditure of time that =
is
>>>>>>> worse than the simple candidate gathering scenarios (reflexive,
>>>>>>> relay), though not "other" gathering methods (UPnP, etc...).
>>>>>> Is there a requirement here that needs to be written down ?
>>>>> The point that Martin is making is that sending an offer and then
>>>>> waiting for the failure may potentially take longer than just gatheri=
ng
>>>>> your candidates. This is true especially in cases where the offerer i=
s
>>>>> only planning on using a single port, on a single interface, with a
>>>>> single STUN/TURN server and no other candidate gathering mechanisms.
>>>> I would agree with that.
>>>>> But then, if this really is the case, the offerer knows it and can
>>>>> choose not to do the probing.
>>>> Agreed.
>>>>>>> half-trickle is a reasonable solution.
>>>>>> Sorry - what's half-trickle ?
>>>>> We had that in the slides on Tuesday but it's not yet in the draft. T=
he
>>>>> idea is that if you can't confirm support for trickle in the answerer=
,
>>>> Do you mean: if you don't know if the answerer supports it before
>>>> sending an offer to him
>>> Yes. That's what I meant. The offerer needs to know that in advance in
>>> order to decide how to form their offer.
>>>=20
>>> Cheers,
>>> Emil
>>>=20
>>>> [since confirming support in the answer should
>>>> be easy]  ?
>>>>> you just send a vanilla ICE offer and you also indicate you do suppor=
t
>>>>> trickle. A trickle ICE answerer can then trickle candidates and in
>>>>> general you get half the optimizations of Trickle ICE. You even get m=
ore
>>>>> than that in case the offerer is able to predictively gather candidat=
es
>>>>> before the call is initiated.
>>>> Agreed.
>>>>> We'll describe this in the next version.
>>>> Ok - thanks for the explanation.
>>>>=20
>>>> -- Flemming
>>>>> Cheers,
>>>>> Emil
>>>>>=20
>>>>>> -- Flemming
>>>>>>=20
>>>>>>=20
>>>>>>> On 8 November 2012 10:12, Emil Ivov <emcho@jitsi.org> wrote:
>>>>>>>> I actually think this was supposed to be a joke. At least that's h=
ow I
>>>>>>>> got it.
>>>>>>>>=20
>>>>>>>> Emil
>>>>>>>>=20
>>>>>>>> On 08.11.12, 10:01, Flemming Andreasen wrote:
>>>>>>>>> This is very different from the way we normally handle extensions=
 in SDP
>>>>>>>>> both from a pure signaling and negotiation point of view. I share=
 the
>>>>>>>>> sentiment of others that this does not look like a good idea but =
would
>>>>>>>>> like to understand if there were other considerations for suggest=
ing
>>>>>>>>> this radically different approach.
>>>>>>>>>=20
>>>>>>>>> Thanks
>>>>>>>>>=20
>>>>>>>>> -- Flemming
>>>>>>>>>=20
>>>>>>>>> On 11/8/12 9:23 AM, Christer Holmberg wrote:
>>>>>>>>>> +1
>>>>>>>>>>=20
>>>>>>>>>> Regards,
>>>>>>>>>>=20
>>>>>>>>>> Christer
>>>>>>>>>>=20
>>>>>>>>>> -----Original Message-----
>>>>>>>>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] O=
n Behalf Of Miguel A. Garcia
>>>>>>>>>> Sent: 8. marraskuuta 2012 9:30
>>>>>>>>>> To: Paul Kyzivat; Martin Thomson; Emil Ivov
>>>>>>>>>> Cc: mmusic@ietf.org
>>>>>>>>>> Subject: Re: [MMUSIC] Signaling trickle support
>>>>>>>>>>=20
>>>>>>>>>> As an individual...
>>>>>>>>>>=20
>>>>>>>>>> I concur with Paul, changing the v=3D version in the SDP is a ve=
ry bad idea.
>>>>>>>>>>=20
>>>>>>>>>> Let met quote RFC 4566:
>>>>>>>>>>=20
>>>>>>>>>>=20
>>>>>>>>>>        The "v=3D" field gives the version of the Session Descrip=
tion Protocol.
>>>>>>>>>>        This memo defines version 0.  There is no minor version n=
umber.
>>>>>>>>>>=20
>>>>>>>>>>=20
>>>>>>>>>> This means that if you define another version different than zer=
o, then, as a side effect, you are creating a different version of SDP than=
 the one specified in RFC 4566. So, you will need to create an RFC specifyi=
ng how such version of SDP works.
>>>>>>>>>>=20
>>>>>>>>>> /Miguel
>>>>>>>>>>=20
>>>>>>>>>> On 07/11/2012 20:27, Paul Kyzivat wrote:
>>>>>>>>>>> On 11/7/12 8:59 AM, Martin Thomson wrote:
>>>>>>>>>>>> Matthew, Justin and I were discussing the problem of signaling
>>>>>>>>>>>> trickle ICE support and the following proposal was made:
>>>>>>>>>>>>=20
>>>>>>>>>>>> The 24th bit in SDP is currently set to zero.  This bit is
>>>>>>>>>>>> specifically reserved for forward compatibility purposes.  We =
can use
>>>>>>>>>>>> this bit.  Set this bit to one to indicate support for trickle=
 ICE.
>>>>>>>>>>>>=20
>>>>>>>>>>>> This meets the "works if supported, fails if it doesn't criter=
ia".
>>>>>>>>>>>>=20
>>>>>>>>>>>> This solution may also be appropriate for bundle/mmt/whatever.
>>>>>>>>>>> After some side discussion, I guess you are talking about bits =
in the
>>>>>>>>>>> value of the v=3D field ???
>>>>>>>>>>>=20
>>>>>>>>>>> If so, I still have no idea what you are talking about. The syn=
tax for
>>>>>>>>>>> the value of v=3D is 1*DIGIT. You can choose to represent that =
in binary
>>>>>>>>>>> and then choose bit 24 if you want, but that has no significanc=
e in 4566.
>>>>>>>>>>>=20
>>>>>>>>>>> Assigning a new SDP version to signal Trickle ICE support seems=
 like
>>>>>>>>>>> an incredibly bad idea.
>>>>>>>>>>>=20
>>>>>>>>>>>         Thanks,
>>>>>>>>>>>         Paul
>>>>>>>>>>>=20
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> mmusic mailing list
>>>>>>>>>>> mmusic@ietf.org
>>>>>>>>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>>>>>>>>>=20
>>>>>>>>>> --
>>>>>>>>>> Miguel A. Garcia
>>>>>>>>>> +34-91-339-3608
>>>>>>>>>> Ericsson Spain
>>>>>>>>>> _______________________________________________
>>>>>>>>>> mmusic mailing list
>>>>>>>>>> mmusic@ietf.org
>>>>>>>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>>>>>>>> _______________________________________________
>>>>>>>>>> mmusic mailing list
>>>>>>>>>> mmusic@ietf.org
>>>>>>>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>>>>>>>> .
>>>>>>>>>>=20
>>>>>>>> --
>>>>>>>> https://jitsi.org
>>>>>>> .
>>>>>>>=20
>>>>=20
>>=20
>>=20
>=20
> --=20
> https://jitsi.org
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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References: <CABkgnnVR-prV8xgzPx9H-MYRM_vTui-8GGfTD0yy1JPvJJzrrQ@mail.gmail.com> <509AB627.1090507@alum.mit.edu> <509B5F87.1010807@ericsson.com> <7594FB04B1934943A5C02806D1A2204B027800@ESESSMB209.ericsson.se> <509BC947.709@cisco.com> <509BCBFB.2020305@jitsi.org> <CABkgnnVMf53Vk_Rd-0erZN4ZLDKM4ePOvMfLJnfOUNVxdWYLqQ@mail.gmail.com> <509BD563.9080808@cisco.com> <509BDE01.9060705@jitsi.org> <509C02F2.10303@cisco.com> <509C0724.3050300@jitsi.org> <509C3066.6020209@cisco.com> <509C3F33.8060809@jitsi.org>
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] Signaling trickle support
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On 11/8/12 6:24 PM, Emil Ivov wrote:
> Hey Flemming,
>
> On 08.11.12, 17:21, Flemming Andreasen wrote:
>>>>> Since protocols like SIP don't really have things like XMPP disco and
>>>>> Cap Neg that would have allowed it, we were considering other options.
>>>>> One such option would be to send a trickle offer and format it in a way
>>>>> that would cause a non-trickle answerer to reject it in a _predictable_
>>>>> way. This would allow the offerer to retry with a regular ICE offer.
>>>> I thought the point with trickle-ICE was to reduce call setup latency
>>> It is.
>>>
>>>> and hence you would want to make sure that things complete in as few
>>>> round-trips as possible. This doesn't seem to line up with that (?).
>>> The thing is that in cases where trickle ICE support cannot be
>>> pre-confirmed you have to fall back to half trickle. The probing
>>> requests would hence allow for an optimisation compared to that.
>> Somewhat unclear to me as it builds upon an assumption that it is
>> somehow "more efficient"to probe than to collect candidates.
> Indeed, this would not always be the case and it would basically come
> down to comparing one signalling RTT to the duration of candidate
> gathering.
>
> As you agreed in one of your previous mails, the relation may vary:
I still agree with that :-)
> Executing a single STUN transaction is likely to be less time consuming
> than a signalling RTT. Candidate gathering however may also last a lot
> longer than a single offer/answer exchange.
>
> However, an agent that has several interfaces and uses them to gather
> candidates for three streams that don't have bundle nor RTCP muxing is
> going to need quite some time to complete a harvest. Add to this UPnP
> allocations and a second STUN server and you could easily find the
> signalling RTT tens of times shorter than the candidate gathering.
Because of the min. 20 ms Ta timer constraints, which apply cross-media 
stream ? Can somebody refresh my mind as to why that is (e.g. are 
residential NATs at this point generally not capable of creating NAT 
bindings faster than at 20 ms intervals) ?

> In all cases the UA would have a fairly good idea of what to expect so
> it could easily determine whether or not a probe would make sense.
>
>> Clarifying
>> the criteria for something being "more efficient" would also be helpful
>> here (what is it truly that we are after; setup delay reduction, fewer
>> NAT bindings, ...)
> I believe we can agree that in the context of trickle the only thing
> that matters is call setup duration.
Ok - so reducing the number of NAT binding allocations is explicitly not 
a concern (only the time it takes to create those bindings, if that time 
is in the critical path for the actual call setup).

>>> With half-trickle the offerer has to gather all candidates anyway so,
>>> while doing so, they might as well try and check if they really need to
>>> complete it before sending their first offer.
>> Unclear to me since you can collect candidates before you know who to
>> send the offer to.
> You could ... but not always. A hard phone can easily do this when
> someone picks it up. A softphone however doesn't have a clear early cue
> about a pending call initiation so the only way would be to always
> maintain a set of active candidates and to keep them alive. Not
> necessarily the most desirable option, especially for large deployments.
I'm not sure I see much difference between hard and soft-phones here. 
For each, there is some sort of interaction before the call is initiated 
giving you a bit of a head-start to gather candidates. If we are trying 
to optimize out several seconds that may not help much, however if we 
are in the 100's ms range, then it will.
> A web page could be even worse, especially in cases where the calling
> functionality is only a side feature (e.g. a support or a customer
> service call option).
Why would a web page be worse ?

Thanks

-- Flemming



From emil@sip-communicator.org  Fri Nov  9 06:43:52 2012
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From: Emil Ivov <emcho@jitsi.org>
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To: Flemming Andreasen <fandreas@cisco.com>
References: <CABkgnnVR-prV8xgzPx9H-MYRM_vTui-8GGfTD0yy1JPvJJzrrQ@mail.gmail.com> <509AB627.1090507@alum.mit.edu> <509B5F87.1010807@ericsson.com> <7594FB04B1934943A5C02806D1A2204B027800@ESESSMB209.ericsson.se> <509BC947.709@cisco.com> <509BCBFB.2020305@jitsi.org> <CABkgnnVMf53Vk_Rd-0erZN4ZLDKM4ePOvMfLJnfOUNVxdWYLqQ@mail.gmail.com> <509BD563.9080808@cisco.com> <509BDE01.9060705@jitsi.org> <509C02F2.10303@cisco.com> <509C0724.3050300@jitsi.org> <509C3066.6020209@cisco.com> <509C3F33.8060809@jitsi.org> <509D1200.3070702@cisco.com>
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] Signaling trickle support
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On 09.11.12, 09:24, Flemming Andreasen wrote:
> 
> On 11/8/12 6:24 PM, Emil Ivov wrote:
>> Hey Flemming,
>>
>> On 08.11.12, 17:21, Flemming Andreasen wrote:
>>>>>> Since protocols like SIP don't really have things like XMPP disco and
>>>>>> Cap Neg that would have allowed it, we were considering other options.
>>>>>> One such option would be to send a trickle offer and format it in a way
>>>>>> that would cause a non-trickle answerer to reject it in a _predictable_
>>>>>> way. This would allow the offerer to retry with a regular ICE offer.
>>>>> I thought the point with trickle-ICE was to reduce call setup latency
>>>> It is.
>>>>
>>>>> and hence you would want to make sure that things complete in as few
>>>>> round-trips as possible. This doesn't seem to line up with that (?).
>>>> The thing is that in cases where trickle ICE support cannot be
>>>> pre-confirmed you have to fall back to half trickle. The probing
>>>> requests would hence allow for an optimisation compared to that.
>>> Somewhat unclear to me as it builds upon an assumption that it is
>>> somehow "more efficient"to probe than to collect candidates.
>> Indeed, this would not always be the case and it would basically come
>> down to comparing one signalling RTT to the duration of candidate
>> gathering.
>>
>> As you agreed in one of your previous mails, the relation may vary:
> I still agree with that :-)
>> Executing a single STUN transaction is likely to be less time consuming
>> than a signalling RTT. Candidate gathering however may also last a lot
>> longer than a single offer/answer exchange.
>>
>> However, an agent that has several interfaces and uses them to gather
>> candidates for three streams that don't have bundle nor RTCP muxing is
>> going to need quite some time to complete a harvest. Add to this UPnP
>> allocations and a second STUN server and you could easily find the
>> signalling RTT tens of times shorter than the candidate gathering.
> Because of the min. 20 ms Ta timer constraints, which apply cross-media 
> stream ?

Yes, although not only. In the case of UPnP you may also have to spend
some time in discovery.

Other gathering mechanisms may involve different delays.

> Can somebody refresh my mind as to why that is (e.g. are 
> residential NATs at this point generally not capable of creating NAT 
> bindings faster than at 20 ms intervals) ?

I believe the idea was to maintain a bitrate that would be similar to
the one you will be using during the call. Otherwise you may end up with
packet bursts and packet drops that could delay conn checks.

>> In all cases the UA would have a fairly good idea of what to expect so
>> it could easily determine whether or not a probe would make sense.
>>
>>> Clarifying
>>> the criteria for something being "more efficient" would also be helpful
>>> here (what is it truly that we are after; setup delay reduction, fewer
>>> NAT bindings, ...)
>> I believe we can agree that in the context of trickle the only thing
>> that matters is call setup duration.
> Ok - so reducing the number of NAT binding allocations is explicitly not 
> a concern (only the time it takes to create those bindings, if that time 
> is in the critical path for the actual call setup).

In the context of Trickle ICE - yes.

>>>> With half-trickle the offerer has to gather all candidates anyway so,
>>>> while doing so, they might as well try and check if they really need to
>>>> complete it before sending their first offer.
>>> Unclear to me since you can collect candidates before you know who to
>>> send the offer to.
>> You could ... but not always. A hard phone can easily do this when
>> someone picks it up. A softphone however doesn't have a clear early cue
>> about a pending call initiation so the only way would be to always
>> maintain a set of active candidates and to keep them alive. Not
>> necessarily the most desirable option, especially for large deployments.
> I'm not sure I see much difference between hard and soft-phones here. 
> For each, there is some sort of interaction before the call is initiated 
> giving you a bit of a head-start to gather candidates. If we are trying 
> to optimize out several seconds that may not help much, however if we 
> are in the 100's ms range, then it will.

Well, in our case for instance (Jitsi) the user is constantly
interacting with the application and the majority of these interactions
do not end with a call. IMs for example are a lot more frequent, and
then you also have other use cases such checking presence for others,
setting a presence status for the user, checking contact details,
viewing call history, etc.

All in all, you only know for sure that a call is being made, once the
call button is pressed.

>> A web page could be even worse, especially in cases where the calling
>> functionality is only a side feature (e.g. a support or a customer
>> service call option).
> Why would a web page be worse ?

Because it could offer even more non-call use cases than a dedicated
application such as a softphone/messenger. Calls could be a relatively
corner use case for many web pages. Imagine for example a product page
on Amazon with a "Call the vendor" button. Calls would be relatively
rare and there would be no way to predict them until the button is pressed.

Emil

-- 
https://jitsi.org

From fandreas@cisco.com  Fri Nov  9 06:58:39 2012
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To: Emil Ivov <emcho@jitsi.org>
References: <CABkgnnVR-prV8xgzPx9H-MYRM_vTui-8GGfTD0yy1JPvJJzrrQ@mail.gmail.com> <509AB627.1090507@alum.mit.edu> <509B5F87.1010807@ericsson.com> <7594FB04B1934943A5C02806D1A2204B027800@ESESSMB209.ericsson.se> <509BC947.709@cisco.com> <509BCBFB.2020305@jitsi.org> <CABkgnnVMf53Vk_Rd-0erZN4ZLDKM4ePOvMfLJnfOUNVxdWYLqQ@mail.gmail.com> <509BD563.9080808@cisco.com> <509BDE01.9060705@jitsi.org> <509C02F2.10303@cisco.com> <509C0724.3050300@jitsi.org> <509C3066.6020209@cisco.com> <509C3F33.8060809@jitsi.org> <509D1200.3070702@cisco.com> <509D16A3.7040007@jitsi.org>
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] Signaling trickle support
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On 11/9/12 9:43 AM, Emil Ivov wrote:
>
>>> Executing a single STUN transaction is likely to be less time consuming
>>> than a signalling RTT. Candidate gathering however may also last a lot
>>> longer than a single offer/answer exchange.
>>>
>>> However, an agent that has several interfaces and uses them to gather
>>> candidates for three streams that don't have bundle nor RTCP muxing is
>>> going to need quite some time to complete a harvest. Add to this UPnP
>>> allocations and a second STUN server and you could easily find the
>>> signalling RTT tens of times shorter than the candidate gathering.
>> Because of the min. 20 ms Ta timer constraints, which apply cross-media
>> stream ?
> Yes, although not only. In the case of UPnP you may also have to spend
> some time in discovery.
>
> Other gathering mechanisms may involve different delays.
>
>> Can somebody refresh my mind as to why that is (e.g. are
>> residential NATs at this point generally not capable of creating NAT
>> bindings faster than at 20 ms intervals) ?
> I believe the idea was to maintain a bitrate that would be similar to
> the one you will be using during the call. Otherwise you may end up with
> packet bursts and packet drops that could delay conn checks.

That only explains why it applies within a media stream (which I can 
understand), not why it applies cross-media stream.

-- Flemming


From harald@alvestrand.no  Fri Nov  9 08:09:22 2012
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Subject: Re: [MMUSIC] Signaling trickle support
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On 11/07/2012 02:59 PM, Martin Thomson wrote:
> Matthew, Justin and I were discussing the problem of signaling trickle
> ICE support and the following proposal was made:
>
> The 24th bit in SDP is currently set to zero.  This bit is
> specifically reserved for forward compatibility purposes.  We can use
> this bit.  Set this bit to one to indicate support for trickle ICE.
hm... the 24th bit in SDP is zero, but setting it would make it SDQ.... 
why not set the 22nd bit and make it SDT (SDP with Trickle)?
>
> This meets the "works if supported, fails if it doesn't criteria".
I don't think it meets the "reverts to old behaviour if it doesn't work" 
criterion that previous attempts to extend SDP have jumped through 
Möbius hoops to maintain. But it's tempting.
>
> This solution may also be appropriate for bundle/mmt/whatever.
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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Dear all,

the full recording (synchronized video, audio, slides and jabber room)
of this WG session at IETF-85 is available.

You can watch it by accessing the following URL:
http://www.meetecho.com/ietf85/recordings

For the chair(s): please feel free to put the link to the recording in 
the minutes, if you think this might be useful.

In case of problems with the playout, just drop an e-mail to 
ietf-team@meetecho.com.

Cheers,
the Meetecho team

-- 
Meetecho s.r.l.
Web Conferencing and Collaboration Tools
www.meetecho.com

From emil@sip-communicator.org  Fri Nov  9 12:16:06 2012
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References: <CABkgnnVR-prV8xgzPx9H-MYRM_vTui-8GGfTD0yy1JPvJJzrrQ@mail.gmail.com> <509AB627.1090507@alum.mit.edu> <509B5F87.1010807@ericsson.com> <7594FB04B1934943A5C02806D1A2204B027800@ESESSMB209.ericsson.se> <509BC947.709@cisco.com> <509BCBFB.2020305@jitsi.org> <CABkgnnVMf53Vk_Rd-0erZN4ZLDKM4ePOvMfLJnfOUNVxdWYLqQ@mail.gmail.com> <509BD563.9080808@cisco.com> <509BDE01.9060705@jitsi.org> <509C02F2.10303@cisco.com> <509C0724.3050300@jitsi.org> <509C3066.6020209@cisco.com> <509C3F33.8060809@jitsi.org> <509D1200.3070702@cisco.com> <509D16A3.7040007@jitsi.org> <509D1A1E.3020303@cisco.com>
In-Reply-To: <509D1A1E.3020303@cisco.com>
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] Signaling trickle support
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On 09.11.12, 09:58, Flemming Andreasen wrote:
> 
> On 11/9/12 9:43 AM, Emil Ivov wrote:
>>
>>>> Executing a single STUN transaction is likely to be less time consuming
>>>> than a signalling RTT. Candidate gathering however may also last a lot
>>>> longer than a single offer/answer exchange.
>>>>
>>>> However, an agent that has several interfaces and uses them to gather
>>>> candidates for three streams that don't have bundle nor RTCP muxing is
>>>> going to need quite some time to complete a harvest. Add to this UPnP
>>>> allocations and a second STUN server and you could easily find the
>>>> signalling RTT tens of times shorter than the candidate gathering.
>>> Because of the min. 20 ms Ta timer constraints, which apply cross-media
>>> stream ?
>> Yes, although not only. In the case of UPnP you may also have to spend
>> some time in discovery.
>>
>> Other gathering mechanisms may involve different delays.
>>
>>> Can somebody refresh my mind as to why that is (e.g. are
>>> residential NATs at this point generally not capable of creating NAT
>>> bindings faster than at 20 ms intervals) ?
>> I believe the idea was to maintain a bitrate that would be similar to
>> the one you will be using during the call. Otherwise you may end up with
>> packet bursts and packet drops that could delay conn checks.
> 
> That only explains why it applies within a media stream (which I can 
> understand), not why it applies cross-media stream.

True. I wasn't around at the time though so I have no idea how it ended
up being the cross-stream minimum.

Emil


-- 
https://jitsi.org

From carl_klatsky@cable.comcast.com  Mon Nov 12 12:46:17 2012
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From: "Klatsky, Carl" <Carl_Klatsky@cable.comcast.com>
To: "'mmusic@ietf.org'" <mmusic@ietf.org>
Thread-Topic: Comments on draft-ietf-mmusic-media-loopback-24
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Subject: [MMUSIC] Comments on draft-ietf-mmusic-media-loopback-24
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Hi,

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


1)      When setting up the media loopback, are there any restrictions on t=
ype of codec used?  Could even telephone-events codec be specified?

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


Regards,

Carl Klatsky
Comcast

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<p class=3D"MsoNormal">Hi,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">A colleague of mine forwarded me this draft and I ha=
d two comments if someone on the distro has input:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-.25in;mso-list:l0 level=
1 lfo1"><![if !supportLists]><span style=3D"mso-list:Ignore">1)<span style=
=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>When setting up the media loopback, are there any r=
estrictions on type of codec used?&nbsp; Could even telephone-events codec =
be specified?<o:p></o:p></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-.25in;mso-list:l0 level=
1 lfo1"><![if !supportLists]><span style=3D"mso-list:Ignore">2)<span style=
=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Although this draft is centered around media, can i=
t be extended to for use as means to test the caller ID set connected to an=
 analog telephony interface?&nbsp; The approach outlined in this draft seem=
s to imply no interaction with the end
 subscriber is needed.&nbsp; If the network operator wanted to silently sen=
d a test pattern to the caller ID set, perhaps this same loopback mechanism=
 can be used with either an additional SDP attribute or some change in the =
SIP header itself.&nbsp; A related application
 would be to advance the timestamp on the caller ID set following Daylight =
Savings Time transition.&nbsp; Typically that is not updated until the next=
 inbound call following the DST change.&nbsp; A loopback call could signal =
to the device to update the caller ID set
 but without ringing / alerting the end subscriber without the need to wait=
 for an inbound call.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">Regards,</span><o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">Carl Klatsky<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">Comcast<o:p></o:p></span></p>
</div>
</body>
</html>

--_000_6C15A6B88541034E912E94C2D8BC3E8772B4C899PACDCEXMB15cabl_--

From harald@alvestrand.no  Tue Nov 13 01:02:09 2012
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References: <CABkgnnVR-prV8xgzPx9H-MYRM_vTui-8GGfTD0yy1JPvJJzrrQ@mail.gmail.com> <509AB627.1090507@alum.mit.edu> <509B5F87.1010807@ericsson.com> <7594FB04B1934943A5C02806D1A2204B027800@ESESSMB209.ericsson.se> <509BC947.709@cisco.com> <509BCBFB.2020305@jitsi.org> <CABkgnnVMf53Vk_Rd-0erZN4ZLDKM4ePOvMfLJnfOUNVxdWYLqQ@mail.gmail.com> <509BD563.9080808@cisco.com> <509BDE01.9060705@jitsi.org> <509C02F2.10303@cisco.com> <509C0724.3050300@jitsi.org> <509C3066.6020209@cisco.com> <509C3F33.8060809@jitsi.org> <509D1200.3070702@cisco.com>
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Subject: Re: [MMUSIC] Signaling trickle support
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On 11/09/2012 03:24 PM, Flemming Andreasen wrote:
>
> On 11/8/12 6:24 PM, Emil Ivov wrote:
>> Hey Flemming,
>>
>> On 08.11.12, 17:21, Flemming Andreasen wrote:
>>>>>> Since protocols like SIP don't really have things like XMPP disco 
>>>>>> and
>>>>>> Cap Neg that would have allowed it, we were considering other 
>>>>>> options.
>>>>>> One such option would be to send a trickle offer and format it in 
>>>>>> a way
>>>>>> that would cause a non-trickle answerer to reject it in a 
>>>>>> _predictable_
>>>>>> way. This would allow the offerer to retry with a regular ICE offer.
>>>>> I thought the point with trickle-ICE was to reduce call setup latency
>>>> It is.
>>>>
>>>>> and hence you would want to make sure that things complete in as few
>>>>> round-trips as possible. This doesn't seem to line up with that (?).
>>>> The thing is that in cases where trickle ICE support cannot be
>>>> pre-confirmed you have to fall back to half trickle. The probing
>>>> requests would hence allow for an optimisation compared to that.
>>> Somewhat unclear to me as it builds upon an assumption that it is
>>> somehow "more efficient"to probe than to collect candidates.
>> Indeed, this would not always be the case and it would basically come
>> down to comparing one signalling RTT to the duration of candidate
>> gathering.
>>
>> As you agreed in one of your previous mails, the relation may vary:
> I still agree with that :-)
>> Executing a single STUN transaction is likely to be less time consuming
>> than a signalling RTT. Candidate gathering however may also last a lot
>> longer than a single offer/answer exchange.
>>
>> However, an agent that has several interfaces and uses them to gather
>> candidates for three streams that don't have bundle nor RTCP muxing is
>> going to need quite some time to complete a harvest. Add to this UPnP
>> allocations and a second STUN server and you could easily find the
>> signalling RTT tens of times shorter than the candidate gathering.
> Because of the min. 20 ms Ta timer constraints, which apply 
> cross-media stream ? Can somebody refresh my mind as to why that is 
> (e.g. are residential NATs at this point generally not capable of 
> creating NAT bindings faster than at 20 ms intervals) ?
I have been told that in BEHAVE testing, NATs were found that would 
behave incoherently at short binding creation intervals, but none were 
found that couldn't handle one new binding every 20 ms, so this was set 
at the minimum interval for potentially binding-creation actions.

             Harald


From dwing@cisco.com  Tue Nov 13 08:39:01 2012
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From: "Dan Wing" <dwing@cisco.com>
To: "'Harald Alvestrand'" <harald@alvestrand.no>, <mmusic@ietf.org>
References: <CABkgnnVR-prV8xgzPx9H-MYRM_vTui-8GGfTD0yy1JPvJJzrrQ@mail.gmail.com>	<509AB627.1090507@alum.mit.edu> <509B5F87.1010807@ericsson.com>	<7594FB04B1934943A5C02806D1A2204B027800@ESESSMB209.ericsson.se>	<509BC947.709@cisco.com> <509BCBFB.2020305@jitsi.org>	<CABkgnnVMf53Vk_Rd-0erZN4ZLDKM4ePOvMfLJnfOUNVxdWYLqQ@mail.gmail.com>	<509BD563.9080808@cisco.com> <509BDE01.9060705@jitsi.org>	<509C02F2.10303@cisco.com> <509C0724.3050300@jitsi.org>	<509C3066.6020209@cisco.com> <509C3F33.8060809@jitsi.org>	<509D1200.3070702@cisco.com> <50A20CE5.5000200@alvestrand.no>
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Date: Tue, 13 Nov 2012 08:38:57 -0800
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Cc: Flemming Andreasen <fandreas@cisco.com>
Subject: Re: [MMUSIC] Signaling trickle support
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Harald Alvestrand wrote:
> On 11/09/2012 03:24 PM, Flemming Andreasen wrote:
...
> > Because of the min. 20 ms Ta timer constraints, which apply
> > cross-media stream ? Can somebody refresh my mind as to why that is
> > (e.g. are residential NATs at this point generally not capable of
> > creating NAT bindings faster than at 20 ms intervals) ?
>
> I have been told that in BEHAVE testing, NATs were found that would
> behave incoherently at short binding creation intervals, but none were
> found that couldn't handle one new binding every 20 ms, so this was set
> at the minimum interval for potentially binding-creation actions.

The behavior was noticed during deployment, not BEHAVE testing -- 
otherwise I would have a nice BEHAVE test report to cite.

And that NAT behavior, combined with ephemeral source ports for DNS
queries, is probably what is tripping up both Microsoft and Chromium, 
causing them to limit themselves to 6 (or so) DNS queries at a time, 
http://code.google.com/p/chromium/issues/detail?id=122566.  I am
told Windows 8 also restricts its number of simultaneous DNS queries.
If that NAT behavior is really what is tripping up Windows and
Chromium, it would be spiffy if -- instead of limiting the number
of outstanding queries -- the software would pace the queries, or
use different behavior when querying the local caching nameserver
which shouldn't suffer that problem.

-d



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From: Flemming Andreasen <fandreas@cisco.com>
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To: Dan Wing <dwing@cisco.com>
References: <CABkgnnVR-prV8xgzPx9H-MYRM_vTui-8GGfTD0yy1JPvJJzrrQ@mail.gmail.com>	<509AB627.1090507@alum.mit.edu> <509B5F87.1010807@ericsson.com>	<7594FB04B1934943A5C02806D1A2204B027800@ESESSMB209.ericsson.se>	<509BC947.709@cisco.com> <509BCBFB.2020305@jitsi.org>	<CABkgnnVMf53Vk_Rd-0erZN4ZLDKM4ePOvMfLJnfOUNVxdWYLqQ@mail.gmail.com>	<509BD563.9080808@cisco.com> <509BDE01.9060705@jitsi.org>	<509C02F2.10303@cisco.com> <509C0724.3050300@jitsi.org>	<509C3066.6020209@cisco.com> <509C3F33.8060809@jitsi.org>	<509D1200.3070702@cisco.com> <50A20CE5.5000200@alvestrand.no> <072901cdc1bd$60d2cd60$22786820$@cisco.com>
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Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] Signaling trickle support
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On 11/13/12 11:38 AM, Dan Wing wrote:
> Harald Alvestrand wrote:
>> On 11/09/2012 03:24 PM, Flemming Andreasen wrote:
> ...
>>> Because of the min. 20 ms Ta timer constraints, which apply
>>> cross-media stream ? Can somebody refresh my mind as to why that is
>>> (e.g. are residential NATs at this point generally not capable of
>>> creating NAT bindings faster than at 20 ms intervals) ?
>> I have been told that in BEHAVE testing, NATs were found that would
>> behave incoherently at short binding creation intervals, but none were
>> found that couldn't handle one new binding every 20 ms, so this was set
>> at the minimum interval for potentially binding-creation actions.
> The behavior was noticed during deployment, not BEHAVE testing --
> otherwise I would have a nice BEHAVE test report to cite.
Does the concern remain today or can we start being more aggressive in 
some of these ?

I've been told (second hand) that Google Maps routinely opens 50 TCP 
connections at a time, but don't have any details as to the speed with 
which they are opened. Anybody have any further insights or similar 
examples (that seem to work in practice) ?

> And that NAT behavior, combined with ephemeral source ports for DNS
> queries, is probably what is tripping up both Microsoft and Chromium,
> causing them to limit themselves to 6 (or so) DNS queries at a time,
> http://code.google.com/p/chromium/issues/detail?id=122566.  I am
> told Windows 8 also restricts its number of simultaneous DNS queries.
> If that NAT behavior is really what is tripping up Windows and
> Chromium, it would be spiffy if -- instead of limiting the number
> of outstanding queries -- the software would pace the queries, or
> use different behavior when querying the local caching nameserver
> which shouldn't suffer that problem.
Interesting - thanks for the data point(er).

-- Flemming
> -d
>
>
> .
>


From harald@alvestrand.no  Thu Nov 15 00:47:33 2012
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Subject: [MMUSIC] Wisdom sought: Prioritization of codecs
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I've been searching RFC 4566 (SDP) for this info, but came up blank....

in an M-line, we typically have

m=video blabla 111 112 113
a=rtpmap:111 ....
a=rtpmap:112 ...
a=rtpmap:113 ....

Section 5.14 says this:

       If the <proto> sub-field is "RTP/AVP" or "RTP/SAVP" the <fmt>
       sub-fields contain RTP payload type numbers.  When a list of
       payload type numbers is given, this implies that all of these
       payload formats MAY be used in the session, but the first of these
       formats SHOULD be used as the default format for the session.

Now, the questions...

1) In offer/answer, does this sequence in an offer mean that the offerer 
prefers that the answerer use codec 111, if supported?

2) in offer/answer, does this sequence in an answer mean that the 
answerer prefers that the offerer use codec 111?

3) is there any preference expressed between codec 112 and codec 113?

4) Should the sequence of a=rtpmap: lines be completely ignored, or does 
it matter?

I'm trying to fix a bug, and want to make sure I fix it according to spec.

                Harald




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<html>
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    <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    I've been searching RFC 4566 (SDP) for this info, but came up
    blank....<br>
    <br>
    in an M-line, we typically have<br>
    <br>
    m=video blabla 111 112 113<br>
    a=rtpmap:111 ....<br>
    a=rtpmap:112 ...<br>
    a=rtpmap:113 ....<br>
    <br>
    Section 5.14 says this:<br>
    <br>
    <meta http-equiv="content-type" content="text/html;
      charset=ISO-8859-1">
    <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always; color: rgb(0, 0, 0); font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: start; text-indent: 0px; text-transform: none; widows: 2; word-spacing: 0px; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px;">      If the &lt;proto&gt; sub-field is "RTP/AVP" or "RTP/SAVP" the &lt;fmt&gt;
      sub-fields contain RTP payload type numbers.  When a list of
      payload type numbers is given, this implies that all of these
      payload formats MAY be used in the session, but the first of these
      formats SHOULD be used as the default format for the session.

</pre>
    Now, the questions...<br>
    <br>
    1) In offer/answer, does this sequence in an offer mean that the
    offerer prefers that the answerer use codec 111, if supported?<br>
    <br>
    2) in offer/answer, does this sequence in an answer mean that the
    answerer prefers that the offerer use codec 111?<br>
    <br>
    3) is there any preference expressed between codec 112 and codec
    113?<br>
    <br>
    4) Should the sequence of a=rtpmap: lines be completely ignored, or
    does it matter?<br>
    <br>
    I'm trying to fix a bug, and want to make sure I fix it according to
    spec.<br>
    <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
    <br>
    <br>
    <br>
  </body>
</html>

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From harald@alvestrand.no  Thu Nov 15 00:57:41 2012
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Subject: Re: [MMUSIC] Wisdom sought: Prioritization of codecs
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Found some answers in RFC 3264.

On 11/15/2012 09:47 AM, Harald Alvestrand wrote:
> I've been searching RFC 4566 (SDP) for this info, but came up blank....
>
> in an M-line, we typically have
>
> m=video blabla 111 112 113
> a=rtpmap:111 ....
> a=rtpmap:112 ...
> a=rtpmap:113 ....
>
> Section 5.14 says this:
>
>        If the <proto> sub-field is "RTP/AVP" or "RTP/SAVP" the <fmt>
>        sub-fields contain RTP payload type numbers.  When a list of
>        payload type numbers is given, this implies that all of these
>        payload formats MAY be used in the session, but the first of these
>        formats SHOULD be used as the default format for the session.
>
> Now, the questions...
>
> 1) In offer/answer, does this sequence in an offer mean that the 
> offerer prefers that the answerer use codec 111, if supported?

3264 5.1:

    In all cases, the formats in the "m=" line MUST be listed in order of
    preference, with the first format listed being preferred.  In this
    case, preferred means that the recipient of the offer SHOULD use the
    format with the highest preference that is acceptable to it.


>
> 2) in offer/answer, does this sequence in an answer mean that the 
> answerer prefers that the offerer use codec 111?
3264 6.1:

   The
    media formats in the "m=" line MUST be listed in order of preference,
    with the first format listed being preferred.  In this case,
    preferred means that the offerer SHOULD use the format with the
    highest preference from the answer.



>
> 3) is there any preference expressed between codec 112 and codec 113?
Clear from 3264. If we ever update 4566, might want to harmonize those.
>
> 4) Should the sequence of a=rtpmap: lines be completely ignored, or 
> does it matter?

I hope the answer is "completely ignored", have found no evidence to the 
contrary.

>
> I'm trying to fix a bug, and want to make sure I fix it according to spec.
>
>                Harald
>
>
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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    <div class="moz-cite-prefix">Found some answers in RFC 3264.<br>
      <br>
      On 11/15/2012 09:47 AM, Harald Alvestrand wrote:<br>
    </div>
    <blockquote cite="mid:50A4AC20.5030306@alvestrand.no" type="cite">
      <meta http-equiv="content-type" content="text/html;
        charset=ISO-8859-1">
      I've been searching RFC 4566 (SDP) for this info, but came up
      blank....<br>
      <br>
      in an M-line, we typically have<br>
      <br>
      m=video blabla 111 112 113<br>
      a=rtpmap:111 ....<br>
      a=rtpmap:112 ...<br>
      a=rtpmap:113 ....<br>
      <br>
      Section 5.14 says this:<br>
      <br>
      <meta http-equiv="content-type" content="text/html;
        charset=ISO-8859-1">
      <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always; color: rgb(0, 0, 0); font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: start; text-indent: 0px; text-transform: none; widows: 2; word-spacing: 0px; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px;">      If the &lt;proto&gt; sub-field is "RTP/AVP" or "RTP/SAVP" the &lt;fmt&gt;
      sub-fields contain RTP payload type numbers.  When a list of
      payload type numbers is given, this implies that all of these
      payload formats MAY be used in the session, but the first of these
      formats SHOULD be used as the default format for the session.

</pre>
      Now, the questions...<br>
      <br>
      1) In offer/answer, does this sequence in an offer mean that the
      offerer prefers that the answerer use codec 111, if supported?<br>
    </blockquote>
    <br>
    3264 5.1:<br>
    <br>
    <meta http-equiv="content-type" content="text/html;
      charset=ISO-8859-1">
    <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always; color: rgb(0, 0, 0); font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: start; text-indent: 0px; text-transform: none; widows: 2; word-spacing: 0px; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px;">   In all cases, the formats in the "m=" line MUST be listed in order of
   preference, with the first format listed being preferred.  In this
   case, preferred means that the recipient of the offer SHOULD use the
   format with the highest preference that is acceptable to it.
</pre>
    <br>
    <blockquote cite="mid:50A4AC20.5030306@alvestrand.no" type="cite"> <br>
      2) in offer/answer, does this sequence in an answer mean that the
      answerer prefers that the offerer use codec 111?<br>
    </blockquote>
    3264 6.1:<br>
    <br>
    <meta http-equiv="content-type" content="text/html;
      charset=ISO-8859-1">
    <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always; color: rgb(0, 0, 0); font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: start; text-indent: 0px; text-transform: none; widows: 2; word-spacing: 0px; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px;">  The
   media formats in the "m=" line MUST be listed in order of preference,
   with the first format listed being preferred.  In this case,
   preferred means that the offerer SHOULD use the format with the
   highest preference from the answer.
</pre>
    <br class="Apple-interchange-newline">
    <br>
    <blockquote cite="mid:50A4AC20.5030306@alvestrand.no" type="cite"> <br>
      3) is there any preference expressed between codec 112 and codec
      113?<br>
    </blockquote>
    Clear from 3264. If we ever update 4566, might want to harmonize
    those.<br>
    <blockquote cite="mid:50A4AC20.5030306@alvestrand.no" type="cite"> <br>
      4) Should the sequence of a=rtpmap: lines be completely ignored,
      or does it matter?<br>
    </blockquote>
    <br>
    I hope the answer is "completely ignored", have found no evidence to
    the contrary.<br>
    <br>
    <blockquote cite="mid:50A4AC20.5030306@alvestrand.no" type="cite"> <br>
      I'm trying to fix a bug, and want to make sure I fix it according
      to spec.<br>
      <br>
      &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
      <br>
      <br>
      <br>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
mmusic mailing list
<a class="moz-txt-link-abbreviated" href="mailto:mmusic@ietf.org">mmusic@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a>
</pre>
    </blockquote>
    <br>
  </body>
</html>

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From ernst.horvath@siemens-enterprise.com  Thu Nov 15 01:05:11 2012
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From: "Horvath, Ernst" <ernst.horvath@siemens-enterprise.com>
To: Harald Alvestrand <harald@alvestrand.no>, mmusic <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Wisdom sought: Prioritization of codecs
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Harald,

RFC 3264 states that codecs are listed in order of preference, most preferr=
ed first, in both offer and answer. When selecting the acceptable codecs fr=
om the offered list the answerer should keep the order unless there is good=
 reason not to do so. The offerer must be prepared to receive an answer wit=
h codecs arranged in a different order. Although both sides can then use an=
y of the negotiated codecs it is recommended they use the first in the list=
.

Regards,
Ernst

________________________________
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 Harald Alvestrand
Sent: Donnerstag, 15. November 2012 09:47
To: mmusic
Subject: [MMUSIC] Wisdom sought: Prioritization of codecs

I've been searching RFC 4566 (SDP) for this info, but came up blank....

in an M-line, we typically have

m=3Dvideo blabla 111 112 113
a=3Drtpmap:111 ....
a=3Drtpmap:112 ...
a=3Drtpmap:113 ....

Section 5.14 says this:


      If the <proto> sub-field is "RTP/AVP" or "RTP/SAVP" the <fmt>
      sub-fields contain RTP payload type numbers.  When a list of
      payload type numbers is given, this implies that all of these
      payload formats MAY be used in the session, but the first of these
      formats SHOULD be used as the default format for the session.



Now, the questions...

1) In offer/answer, does this sequence in an offer mean that the offerer pr=
efers that the answerer use codec 111, if supported?

2) in offer/answer, does this sequence in an answer mean that the answerer =
prefers that the offerer use codec 111?

3) is there any preference expressed between codec 112 and codec 113?

4) Should the sequence of a=3Drtpmap: lines be completely ignored, or does =
it matter?

I'm trying to fix a bug, and want to make sure I fix it according to spec.

               Harald




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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
<meta name=3D"GENERATOR" content=3D"MSHTML 8.00.6001.19328">
</head>
<body bgcolor=3D"#ffffff" text=3D"#000000">
<div><span class=3D"742515308-15112012"><font color=3D"#0000ff" size=3D"2" =
face=3D"Arial">Harald,</font></span></div>
<div><span class=3D"742515308-15112012"><font color=3D"#0000ff" size=3D"2" =
face=3D"Arial"></font></span>&nbsp;</div>
<div><span class=3D"742515308-15112012"><font color=3D"#0000ff" size=3D"2" =
face=3D"Arial">RFC 3264 states that codecs are listed in order of preferenc=
e, most preferred first, in both offer and answer. When selecting the accep=
table codecs from the offered list the answerer
 should keep the order unless there is good reason not to do so. The offere=
r must be prepared to receive an answer with codecs arranged in a different=
 order. Although both sides can then use any of the&nbsp;negotiated codecs =
it is recommended they use the first
 in the list.</font></span></div>
<div><span class=3D"742515308-15112012"><font color=3D"#0000ff" size=3D"2" =
face=3D"Arial"></font></span>&nbsp;</div>
<div><span class=3D"742515308-15112012"><font color=3D"#0000ff" size=3D"2" =
face=3D"Arial">Regards,</font></span></div>
<div><span class=3D"742515308-15112012"><font color=3D"#0000ff" size=3D"2" =
face=3D"Arial">Ernst</font></span></div>
<br>
<blockquote style=3D"BORDER-LEFT: #0000ff 2px solid; PADDING-LEFT: 5px; MAR=
GIN-LEFT: 5px; MARGIN-RIGHT: 0px">
<div dir=3D"ltr" lang=3D"en-us" class=3D"OutlookMessageHeader" align=3D"lef=
t">
<hr tabindex=3D"-1">
<font size=3D"2" face=3D"Tahoma"><b>From:</b> mmusic-bounces@ietf.org [mail=
to:mmusic-bounces@ietf.org]
<b>On Behalf Of </b>Harald Alvestrand<br>
<b>Sent:</b> Donnerstag, 15. November 2012 09:47<br>
<b>To:</b> mmusic<br>
<b>Subject:</b> [MMUSIC] Wisdom sought: Prioritization of codecs<br>
</font><br>
</div>
<div></div>
I've been searching RFC 4566 (SDP) for this info, but came up blank....<br>
<br>
in an M-line, we typically have<br>
<br>
m=3Dvideo blabla 111 112 113<br>
a=3Drtpmap:111 ....<br>
a=3Drtpmap:112 ...<br>
a=3Drtpmap:113 ....<br>
<br>
Section 5.14 says this:<br>
<br>
<pre style=3D"PAGE-BREAK-BEFORE: always; LINE-HEIGHT: normal; WIDOWS: 2; TE=
XT-TRANSFORM: none; FONT-VARIANT: normal; FONT-STYLE: normal; MARGIN-TOP: 0=
px; TEXT-INDENT: 0px; ORPHANS: 2; MARGIN-BOTTOM: 0px; LETTER-SPACING: norma=
l; COLOR: rgb(0,0,0); FONT-SIZE: 1em; FONT-WEIGHT: normal; WORD-SPACING: 0p=
x; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px" class=3D=
"newpage">      If the &lt;proto&gt; sub-field is &quot;RTP/AVP&quot; or &q=
uot;RTP/SAVP&quot; the &lt;fmt&gt;
      sub-fields contain RTP payload type numbers.  When a list of
      payload type numbers is given, this implies that all of these
      payload formats MAY be used in the session, but the first of these
      formats SHOULD be used as the default format for the session.

</pre>
Now, the questions...<br>
<br>
1) In offer/answer, does this sequence in an offer mean that the offerer pr=
efers that the answerer use codec 111, if supported?<br>
<br>
2) in offer/answer, does this sequence in an answer mean that the answerer =
prefers that the offerer use codec 111?<br>
<br>
3) is there any preference expressed between codec 112 and codec 113?<br>
<br>
4) Should the sequence of a=3Drtpmap: lines be completely ignored, or does =
it matter?<br>
<br>
I'm trying to fix a bug, and want to make sure I fix it according to spec.<=
br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp; Harald<br>
<br>
<br>
<br>
</blockquote>
</body>
</html>

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From Simo.Veikkolainen@nokia.com  Thu Nov 15 01:06:18 2012
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From: <Simo.Veikkolainen@nokia.com>
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Subject: [MMUSIC] E.164 and visual separators in draft-ietf-mmusic-sdp-cs
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Hi Flemming, Paul, Keith, and all,

When going through http://tools.ietf.org/id/draft-ietf-mmusic-sdp-cs-13.txt=
 for publication, Flemming had a comment on the format of global telephone =
numbers.

The draft currently says in Section 5.2.1:

  This memo exclusively uses the international representation of E.164
   numbers, i.e., those initiated with a '+' sign.  The syntax (see
   Section 5.7) refers to the representation of a 'global-number'
   construction already specified in RFC 3966 [RFC3966].  This
   representation requires the presence of the '+' sign.  Additionally,
   this representation allows for the presence of one or more 'visual-
   separator' constructions.  Implementations confirming to this
   specification and using the "E164" address type together with the
   "PSTN" network type MUST only use international E.164, i.e., those
   starting with a '+' sign and SHOULD NOT use visual-separators.

This was discussed earlier on the list, see  http://www.ietf.org/mail-archi=
ve/web/mmusic/current/msg09630.html

Flemming's comment was that referencing to RFC3966 but saying that the synt=
ax is invalid because we only use parts of it is not a good idea.

The question is, could we just refer to RFC3966 and allow visual separators=
?

In my understanding the tradeoff is between slightly increased complexity i=
n parser implementation versus being clear on the syntax. I would be inclin=
ed to prefer the latter.

Regards,
Simo





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<p class=3D"MsoNormal">Hi Flemming, Paul, Keith, and all,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">When going through <a href=3D"http://tools.ietf.org/=
id/draft-ietf-mmusic-sdp-cs-13.txt">
http://tools.ietf.org/id/draft-ietf-mmusic-sdp-cs-13.txt</a> for publicatio=
n, Flemming had a comment on the format of global telephone numbers.<o:p></=
o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The draft currently says in Section 5.2.1:<o:p></o:p=
></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp;This memo exclusively uses the international r=
epresentation of E.164<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; numbers, i.e., those initiated with a '&#43;'=
 sign.&nbsp; The syntax (see<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; Section 5.7) refers to the representation of =
a 'global-number'<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; construction already specified in RFC 3966 [R=
FC3966].&nbsp; This<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; representation requires the presence of the '=
&#43;' sign.&nbsp; Additionally,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; this representation allows for the presence o=
f one or more 'visual-<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; separator' constructions.&nbsp; Implementatio=
ns confirming to this<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; specification and using the &quot;E164&quot; =
address type together with the<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; &quot;PSTN&quot; network type MUST only use i=
nternational E.164, i.e., those<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; starting with a '&#43;' sign and SHOULD NOT u=
se visual-separators.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">This was discussed earlier on the list, see &nbsp;<a=
 href=3D"http://www.ietf.org/mail-archive/web/mmusic/current/msg09630.html"=
>http://www.ietf.org/mail-archive/web/mmusic/current/msg09630.html</a><o:p>=
</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Flemming&#8217;s comment was that referencing to RFC=
3966 but saying that the syntax is invalid because we only use parts of it =
is not a good idea.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The question is, could we just refer to RFC3966 and =
allow visual separators?
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">In my understanding the tradeoff is between slightly=
 increased complexity in parser implementation versus being clear on the sy=
ntax. I would be inclined to prefer the latter.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal">Simo<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</body>
</html>

--_000_D09DAE6B636851459F7575D146EFB54B1A3BD4C3008AM1MPN1025mg_--

From pkyzivat@alum.mit.edu  Thu Nov 15 06:56:19 2012
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Subject: Re: [MMUSIC] E.164 and visual separators in draft-ietf-mmusic-sdp-cs
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On 11/15/12 4:06 AM, Simo.Veikkolainen@nokia.com wrote:
> Hi Flemming, Paul, Keith, and all,
>
> When going through
> http://tools.ietf.org/id/draft-ietf-mmusic-sdp-cs-13.txt for
> publication, Flemming had a comment on the format of global telephone
> numbers.
>
> The draft currently says in Section 5.2.1:
>
>    This memo exclusively uses the international representation of E.164
>
>     numbers, i.e., those initiated with a '+' sign.  The syntax (see
>
>     Section 5.7) refers to the representation of a 'global-number'
>
>     construction already specified in RFC 3966 [RFC3966].  This
>
>     representation requires the presence of the '+' sign.  Additionally,
>
>     this representation allows for the presence of one or more 'visual-
>
>     separator' constructions.  Implementations confirming to this
>
>     specification and using the "E164" address type together with the
>
>     "PSTN" network type MUST only use international E.164, i.e., those
>
>     starting with a '+' sign and SHOULD NOT use visual-separators.
>
> This was discussed earlier on the list, see
> http://www.ietf.org/mail-archive/web/mmusic/current/msg09630.html
>
> Flemming’s comment was that referencing to RFC3966 but saying that the
> syntax is invalid because we only use parts of it is not a good idea.
>
> The question is, could we just refer to RFC3966 and allow visual
> separators?
>
> In my understanding the tradeoff is between slightly increased
> complexity in parser implementation versus being clear on the syntax. I
> would be inclined to prefer the latter.

My personal preference is, like yours, to go with 3966 and permit 
separators. IIRC the reason for text saying not to use separators was 
that some others were opposed to having them. (Early versions of the 
draft allowed only digits and didn't refer to 3966.)

	Thanks,
	Paul


From magnus.westerlund@ericsson.com  Thu Nov 15 07:08:23 2012
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Hi Flemming,

Thanks for the comments.

On 2012-10-11 21:35, Flemming Andreasen wrote:
> I have reviewed the RTSP NAT Evaluation document and have several
> comments on it, both technical and editorial. I will provide the main
> technical comments below and send the minor and editorial comments
> directly to the authors in a more "user-friendly" format.
> 
> 
> Major comments
> 
> - STUN based on either RFC 3489 or RFC 5389 (or both). The draft
> discusses use of STUN as a NAT traversal mechanism (with some RTSP
> operational "extensions"), and in so doing it references both RFC 3489
> (which is obsolete) and RFC 5389 (the replacement). Furthermore, in
> several places, it bases and describes operation on now defunct
> (obsolete) RFC 3489 logic and messages instead of RFC 5389. This either
> needs to be rectified, or we should have a very good explanation in the
> document as to why we are using RFC 3489 (and then be clear which parts
> are based on RFC 3489 and which are based on RFC 5389).

I intended to go with an background explanation. This document was
evaluating RTSP NAT proposals at a time and in a context which is
significantly before RFC 5389. Yes, the limits to the classification
techniques exist and should be correctly described. But the fact that
they was considered can not be undone in a document intended to document
the considerations done at that time of picking ICE as a solution. Which
is now 7+ year into history.

> 
> - "Symmetric RTP" is called out as a potential solution for NAT
> traversal with 3 different variations. Note that symmetric RTP is
> defined in RFC 4961, and what is described in this document as
> "symmetric RTP" looks a lot more like latching to me, with all the
> potential issues that brings (and then some due to the shared use of
> ports and hence shared SSRC space between different clients, as noted in
> the document). The document needs to clear up its terminology in this
> area ("connect", "binding", latch, symmetric RTP, etc.) and be clearer
> about what is being proposed (which is latching as far as I can tell,
> however it makes reference to certain mechanisms that aren't fully
> described, e.g. binding packets with random nonces).

Yes, it is what called latching today. When this was originally written
latching wasn't called latching in this community. But I will work
through the text to try avoid any missunderstandings. There is after all
a basic requirement on latching that one uses symmetric RTP after the
latching or things don't work.

> 
> 
> Other significant comments
> 
> - The draft intends to investigate and outline different RTSP NAT
> traversal proposals with the goal of picking one for further detailed
> specification (which can be found in the draft-ietf-mmusic-rtsp-nat
> spec). There is a mixture of RFC 2119 language and more informal
> language in several of these descriptions. Given that these are
> high-level descriptions and have not been specified in detail, I do not
> think we should give any other impression and hence I think RFC 2119
> language should be avoided.

Fully agreed. There should be no RFC 2119 language in this doc.

> 
> - The draft suggests that the RTSP server needs to enforce that
> signaling and media come from the same IP-address as a security remedy
> in several places, however this may not work with larger scale NATs
> (e.g. carrier-grade NATs) where the NAT may be using more than one
> external IP-address.

It is more than a suggestion. It is an fact that servers implements this
protection. If a CGN fails to give a host the same IP address then
unfortunately that client will not get any content. I will claim it is
the CGN that is more broken than the security mechanism.


> 
> - The comparison section is missing two of the alternatives considered
> (namely ALG and TCP tunneling). Also, the comparison chart would lead
> most people to a different solution than the one chosen. To some extent,
> I think it's because the requirements list is a simplified list of all
> the considerations behind each of the proposals (as more fully explained
> in the document overall). There are additional drawbacks to some of
> these other solutions that do not show up in the table (e.g.
> unauthenticated "latching" and requirement for RTSP server to have
> public IP-address). It would be worthwhile briefly recapping those and
> hence further motivate the choice of ICE in this section.

Ok, will see what I can do with reasonable work effort.

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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On 11/15/12 4:06 AM, Simo.Veikkolainen@nokia.com wrote:
>
> Hi Flemming, Paul, Keith, and all,
>
> When going through 
> http://tools.ietf.org/id/draft-ietf-mmusic-sdp-cs-13.txt 
> <http://tools.ietf.org/id/draft-ietf-mmusic-sdp-cs-13.txt> for 
> publication, Flemming had a comment on the format of global telephone 
> numbers.
>
> The draft currently says in Section 5.2.1:
>
>   This memo exclusively uses the international representation of E.164
>
> numbers, i.e., those initiated with a '+' sign.  The syntax (see
>
> Section 5.7) refers to the representation of a 'global-number'
>
> construction already specified in RFC 3966 [RFC3966].  This
>
> representation requires the presence of the '+' sign. Additionally,
>
> this representation allows for the presence of one or more 'visual-
>
> separator' constructions.  Implementations confirming to this
>
> specification and using the "E164" address type together with the
>
> "PSTN" network type MUST only use international E.164, i.e., those
>
> starting with a '+' sign and SHOULD NOT use visual-separators.
>
> This was discussed earlier on the list, see 
> http://www.ietf.org/mail-archive/web/mmusic/current/msg09630.html
>
> Flemming's comment was that referencing to RFC3966 but saying that the 
> syntax is invalid because we only use parts of it is not a good idea.
>
> The question is, could we just refer to RFC3966 and allow visual 
> separators?
>
To be clear, the "global-number" production in RFC 3966 allows for a 
number of parameters that are not part of the International E.164 format 
and thus I have two concerns with the current text
1) The visual-separator exclusion although the ABNF allows for it
2) The apparent disconnect between all of these optional extra 
parameters and the above text suggesting they are not allowed.

Thanks

-- Flemming

> In my understanding the tradeoff is between slightly increased 
> complexity in parser implementation versus being clear on the syntax. 
> I would be inclined to prefer the latter.
>
> Regards,
>
> Simo
>


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    <br>
    <div class="moz-cite-prefix">On 11/15/12 4:06 AM,
      <a class="moz-txt-link-abbreviated" href="mailto:Simo.Veikkolainen@nokia.com">Simo.Veikkolainen@nokia.com</a> wrote:<br>
    </div>
    <blockquote
cite="mid:D09DAE6B636851459F7575D146EFB54B1A3BD4C3@008-AM1MPN1-025.mgdnok.nokia.com"
      type="cite">
      <meta http-equiv="Content-Type" content="text/html;
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      <div class="WordSection1">
        <p class="MsoNormal">Hi Flemming, Paul, Keith, and all,<o:p></o:p></p>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        <p class="MsoNormal">When going through <a
            moz-do-not-send="true"
            href="http://tools.ietf.org/id/draft-ietf-mmusic-sdp-cs-13.txt">
            http://tools.ietf.org/id/draft-ietf-mmusic-sdp-cs-13.txt</a>
          for publication, Flemming had a comment on the format of
          global telephone numbers.<o:p></o:p></p>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        <p class="MsoNormal">The draft currently says in Section 5.2.1:<o:p></o:p></p>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier New&quot;">&nbsp;&nbsp;This
            memo exclusively uses the international representation of
            E.164<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier New&quot;">&nbsp;&nbsp;
            numbers, i.e., those initiated with a '+' sign.&nbsp; The syntax
            (see<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier New&quot;">&nbsp;&nbsp;
            Section 5.7) refers to the representation of a
            'global-number'<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier New&quot;">&nbsp;&nbsp;
            construction already specified in RFC 3966 [RFC3966].&nbsp; This<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier New&quot;">&nbsp;&nbsp;
            representation requires the presence of the '+' sign.&nbsp;
            Additionally,<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier New&quot;">&nbsp;&nbsp;
            this representation allows for the presence of one or more
            'visual-<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier New&quot;">&nbsp;&nbsp;
            separator' constructions.&nbsp; Implementations confirming to
            this<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier New&quot;">&nbsp;&nbsp;
            specification and using the "E164" address type together
            with the<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier New&quot;">&nbsp;&nbsp;
            "PSTN" network type MUST only use international E.164, i.e.,
            those<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier New&quot;">&nbsp;&nbsp;
            starting with a '+' sign and SHOULD NOT use
            visual-separators.<o:p></o:p></span></p>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        <p class="MsoNormal">This was discussed earlier on the list, see
          &nbsp;<a moz-do-not-send="true"
            href="http://www.ietf.org/mail-archive/web/mmusic/current/msg09630.html">http://www.ietf.org/mail-archive/web/mmusic/current/msg09630.html</a><o:p></o:p></p>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        <p class="MsoNormal">Flemming&#8217;s comment was that referencing to
          RFC3966 but saying that the syntax is invalid because we only
          use parts of it is not a good idea.<o:p></o:p></p>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        <p class="MsoNormal">The question is, could we just refer to
          RFC3966 and allow visual separators?
          <o:p></o:p></p>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
      </div>
    </blockquote>
    To be clear, the "global-number" production in RFC 3966 allows for a
    number of parameters that are not part of the International E.164
    format and thus I have two concerns with the current text<br>
    1) The visual-separator exclusion although the ABNF allows for it<br>
    2) The apparent disconnect between all of these optional extra
    parameters and the above text suggesting they are not allowed.<br>
    <br>
    Thanks <br>
    <br>
    -- Flemming <br>
    <br>
    <blockquote
cite="mid:D09DAE6B636851459F7575D146EFB54B1A3BD4C3@008-AM1MPN1-025.mgdnok.nokia.com"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal">In my understanding the tradeoff is between
          slightly increased complexity in parser implementation versus
          being clear on the syntax. I would be inclined to prefer the
          latter.<o:p></o:p></p>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        <p class="MsoNormal">Regards,<o:p></o:p></p>
        <p class="MsoNormal">Simo<o:p></o:p></p>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
      </div>
    </blockquote>
    <br>
  </body>
</html>

--------------080405050107010403050803--

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Subject: Re: [MMUSIC] WGLC for draft-ietf-mmusic-rtsp-nat-evaluation-05
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On 11/15/12 10:08 AM, Magnus Westerlund wrote:
> Hi Flemming,
>
> Thanks for the comments.
>
> On 2012-10-11 21:35, Flemming Andreasen wrote:
>> I have reviewed the RTSP NAT Evaluation document and have several
>> comments on it, both technical and editorial. I will provide the main
>> technical comments below and send the minor and editorial comments
>> directly to the authors in a more "user-friendly" format.
>>
>>
>> Major comments
>>
>> - STUN based on either RFC 3489 or RFC 5389 (or both). The draft
>> discusses use of STUN as a NAT traversal mechanism (with some RTSP
>> operational "extensions"), and in so doing it references both RFC 3489
>> (which is obsolete) and RFC 5389 (the replacement). Furthermore, in
>> several places, it bases and describes operation on now defunct
>> (obsolete) RFC 3489 logic and messages instead of RFC 5389. This either
>> needs to be rectified, or we should have a very good explanation in the
>> document as to why we are using RFC 3489 (and then be clear which parts
>> are based on RFC 3489 and which are based on RFC 5389).
> I intended to go with an background explanation. This document was
> evaluating RTSP NAT proposals at a time and in a context which is
> significantly before RFC 5389. Yes, the limits to the classification
> techniques exist and should be correctly described. But the fact that
> they was considered can not be undone in a document intended to document
> the considerations done at that time of picking ICE as a solution. Which
> is now 7+ year into history.
Perhaps this is a more general question since we have at least other 
document in the WG with a similar issue:

- For documents that have been underway for a long time, and the 
technology/references they describe has since evolved, should we base 
this on the original technology and knowledge at the time or should it 
reflect the state of affairs at time of publication ?

I tend to think it should be the latter, although I don't see anything 
with also including references to older technology, as long as the 
limitations are noted, the new technology is called out, and we don't 
make any normative recommendations around use of the old technology 
unless there is a very good reason to do so.

In the specific case, is there any particular reason (apart from this 
obviously being more/new work again) why we wouldn't reference RFC 5389 
at this point and possibly note the limitations of the RFC 3489 approach 
as well (for background) ?

>> - "Symmetric RTP" is called out as a potential solution for NAT
>> traversal with 3 different variations. Note that symmetric RTP is
>> defined in RFC 4961, and what is described in this document as
>> "symmetric RTP" looks a lot more like latching to me, with all the
>> potential issues that brings (and then some due to the shared use of
>> ports and hence shared SSRC space between different clients, as noted in
>> the document). The document needs to clear up its terminology in this
>> area ("connect", "binding", latch, symmetric RTP, etc.) and be clearer
>> about what is being proposed (which is latching as far as I can tell,
>> however it makes reference to certain mechanisms that aren't fully
>> described, e.g. binding packets with random nonces).
> Yes, it is what called latching today. When this was originally written
> latching wasn't called latching in this community. But I will work
> through the text to try avoid any missunderstandings. There is after all
> a basic requirement on latching that one uses symmetric RTP after the
> latching or things don't work.
Ok.
>>
>> Other significant comments
>>
>> - The draft intends to investigate and outline different RTSP NAT
>> traversal proposals with the goal of picking one for further detailed
>> specification (which can be found in the draft-ietf-mmusic-rtsp-nat
>> spec). There is a mixture of RFC 2119 language and more informal
>> language in several of these descriptions. Given that these are
>> high-level descriptions and have not been specified in detail, I do not
>> think we should give any other impression and hence I think RFC 2119
>> language should be avoided.
> Fully agreed. There should be no RFC 2119 language in this doc.
>
>> - The draft suggests that the RTSP server needs to enforce that
>> signaling and media come from the same IP-address as a security remedy
>> in several places, however this may not work with larger scale NATs
>> (e.g. carrier-grade NATs) where the NAT may be using more than one
>> external IP-address.
> It is more than a suggestion. It is an fact that servers implements this
> protection. If a CGN fails to give a host the same IP address then
> unfortunately that client will not get any content. I will claim it is
> the CGN that is more broken than the security mechanism.
>
Ok - I think it's worth calling out explicitly.
>> - The comparison section is missing two of the alternatives considered
>> (namely ALG and TCP tunneling). Also, the comparison chart would lead
>> most people to a different solution than the one chosen. To some extent,
>> I think it's because the requirements list is a simplified list of all
>> the considerations behind each of the proposals (as more fully explained
>> in the document overall). There are additional drawbacks to some of
>> these other solutions that do not show up in the table (e.g.
>> unauthenticated "latching" and requirement for RTSP server to have
>> public IP-address). It would be worthwhile briefly recapping those and
>> hence further motivate the choice of ICE in this section.
> Ok, will see what I can do with reasonable work effort.
Sounds good.

Thanks

-- Flemming


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


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Subject: Re: [MMUSIC] Wisdom sought: Prioritization of codecs
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On 11/15/12 3:57 AM, Harald Alvestrand wrote:
> Found some answers in RFC 3264.
>
That's the right place to look assuming this is all for use in an 
Offer/Answer context.


> On 11/15/2012 09:47 AM, Harald Alvestrand wrote:
>> I've been searching RFC 4566 (SDP) for this info, but came up blank....
>>
>> in an M-line, we typically have
>>
>> m=video blabla 111 112 113
>> a=rtpmap:111 ....
>> a=rtpmap:112 ...
>> a=rtpmap:113 ....
>>
>> Section 5.14 says this:
>>
>>        If the <proto> sub-field is "RTP/AVP" or "RTP/SAVP" the <fmt>
>>        sub-fields contain RTP payload type numbers.  When a list of
>>        payload type numbers is given, this implies that all of these
>>        payload formats MAY be used in the session, but the first of these
>>        formats SHOULD be used as the default format for the session.
>>
>> Now, the questions...
>>
>> 1) In offer/answer, does this sequence in an offer mean that the 
>> offerer prefers that the answerer use codec 111, if supported?
>
> 3264 5.1:
>
>     In all cases, the formats in the "m=" line MUST be listed in order of
>     preference, with the first format listed being preferred.  In this
>     case, preferred means that the recipient of the offer SHOULD use the
>     format with the highest preference that is acceptable to it.
>
>>
>> 2) in offer/answer, does this sequence in an answer mean that the 
>> answerer prefers that the offerer use codec 111?
> 3264 6.1:
>
>    The
>     media formats in the "m=" line MUST be listed in order of preference,
>     with the first format listed being preferred.  In this case,
>     preferred means that the offerer SHOULD use the format with the
>     highest preference from the answer.
>
>
>>
>> 3) is there any preference expressed between codec 112 and codec 113?
> Clear from 3264. If we ever update 4566, might want to harmonize those.
4566bis is work-in-progress - do you see a disconnect between 4566 and 
3264 (or are you just looking to repeat the information in 4566) ?

>>
>> 4) Should the sequence of a=rtpmap: lines be completely ignored, or 
>> does it matter?
>
> I hope the answer is "completely ignored", have found no evidence to 
> the contrary.
>
It should not matter, but I don't think we have any text explicitly 
stating that anywhere.

-- Flemming


>>
>> I'm trying to fix a bug, and want to make sure I fix it according to 
>> spec.
>>
>>                Harald
>>
>>
>>
>>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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      http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    <br>
    <div class="moz-cite-prefix">On 11/15/12 3:57 AM, Harald Alvestrand
      wrote:<br>
    </div>
    <blockquote cite="mid:50A4AE7F.4040403@alvestrand.no" type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=ISO-8859-1">
      <div class="moz-cite-prefix">Found some answers in RFC 3264.<br>
        <br>
      </div>
    </blockquote>
    That's the right place to look assuming this is all for use in an
    Offer/Answer context. <br>
    <br>
    <br>
    <blockquote cite="mid:50A4AE7F.4040403@alvestrand.no" type="cite">
      <div class="moz-cite-prefix"> On 11/15/2012 09:47 AM, Harald
        Alvestrand wrote:<br>
      </div>
      <blockquote cite="mid:50A4AC20.5030306@alvestrand.no" type="cite">
        I've been searching RFC 4566 (SDP) for this info, but came up
        blank....<br>
        <br>
        in an M-line, we typically have<br>
        <br>
        m=video blabla 111 112 113<br>
        a=rtpmap:111 ....<br>
        a=rtpmap:112 ...<br>
        a=rtpmap:113 ....<br>
        <br>
        Section 5.14 says this:<br>
        <br>
        <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always; color: rgb(0, 0, 0); font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: start; text-indent: 0px; text-transform: none; widows: 2; word-spacing: 0px; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px;">      If the &lt;proto&gt; sub-field is "RTP/AVP" or "RTP/SAVP" the &lt;fmt&gt;
      sub-fields contain RTP payload type numbers.  When a list of
      payload type numbers is given, this implies that all of these
      payload formats MAY be used in the session, but the first of these
      formats SHOULD be used as the default format for the session.

</pre>
        Now, the questions...<br>
        <br>
        1) In offer/answer, does this sequence in an offer mean that the
        offerer prefers that the answerer use codec 111, if supported?<br>
      </blockquote>
      <br>
      3264 5.1:<br>
      <br>
      <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always; color: rgb(0, 0, 0); font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: start; text-indent: 0px; text-transform: none; widows: 2; word-spacing: 0px; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px;">   In all cases, the formats in the "m=" line MUST be listed in order of
   preference, with the first format listed being preferred.  In this
   case, preferred means that the recipient of the offer SHOULD use the
   format with the highest preference that is acceptable to it.
</pre>
      <br>
      <blockquote cite="mid:50A4AC20.5030306@alvestrand.no" type="cite">
        <br>
        2) in offer/answer, does this sequence in an answer mean that
        the answerer prefers that the offerer use codec 111?<br>
      </blockquote>
      3264 6.1:<br>
      <br>
      <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always; color: rgb(0, 0, 0); font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: start; text-indent: 0px; text-transform: none; widows: 2; word-spacing: 0px; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px;">  The
   media formats in the "m=" line MUST be listed in order of preference,
   with the first format listed being preferred.  In this case,
   preferred means that the offerer SHOULD use the format with the
   highest preference from the answer.
</pre>
      <br class="Apple-interchange-newline">
      <br>
      <blockquote cite="mid:50A4AC20.5030306@alvestrand.no" type="cite">
        <br>
        3) is there any preference expressed between codec 112 and codec
        113?<br>
      </blockquote>
      Clear from 3264. If we ever update 4566, might want to harmonize
      those.<br>
    </blockquote>
    4566bis is work-in-progress - do you see a disconnect between 4566
    and 3264 (or are you just looking to repeat the information in 4566)
    ? <br>
    <br>
    <blockquote cite="mid:50A4AE7F.4040403@alvestrand.no" type="cite">
      <blockquote cite="mid:50A4AC20.5030306@alvestrand.no" type="cite">
        <br>
        4) Should the sequence of a=rtpmap: lines be completely ignored,
        or does it matter?<br>
      </blockquote>
      <br>
      I hope the answer is "completely ignored", have found no evidence
      to the contrary.<br>
      <br>
    </blockquote>
    It should not matter, but I don't think we have any text explicitly
    stating that anywhere. <br>
    <br>
    -- Flemming<br>
    <br>
    <br>
    <blockquote cite="mid:50A4AE7F.4040403@alvestrand.no" type="cite">
      <blockquote cite="mid:50A4AC20.5030306@alvestrand.no" type="cite">
        <br>
        I'm trying to fix a bug, and want to make sure I fix it
        according to spec.<br>
        <br>
        &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
        <br>
        <br>
        <br>
        <br>
        <fieldset class="mimeAttachmentHeader"></fieldset>
        <br>
        <pre wrap="">_______________________________________________
mmusic mailing list
<a moz-do-not-send="true" class="moz-txt-link-abbreviated" href="mailto:mmusic@ietf.org">mmusic@ietf.org</a>
<a moz-do-not-send="true" class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a>
</pre>
      </blockquote>
      <br>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
mmusic mailing list
<a class="moz-txt-link-abbreviated" href="mailto:mmusic@ietf.org">mmusic@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a>
</pre>
    </blockquote>
    <br>
  </body>
</html>

--------------060506010702050700040409--

From adam@nostrum.com  Thu Nov 15 08:38:00 2012
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Subject: Re: [MMUSIC] E.164 and visual separators in draft-ietf-mmusic-sdp-cs
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On 11/15/12 09:51, Flemming Andreasen wrote:
> To be clear, the "global-number" production in RFC 3966 allows for a 
> number of parameters that are not part of the International E.164 
> format and thus I have two concerns with the current text
> 1) The visual-separator exclusion although the ABNF allows for it
> 2) The apparent disconnect between all of these optional extra 
> parameters and the above text suggesting they are not allowed.

I don't think these really are issues. There are a lot of specifications 
where the ABNF would allow for certain productions, while the normative 
specification making use of that ABNF restricts those productions. For 
example, there is nothing in the ABNF of RFC 3261 that restricts SIP 
messages from having multiple CSeq header fields, although it doing so 
certainly isn't allowed.

/a

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    <div class="moz-cite-prefix">On 11/15/12 09:51, Flemming Andreasen
      wrote:<br>
    </div>
    <blockquote cite="mid:50A50F77.7060906@cisco.com" type="cite">
      <blockquote
cite="mid:D09DAE6B636851459F7575D146EFB54B1A3BD4C3@008-AM1MPN1-025.mgdnok.nokia.com"
        type="cite">
        <div class="WordSection1"> </div>
      </blockquote>
      To be clear, the "global-number" production in RFC 3966 allows for
      a number of parameters that are not part of the International
      E.164 format and thus I have two concerns with the current text<br>
      <o:p>&nbsp;</o:p> 1) The visual-separator exclusion although the ABNF
      allows for it<br>
      2) The apparent disconnect between all of these optional extra
      parameters and the above text suggesting they are not allowed.<br>
    </blockquote>
    <br>
    I don't think these really are issues. There are a lot of
    specifications where the ABNF would allow for certain productions,
    while the normative specification making use of that ABNF restricts
    those productions. For example, there is nothing in the ABNF of RFC
    3261 that restricts SIP messages from having multiple CSeq header
    fields, although it doing so certainly isn't allowed.<br>
    <br>
    /a<br>
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From pkyzivat@alum.mit.edu  Thu Nov 15 08:43:50 2012
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Subject: Re: [MMUSIC] E.164 and visual separators in draft-ietf-mmusic-sdp-cs
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At end...

On 11/15/12 10:51 AM, Flemming Andreasen wrote:
>
> On 11/15/12 4:06 AM, Simo.Veikkolainen@nokia.com wrote:
>>
>> Hi Flemming, Paul, Keith, and all,
>>
>> When going through
>> http://tools.ietf.org/id/draft-ietf-mmusic-sdp-cs-13.txt
>> <http://tools.ietf.org/id/draft-ietf-mmusic-sdp-cs-13.txt> for
>> publication, Flemming had a comment on the format of global telephone
>> numbers.
>>
>> The draft currently says in Section 5.2.1:
>>
>>   This memo exclusively uses the international representation of E.164
>>
>> numbers, i.e., those initiated with a '+' sign.  The syntax (see
>>
>> Section 5.7) refers to the representation of a 'global-number'
>>
>> construction already specified in RFC 3966 [RFC3966].  This
>>
>> representation requires the presence of the '+' sign. Additionally,
>>
>> this representation allows for the presence of one or more 'visual-
>>
>> separator' constructions.  Implementations confirming to this
>>
>> specification and using the "E164" address type together with the
>>
>> "PSTN" network type MUST only use international E.164, i.e., those
>>
>> starting with a '+' sign and SHOULD NOT use visual-separators.
>>
>> This was discussed earlier on the list, see
>> http://www.ietf.org/mail-archive/web/mmusic/current/msg09630.html
>>
>> Flemming’s comment was that referencing to RFC3966 but saying that the
>> syntax is invalid because we only use parts of it is not a good idea.
>>
>> The question is, could we just refer to RFC3966 and allow visual
>> separators?
>>
> To be clear, the "global-number" production in RFC 3966 allows for a
> number of parameters that are not part of the International E.164 format
> and thus I have two concerns with the current text
> 1) The visual-separator exclusion although the ABNF allows for it
> 2) The apparent disconnect between all of these optional extra
> parameters and the above text suggesting they are not allowed.

IIRC there was discussion on this. There are a number of symbols defined 
in 3966 that *could* be used here: telephone-subscriber, global-number, 
global-number-digits. All of them permit separators. Which one to choose 
depends on how much flexibility one wants to permit. The flexibility 
brings with it more parsing demands which people might object to if the 
flexibility will never be used.

I don't really care which one it is. global-number-digits is probably 
enough for common uses.

	Thanks,
	Paul


From internet-drafts@ietf.org  Thu Nov 15 08:43:56 2012
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-rtsp-nat-14.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Multiparty Multimedia Session Control Wor=
king Group of the IETF.

	Title           : A Network Address Translator (NAT) Traversal mechanism f=
or media controlled by Real-Time Streaming Protocol (RTSP)
	Author(s)       : Jeff Goldberg
                          Magnus Westerlund
                          Thomas Zeng
	Filename        : draft-ietf-mmusic-rtsp-nat-14.txt
	Pages           : 31
	Date            : 2012-11-15

Abstract:
   This document defines a solution for Network Address Translation
   (NAT) traversal for datagram based media streams setup and controlled
   with Real-time Streaming Protocol version 2 (RTSP 2.0).  It uses
   Interactive Connectivity Establishment (ICE) adapted to use RTSP as a
   signalling channel, defining the necessary extra RTSP extensions and
   procedures.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-rtsp-nat-14

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-rtsp-nat-14


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


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Date: Thu, 15 Nov 2012 11:57:40 -0500
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [MMUSIC] Wisdom sought: Prioritization of codecs
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Harald,

While the specs aren't entirely clear, after some searching and study 
they become *reasonably* clear as you have found. But this is only true 
in the context of a particular usage profile - that the RTP session will 
contain only one RTP stream, and that stream will use only a single codec.

If you move outside of that usage profile then the existing text is 
really deficient. And we are often outside that profile. One example is 
telephone-events. That is a separate "codec" that is never used alone, 
it is always used in conjunction with another codec. You may prefer G711 
for most audio, but still prefer telephone-events for DTMF tones.

The whole notion of codec preference breaks down even more for 
multiplexed streams. You may very explicitly prefer/require different 
streams to use different codecs.

So I don't think we can rely too much on parsing the existing text for 
guidance on how to use SDP to represent multiplexed sessions. (Unless of 
course we use bundle, with a separate m-line for each stream.) AFAIK we 
are breaking new ground here.

	Thanks,
	Paul

On 11/15/12 3:57 AM, Harald Alvestrand wrote:
> Found some answers in RFC 3264.
>
> On 11/15/2012 09:47 AM, Harald Alvestrand wrote:
>> I've been searching RFC 4566 (SDP) for this info, but came up blank....
>>
>> in an M-line, we typically have
>>
>> m=video blabla 111 112 113
>> a=rtpmap:111 ....
>> a=rtpmap:112 ...
>> a=rtpmap:113 ....
>>
>> Section 5.14 says this:
>>
>>        If the <proto> sub-field is "RTP/AVP" or "RTP/SAVP" the <fmt>
>>        sub-fields contain RTP payload type numbers.  When a list of
>>        payload type numbers is given, this implies that all of these
>>        payload formats MAY be used in the session, but the first of these
>>        formats SHOULD be used as the default format for the session.
>>
>> Now, the questions...
>>
>> 1) In offer/answer, does this sequence in an offer mean that the
>> offerer prefers that the answerer use codec 111, if supported?
>
> 3264 5.1:
>
>     In all cases, the formats in the "m=" line MUST be listed in order of
>     preference, with the first format listed being preferred.  In this
>     case, preferred means that the recipient of the offer SHOULD use the
>     format with the highest preference that is acceptable to it.
>
>
>>
>> 2) in offer/answer, does this sequence in an answer mean that the
>> answerer prefers that the offerer use codec 111?
> 3264 6.1:
>
>    The
>     media formats in the "m=" line MUST be listed in order of preference,
>     with the first format listed being preferred.  In this case,
>     preferred means that the offerer SHOULD use the format with the
>     highest preference from the answer.
>
>
>
>>
>> 3) is there any preference expressed between codec 112 and codec 113?
> Clear from 3264. If we ever update 4566, might want to harmonize those.
>>
>> 4) Should the sequence of a=rtpmap: lines be completely ignored, or
>> does it matter?
>
> I hope the answer is "completely ignored", have found no evidence to the
> contrary.
>
>>
>> I'm trying to fix a bug, and want to make sure I fix it according to spec.
>>
>>                Harald
>>
>>
>>
>>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


From carl_klatsky@cable.comcast.com  Thu Nov 15 09:48:38 2012
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To: "'mmusic@ietf.org'" <mmusic@ietf.org>
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Hi,

Question on the use of multiple media stream in SDP.  When multiple streams=
 each of  (more than one m=3D line) of type RTP/AVP are offered and answere=
d, does that imply that the RTP media packets for each stream gets their ow=
n unique synchronization source (SSRC) identifier?

Regards,

Carl Klatsky


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<p class=3D"MsoNormal">Hi,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Question on the use of multiple media stream in SDP.=
&nbsp; When multiple streams each of &nbsp;(more than one m=3D line) of typ=
e RTP/AVP are offered and answered, does that imply that the RTP media pack=
ets for each stream gets their own unique synchronization
 source (SSRC) identifier?<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">Regards,</span><o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">Carl Klatsky<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
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Subject: Re: [MMUSIC] E.164 and visual separators in draft-ietf-mmusic-sdp-cs
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On 11/15/12 11:37 AM, Adam Roach wrote:
> On 11/15/12 09:51, Flemming Andreasen wrote:
>> To be clear, the "global-number" production in RFC 3966 allows for a 
>> number of parameters that are not part of the International E.164 
>> format and thus I have two concerns with the current text
>> 1) The visual-separator exclusion although the ABNF allows for it
>> 2) The apparent disconnect between all of these optional extra 
>> parameters and the above text suggesting they are not allowed.
>
> I don't think these really are issues. There are a lot of 
> specifications where the ABNF would allow for certain productions, 
> while the normative specification making use of that ABNF restricts 
> those productions. For example, there is nothing in the ABNF of RFC 
> 3261 that restricts SIP messages from having multiple CSeq header 
> fields, although it doing so certainly isn't allowed.
>

I think there is a big difference between having an ABNF that includes 
terminals and non-terminals in productions that may not be used at all 
as opposed to having an ABNF with certain semantic restrictions on what 
constitutes valid semantic values based on otherwise valid syntax. The 
former is generally well understood and accepted whereas the latter is 
not (and for good reason IMO - what's the point of having an ABNF 
otherwise ?).

-- Flemming

> /a


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    <br>
    <div class="moz-cite-prefix">On 11/15/12 11:37 AM, Adam Roach wrote:<br>
    </div>
    <blockquote cite="mid:50A51A61.1070905@nostrum.com" type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=ISO-8859-1">
      <div class="moz-cite-prefix">On 11/15/12 09:51, Flemming Andreasen
        wrote:<br>
      </div>
      <blockquote cite="mid:50A50F77.7060906@cisco.com" type="cite">
        <blockquote
cite="mid:D09DAE6B636851459F7575D146EFB54B1A3BD4C3@008-AM1MPN1-025.mgdnok.nokia.com"
          type="cite">
          <div class="WordSection1"> </div>
        </blockquote>
        To be clear, the "global-number" production in RFC 3966 allows
        for a number of parameters that are not part of the
        International E.164 format and thus I have two concerns with the
        current text<br>
        <o:p>&nbsp;</o:p> 1) The visual-separator exclusion although the ABNF
        allows for it<br>
        2) The apparent disconnect between all of these optional extra
        parameters and the above text suggesting they are not allowed.<br>
      </blockquote>
      <br>
      I don't think these really are issues. There are a lot of
      specifications where the ABNF would allow for certain productions,
      while the normative specification making use of that ABNF
      restricts those productions. For example, there is nothing in the
      ABNF of RFC 3261 that restricts SIP messages from having multiple
      CSeq header fields, although it doing so certainly isn't allowed.<br>
      <br>
    </blockquote>
    <br>
    I think there is a big difference between having an ABNF that
    includes terminals and non-terminals in productions that may not be
    used at all as opposed to having an ABNF with certain semantic
    restrictions on what constitutes valid semantic values based on
    otherwise valid syntax. The former is generally well understood and
    accepted whereas the latter is not (and for good reason IMO - what's
    the point of having an ABNF otherwise ?). <br>
    <br>
    -- Flemming <br>
    <br>
    <blockquote cite="mid:50A51A61.1070905@nostrum.com" type="cite"> /a<br>
    </blockquote>
    <br>
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Subject: Re: [MMUSIC] E.164 and visual separators in draft-ietf-mmusic-sdp-cs
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On 11/15/12 11:58, Flemming Andreasen wrote:
>
> On 11/15/12 11:37 AM, Adam Roach wrote:
>>
>> I don't think these really are issues. There are a lot of 
>> specifications where the ABNF would allow for certain productions, 
>> while the normative specification making use of that ABNF restricts 
>> those productions. For example, there is nothing in the ABNF of RFC 
>> 3261 that restricts SIP messages from having multiple CSeq header 
>> fields, although it doing so certainly isn't allowed.
>>
>
> I think there is a big difference between having an ABNF that includes 
> terminals and non-terminals in productions that may not be used at all 
> as opposed to having an ABNF with certain semantic restrictions on 
> what constitutes valid semantic values based on otherwise valid 
> syntax. The former is generally well understood and accepted whereas 
> the latter is not (and for good reason IMO - what's the point of 
> having an ABNF otherwise ?).
You raise a good point. So, yeah, it looks like we probably want to take 
on an existing ABNF (with all its warts and optional things) or define 
our own (which would be trivial -- a plus sign followed by one or more 
digits is some of the most straightforward ABNF you'll ever see written).

/a

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    <div class="moz-cite-prefix">On 11/15/12 11:58, Flemming Andreasen
      wrote:<br>
    </div>
    <blockquote cite="mid:50A52D46.5020803@cisco.com" type="cite">
      <meta content="text/html; charset=ISO-8859-1"
        http-equiv="Content-Type">
      <br>
      <div class="moz-cite-prefix">On 11/15/12 11:37 AM, Adam Roach
        wrote:<br>
      </div>
      <blockquote cite="mid:50A51A61.1070905@nostrum.com" type="cite">
        <meta http-equiv="Content-Type" content="text/html;
          charset=ISO-8859-1">
        <br>
        I don't think these really are issues. There are a lot of
        specifications where the ABNF would allow for certain
        productions, while the normative specification making use of
        that ABNF restricts those productions. For example, there is
        nothing in the ABNF of RFC 3261 that restricts SIP messages from
        having multiple CSeq header fields, although it doing so
        certainly isn't allowed.<br>
        <br>
      </blockquote>
      <br>
      I think there is a big difference between having an ABNF that
      includes terminals and non-terminals in productions that may not
      be used at all as opposed to having an ABNF with certain semantic
      restrictions on what constitutes valid semantic values based on
      otherwise valid syntax. The former is generally well understood
      and accepted whereas the latter is not (and for good reason IMO -
      what's the point of having an ABNF otherwise ?). <br>
    </blockquote>
    You raise a good point. So, yeah, it looks like we probably want to
    take on an existing ABNF (with all its warts and optional things) or
    define our own (which would be trivial -- a plus sign followed by
    one or more digits is some of the most straightforward ABNF you'll
    ever see written).<br>
    <br>
    /a<br>
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From adam@nostrum.com  Thu Nov 15 11:55:15 2012
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Subject: Re: [MMUSIC] Wisdom sought: Prioritization of codecs
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On 11/15/12 10:57, Paul Kyzivat wrote:
> The whole notion of codec preference breaks down even more for 
> multiplexed streams. You may very explicitly prefer/require different 
> streams to use different codecs.
>
> So I don't think we can rely too much on parsing the existing text for 
> guidance on how to use SDP to represent multiplexed sessions. (Unless 
> of course we use bundle, with a separate m-line for each stream.) 
> AFAIK we are breaking new ground here.

This kind of "we fixed this problem once and get to fix it all over 
again" issue is exactly the kind of thing I was speaking to at the 
microphone in MMUSIC in Atlanta. The further we drift from the 
established semantics of SDP, the less applicable previously developed 
mechanisms become. Bundle and similar approaches that maintain one m= 
line per stream allow us to leverage 14 years of extensions and 
clarifications. If we decide to split from that (rather fundamental, 
IMHO) characteristic of SDP, we are effectively forking the protocol, 
and throwing away years of prior work.

/a

From pkyzivat@alum.mit.edu  Thu Nov 15 12:09:38 2012
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Subject: Re: [MMUSIC] E.164 and visual separators in draft-ietf-mmusic-sdp-cs
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On 11/15/12 12:58 PM, Flemming Andreasen wrote:
>
> On 11/15/12 11:37 AM, Adam Roach wrote:
>> On 11/15/12 09:51, Flemming Andreasen wrote:
>>> To be clear, the "global-number" production in RFC 3966 allows for a
>>> number of parameters that are not part of the International E.164
>>> format and thus I have two concerns with the current text
>>> 1) The visual-separator exclusion although the ABNF allows for it
>>> 2) The apparent disconnect between all of these optional extra
>>> parameters and the above text suggesting they are not allowed.
>>
>> I don't think these really are issues. There are a lot of
>> specifications where the ABNF would allow for certain productions,
>> while the normative specification making use of that ABNF restricts
>> those productions. For example, there is nothing in the ABNF of RFC
>> 3261 that restricts SIP messages from having multiple CSeq header
>> fields, although it doing so certainly isn't allowed.
>>
>
> I think there is a big difference between having an ABNF that includes
> terminals and non-terminals in productions that may not be used at all
> as opposed to having an ABNF with certain semantic restrictions on what
> constitutes valid semantic values based on otherwise valid syntax. The
> former is generally well understood and accepted whereas the latter is
> not (and for good reason IMO - what's the point of having an ABNF
> otherwise ?).

Yes, I agree that this kind of constraint is a bit more of a stretch 
than the usual sort of things we revert to text for. (But I often have 
to make a tradeoff when constructing ABNF, between constraints that 
*can* be done syntactically with complex ANBF, and more forgiving ABNF 
and textual constraints.)

This is at least the third place I have come across in the last few 
months that had a need to represent a phone number, and where a 
reference into the 3966 ABNF could *almost* fit the bill. The last time 
was with TERQ. There I raised the possibility to Jon P that maybe it is 
time to revise 3966, so that it is more suited to being a source of 
definitions of phone number stuff that can be referenced. I have mixed 
feelings about that - I think it would be a good thing to do, but I 
don't know how many will consider it worth the trouble.

	Thanks,
	Paul


From pkyzivat@alum.mit.edu  Thu Nov 15 12:16:21 2012
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Cc: Flemming Andreasen <fandreas@cisco.com>, keith.drage@ALCATEL-LUCENT.COM, mmusic@ietf.org
Subject: Re: [MMUSIC] E.164 and visual separators in draft-ietf-mmusic-sdp-cs
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On 11/15/12 1:24 PM, Adam Roach wrote:
> On 11/15/12 11:58, Flemming Andreasen wrote:
>>
>> On 11/15/12 11:37 AM, Adam Roach wrote:
>>>
>>> I don't think these really are issues. There are a lot of
>>> specifications where the ABNF would allow for certain productions,
>>> while the normative specification making use of that ABNF restricts
>>> those productions. For example, there is nothing in the ABNF of RFC
>>> 3261 that restricts SIP messages from having multiple CSeq header
>>> fields, although it doing so certainly isn't allowed.
>>>
>>
>> I think there is a big difference between having an ABNF that includes
>> terminals and non-terminals in productions that may not be used at all
>> as opposed to having an ABNF with certain semantic restrictions on
>> what constitutes valid semantic values based on otherwise valid
>> syntax. The former is generally well understood and accepted whereas
>> the latter is not (and for good reason IMO - what's the point of
>> having an ABNF otherwise ?).
> You raise a good point. So, yeah, it looks like we probably want to take
> on an existing ABNF (with all its warts and optional things) or define
> our own (which would be trivial -- a plus sign followed by one or more
> digits is some of the most straightforward ABNF you'll ever see written).

Yes, that is where this draft started. (Without the "+".) Certainly it 
would "work". But it seems wrong to me. There ought to be an 
authoritative source for the syntax.

Also, I wonder if some of that extra syntax might actually be useful in 
getting the CS call established in special situations. Maybe even local 
numbers might be useful in some situations.

I'll admit that the separators are not *necessary* in any case. But 
forbidding those might just mean that the application will need to add 
code to strip them from whatever source it uses for its local number. At 
least this way its better defined what should be stripped.

	Thanks,
	Paul


From pkyzivat@alum.mit.edu  Thu Nov 15 13:32:55 2012
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Subject: Re: [MMUSIC] Wisdom sought: Prioritization of codecs
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On 11/15/12 2:55 PM, Adam Roach wrote:
> On 11/15/12 10:57, Paul Kyzivat wrote:
>> The whole notion of codec preference breaks down even more for
>> multiplexed streams. You may very explicitly prefer/require different
>> streams to use different codecs.
>>
>> So I don't think we can rely too much on parsing the existing text for
>> guidance on how to use SDP to represent multiplexed sessions. (Unless
>> of course we use bundle, with a separate m-line for each stream.)
>> AFAIK we are breaking new ground here.
>
> This kind of "we fixed this problem once and get to fix it all over
> again" issue is exactly the kind of thing I was speaking to at the
> microphone in MMUSIC in Atlanta. The further we drift from the
> established semantics of SDP, the less applicable previously developed
> mechanisms become. Bundle and similar approaches that maintain one m=
> line per stream allow us to leverage 14 years of extensions and
> clarifications. If we decide to split from that (rather fundamental,
> IMHO) characteristic of SDP, we are effectively forking the protocol,
> and throwing away years of prior work.

I would like to agree with you. But when multiplexing RTP streams it is 
problematic to prearrange every stream in SDP. That requires knowing in 
advance how many there will be, what their SSRCs will be, and what the 
mapping from SSRC to codec will be.

We are still thrashing that out for CLUE. We have some use cases where 
it seems we can't do all of that in advance, and doing an O/A after we 
get packets is too late. Maybe we will find a way around that. Too early 
to tell.

And AFAIK RTCWEB has only been planning to use bundle to give a separate 
m-line for audio and one for video, not necessarily a separate one for 
every media stream. I think there is still an implication that a given 
m-line might be multiplexed, and might use different codecs (among those 
that are negotiated) for the different multiplexed streams.

And even our 14 years of experience on this haven't resulted in a spec 
that fully lines up with what is deployed in the wild. There is an 
assumption in the specs that if you offer multiple codecs and the answer 
accepts multiple codecs, then you can accept packets in any of the 
negotiated codecs, and switch among them, even though we know 
implementations that depend on only one codec being used. (With a 
special case for telephone-events.)

We are now encountering applications that have more complex needs than 
were anticipated over those 14 years. It's nice if we can reuse what we 
have without change to meet those needs. But lets not constrain new 
applications to the sorts of things we had in mind 14 years ago.

	Thanks,
	Paul


From adam@nostrum.com  Thu Nov 15 13:53:25 2012
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Subject: Re: [MMUSIC] Wisdom sought: Prioritization of codecs
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On 11/15/12 15:32, Paul Kyzivat wrote:
> On 11/15/12 2:55 PM, Adam Roach wrote:
>> On 11/15/12 10:57, Paul Kyzivat wrote:
>>> The whole notion of codec preference breaks down even more for
>>> multiplexed streams. You may very explicitly prefer/require different
>>> streams to use different codecs.
>>>
>>> So I don't think we can rely too much on parsing the existing text for
>>> guidance on how to use SDP to represent multiplexed sessions. (Unless
>>> of course we use bundle, with a separate m-line for each stream.)
>>> AFAIK we are breaking new ground here.
>>
>> This kind of "we fixed this problem once and get to fix it all over
>> again" issue is exactly the kind of thing I was speaking to at the
>> microphone in MMUSIC in Atlanta. The further we drift from the
>> established semantics of SDP, the less applicable previously developed
>> mechanisms become. Bundle and similar approaches that maintain one m=
>> line per stream allow us to leverage 14 years of extensions and
>> clarifications. If we decide to split from that (rather fundamental,
>> IMHO) characteristic of SDP, we are effectively forking the protocol,
>> and throwing away years of prior work.
>
> I would like to agree with you. But when multiplexing RTP streams it 
> is problematic to prearrange every stream in SDP. That requires 
> knowing in advance how many there will be, what their SSRCs will be, 
> and what the mapping from SSRC to codec will be.
>
> We are still thrashing that out for CLUE. We have some use cases where 
> it seems we can't do all of that in advance, and doing an O/A after we 
> get packets is too late. Maybe we will find a way around that. Too 
> early to tell.

I need to catch up on CLUE before I can talk to that very much. I'd been 
pretty much ignoring CLUE; but in Atlanta, it became clear that CLUE's 
impact was going to seep out into every other working group in RAI.

> And AFAIK RTCWEB has only been planning to use bundle to give a 
> separate m-line for audio and one for video, not necessarily a 
> separate one for every media stream.

I don't think that's a settled question.

> And even our 14 years of experience on this haven't resulted in a spec 
> that fully lines up with what is deployed in the wild. There is an 
> assumption in the specs that if you offer multiple codecs and the 
> answer accepts multiple codecs, then you can accept packets in any of 
> the negotiated codecs, and switch among them, even though we know 
> implementations that depend on only one codec being used. (With a 
> special case for telephone-events.)

True. It would be good if we could document this set of unstated 
assumptions as a "current practices" document at some point (note the 
lack of "best" before "current practices").

> We are now encountering applications that have more complex needs than 
> were anticipated over those 14 years. It's nice if we can reuse what 
> we have without change to meet those needs. But lets not constrain new 
> applications to the sorts of things we had in mind 14 years ago.

I don't think it's necessary to do so. I've spent some time thinking on 
this problem, and have a half-formed proposal that I plan to bring forth 
once I'm more comfortable that it provides a satisfying user experience.

I just think we're rushing down a path of throwing out a lot of good to 
get rid of a little bit of bad. I'd like to see more discussion around 
less radical changes to SDP to achieve the goals of CLUE and RTCWEB.

/a

From eckelcu@cisco.com  Thu Nov 15 14:59:20 2012
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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: "Klatsky, Carl" <Carl_Klatsky@cable.comcast.com>, "'mmusic@ietf.org'" <mmusic@ietf.org>
Thread-Topic: Multiple media streams in SDP
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Subject: Re: [MMUSIC] Multiple media streams in SDP
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As currently defined (i.e. without SHIM or BUNDLE), each m-line corresponds=
 to a separate RTP session, each with its own SSRC space. An implementation=
 would be free to use the same or different SSRCs across RTP sessions, depe=
nding on the circumstances.

Cheers,
Charles

> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Klatsky, Carl
> Sent: Thursday, November 15, 2012 9:48 AM
> To: 'mmusic@ietf.org'
> Subject: [MMUSIC] Multiple media streams in SDP
>=20
> Hi,
>=20
>=20
>=20
> Question on the use of multiple media stream in SDP.  When multiple
> streams each of  (more than one m=3D line) of type RTP/AVP are offered an=
d
> answered, does that imply that the RTP media packets for each stream gets
> their own unique synchronization source (SSRC) identifier?
>=20
>=20
>=20
> Regards,
>=20
>=20
>=20
> Carl Klatsky
>=20
>=20


From magnus.westerlund@ericsson.com  Fri Nov 16 02:19:55 2012
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On 2012-11-15 16:59, Flemming Andreasen wrote:
> 
> On 11/15/12 10:08 AM, Magnus Westerlund wrote:
>> Hi Flemming,
>>
>> Thanks for the comments.
>>
>> On 2012-10-11 21:35, Flemming Andreasen wrote:

>>> - STUN based on either RFC 3489 or RFC 5389 (or both). The draft
>>> discusses use of STUN as a NAT traversal mechanism (with some RTSP
>>> operational "extensions"), and in so doing it references both RFC 3489
>>> (which is obsolete) and RFC 5389 (the replacement). Furthermore, in
>>> several places, it bases and describes operation on now defunct
>>> (obsolete) RFC 3489 logic and messages instead of RFC 5389. This either
>>> needs to be rectified, or we should have a very good explanation in the
>>> document as to why we are using RFC 3489 (and then be clear which parts
>>> are based on RFC 3489 and which are based on RFC 5389).
>> I intended to go with an background explanation. This document was
>> evaluating RTSP NAT proposals at a time and in a context which is
>> significantly before RFC 5389. Yes, the limits to the classification
>> techniques exist and should be correctly described. But the fact that
>> they was considered can not be undone in a document intended to document
>> the considerations done at that time of picking ICE as a solution. Which
>> is now 7+ year into history.

> Perhaps this is a more general question since we have at least other
> document in the WG with a similar issue:
> 
> - For documents that have been underway for a long time, and the
> technology/references they describe has since evolved, should we base
> this on the original technology and knowledge at the time or should it
> reflect the state of affairs at time of publication ?
> 
> I tend to think it should be the latter, although I don't see anything
> with also including references to older technology, as long as the
> limitations are noted, the new technology is called out, and we don't
> make any normative recommendations around use of the old technology
> unless there is a very good reason to do so.

I would tend to agree with you. At the same time I am feeling the
authors pain with having to make substantial changes to the document
which likely was correct, but which requires more work due to the slow
publication of the document. And in retrospect we probably should have
published this document a long time ago. It isn't really dependent on
the technical specifications.

> 
> In the specific case, is there any particular reason (apart from this
> obviously being more/new work again) why we wouldn't reference RFC 5389
> at this point and possibly note the limitations of the RFC 3489 approach
> as well (for background) ?

Not for the functionality that is present in RFC 5389 I don't see any
issues. But for this document, I will clearly be required to reference both.


>>>
>>> Other significant comments
>>>
>>> - The draft intends to investigate and outline different RTSP NAT
>>> traversal proposals with the goal of picking one for further detailed
>>> specification (which can be found in the draft-ietf-mmusic-rtsp-nat
>>> spec). There is a mixture of RFC 2119 language and more informal
>>> language in several of these descriptions. Given that these are
>>> high-level descriptions and have not been specified in detail, I do not
>>> think we should give any other impression and hence I think RFC 2119
>>> language should be avoided.
>> Fully agreed. There should be no RFC 2119 language in this doc.
>>
>>> - The draft suggests that the RTSP server needs to enforce that
>>> signaling and media come from the same IP-address as a security remedy
>>> in several places, however this may not work with larger scale NATs
>>> (e.g. carrier-grade NATs) where the NAT may be using more than one
>>> external IP-address.
>> It is more than a suggestion. It is an fact that servers implements this
>> protection. If a CGN fails to give a host the same IP address then
>> unfortunately that client will not get any content. I will claim it is
>> the CGN that is more broken than the security mechanism.
>>
> Ok - I think it's worth calling out explicitly.

Sure.

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: Daniel Park <soohongp@gmail.com>
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I do not know why this mail is forwarded...it is spam..please ignore it.
sorry.

Soohong Daniel Park from my Galaxy Note
2012. 11. 17. =BF=C0=C0=FC 11:41=BF=A1 "Daniel Park" <soohongp@gmail.com>=
=B4=D4=C0=CC =C0=DB=BC=BA:

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<p>I do not know why this mail is forwarded...it is spam..please ignore it.=
 sorry.</p>
<p>Soohong Daniel Park from my Galaxy Note</p>
<div class=3D"gmail_quote">2012. 11. 17. =BF=C0=C0=FC 11:41=BF=A1 &quot;Dan=
iel Park&quot; &lt;<a href=3D"mailto:soohongp@gmail.com">soohongp@gmail.com=
</a>&gt;=B4=D4=C0=CC =C0=DB=BC=BA:<br type=3D"attribution"></div>

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From harald@alvestrand.no  Sun Nov 18 21:13:06 2012
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Subject: [MMUSIC] M-line philosophy (Re: Wisdom sought: Prioritization of codecs)
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On 11/15/2012 08:55 PM, Adam Roach wrote:
> On 11/15/12 10:57, Paul Kyzivat wrote:
>> The whole notion of codec preference breaks down even more for 
>> multiplexed streams. You may very explicitly prefer/require different 
>> streams to use different codecs.
>>
>> So I don't think we can rely too much on parsing the existing text 
>> for guidance on how to use SDP to represent multiplexed sessions. 
>> (Unless of course we use bundle, with a separate m-line for each 
>> stream.) AFAIK we are breaking new ground here.
>
> This kind of "we fixed this problem once and get to fix it all over 
> again" issue is exactly the kind of thing I was speaking to at the 
> microphone in MMUSIC in Atlanta. The further we drift from the 
> established semantics of SDP, the less applicable previously developed 
> mechanisms become. Bundle and similar approaches that maintain one m= 
> line per stream allow us to leverage 14 years of extensions and 
> clarifications. If we decide to split from that (rather fundamental, 
> IMHO) characteristic of SDP, we are effectively forking the protocol, 
> and throwing away years of prior work.
actually my caricature of the history of SDP is 4 years of multicast, 
where multiple streams per RTP session was the norm, 10 years of 
telephony, with only one stream of each type that mattered in each 
direction, and 12 months of trying to get multiple streams working again.

While beating down the resistance of the RTP people against having 
different media types in one RTP session, I've been told over and over 
again that having multiple media streams of the *same* media type in one 
RTP session is just fine and dandy "because that's what we designed RTP 
to do", and now we're finally getting explicit about addressing the idea 
of whether we tolerate more than one media stream per <control surface 
unit> - where the only control surface unit that's been used for the 10 
years of telephony is the M-line.

The M-line is problematic because it binds together multiple things:

- A top level media type
- A set of parameters for media transmission *in all directions* (bound 
to payload type)
- Transport information (the "RTP session" data)


SDP M-lines are already problematic as soon as you want different 
parameters in the two directions. It only goes downhill from there when 
one does multiple flows with varying characteristics.

As I said in Atlanta: SDP M-lines have been walking with one foot on the 
"I'm an RTP session" bank of the canal and with the other foot on the 
"I'm a media stream description" bank of the canal.

The canal is growing wider. Fast.




From bernard_aboba@hotmail.com  Sun Nov 18 22:17:49 2012
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Subject: Re: [MMUSIC] Wisdom sought: Prioritization of codecs
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Adam Roach said:

"I need to catch up on CLUE before I can talk to that very much. I'd been
pretty much ignoring CLUE; but in Atlanta, it became clear that CLUE's
impact was going to seep out into every other working group in RAI."

[BA] I think it is probably correct to say that the questions asked in CLUE
are relevant to many other working groups in RAI, including RTCWEB.   Today
even a moderately priced computer can support multiple monitors, as well as
multiple cameras, each of which can support a layered codec, and the cost of
business-grade telepresence systems has come down by an order of magnitude
in the last few years.  Given these trends, it is inevitable that WebRTC
will eventually support these scenarios.  Thinking that through now will
avoid a lot of pain later. 

Adam said:

"I just think we're rushing down a path of throwing out a lot of good to get
rid of a little bit of bad. I'd like to see more discussion around less
radical changes to SDP to achieve the goals of CLUE and RTCWEB."

[BA]  It certainly makes sense to ensure that the WebRTC architecture can
handle CLUE scenarios.   In fact, I do believe that some basic CLUE
scenarios (such as support for multiple cameras and monitors) are already
implementable in some WebRTC implementations, without much in the way of SDP
or API changes. 


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See at the end.

-----Original Message-----
From: ext Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]=20
Sent: 15. marraskuuta 2012 22:16
To: Adam Roach
Cc: Flemming Andreasen; Veikkolainen Simo (Nokia-CTO/Espoo); keith.drage@AL=
CATEL-LUCENT.COM; mmusic@ietf.org
Subject: Re: [MMUSIC] E.164 and visual separators in draft-ietf-mmusic-sdp-=
cs

On 11/15/12 1:24 PM, Adam Roach wrote:
> On 11/15/12 11:58, Flemming Andreasen wrote:
>>
>> On 11/15/12 11:37 AM, Adam Roach wrote:
>>>
>>> I don't think these really are issues. There are a lot of=20
>>> specifications where the ABNF would allow for certain productions,=20
>>> while the normative specification making use of that ABNF restricts=20
>>> those productions. For example, there is nothing in the ABNF of RFC
>>> 3261 that restricts SIP messages from having multiple CSeq header=20
>>> fields, although it doing so certainly isn't allowed.
>>>
>>
>> I think there is a big difference between having an ABNF that=20
>> includes terminals and non-terminals in productions that may not be=20
>> used at all as opposed to having an ABNF with certain semantic=20
>> restrictions on what constitutes valid semantic values based on=20
>> otherwise valid syntax. The former is generally well understood and=20
>> accepted whereas the latter is not (and for good reason IMO - what's=20
>> the point of having an ABNF otherwise ?).
> You raise a good point. So, yeah, it looks like we probably want to=20
> take on an existing ABNF (with all its warts and optional things) or=20
> define our own (which would be trivial -- a plus sign followed by one=20
> or more digits is some of the most straightforward ABNF you'll ever see w=
ritten).

Yes, that is where this draft started. (Without the "+".) Certainly it woul=
d "work". But it seems wrong to me. There ought to be an authoritative sour=
ce for the syntax.

Also, I wonder if some of that extra syntax might actually be useful in get=
ting the CS call established in special situations. Maybe even local number=
s might be useful in some situations.

I'll admit that the separators are not *necessary* in any case. But forbidd=
ing those might just mean that the application will need to add code to str=
ip them from whatever source it uses for its local number. At least this wa=
y its better defined what should be stripped.

[Simo] Referencing RFC3966 can be done in several ways: reusing the entire =
telephone-uri definition, the global-number, or only the global-number-digi=
ts (which allows visual separators). What the draft is using the number for=
 (correlation of SDP with a CS call), just referencing the global-number-di=
gits part would suffice. However, global-number allows parameters to be add=
ed to global-number-digits, which I agree might be useful in some scenarios=
 - basic interoperability would be achieved by parsing the global-number-di=
gits portion. local-number in RFC3966 include things like HEXDIG, "*" and "=
#" which I don't think provide much additional help in achieving our goal.

-Simo


	Thanks,
	Paul


From pkyzivat@alum.mit.edu  Mon Nov 19 07:12:18 2012
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Subject: Re: [MMUSIC] M-line philosophy (Re: Wisdom sought: Prioritization of codecs)
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On 11/19/12 12:12 AM, Harald Alvestrand wrote:
> On 11/15/2012 08:55 PM, Adam Roach wrote:
>> On 11/15/12 10:57, Paul Kyzivat wrote:
>>> The whole notion of codec preference breaks down even more for
>>> multiplexed streams. You may very explicitly prefer/require different
>>> streams to use different codecs.
>>>
>>> So I don't think we can rely too much on parsing the existing text
>>> for guidance on how to use SDP to represent multiplexed sessions.
>>> (Unless of course we use bundle, with a separate m-line for each
>>> stream.) AFAIK we are breaking new ground here.
>>
>> This kind of "we fixed this problem once and get to fix it all over
>> again" issue is exactly the kind of thing I was speaking to at the
>> microphone in MMUSIC in Atlanta. The further we drift from the
>> established semantics of SDP, the less applicable previously developed
>> mechanisms become. Bundle and similar approaches that maintain one m=
>> line per stream allow us to leverage 14 years of extensions and
>> clarifications. If we decide to split from that (rather fundamental,
>> IMHO) characteristic of SDP, we are effectively forking the protocol,
>> and throwing away years of prior work.
> actually my caricature of the history of SDP is 4 years of multicast,
> where multiple streams per RTP session was the norm, 10 years of
> telephony, with only one stream of each type that mattered in each
> direction, and 12 months of trying to get multiple streams working again.

:-)  :-(

> While beating down the resistance of the RTP people against having
> different media types in one RTP session, I've been told over and over
> again that having multiple media streams of the *same* media type in one
> RTP session is just fine and dandy "because that's what we designed RTP
> to do", and now we're finally getting explicit about addressing the idea
> of whether we tolerate more than one media stream per <control surface
> unit> - where the only control surface unit that's been used for the 10
> years of telephony is the M-line.
>
> The M-line is problematic because it binds together multiple things:
>
> - A top level media type
> - A set of parameters for media transmission *in all directions* (bound
> to payload type)
> - Transport information (the "RTP session" data)
>
>
> SDP M-lines are already problematic as soon as you want different
> parameters in the two directions. It only goes downhill from there when
> one does multiple flows with varying characteristics.
>
> As I said in Atlanta: SDP M-lines have been walking with one foot on the
> "I'm an RTP session" bank of the canal and with the other foot on the
> "I'm a media stream description" bank of the canal.
>
> The canal is growing wider. Fast.

The *in all directions* is itself far from clear.
- there is a bias towards using the same payload number in both
   directions but it is allowed to negotiate a different payload
   number in each direction for the same payload type.
- there seems to be an implicit assumption that the same payload type
   will be used in both directions, but it is still possible to use
   different payload types for the two directions
- you can negotiate asymmetric codec use via sendonly/recvonly sessions,
   but common usage typically doesn't support two one-way sessions as an
   alternative to one two-way session.

This just further illustrates the schizophrenic nature of the existing 
SDP support that you are pointing out.

	Thanks,
	Paul

From adam@nostrum.com  Mon Nov 19 07:39:03 2012
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Subject: Re: [MMUSIC] M-line philosophy (Re: Wisdom sought: Prioritization of codecs)
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On 11/18/12 23:12, Harald Alvestrand wrote:
>
> The M-line is problematic because it binds together multiple things:
>
> - A top level media type
> - A set of parameters for media transmission *in all directions* 
> (bound to payload type)
> - Transport information (the "RTP session" data)
>
>
> SDP M-lines are already problematic as soon as you want different 
> parameters in the two directions. It only goes downhill from there 
> when one does multiple flows with varying characteristics.
>
> As I said in Atlanta: SDP M-lines have been walking with one foot on 
> the "I'm an RTP session" bank of the canal and with the other foot on 
> the "I'm a media stream description" bank of the canal.

I agree with your characterization of the situation. I'm a little 
surprised that it doesn't lead you to the same conclusion as I've reached.

There are, at the moment, 176 attributes in the IANA registry that are 
applicable to m= sections:

3GPP-Adaptation-Support 3GPP-Asset-Information 3GPP-QoE-Metrics
3GPP-SRTP-Config 3gpp-videopostdecbufsize FEC FEC-OTI-extension
FEC-declaration SRTPAuthentication SRTPROCTxRate T38FaxFillBitRemoval
T38FaxMaxBuffer T38FaxMaxDatagram T38FaxRateManagement
T38FaxTranscodingJBIG T38FaxTranscodingMMR T38FaxUdpEC T38FaxVersion
T38MaxBitRate T38VendorInfo X-decbyterate X-initpostdecbufperiod
X-initpredecbufperiod X-predecbufsize aal2CPS aal2CPSSDUrate
aal2sscs3661assured aal2sscs3661unassured aal2sscs3662 aal5sscop
aalApp aalType abrParms abrSetup acap accept-types accept-wrapped-types
acfg alt alt-default-id anycast atmQOSparms atmTrfcDesc atmmap bcob
bearerSigIE bearerType cache candidate capability cbrRate cdsc
cfw-id chain channel clkrec cmid codecconfig conf confid connection
content-desc cpar cparmax cparmin cpsSDUsize creq crypto csup curr
dccp-port dccp-service-code depend des dsel ecan ecn-capable-rtp
eecid extmap fec fec-repair-flow fec-source-flow file-date
file-disposition file-icon file-range file-selector file-transfer-id
fingerprint floorctrl floorid flute-ch flute-tsi fmtp framerate
framesize fsel g.3gpp.cat g.3gpp.crs gc h248item ice-mismatch ice-pwd
ice-ufrag ike-setup imageattr inactive isup_usi key-mgmt label lang
lij max-size maxprate maxptime mbms-flowid mbms-mode mbms-repair
mid msrp-cema multicast-rtcp omr-codecs omr-m-att omr-m-bw omr-m-cksum
omr-s-att omr-s-bw omr-s-cksum onewaySel orient path pcfg portmapping-req
profileDesc prtfl psk-fingerprint ptime qos-mech-recv qos-mech-send
qosClass quality rams-updates recvonly remote-candidates repair-window
resource rtcp rtcp-fb rtcp-mux rtcp-rsize rtcp-unicast rtcp-xr
rtpmap rtpred1 rtpred2 sbc sdplang secondary-realm sendonly sendrecv
setup silenceSupp source-filter sqn ssrc ssrc-group stc stkmstream
structure tcap uiLayer1_Prot upcc userid visited-realm vsel zrtp-hash

Some of these are applicable to the RTP session; others only make sense 
as applied to a single media stream.

If we choose to make each m= section represent one media stream, then 
these attributes are applicable as-is. We get to take advantage of the 
work put in place by the 50 to 60 documents that defined the 
corresponding mechanisms with no changes whatsoever to this existing 
corpus of work.

If we choose to make each m= section represent one RTP session, in which 
multiple media streams can be placed, then the use of any of those 176 
attributes that are to be applied on the stream level becomes ambiguous. 
If we want to take advantage of them for RTCWEB (or anything else that 
leverages the work of untangling this mess), then we have to come up 
with an alternate, non-backwards-compatible syntax for conveying each of 
those attributes (cf. draft-lennox-mmusic-sdp-source-selection).

And that's *just* the issues associated with attributes -- the problem 
is even worse once you consider the more subtle (and difficult to 
predict) damage to established mechanisms, like your codec priority 
conundrum.

/a

From harald@alvestrand.no  Mon Nov 19 08:04:43 2012
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Subject: Re: [MMUSIC] M-line philosophy (Re: Wisdom sought: Prioritization of codecs)
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On 11/19/2012 04:38 PM, Adam Roach wrote:
> On 11/18/12 23:12, Harald Alvestrand wrote:
>>
>> The M-line is problematic because it binds together multiple things:
>>
>> - A top level media type
>> - A set of parameters for media transmission *in all directions* 
>> (bound to payload type)
>> - Transport information (the "RTP session" data)
>>
>>
>> SDP M-lines are already problematic as soon as you want different 
>> parameters in the two directions. It only goes downhill from there 
>> when one does multiple flows with varying characteristics.
>>
>> As I said in Atlanta: SDP M-lines have been walking with one foot on 
>> the "I'm an RTP session" bank of the canal and with the other foot on 
>> the "I'm a media stream description" bank of the canal.
>
> I agree with your characterization of the situation. I'm a little 
> surprised that it doesn't lead you to the same conclusion as I've 
> reached.
>
> There are, at the moment, 176 attributes in the IANA registry that are 
> applicable to m= sections:
>
> 3GPP-Adaptation-Support 3GPP-Asset-Information 3GPP-QoE-Metrics
> 3GPP-SRTP-Config 3gpp-videopostdecbufsize FEC FEC-OTI-extension
> FEC-declaration SRTPAuthentication SRTPROCTxRate T38FaxFillBitRemoval
> T38FaxMaxBuffer T38FaxMaxDatagram T38FaxRateManagement
> T38FaxTranscodingJBIG T38FaxTranscodingMMR T38FaxUdpEC T38FaxVersion
> T38MaxBitRate T38VendorInfo X-decbyterate X-initpostdecbufperiod
> X-initpredecbufperiod X-predecbufsize aal2CPS aal2CPSSDUrate
> aal2sscs3661assured aal2sscs3661unassured aal2sscs3662 aal5sscop
> aalApp aalType abrParms abrSetup acap accept-types accept-wrapped-types
> acfg alt alt-default-id anycast atmQOSparms atmTrfcDesc atmmap bcob
> bearerSigIE bearerType cache candidate capability cbrRate cdsc
> cfw-id chain channel clkrec cmid codecconfig conf confid connection
> content-desc cpar cparmax cparmin cpsSDUsize creq crypto csup curr
> dccp-port dccp-service-code depend des dsel ecan ecn-capable-rtp
> eecid extmap fec fec-repair-flow fec-source-flow file-date
> file-disposition file-icon file-range file-selector file-transfer-id
> fingerprint floorctrl floorid flute-ch flute-tsi fmtp framerate
> framesize fsel g.3gpp.cat g.3gpp.crs gc h248item ice-mismatch ice-pwd
> ice-ufrag ike-setup imageattr inactive isup_usi key-mgmt label lang
> lij max-size maxprate maxptime mbms-flowid mbms-mode mbms-repair
> mid msrp-cema multicast-rtcp omr-codecs omr-m-att omr-m-bw omr-m-cksum
> omr-s-att omr-s-bw omr-s-cksum onewaySel orient path pcfg portmapping-req
> profileDesc prtfl psk-fingerprint ptime qos-mech-recv qos-mech-send
> qosClass quality rams-updates recvonly remote-candidates repair-window
> resource rtcp rtcp-fb rtcp-mux rtcp-rsize rtcp-unicast rtcp-xr
> rtpmap rtpred1 rtpred2 sbc sdplang secondary-realm sendonly sendrecv
> setup silenceSupp source-filter sqn ssrc ssrc-group stc stkmstream
> structure tcap uiLayer1_Prot upcc userid visited-realm vsel zrtp-hash
>
> Some of these are applicable to the RTP session; others only make 
> sense as applied to a single media stream.
And some make absolutely no sense at all in the RTCWEB context. Just for 
grins, let's take every single attribute defined for ATM bearer 
connection attributes and just remove them from discussion. Up to 35 
attributes down at a single blow.

I have at one point in my life dealt with T-series fax (RFC 2159 is long 
forgotten, I am happy to say). 'Nuff said.

And those were just the 2 first ones I looked at.
>
> If we choose to make each m= section represent one media stream, then 
> these attributes are applicable as-is. We get to take advantage of the 
> work put in place by the 50 to 60 documents that defined the 
> corresponding mechanisms with no changes whatsoever to this existing 
> corpus of work.
Except that, for instance, ice-pwd becomes ambiguous, and has to be 
dealt with using a bundle mechanism.

>
> If we choose to make each m= section represent one RTP session, in 
> which multiple media streams can be placed, then the use of any of 
> those 176 attributes that are to be applied on the stream level 
> becomes ambiguous. If we want to take advantage of them for RTCWEB (or 
> anything else that leverages the work of untangling this mess), then 
> we have to come up with an alternate, non-backwards-compatible syntax 
> for conveying each of those attributes (cf. 
> draft-lennox-mmusic-sdp-source-selection).
>
> And that's *just* the issues associated with attributes -- the problem 
> is even worse once you consider the more subtle (and difficult to 
> predict) damage to established mechanisms, like your codec priority 
> conundrum.
That was an easy one.


From adam@nostrum.com  Mon Nov 19 08:25:18 2012
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Subject: Re: [MMUSIC] M-line philosophy (Re: Wisdom sought: Prioritization of codecs)
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On 11/19/12 10:04, Harald Alvestrand wrote:
> On 11/19/2012 04:38 PM, Adam Roach wrote:
>>
>>
>> There are, at the moment, 176 attributes in the IANA registry that 
>> are applicable to m= sections:
>> ...
>> Some of these are applicable to the RTP session; others only make 
>> sense as applied to a single media stream.
> And some make absolutely no sense at all in the RTCWEB context.

Perhaps not for RTCWEB, but I'm fairly certain that we'd like to see the 
same solutions applies to CLUE as RTCWEB. I concede that some of these 
are applicable only to technologies that are dead or dying; however...

> I have at one point in my life dealt with T-series fax

...T.38 fax isn't one of them. Sadly.

It sure would be nice to discount all of the existing mechanisms, but I 
suspect that they're not as wholly worthless are you're implying. 
Perhaps when I have a bit more time, I'll sit down and do a 
mechanism-by-mechanism evaluation of applicability of each to RTCWEB and 
CLUE. Until then, I expect it's relatively safe to assume that people 
haven't been putting values into that registry by performing utterly 
meaningless work; and, further, that only a fraction of it has been 
overtaken by technological progress.

/a

From pkyzivat@alum.mit.edu  Mon Nov 19 08:37:33 2012
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Subject: Re: [MMUSIC] M-line philosophy (Re: Wisdom sought: Prioritization of codecs)
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Adam,

I'm open to different approaches for addressing multiple RTP streams. So 
far ISTM there are drawbacks to all the ones I have seen discussed.

The m-line per stream approach has problems if you don't know a priori 
how many, or which, streams you will have need to send over the session.

I think there is also a problem with how RTCP works, and how the 
attributes that affect it apply, in the m-line per session approach. 
(And there are also problems here if you allow multiple streams per m-line.)

	Thanks,
	Paul

On 11/19/12 11:24 AM, Adam Roach wrote:
> On 11/19/12 10:04, Harald Alvestrand wrote:
>> On 11/19/2012 04:38 PM, Adam Roach wrote:
>>>
>>>
>>> There are, at the moment, 176 attributes in the IANA registry that
>>> are applicable to m= sections:
>>> ...
>>> Some of these are applicable to the RTP session; others only make
>>> sense as applied to a single media stream.
>> And some make absolutely no sense at all in the RTCWEB context.
>
> Perhaps not for RTCWEB, but I'm fairly certain that we'd like to see the
> same solutions applies to CLUE as RTCWEB. I concede that some of these
> are applicable only to technologies that are dead or dying; however...
>
>> I have at one point in my life dealt with T-series fax
>
> ...T.38 fax isn't one of them. Sadly.
>
> It sure would be nice to discount all of the existing mechanisms, but I
> suspect that they're not as wholly worthless are you're implying.
> Perhaps when I have a bit more time, I'll sit down and do a
> mechanism-by-mechanism evaluation of applicability of each to RTCWEB and
> CLUE. Until then, I expect it's relatively safe to assume that people
> haven't been putting values into that registry by performing utterly
> meaningless work; and, further, that only a fraction of it has been
> overtaken by technological progress.
>
> /a
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


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On 11/19/2012 05:24 PM, Adam Roach wrote:
> On 11/19/12 10:04, Harald Alvestrand wrote:
>> On 11/19/2012 04:38 PM, Adam Roach wrote:
>>>
>>>
>>> There are, at the moment, 176 attributes in the IANA registry that 
>>> are applicable to m= sections:
>>> ...
>>> Some of these are applicable to the RTP session; others only make 
>>> sense as applied to a single media stream.
>> And some make absolutely no sense at all in the RTCWEB context.
>
> Perhaps not for RTCWEB, but I'm fairly certain that we'd like to see 
> the same solutions applies to CLUE as RTCWEB. I concede that some of 
> these are applicable only to technologies that are dead or dying; 
> however...
>
>> I have at one point in my life dealt with T-series fax
>
> ...T.38 fax isn't one of them. Sadly.
What's your scenario where T.38 fax over RTP is relevant when file 
transfer over a reliable channel is available between the same 2 devices?
>
> It sure would be nice to discount all of the existing mechanisms, but 
> I suspect that they're not as wholly worthless are you're implying. 
> Perhaps when I have a bit more time, I'll sit down and do a 
> mechanism-by-mechanism evaluation of applicability of each to RTCWEB 
> and CLUE. Until then, I expect it's relatively safe to assume that 
> people haven't been putting values into that registry by performing 
> utterly meaningless work; and, further, that only a fraction of it has 
> been overtaken by technological progress.
90% is also a fraction. But I agree it's unlikely to be that high.

>
> /a


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On 11/19/12 10:50, Harald Alvestrand wrote:
>> ...T.38 fax isn't one of them. Sadly.
>
> What's your scenario where T.38 fax over RTP is relevant when file 
> transfer over a reliable channel is available between the same 2 devices?

This is a rat hole that I'm not going to dive too far into, but my 
understanding is that the tenacity of fax as an underlying technology 
has to do with a substantial body of U.S. case law that establishes that 
faxed legal documents are binding, while leaving the issue of 
transmission using a technology known by any other name unsettled.

Is this a mind-numbingly stupid reason for it to hang on? Yeah, maybe. 
Does that change the facts on the ground? No. Not even a little bit.

ObHumor: 
http://www.theonion.com/articles/report-fax-machines-still-pretty-impressive-if-you,21256/

/a

From pkyzivat@alum.mit.edu  Mon Nov 19 08:57:52 2012
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On 11/19/12 11:50 AM, Harald Alvestrand wrote:

>>> I have at one point in my life dealt with T-series fax
>>
>> ...T.38 fax isn't one of them. Sadly.
> What's your scenario where T.38 fax over RTP is relevant when file
> transfer over a reliable channel is available between the same 2 devices?

Sadly, I gather that the legal world hasn't yet caught up with 
technology. E.g., I think that faxed copies of documents with signatures 
have more legal validity than do pdf files containing signatures. 
Stupid, but it can still drive demand for continued support of fax 
protocols.

[I'm not an expert on this - this comment is mostly based on 2nd hand data.]

	Thanks,
	Paul



From harald@alvestrand.no  Tue Nov 20 00:47:48 2012
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Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] T.38 (Re: M-line philosophy (Re: Wisdom sought: Prioritization of codecs))
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On 11/19/2012 05:57 PM, Adam Roach wrote:
> On 11/19/12 10:50, Harald Alvestrand wrote:
>>> ...T.38 fax isn't one of them. Sadly.
>>
>> What's your scenario where T.38 fax over RTP is relevant when file 
>> transfer over a reliable channel is available between the same 2 
>> devices?
>
> This is a rat hole that I'm not going to dive too far into, but my 
> understanding is that the tenacity of fax as an underlying technology 
> has to do with a substantial body of U.S. case law that establishes 
> that faxed legal documents are binding, while leaving the issue of 
> transmission using a technology known by any other name unsettled.
>
> Is this a mind-numbingly stupid reason for it to hang on? Yeah, maybe. 
> Does that change the facts on the ground? No. Not even a little bit.
Sympathy. I've read the original judge's opinion in the English court 
case that established telex as legally binding - he explicitly said "I 
have this opinion not because it is technically sound, but because it is 
necessary to keep commerce flowing". We're still there, and the fax 
industry keeps chugging along because of it.

Still, I don't see the compelling use case for T.38 fax over RTP *into 
browsers*.
>
> ObHumor: 
> http://www.theonion.com/articles/report-fax-machines-still-pretty-impressive-if-you,21256/
>
> /a


From pkyzivat@alum.mit.edu  Tue Nov 20 07:42:39 2012
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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: [MMUSIC] SDP Directorate review of draft-ietf-xrblock-rtcp-xr-burst-gap-discard-06
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I am the assigned SDP directorate reviewer for 
draft-ietf-xrblock-rtcp-xr-burst-gap-discard-06. For background on SDP 
directorate, please see the FAQ at 
<http://www.ietf.org/iesg/directorate/sdp.html>.

Please wait for direction from your document shepherd or AD before 
posting a new version of the draft.

* Summary:

The SDP syntax defined in this draft is trivial and fine.
It exercises a defined extension point in RFC3611. That RFC defines an 
IANA registry for new values using this extension point. This document 
nominally supplies the information called for when registering. However, 
the semantics for processing the new syntax is underspecified, and needs 
to be further elaborated.

* New SDP information elements:

Section 5 of this draft augments the SDP attribute "rtcp-xr" defined in 
RFC3611, providing an additional value: "brst-gap-dscrd".

* Technical Issues:

RFC3611 establishes an IANA registry for new values, and requires that 
the following information be supplied when registering a new value:

    -  Contact information of the one doing the registration, including
       at least name, address, and email.

    -  An Augmented Backus-Naur Form (ABNF) [2] definition of the
       parameter, in accordance with the "format-ext" definition of
       Section 5.1.

    -  A description of what the parameter represents and how it shall be
       interpreted, both normally and in Offer/Answer.

The first two of the above are properly supplied. But the offer/answer 
behavior is insufficiently specified. ("When SDP is used in offer-answer 
context, the SDP Offer/Answer usage defined in [RFC3611] applies.")

RFC3611 states:

    Each "rtcp-xr" parameter belongs to one of two categories.  The first
    category, the unilateral parameters, are for report blocks that
    simply report on the RTP stream and related metrics.  The second
    category, collaborative parameters, are for XR blocks that involve
    actions by more than one party in order to carry out their functions.

It then goes on to give different offer/answer procedures for each 
category of parameter. The reference in *this* draft ("the SDP 
Offer/Answer usage defined in [RFC3611] applies") is insufficient 
because there is no indication in this draft whether the new value is a 
"unilateral" or "collaborative" parameter.

The draft needs to be revised to address this issue. I *think* this new 
parameter is a "unilateral" parameter. If so, it MAY be sufficient to 
revise section 5.2 as follows:

OLD:
    When SDP is used in offer-answer context, the SDP Offer/Answer usage
    defined in [RFC3611] applies.

NEW:
    When SDP is used in offer-answer context, the SDP Offer/Answer usage
    defined in [RFC3611] for unilateral "rtcp-xr" attribute parameters
    applies.

It may be useful to provide an expanded explanation.

	Regards,
	Paul Kyzivat

From btv1==6713412e283==HKaplan@acmepacket.com  Tue Nov 20 10:16:06 2012
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
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Cc: Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>, "draft-ietf-mmusic-media-loopback@tools.ietf.org" <draft-ietf-mmusic-media-loopback@tools.ietf.org>
Subject: [MMUSIC] NEED WG CONSENSUS: media-loopback is all-or-nothing
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Howdy,
during IESG review of the media-loopback draft, a discussion arose regardin=
g the language around indicating media-loopback support per media descripti=
on (i.e., per m-line), in sections 3.1 and 3.2.

My interpretation/understanding was we were requiring ALL m-lines to indica=
te media-loopback; in other words it's an all-or-nothing in both the offer+=
answer, and either every media stream is looped back, or none are.  However=
 the text currently doesn't make this clear, and it could be argued either =
way, with pro's/con's either way too; so I'd like the WG's consensus on whe=
ther it should in fact be an all-or-none approach, or not.

Please respond to this email by **Tuesday, December 4th** with a "YES" if t=
he draft should mandate ALL media be looped-back, or "NO" if it should allo=
w a hybrid approach of some streams being looped while others are not.

For reference, the draft is here:
http://tools.ietf.org/html/draft-ietf-mmusic-media-loopback-24

-hadriel


From pkyzivat@alum.mit.edu  Tue Nov 20 10:42:46 2012
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Subject: Re: [MMUSIC] NEED WG CONSENSUS: media-loopback is all-or-nothing
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NO

*Why* should the draft be limited this way if it need not be?

	Thanks,
	Paul

On 11/20/12 1:16 PM, Hadriel Kaplan wrote:
> Howdy,
> during IESG review of the media-loopback draft, a discussion arose regarding the language around indicating media-loopback support per media description (i.e., per m-line), in sections 3.1 and 3.2.
>
> My interpretation/understanding was we were requiring ALL m-lines to indicate media-loopback; in other words it's an all-or-nothing in both the offer+answer, and either every media stream is looped back, or none are.  However the text currently doesn't make this clear, and it could be argued either way, with pro's/con's either way too; so I'd like the WG's consensus on whether it should in fact be an all-or-none approach, or not.
>
> Please respond to this email by **Tuesday, December 4th** with a "YES" if the draft should mandate ALL media be looped-back, or "NO" if it should allow a hybrid approach of some streams being looped while others are not.
>
> For reference, the draft is here:
> http://tools.ietf.org/html/draft-ietf-mmusic-media-loopback-24
>
> -hadriel
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


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To Paul's point=2C there may be a scenario in which some m lines would be l=
ooped back and others would not.  For example=2C what if I want to do a loo=
pback test of audio/video but also want to be able to converse via RTT to d=
iscuss the results of the test in progress.  In that case the audio and vid=
eo m lines would be looped back but the text line would not be.=20

> Date: Tue=2C 20 Nov 2012 13:42:43 -0500
> From: pkyzivat@alum.mit.edu
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] NEED WG CONSENSUS: media-loopback is all-or-nothing
>=20
> NO
>=20
> *Why* should the draft be limited this way if it need not be?
>=20
> 	Thanks=2C
> 	Paul
>=20
> On 11/20/12 1:16 PM=2C Hadriel Kaplan wrote:
> > Howdy=2C
> > during IESG review of the media-loopback draft=2C a discussion arose re=
garding the language around indicating media-loopback support per media des=
cription (i.e.=2C per m-line)=2C in sections 3.1 and 3.2.
> >
> > My interpretation/understanding was we were requiring ALL m-lines to in=
dicate media-loopback=3B in other words it's an all-or-nothing in both the =
offer+answer=2C and either every media stream is looped back=2C or none are=
.  However the text currently doesn't make this clear=2C and it could be ar=
gued either way=2C with pro's/con's either way too=3B so I'd like the WG's =
consensus on whether it should in fact be an all-or-none approach=2C or not=
.
> >
> > Please respond to this email by **Tuesday=2C December 4th** with a "YES=
" if the draft should mandate ALL media be looped-back=2C or "NO" if it sho=
uld allow a hybrid approach of some streams being looped while others are n=
ot.
> >
> > For reference=2C the draft is here:
> > http://tools.ietf.org/html/draft-ietf-mmusic-media-loopback-24
> >
> > -hadriel
> >
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www.ietf.org/mailman/listinfo/mmusic
> >
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>To Paul's point=2C there may be =
a scenario in which some m lines would be looped back and others would not.=
&nbsp=3B For example=2C what if I want to do a loopback test of audio/video=
 but also want to be able to converse via RTT to discuss the results of the=
 test in progress.&nbsp=3B In that case the audio and video m lines would b=
e looped back but the text line would not be. <br><br><div><div id=3D"SkyDr=
ivePlaceholder"></div>&gt=3B Date: Tue=2C 20 Nov 2012 13:42:43 -0500<br>&gt=
=3B From: pkyzivat@alum.mit.edu<br>&gt=3B To: mmusic@ietf.org<br>&gt=3B Sub=
ject: Re: [MMUSIC] NEED WG CONSENSUS: media-loopback is all-or-nothing<br>&=
gt=3B <br>&gt=3B NO<br>&gt=3B <br>&gt=3B *Why* should the draft be limited =
this way if it need not be?<br>&gt=3B <br>&gt=3B 	Thanks=2C<br>&gt=3B 	Paul=
<br>&gt=3B <br>&gt=3B On 11/20/12 1:16 PM=2C Hadriel Kaplan wrote:<br>&gt=
=3B &gt=3B Howdy=2C<br>&gt=3B &gt=3B during IESG review of the media-loopba=
ck draft=2C a discussion arose regarding the language around indicating med=
ia-loopback support per media description (i.e.=2C per m-line)=2C in sectio=
ns 3.1 and 3.2.<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B My interpretation/underst=
anding was we were requiring ALL m-lines to indicate media-loopback=3B in o=
ther words it's an all-or-nothing in both the offer+answer=2C and either ev=
ery media stream is looped back=2C or none are.  However the text currently=
 doesn't make this clear=2C and it could be argued either way=2C with pro's=
/con's either way too=3B so I'd like the WG's consensus on whether it shoul=
d in fact be an all-or-none approach=2C or not.<br>&gt=3B &gt=3B<br>&gt=3B =
&gt=3B Please respond to this email by **Tuesday=2C December 4th** with a "=
YES" if the draft should mandate ALL media be looped-back=2C or "NO" if it =
should allow a hybrid approach of some streams being looped while others ar=
e not.<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B For reference=2C the draft is here=
:<br>&gt=3B &gt=3B http://tools.ietf.org/html/draft-ietf-mmusic-media-loopb=
ack-24<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B -hadriel<br>&gt=3B &gt=3B<br>&gt=
=3B &gt=3B _______________________________________________<br>&gt=3B &gt=3B=
 mmusic mailing list<br>&gt=3B &gt=3B mmusic@ietf.org<br>&gt=3B &gt=3B http=
s://www.ietf.org/mailman/listinfo/mmusic<br>&gt=3B &gt=3B<br>&gt=3B <br>&gt=
=3B _______________________________________________<br>&gt=3B mmusic mailin=
g list<br>&gt=3B mmusic@ietf.org<br>&gt=3B https://www.ietf.org/mailman/lis=
tinfo/mmusic<br></div> 		 	   		  </div></body>
</html>=

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Date: Tue, 20 Nov 2012 11:12:23 -0800
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Subject: Re: [MMUSIC] M-line philosophy (Re: Wisdom sought: Prioritization of codecs)
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> The M-line is problematic because it binds together multiple things:
>=20
> - A set of parameters for media transmission *in all directions* (bound t=
o payload type)

[BA] RFC 3264 Section 5.1 makes it clear that the parameters need not apply=
 in all directions (though sendrecv is the default):=20

   The list of media formats for each media stream conveys two pieces of=0A=
   information=2C namely the set of formats (codecs and any parameters=0A=
   associated with the codec=2C in the case of RTP) that the offerer is=0A=
   capable of sending and/or receiving (depending on the direction=0A=
   attributes)=2C and=2C in the case of RTP=2C the RTP payload type numbers=
=0A=
   used to identify those formats.  If multiple formats are listed=2C it=0A=
   means that the offerer is capable of making use of any of those=0A=
   formats during the session.  In other words=2C the answerer MAY change=
=0A=
   formats in the middle of the session=2C making use of any of the=0A=
   formats listed=2C without sending a new offer.  For a sendonly stream=2C=
=0A=
   the offer SHOULD indicate those formats the offerer is willing to=0A=
   send for this stream.  For a recvonly stream=2C the offer SHOULD=0A=
   indicate those formats the offerer is willing to receive for this=0A=
   stream.  For a sendrecv stream=2C the offer SHOULD indicate those=0A=
   codecs that the offerer is willing to send and receive with.
 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>&gt=3B The M-line is problematic=
 because it binds together multiple things:<br><div>&gt=3B <br>&gt=3B - A s=
et of parameters for media transmission *in all directions* (bound to paylo=
ad type)<br><br>[BA] RFC 3264 Section 5.1 makes it clear that the parameter=
s need not apply in all directions (though sendrecv is the default): <br><b=
r><pre class=3D"newpage">   The list of media formats for each media stream=
 conveys two pieces of=0A=
   information=2C namely the set of formats (codecs and any parameters=0A=
   associated with the codec=2C in the case of RTP) that the offerer is=0A=
   capable of sending and/or receiving (depending on the direction=0A=
   attributes)=2C and=2C in the case of RTP=2C the RTP payload type numbers=
=0A=
   used to identify those formats.  If multiple formats are listed=2C it=0A=
   means that the offerer is capable of making use of any of those=0A=
   formats during the session.  In other words=2C the answerer MAY change=
=0A=
   formats in the middle of the session=2C making use of any of the=0A=
   formats listed=2C without sending a new offer.  For a sendonly stream=2C=
=0A=
   the offer SHOULD indicate those formats the offerer is willing to=0A=
   send for this stream.  For a recvonly stream=2C the offer SHOULD=0A=
   indicate those formats the offerer is willing to receive for this=0A=
   stream.  For a sendrecv stream=2C the offer SHOULD indicate those=0A=
   codecs that the offerer is willing to send and receive with.</pre><br></=
div> 		 	   		  </div></body>
</html>=

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From btv1==6713412e283==HKaplan@acmepacket.com  Tue Nov 20 11:51:49 2012
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Bernard Aboba <bernard_aboba@hotmail.com>
Thread-Topic: [MMUSIC] NEED WG CONSENSUS: media-loopback is all-or-nothing
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] NEED WG CONSENSUS: media-loopback is all-or-nothing
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I take this is a "NO" vote?

-hadriel


On Nov 20, 2012, at 1:56 PM, Bernard Aboba <bernard_aboba@hotmail.com<mailt=
o:bernard_aboba@hotmail.com>> wrote:

To Paul's point, there may be a scenario in which some m lines would be loo=
ped back and others would not.  For example, what if I want to do a loopbac=
k test of audio/video but also want to be able to converse via RTT to discu=
ss the results of the test in progress.  In that case the audio and video m=
 lines would be looped back but the text line would not be.

> Date: Tue, 20 Nov 2012 13:42:43 -0500
> From: pkyzivat@alum.mit.edu<mailto:pkyzivat@alum.mit.edu>
> To: mmusic@ietf.org<mailto:mmusic@ietf.org>
> Subject: Re: [MMUSIC] NEED WG CONSENSUS: media-loopback is all-or-nothing
>
> NO
>
> *Why* should the draft be limited this way if it need not be?
>
> Thanks,
> Paul
>
> On 11/20/12 1:16 PM, Hadriel Kaplan wrote:
> > Howdy,
> > during IESG review of the media-loopback draft, a discussion arose rega=
rding the language around indicating media-loopback support per media descr=
iption (i.e., per m-line), in sections 3.1 and 3.2.
> >
> > My interpretation/understanding was we were requiring ALL m-lines to in=
dicate media-loopback; in other words it's an all-or-nothing in both the of=
fer+answer, and either every media stream is looped back, or none are. Howe=
ver the text currently doesn't make this clear, and it could be argued eith=
er way, with pro's/con's either way too; so I'd like the WG's consensus on =
whether it should in fact be an all-or-none approach, or not.
> >
> > Please respond to this email by **Tuesday, December 4th** with a "YES" =
if the draft should mandate ALL media be looped-back, or "NO" if it should =
allow a hybrid approach of some streams being looped while others are not.
> >
> > For reference, the draft is here:
> > http://tools.ietf.org/html/draft-ietf-mmusic-media-loopback-24
> >
> > -hadriel
> >
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org<mailto:mmusic@ietf.org>
> > https://www.ietf.org/mailman/listinfo/mmusic
> >
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org<mailto:mmusic@ietf.org>
> https://www.ietf.org/mailman/listinfo/mmusic
_______________________________________________
mmusic mailing list
mmusic@ietf.org<mailto:mmusic@ietf.org>
https://www.ietf.org/mailman/listinfo/mmusic


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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Diso-8859-=
1">
<base href=3D"x-msg://368/">
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; ">
<div><br>
</div>
<div>I take this is a &quot;NO&quot; vote?&nbsp;</div>
<div><br>
</div>
<div>-hadriel</div>
<div><br>
</div>
<div><br>
</div>
<div>
<div>On Nov 20, 2012, at 1:56 PM, Bernard Aboba &lt;<a href=3D"mailto:berna=
rd_aboba@hotmail.com">bernard_aboba@hotmail.com</a>&gt; wrote:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite">
<div class=3D"hmmessage" style=3D"font-size: 12pt; font-family: Calibri; fo=
nt-style: normal; font-variant: normal; font-weight: normal; letter-spacing=
: normal; line-height: normal; orphans: 2; text-align: -webkit-auto; text-i=
ndent: 0px; text-transform: none; white-space: normal; widows: 2; word-spac=
ing: 0px; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "=
>
<div dir=3D"ltr">To Paul's point, there may be a scenario in which some m l=
ines would be looped back and others would not.&nbsp; For example, what if =
I want to do a loopback test of audio/video but also want to be able to con=
verse via RTT to discuss the results of
 the test in progress.&nbsp; In that case the audio and video m lines would=
 be looped back but the text line would not be.<span class=3D"Apple-convert=
ed-space">&nbsp;</span><br>
<br>
<div>
<div id=3D"SkyDrivePlaceholder"></div>
&gt; Date: Tue, 20 Nov 2012 13:42:43 -0500<br>
&gt; From:<span class=3D"Apple-converted-space">&nbsp;</span><a href=3D"mai=
lto:pkyzivat@alum.mit.edu">pkyzivat@alum.mit.edu</a><br>
&gt; To:<span class=3D"Apple-converted-space">&nbsp;</span><a href=3D"mailt=
o:mmusic@ietf.org">mmusic@ietf.org</a><br>
&gt; Subject: Re: [MMUSIC] NEED WG CONSENSUS: media-loopback is all-or-noth=
ing<br>
&gt;<span class=3D"Apple-converted-space">&nbsp;</span><br>
&gt; NO<br>
&gt;<span class=3D"Apple-converted-space">&nbsp;</span><br>
&gt; *Why* should the draft be limited this way if it need not be?<br>
&gt;<span class=3D"Apple-converted-space">&nbsp;</span><br>
&gt; Thanks,<br>
&gt; Paul<br>
&gt;<span class=3D"Apple-converted-space">&nbsp;</span><br>
&gt; On 11/20/12 1:16 PM, Hadriel Kaplan wrote:<br>
&gt; &gt; Howdy,<br>
&gt; &gt; during IESG review of the media-loopback draft, a discussion aros=
e regarding the language around indicating media-loopback support per media=
 description (i.e., per m-line), in sections 3.1 and 3.2.<br>
&gt; &gt;<br>
&gt; &gt; My interpretation/understanding was we were requiring ALL m-lines=
 to indicate media-loopback; in other words it's an all-or-nothing in both =
the offer&#43;answer, and either every media stream is looped back, or none=
 are. However the text currently doesn't make
 this clear, and it could be argued either way, with pro's/con's either way=
 too; so I'd like the WG's consensus on whether it should in fact be an all=
-or-none approach, or not.<br>
&gt; &gt;<br>
&gt; &gt; Please respond to this email by **Tuesday, December 4th** with a =
&quot;YES&quot; if the draft should mandate ALL media be looped-back, or &q=
uot;NO&quot; if it should allow a hybrid approach of some streams being loo=
ped while others are not.<br>
&gt; &gt;<br>
&gt; &gt; For reference, the draft is here:<br>
&gt; &gt;<span class=3D"Apple-converted-space">&nbsp;</span><a href=3D"http=
://tools.ietf.org/html/draft-ietf-mmusic-media-loopback-24">http://tools.ie=
tf.org/html/draft-ietf-mmusic-media-loopback-24</a><br>
&gt; &gt;<br>
&gt; &gt; -hadriel<br>
&gt; &gt;<br>
&gt; &gt; _______________________________________________<br>
&gt; &gt; mmusic mailing list<br>
&gt; &gt;<span class=3D"Apple-converted-space">&nbsp;</span><a href=3D"mail=
to:mmusic@ietf.org">mmusic@ietf.org</a><br>
&gt; &gt;<span class=3D"Apple-converted-space">&nbsp;</span><a href=3D"http=
s://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/list=
info/mmusic</a><br>
&gt; &gt;<br>
&gt;<span class=3D"Apple-converted-space">&nbsp;</span><br>
&gt; _______________________________________________<br>
&gt; mmusic mailing list<br>
&gt;<span class=3D"Apple-converted-space">&nbsp;</span><a href=3D"mailto:mm=
usic@ietf.org">mmusic@ietf.org</a><br>
&gt;<span class=3D"Apple-converted-space">&nbsp;</span><a href=3D"https://w=
ww.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/=
mmusic</a><br>
</div>
</div>
_______________________________________________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.o=
rg/mailman/listinfo/mmusic</a><br>
</div>
</blockquote>
</div>
<br>
</body>
</html>

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From btv1==6713412e283==HKaplan@acmepacket.com  Tue Nov 20 11:54:17 2012
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
Thread-Topic: [MMUSIC] NEED WG CONSENSUS: media-loopback is all-or-nothing
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Cc: "<mmusic@ietf.org>" <mmusic@ietf.org>
Subject: Re: [MMUSIC] NEED WG CONSENSUS: media-loopback is all-or-nothing
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Are you asking for someone to defend/explain the advantages are for limitin=
g it, or just giving that as the rationale for your NO vote?
(I don't mean this in a negative way - I'm literally not sure if you're ask=
ing a question or being rhetorical)

-hadriel


On Nov 20, 2012, at 1:42 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> NO
>=20
> *Why* should the draft be limited this way if it need not be?
>=20
> 	Thanks,
> 	Paul
>=20
> On 11/20/12 1:16 PM, Hadriel Kaplan wrote:
>> Howdy,
>> during IESG review of the media-loopback draft, a discussion arose regar=
ding the language around indicating media-loopback support per media descri=
ption (i.e., per m-line), in sections 3.1 and 3.2.
>>=20
>> My interpretation/understanding was we were requiring ALL m-lines to ind=
icate media-loopback; in other words it's an all-or-nothing in both the off=
er+answer, and either every media stream is looped back, or none are.  Howe=
ver the text currently doesn't make this clear, and it could be argued eith=
er way, with pro's/con's either way too; so I'd like the WG's consensus on =
whether it should in fact be an all-or-none approach, or not.
>>=20
>> Please respond to this email by **Tuesday, December 4th** with a "YES" i=
f the draft should mandate ALL media be looped-back, or "NO" if it should a=
llow a hybrid approach of some streams being looped while others are not.
>>=20
>> For reference, the draft is here:
>> http://tools.ietf.org/html/draft-ietf-mmusic-media-loopback-24
>>=20
>> -hadriel
>>=20
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>>=20
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From bernard_aboba@hotmail.com  Tue Nov 20 12:02:25 2012
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] NEED WG CONSENSUS: media-loopback is all-or-nothing
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"NO".

From: HKaplan@acmepacket.com
To: bernard_aboba@hotmail.com
CC: pkyzivat@alum.mit.edu=3B mmusic@ietf.org
Subject: Re: [MMUSIC] NEED WG CONSENSUS: media-loopback is all-or-nothing
Date: Tue=2C 20 Nov 2012 19:51:33 +0000

=0A=
=0A=
=0A=
=0A=
=0A=
=0A=

=0A=
=0A=
I take this is a "NO" vote? =0A=

=0A=
=0A=
-hadriel=0A=

=0A=
=0A=

=0A=
=0A=
=0A=
On Nov 20=2C 2012=2C at 1:56 PM=2C Bernard Aboba <bernard_aboba@hotmail.com=
> wrote:=0A=
=0A=
=0A=
=0A=
To Paul's point=2C there may be a scenario in which some m lines would be l=
ooped back and others would not.  For example=2C what if I want to do a loo=
pback test of audio/video but also want to be able to converse via RTT to d=
iscuss the results of=0A=
 the test in progress.  In that case the audio and video m lines would be l=
ooped back but the text line would not be.=20
=0A=

=0A=
=0A=
=0A=
> Date: Tue=2C 20 Nov 2012 13:42:43 -0500
=0A=
> From: pkyzivat@alum.mit.edu
=0A=
> To: mmusic@ietf.org
=0A=
> Subject: Re: [MMUSIC] NEED WG CONSENSUS: media-loopback is all-or-nothing
=0A=
>=20
=0A=
> NO
=0A=
>=20
=0A=
> *Why* should the draft be limited this way if it need not be?
=0A=
>=20
=0A=
> Thanks=2C
=0A=
> Paul
=0A=
>=20
=0A=
> On 11/20/12 1:16 PM=2C Hadriel Kaplan wrote:
=0A=
> > Howdy=2C
=0A=
> > during IESG review of the media-loopback draft=2C a discussion arose re=
garding the language around indicating media-loopback support per media des=
cription (i.e.=2C per m-line)=2C in sections 3.1 and 3.2.
=0A=
> >
=0A=
> > My interpretation/understanding was we were requiring ALL m-lines to in=
dicate media-loopback=3B in other words it's an all-or-nothing in both the =
offer+answer=2C and either every media stream is looped back=2C or none are=
. However the text currently doesn't make=0A=
 this clear=2C and it could be argued either way=2C with pro's/con's either=
 way too=3B so I'd like the WG's consensus on whether it should in fact be =
an all-or-none approach=2C or not.
=0A=
> >
=0A=
> > Please respond to this email by **Tuesday=2C December 4th** with a "YES=
" if the draft should mandate ALL media be looped-back=2C or "NO" if it sho=
uld allow a hybrid approach of some streams being looped while others are n=
ot.
=0A=
> >
=0A=
> > For reference=2C the draft is here:
=0A=
> > http://tools.ietf.org/html/draft-ietf-mmusic-media-loopback-24
=0A=
> >
=0A=
> > -hadriel
=0A=
> >
=0A=
> > _______________________________________________
=0A=
> > mmusic mailing list
=0A=
> > mmusic@ietf.org
=0A=
> > https://www.ietf.org/mailman/listinfo/mmusic
=0A=
> >
=0A=
>=20
=0A=
> _______________________________________________
=0A=
> mmusic mailing list
=0A=
> mmusic@ietf.org
=0A=
> https://www.ietf.org/mailman/listinfo/mmusic
=0A=
=0A=
=0A=
_______________________________________________
=0A=
mmusic mailing list
=0A=
mmusic@ietf.org
=0A=
https://www.ietf.org/mailman/listinfo/mmusic
=0A=
=0A=
=0A=
=0A=

 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>"NO".<br><br><div><hr id=3D"stop=
Spelling">From: HKaplan@acmepacket.com<br>To: bernard_aboba@hotmail.com<br>=
CC: pkyzivat@alum.mit.edu=3B mmusic@ietf.org<br>Subject: Re: [MMUSIC] NEED =
WG CONSENSUS: media-loopback is all-or-nothing<br>Date: Tue=2C 20 Nov 2012 =
19:51:33 +0000<br><br>=0A=
=0A=
=0A=
=0A=
=0A=
=0A=
<div><br>=0A=
</div>=0A=
<div>I take this is a "NO" vote?&nbsp=3B</div>=0A=
<div><br>=0A=
</div>=0A=
<div>-hadriel</div>=0A=
<div><br>=0A=
</div>=0A=
<div><br>=0A=
</div>=0A=
<div>=0A=
<div>On Nov 20=2C 2012=2C at 1:56 PM=2C Bernard Aboba &lt=3B<a href=3D"mail=
to:bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a>&gt=3B wrote:</d=
iv>=0A=
<br class=3D"ecxApple-interchange-newline">=0A=
<blockquote>=0A=
<div class=3D"ecxhmmessage" style=3D"font-size:12pt=3Bfont-family:Calibri=
=3Bfont-style:normal=3Bfont-variant:normal=3Bfont-weight:normal=3Bletter-sp=
acing:normal=3Bline-height:normal=3Borphans:2=3Btext-align:-webkit-auto=3Bt=
ext-indent:0px=3Btext-transform:none=3Bwhite-space:normal=3Bwidows:2=3Bword=
-spacing:0px">=0A=
<div dir=3D"ltr">To Paul's point=2C there may be a scenario in which some m=
 lines would be looped back and others would not.&nbsp=3B For example=2C wh=
at if I want to do a loopback test of audio/video but also want to be able =
to converse via RTT to discuss the results of=0A=
 the test in progress.&nbsp=3B In that case the audio and video m lines wou=
ld be looped back but the text line would not be.<span class=3D"ecxApple-co=
nverted-space">&nbsp=3B</span><br>=0A=
<br>=0A=
<div>=0A=
=0A=
&gt=3B Date: Tue=2C 20 Nov 2012 13:42:43 -0500<br>=0A=
&gt=3B From:<span class=3D"ecxApple-converted-space">&nbsp=3B</span><a href=
=3D"mailto:pkyzivat@alum.mit.edu">pkyzivat@alum.mit.edu</a><br>=0A=
&gt=3B To:<span class=3D"ecxApple-converted-space">&nbsp=3B</span><a href=
=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>=0A=
&gt=3B Subject: Re: [MMUSIC] NEED WG CONSENSUS: media-loopback is all-or-no=
thing<br>=0A=
&gt=3B<span class=3D"ecxApple-converted-space">&nbsp=3B</span><br>=0A=
&gt=3B NO<br>=0A=
&gt=3B<span class=3D"ecxApple-converted-space">&nbsp=3B</span><br>=0A=
&gt=3B *Why* should the draft be limited this way if it need not be?<br>=0A=
&gt=3B<span class=3D"ecxApple-converted-space">&nbsp=3B</span><br>=0A=
&gt=3B Thanks=2C<br>=0A=
&gt=3B Paul<br>=0A=
&gt=3B<span class=3D"ecxApple-converted-space">&nbsp=3B</span><br>=0A=
&gt=3B On 11/20/12 1:16 PM=2C Hadriel Kaplan wrote:<br>=0A=
&gt=3B &gt=3B Howdy=2C<br>=0A=
&gt=3B &gt=3B during IESG review of the media-loopback draft=2C a discussio=
n arose regarding the language around indicating media-loopback support per=
 media description (i.e.=2C per m-line)=2C in sections 3.1 and 3.2.<br>=0A=
&gt=3B &gt=3B<br>=0A=
&gt=3B &gt=3B My interpretation/understanding was we were requiring ALL m-l=
ines to indicate media-loopback=3B in other words it's an all-or-nothing in=
 both the offer+answer=2C and either every media stream is looped back=2C o=
r none are. However the text currently doesn't make=0A=
 this clear=2C and it could be argued either way=2C with pro's/con's either=
 way too=3B so I'd like the WG's consensus on whether it should in fact be =
an all-or-none approach=2C or not.<br>=0A=
&gt=3B &gt=3B<br>=0A=
&gt=3B &gt=3B Please respond to this email by **Tuesday=2C December 4th** w=
ith a "YES" if the draft should mandate ALL media be looped-back=2C or "NO"=
 if it should allow a hybrid approach of some streams being looped while ot=
hers are not.<br>=0A=
&gt=3B &gt=3B<br>=0A=
&gt=3B &gt=3B For reference=2C the draft is here:<br>=0A=
&gt=3B &gt=3B<span class=3D"ecxApple-converted-space">&nbsp=3B</span><a hre=
f=3D"http://tools.ietf.org/html/draft-ietf-mmusic-media-loopback-24" target=
=3D"_blank">http://tools.ietf.org/html/draft-ietf-mmusic-media-loopback-24<=
/a><br>=0A=
&gt=3B &gt=3B<br>=0A=
&gt=3B &gt=3B -hadriel<br>=0A=
&gt=3B &gt=3B<br>=0A=
&gt=3B &gt=3B _______________________________________________<br>=0A=
&gt=3B &gt=3B mmusic mailing list<br>=0A=
&gt=3B &gt=3B<span class=3D"ecxApple-converted-space">&nbsp=3B</span><a hre=
f=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>=0A=
&gt=3B &gt=3B<span class=3D"ecxApple-converted-space">&nbsp=3B</span><a hre=
f=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_blank">https:=
//www.ietf.org/mailman/listinfo/mmusic</a><br>=0A=
&gt=3B &gt=3B<br>=0A=
&gt=3B<span class=3D"ecxApple-converted-space">&nbsp=3B</span><br>=0A=
&gt=3B _______________________________________________<br>=0A=
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From gunnar.hellstrom@omnitor.se  Tue Nov 20 12:30:50 2012
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Subject: Re: [MMUSIC] NEED WG CONSENSUS: media-loopback is all-or-nothing
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NO

There are unforeseeable effects of requiring all or nothing.

What if a new RTP based medium is registered, e.g. m=aroma.
Then the logic would require that the media-loopback specification would 
need to be updated before m=aroma could be used at all.

There is also an inconsistency in that 3.1 requires all m-lines in the 
sdp to include loopback payload type and the draft is valid only for RTP 
media. There are other media also declared by m-lines, e.g. m=message 
for MSRP, and many m=application are not RTP-based.
A proposal taking care of this problem, but not the one above about new 
media would be to in 3.1 in the phrase
"

each individual media description"
say
"each individual RTP-based media description"

/Gunnar



On 2012-11-20 19:42, Paul Kyzivat wrote:
> NO
>
> *Why* should the draft be limited this way if it need not be?
>
>     Thanks,
>     Paul
>
> On 11/20/12 1:16 PM, Hadriel Kaplan wrote:
>> Howdy,
>> during IESG review of the media-loopback draft, a discussion arose 
>> regarding the language around indicating media-loopback support per 
>> media description (i.e., per m-line), in sections 3.1 and 3.2.
>>
>> My interpretation/understanding was we were requiring ALL m-lines to 
>> indicate media-loopback; in other words it's an all-or-nothing in 
>> both the offer+answer, and either every media stream is looped back, 
>> or none are.  However the text currently doesn't make this clear, and 
>> it could be argued either way, with pro's/con's either way too; so 
>> I'd like the WG's consensus on whether it should in fact be an 
>> all-or-none approach, or not.
>>
>> Please respond to this email by **Tuesday, December 4th** with a 
>> "YES" if the draft should mandate ALL media be looped-back, or "NO" 
>> if it should allow a hybrid approach of some streams being looped 
>> while others are not.
>>
>> For reference, the draft is here:
>> http://tools.ietf.org/html/draft-ietf-mmusic-media-loopback-24
>>
>> -hadriel
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From pkyzivat@alum.mit.edu  Tue Nov 20 12:50:45 2012
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On 11/20/12 2:54 PM, Hadriel Kaplan wrote:
>
> Are you asking for someone to defend/explain the advantages are for limiting it, or just giving that as the rationale for your NO vote?
> (I don't mean this in a negative way - I'm literally not sure if you're asking a question or being rhetorical)

It was primarily a justification for my NO vote.
But it is also an invitation for someone who votes yes to convince me of 
why.

	Thanks,
	Paul

> -hadriel
>
>
> On Nov 20, 2012, at 1:42 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>
>> NO
>>
>> *Why* should the draft be limited this way if it need not be?
>>
>> 	Thanks,
>> 	Paul
>>
>> On 11/20/12 1:16 PM, Hadriel Kaplan wrote:
>>> Howdy,
>>> during IESG review of the media-loopback draft, a discussion arose regarding the language around indicating media-loopback support per media description (i.e., per m-line), in sections 3.1 and 3.2.
>>>
>>> My interpretation/understanding was we were requiring ALL m-lines to indicate media-loopback; in other words it's an all-or-nothing in both the offer+answer, and either every media stream is looped back, or none are.  However the text currently doesn't make this clear, and it could be argued either way, with pro's/con's either way too; so I'd like the WG's consensus on whether it should in fact be an all-or-none approach, or not.
>>>
>>> Please respond to this email by **Tuesday, December 4th** with a "YES" if the draft should mandate ALL media be looped-back, or "NO" if it should allow a hybrid approach of some streams being looped while others are not.
>>>
>>> For reference, the draft is here:
>>> http://tools.ietf.org/html/draft-ietf-mmusic-media-loopback-24
>>>
>>> -hadriel
>>>
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>
>


From fandreas@cisco.com  Tue Nov 20 13:31:54 2012
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One more "NO" vote - I also don't see a reason to mandate such a 
restriction as it may limit the use cases.

-- Flemming

On 11/20/12 1:42 PM, Paul Kyzivat wrote:
> NO
>
> *Why* should the draft be limited this way if it need not be?
>
>     Thanks,
>     Paul
>
> On 11/20/12 1:16 PM, Hadriel Kaplan wrote:
>> Howdy,
>> during IESG review of the media-loopback draft, a discussion arose 
>> regarding the language around indicating media-loopback support per 
>> media description (i.e., per m-line), in sections 3.1 and 3.2.
>>
>> My interpretation/understanding was we were requiring ALL m-lines to 
>> indicate media-loopback; in other words it's an all-or-nothing in 
>> both the offer+answer, and either every media stream is looped back, 
>> or none are.  However the text currently doesn't make this clear, and 
>> it could be argued either way, with pro's/con's either way too; so 
>> I'd like the WG's consensus on whether it should in fact be an 
>> all-or-none approach, or not.
>>
>> Please respond to this email by **Tuesday, December 4th** with a 
>> "YES" if the draft should mandate ALL media be looped-back, or "NO" 
>> if it should allow a hybrid approach of some streams being looped 
>> while others are not.
>>
>> For reference, the draft is here:
>> http://tools.ietf.org/html/draft-ietf-mmusic-media-loopback-24
>>
>> -hadriel
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
> .
>


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Subject: Re: [MMUSIC] T.38 (Re: M-line philosophy (Re: Wisdom sought: Prioritization of codecs))
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On 11/20/12 3:47 AM, Harald Alvestrand wrote:
> On 11/19/2012 05:57 PM, Adam Roach wrote:
>> On 11/19/12 10:50, Harald Alvestrand wrote:
>>>> ...T.38 fax isn't one of them. Sadly.
>>>
>>> What's your scenario where T.38 fax over RTP is relevant when file 
>>> transfer over a reliable channel is available between the same 2 
>>> devices?
>>
>> This is a rat hole that I'm not going to dive too far into, but my 
>> understanding is that the tenacity of fax as an underlying technology 
>> has to do with a substantial body of U.S. case law that establishes 
>> that faxed legal documents are binding, while leaving the issue of 
>> transmission using a technology known by any other name unsettled.
>>
>> Is this a mind-numbingly stupid reason for it to hang on? Yeah, 
>> maybe. Does that change the facts on the ground? No. Not even a 
>> little bit.
> Sympathy. I've read the original judge's opinion in the English court 
> case that established telex as legally binding - he explicitly said "I 
> have this opinion not because it is technically sound, but because it 
> is necessary to keep commerce flowing". We're still there, and the fax 
> industry keeps chugging along because of it.
>
> Still, I don't see the compelling use case for T.38 fax over RTP *into 
> browsers*.

Why not if that's the only client I have ?

-- Flemming



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Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] M-line philosophy (Re: Wisdom sought: Prioritization of codecs)
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On 11/19/12 10:38 AM, Adam Roach wrote:
> On 11/18/12 23:12, Harald Alvestrand wrote:
>>
>> The M-line is problematic because it binds together multiple things:
>>
>> - A top level media type
>> - A set of parameters for media transmission *in all directions* 
>> (bound to payload type)
>> - Transport information (the "RTP session" data)
>>
>>
>> SDP M-lines are already problematic as soon as you want different 
>> parameters in the two directions. It only goes downhill from there 
>> when one does multiple flows with varying characteristics.
>>
>> As I said in Atlanta: SDP M-lines have been walking with one foot on 
>> the "I'm an RTP session" bank of the canal and with the other foot on 
>> the "I'm a media stream description" bank of the canal.
>
> I agree with your characterization of the situation. I'm a little 
> surprised that it doesn't lead you to the same conclusion as I've 
> reached.
>
> There are, at the moment, 176 attributes in the IANA registry that are 
> applicable to m= sections:
>
> 3GPP-Adaptation-Support 3GPP-Asset-Information 3GPP-QoE-Metrics
> 3GPP-SRTP-Config 3gpp-videopostdecbufsize FEC FEC-OTI-extension
> FEC-declaration SRTPAuthentication SRTPROCTxRate T38FaxFillBitRemoval
> T38FaxMaxBuffer T38FaxMaxDatagram T38FaxRateManagement
> T38FaxTranscodingJBIG T38FaxTranscodingMMR T38FaxUdpEC T38FaxVersion
> T38MaxBitRate T38VendorInfo X-decbyterate X-initpostdecbufperiod
> X-initpredecbufperiod X-predecbufsize aal2CPS aal2CPSSDUrate
> aal2sscs3661assured aal2sscs3661unassured aal2sscs3662 aal5sscop
> aalApp aalType abrParms abrSetup acap accept-types accept-wrapped-types
> acfg alt alt-default-id anycast atmQOSparms atmTrfcDesc atmmap bcob
> bearerSigIE bearerType cache candidate capability cbrRate cdsc
> cfw-id chain channel clkrec cmid codecconfig conf confid connection
> content-desc cpar cparmax cparmin cpsSDUsize creq crypto csup curr
> dccp-port dccp-service-code depend des dsel ecan ecn-capable-rtp
> eecid extmap fec fec-repair-flow fec-source-flow file-date
> file-disposition file-icon file-range file-selector file-transfer-id
> fingerprint floorctrl floorid flute-ch flute-tsi fmtp framerate
> framesize fsel g.3gpp.cat g.3gpp.crs gc h248item ice-mismatch ice-pwd
> ice-ufrag ike-setup imageattr inactive isup_usi key-mgmt label lang
> lij max-size maxprate maxptime mbms-flowid mbms-mode mbms-repair
> mid msrp-cema multicast-rtcp omr-codecs omr-m-att omr-m-bw omr-m-cksum
> omr-s-att omr-s-bw omr-s-cksum onewaySel orient path pcfg portmapping-req
> profileDesc prtfl psk-fingerprint ptime qos-mech-recv qos-mech-send
> qosClass quality rams-updates recvonly remote-candidates repair-window
> resource rtcp rtcp-fb rtcp-mux rtcp-rsize rtcp-unicast rtcp-xr
> rtpmap rtpred1 rtpred2 sbc sdplang secondary-realm sendonly sendrecv
> setup silenceSupp source-filter sqn ssrc ssrc-group stc stkmstream
> structure tcap uiLayer1_Prot upcc userid visited-realm vsel zrtp-hash
>
> Some of these are applicable to the RTP session; others only make 
> sense as applied to a single media stream.
>
> If we choose to make each m= section represent one media stream, then 
> these attributes are applicable as-is.
Unfortunately not; some of them are, some of them are not (typically 
because they are attributes related to the "stream" part more than the 
"media format" part). SDES and ICE are examples that require some form 
of selection.

There was an off-line discussion between a few folks in Atlanta about 
taking a look at every single SDP attribute defined today (and the other 
SDP parameters) and determine for each if it would work "as-is" in a 
bundle scenario or whether it would require some sort of special 
treatment and what that would be (e.g "aggregation" or "pick-one"). I'm 
not sure that work has been started yet, but it's probably a good way of 
making progress on this part of the discussion.

Thanks

-- Flemming



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This is a multi-part message in MIME format.
--------------060209080304050108030605
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This is to announce a 2 week Working Group Last Call for

     draft-ietf-mmusic-rtsp-nat-14

as Proposed Standard. Please review and provide any comments you may 
have on the document by December 5, 2012. Comments should be sent to the 
document authors and the MMUSIC WG list. If you review the document but 
do not have any comments, please send a note to that effect as well.


Thanks

        Flemming


Draft Info:

  This draft is a work item of the Multiparty Multimedia Session Control Working Group of the IETF.

	Title           : A Network Address Translator (NAT) Traversal mechanism for media controlled by Real-Time Streaming Protocol (RTSP)
	Author(s)       : Jeff Goldberg
                           Magnus Westerlund
                           Thomas Zeng
	Filename        : draft-ietf-mmusic-rtsp-nat-14.txt
	Pages           : 31
	Date            : 2012-11-15

Abstract:
    This document defines a solution for Network Address Translation
    (NAT) traversal for datagram based media streams setup and controlled
    with Real-time Streaming Protocol version 2 (RTSP 2.0).  It uses
    Interactive Connectivity Establishment (ICE) adapted to use RTSP as a
    signalling channel, defining the necessary extra RTSP extensions and
    procedures.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-rtsp-nat-14

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-rtsp-nat-14


Internet-Drafts are also available by anonymous FTP at:
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    &nbsp;&nbsp;&nbsp; draft-ietf-mmusic-rtsp-nat-14<span class="Apple-style-span"
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                  <br>
                  as Proposed Standard. Please review and provide any
                  comments you may have on the document by December 5,
                  2012. Comments should be sent to the document authors
                  and the MMUSIC WG list. If you review the document but
                  do not have any comments, please send a note to that
                  effect as well. <br>
                  <br>
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                                Thanks <br>
                                <br>
                                &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Flemming<br>
                                <br>
                                <br>
                                Draft Info:<br>
                                <br>
                              </span></span></span></span></span></span></span></span></span></span></span></span></span></span>
    <pre wrap=""> This draft is a work item of the Multiparty Multimedia Session Control Working Group of the IETF.

	Title           : A Network Address Translator (NAT) Traversal mechanism for media controlled by Real-Time Streaming Protocol (RTSP)
	Author(s)       : Jeff Goldberg
                          Magnus Westerlund
                          Thomas Zeng
	Filename        : draft-ietf-mmusic-rtsp-nat-14.txt
	Pages           : 31
	Date            : 2012-11-15

Abstract:
   This document defines a solution for Network Address Translation
   (NAT) traversal for datagram based media streams setup and controlled
   with Real-time Streaming Protocol version 2 (RTSP 2.0).  It uses
   Interactive Connectivity Establishment (ICE) adapted to use RTSP as a
   signalling channel, defining the necessary extra RTSP extensions and
   procedures.


The IETF datatracker status page for this draft is:
<a class="moz-txt-link-freetext" href="https://datatracker.ietf.org/doc/draft-ietf-mmusic-rtsp-nat">https://datatracker.ietf.org/doc/draft-ietf-mmusic-rtsp-nat</a>

There's also a htmlized version available at:
<a class="moz-txt-link-freetext" href="http://tools.ietf.org/html/draft-ietf-mmusic-rtsp-nat-14">http://tools.ietf.org/html/draft-ietf-mmusic-rtsp-nat-14</a>

A diff from the previous version is available at:
<a class="moz-txt-link-freetext" href="http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-rtsp-nat-14">http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-rtsp-nat-14</a>


Internet-Drafts are also available by anonymous FTP at:
<a class="moz-txt-link-freetext" href="ftp://ftp.ietf.org/internet-drafts/">ftp://ftp.ietf.org/internet-drafts/</a>
</pre>
    <br>
  </body>
</html>

--------------060209080304050108030605--

From adam@nostrum.com  Tue Nov 20 13:56:25 2012
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On 11/20/12 02:47, Harald Alvestrand wrote:
> On 11/19/2012 05:57 PM, Adam Roach wrote:
>> On 11/19/12 10:50, Harald Alvestrand wrote:
>>>> ...T.38 fax isn't one of them. Sadly.
>>>
>>> What's your scenario where T.38 fax over RTP is relevant when file 
>>> transfer over a reliable channel is available between the same 2 
>>> devices?
>>
>> This is a rat hole that I'm not going to dive too far into, but my 
>> understanding is that the tenacity of fax as an underlying technology 
>> has to do with a substantial body of U.S. case law that establishes 
>> that faxed legal documents are binding, while leaving the issue of 
>> transmission using a technology known by any other name unsettled.
>>
>> Is this a mind-numbingly stupid reason for it to hang on? Yeah, 
>> maybe. Does that change the facts on the ground? No. Not even a 
>> little bit.
> Sympathy. I've read the original judge's opinion in the English court 
> case that established telex as legally binding - he explicitly said "I 
> have this opinion not because it is technically sound, but because it 
> is necessary to keep commerce flowing". We're still there, and the fax 
> industry keeps chugging along because of it.
>
> Still, I don't see the compelling use case for T.38 fax over RTP *into 
> browsers*.

   +---------+   +------------+   +---------+
   | Web App |---| SIP Server |---|         |
   +---------+   +------------+   |         | +-------------+
        |                         | PSTN GW |--| Fax Machine |
   +---------+      media         |         | +-------------+
   | Browser |--------------------|         |
   +---------+                    +---------+


/a



From rbarnes@bbn.com  Tue Nov 20 14:09:41 2012
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Cc: mmusic@ietf.org, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] NEED WG CONSENSUS: media-loopback is all-or-nothing
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I lean in favor of YES, because it seems to be much simpler and address =
the most pressing use cases.

(I haven't been involved deeply in the discussion, though; is there =
something I'm missing?)



On Nov 20, 2012, at 4:31 PM, Flemming Andreasen <fandreas@cisco.com> =
wrote:

> One more "NO" vote - I also don't see a reason to mandate such a =
restriction as it may limit the use cases.
>=20
> -- Flemming
>=20
> On 11/20/12 1:42 PM, Paul Kyzivat wrote:
>> NO
>>=20
>> *Why* should the draft be limited this way if it need not be?
>>=20
>>    Thanks,
>>    Paul
>>=20
>> On 11/20/12 1:16 PM, Hadriel Kaplan wrote:
>>> Howdy,
>>> during IESG review of the media-loopback draft, a discussion arose =
regarding the language around indicating media-loopback support per =
media description (i.e., per m-line), in sections 3.1 and 3.2.
>>>=20
>>> My interpretation/understanding was we were requiring ALL m-lines to =
indicate media-loopback; in other words it's an all-or-nothing in both =
the offer+answer, and either every media stream is looped back, or none =
are.  However the text currently doesn't make this clear, and it could =
be argued either way, with pro's/con's either way too; so I'd like the =
WG's consensus on whether it should in fact be an all-or-none approach, =
or not.
>>>=20
>>> Please respond to this email by **Tuesday, December 4th** with a =
"YES" if the draft should mandate ALL media be looped-back, or "NO" if =
it should allow a hybrid approach of some streams being looped while =
others are not.
>>>=20
>>> For reference, the draft is here:
>>> http://tools.ietf.org/html/draft-ietf-mmusic-media-loopback-24
>>>=20
>>> -hadriel
>>>=20
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>=20
>>=20
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>> .
>>=20
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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Cc: mmusic@ietf.org, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] NEED WG CONSENSUS: media-loopback is all-or-nothing
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We normally try and provide general solutions rather than focus on 
specific use cases. Looking at media streams in general, we negotiate 
and accept/reject them independently (and likewise for preconditions). 
Why should this be any different ?

Thanks

-- Flemming

On 11/20/12 5:09 PM, Richard L. Barnes wrote:
> I lean in favor of YES, because it seems to be much simpler and address the most pressing use cases.
>
> (I haven't been involved deeply in the discussion, though; is there something I'm missing?)
>
>
>
> On Nov 20, 2012, at 4:31 PM, Flemming Andreasen <fandreas@cisco.com> wrote:
>
>> One more "NO" vote - I also don't see a reason to mandate such a restriction as it may limit the use cases.
>>
>> -- Flemming
>>
>> On 11/20/12 1:42 PM, Paul Kyzivat wrote:
>>> NO
>>>
>>> *Why* should the draft be limited this way if it need not be?
>>>
>>>     Thanks,
>>>     Paul
>>>
>>> On 11/20/12 1:16 PM, Hadriel Kaplan wrote:
>>>> Howdy,
>>>> during IESG review of the media-loopback draft, a discussion arose regarding the language around indicating media-loopback support per media description (i.e., per m-line), in sections 3.1 and 3.2.
>>>>
>>>> My interpretation/understanding was we were requiring ALL m-lines to indicate media-loopback; in other words it's an all-or-nothing in both the offer+answer, and either every media stream is looped back, or none are.  However the text currently doesn't make this clear, and it could be argued either way, with pro's/con's either way too; so I'd like the WG's consensus on whether it should in fact be an all-or-none approach, or not.
>>>>
>>>> Please respond to this email by **Tuesday, December 4th** with a "YES" if the draft should mandate ALL media be looped-back, or "NO" if it should allow a hybrid approach of some streams being looped while others are not.
>>>>
>>>> For reference, the draft is here:
>>>> http://tools.ietf.org/html/draft-ietf-mmusic-media-loopback-24
>>>>
>>>> -hadriel
>>>>
>>>> _______________________________________________
>>>> mmusic mailing list
>>>> mmusic@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>>
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>> .
>>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
> .
>


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Subject: Re: [MMUSIC] RFC 5576 is da answah
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Christer said:

"RFC 5576 is da answah :)=0A=
=0A=
http://tools.ietf.org/rfc/rfc5576.txt=0A=
=0A=
Or=2C do you have specific examples/use-cases where it wouldn't work?=0A=
"

[BA] In the WebRTC context=2C IMHO there are a lot of unanswered questions =
in RFC 5576.=20

>From Section 5:

   Endpoints MUST NOT assume that only the sources mentioned in SDP will
   be present in an RTP session=3B additional sources=2C with previously
   unmentioned SSRC IDs=2C can be added at any time=2C and endpoints MUST b=
e
   prepared to receive packets from these sources.  (How endpoints
   handle such packets is not specified here=3B they SHOULD be handled in
   the same manner as packets from additional sources would be handled
   had the endpoint not received any a=3Dssrc: attributes at all.)

[BA] While the advice above seems generally sensible=2C the meaning of "pre=
pared to receive" isn't very clear.  Implementations may differ in how they=
 respond if they don't receive any a=3Dssrc attributes.   For example=2C an=
 implementation might only be able to deal with a single undeclared source =
or perhaps not at all.  So being "prepared to receive" isn't enough to be a=
ble to deal with an RTP translator=2C for example.=20

>From Section 8:

   When used with the SDP Offer/Answer Model [RFC3264]=2C SDP source-
   specific attributes describe only the sources that each party is
   willing to send (whether it is sending RTP data or RTCP report
   blocks).  No mechanism is provided by which an answer can accept or
   reject individual sources within a media stream=3B if the set of
   sources in a media stream is unacceptable=2C the answerer's only option
   is to reject the media stream or the entire multimedia session.

[BA] The advice above appears to only apply to an answerer that supports RF=
C 5576.  If it doesn't support RFC 5576=2C then it will presumably ignore t=
he a=3Dssrc lines=2C but may not reject the media stream (by setting the po=
rt to zero) or the entire session.   In this case=2C it may not be clear to=
 the Offerer whether in fact it can send all the declared ssrc's or not.  C=
ontrast this with the CLUE approach in which receiver gets to indicate what=
 media captures it would like to receive.=20

>From Section 9:

   Source descriptions are specified in this document such that RTP
   endpoints that are compliant with the RTP specification [RFC3550]
   will be able to decode the media streams they describe whether or not
   they support explicit source descriptions.  However=2C some deployed
   RTP implementations may not actually support multiple media sources
   in a media stream.  Media senders MAY wish to restrict themselves to
   a single source at a time unless they have some means of concluding
   that the receivers of the media stream support source multiplexing.

[BA]  The obvious way for an answerer to indicate support for RFC 5576 woul=
d be to include a=3Dssrc lines in the answer.   If the Answer doesn't inclu=
de a=3Dssrc lines=2C should the Offerer conclude that only one source can b=
e sent?   The above paragraph indicates this is optional.=20
 		 	   		  =

--_1375a7c9-5dab-4edc-b8cc-c38d1fadb17a_
Content-Type: text/html; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<html>
<head>
<style><!--
.hmmessage P
{
margin:0px=3B
padding:0px
}
body.hmmessage
{
font-size: 12pt=3B
font-family:Calibri
}
--></style></head>
<body class=3D'hmmessage'><div dir=3D'ltr'>Christer said:<br><br>"RFC 5576 =
is da answah :)=0A=
=0A=
<a rel=3D"nofollow" href=3D"http://tools.ietf.org/rfc/rfc5576.txt">http://t=
ools.ietf.org/rfc/rfc5576.txt</a>=0A=
=0A=
Or=2C do you have specific examples/use-cases where it wouldn't work?=0A=
"<br><br>[BA] In the WebRTC context=2C IMHO there are a lot of unanswered q=
uestions in RFC 5576. <br><br>From Section 5:<br><br>&nbsp=3B&nbsp=3B Endpo=
ints MUST NOT assume that only the sources mentioned in SDP will<br>&nbsp=
=3B&nbsp=3B be present in an RTP session=3B additional sources=2C with prev=
iously<br>&nbsp=3B&nbsp=3B unmentioned SSRC IDs=2C can be added at any time=
=2C and endpoints MUST be<br>&nbsp=3B&nbsp=3B prepared to receive packets f=
rom these sources.&nbsp=3B (How endpoints<br>&nbsp=3B&nbsp=3B handle such p=
ackets is not specified here=3B they SHOULD be handled in<br>&nbsp=3B&nbsp=
=3B the same manner as packets from additional sources would be handled<br>=
&nbsp=3B&nbsp=3B had the endpoint not received any a=3Dssrc: attributes at =
all.)<br><br>[BA] While the advice above seems generally sensible=2C the me=
aning of "prepared to receive" isn't very clear.&nbsp=3B Implementations ma=
y differ in how they respond if they don't receive any a=3Dssrc attributes.=
&nbsp=3B&nbsp=3B For example=2C an implementation might only be able to dea=
l with a single undeclared source or perhaps not at all.&nbsp=3B So being "=
prepared to receive" isn't enough to be able to deal with an RTP translator=
=2C for example. <br><br>From Section 8:<br><br>&nbsp=3B&nbsp=3B When used =
with the SDP Offer/Answer Model [RFC3264]=2C SDP source-<br>&nbsp=3B&nbsp=
=3B specific attributes describe only the sources that each party is<br>&nb=
sp=3B&nbsp=3B willing to send (whether it is sending RTP data or RTCP repor=
t<br>&nbsp=3B&nbsp=3B blocks).&nbsp=3B No mechanism is provided by which an=
 answer can accept or<br>&nbsp=3B&nbsp=3B reject individual sources within =
a media stream=3B if the set of<br>&nbsp=3B&nbsp=3B sources in a media stre=
am is unacceptable=2C the answerer's only option<br>&nbsp=3B&nbsp=3B is to =
reject the media stream or the entire multimedia session.<br><br>[BA] The a=
dvice above appears to only apply to an answerer that supports RFC 5576.&nb=
sp=3B If it doesn't support RFC 5576=2C then it will presumably ignore the =
a=3Dssrc lines=2C but may not reject the media stream (by setting the port =
to zero) or the entire session.&nbsp=3B&nbsp=3B In this case=2C it may not =
be clear to the Offerer whether in fact it can send all the declared ssrc's=
 or not.&nbsp=3B Contrast this with the CLUE approach in which receiver get=
s to indicate what media captures it would like to receive. <br><br>From Se=
ction 9:<br><br>&nbsp=3B&nbsp=3B Source descriptions are specified in this =
document such that RTP<br>&nbsp=3B&nbsp=3B endpoints that are compliant wit=
h the RTP specification [RFC3550]<br>&nbsp=3B&nbsp=3B will be able to decod=
e the media streams they describe whether or not<br>&nbsp=3B&nbsp=3B they s=
upport explicit source descriptions.&nbsp=3B However=2C some deployed<br>&n=
bsp=3B&nbsp=3B RTP implementations may not actually support multiple media =
sources<br>&nbsp=3B&nbsp=3B in a media stream.&nbsp=3B Media senders MAY wi=
sh to restrict themselves to<br>&nbsp=3B&nbsp=3B a single source at a time =
unless they have some means of concluding<br>&nbsp=3B&nbsp=3B that the rece=
ivers of the media stream support source multiplexing.<br><br>[BA]&nbsp=3B =
The obvious way for an answerer to indicate support for RFC 5576 would be t=
o include a=3Dssrc lines in the answer.&nbsp=3B&nbsp=3B If the Answer doesn=
't include a=3Dssrc lines=2C should the Offerer conclude that only one sour=
ce can be sent?&nbsp=3B&nbsp=3B The above paragraph indicates this is optio=
nal. <br> 		 	   		  </div></body>
</html>=

--_1375a7c9-5dab-4edc-b8cc-c38d1fadb17a_--

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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>, "mmusic@ietf.org (E-mail)" <mmusic@ietf.org>
Thread-Topic: NEED WG CONSENSUS: media-loopback is all-or-nothing
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Date: Wed, 21 Nov 2012 04:30:59 +0000
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Cc: Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>, "draft-ietf-mmusic-media-loopback@tools.ietf.org" <draft-ietf-mmusic-media-loopback@tools.ietf.org>
Subject: Re: [MMUSIC] NEED WG CONSENSUS: media-loopback is all-or-nothing
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|Please respond to this email by **Tuesday, December 4th** with a "YES" if =
the=20
|draft should mandate ALL media be looped-back, or "NO" if it should allow =
a=20
|hybrid approach of some streams being looped while others are not.

NO.

Muthu

|-----Original Message-----
|From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf O=
f Hadriel Kaplan
|Sent: Tuesday, November 20, 2012 11:46 PM
|To: mmusic@ietf.org (E-mail)
|Cc: Gonzalo Camarillo; draft-ietf-mmusic-media-loopback@tools.ietf.org
|Subject: [MMUSIC] NEED WG CONSENSUS: media-loopback is all-or-nothing
|
|Howdy,
|during IESG review of the media-loopback draft, a discussion arose regardi=
ng the language around
|indicating media-loopback support per media description (i.e., per m-line)=
, in sections 3.1 and 3.2.
|
|My interpretation/understanding was we were requiring ALL m-lines to indic=
ate media-loopback; in other
|words it's an all-or-nothing in both the offer+answer, and either every me=
dia stream is looped back,
|or none are.  However the text currently doesn't make this clear, and it c=
ould be argued either way,
|with pro's/con's either way too; so I'd like the WG's consensus on whether=
 it should in fact be an
|all-or-none approach, or not.
|
|Please respond to this email by **Tuesday, December 4th** with a "YES" if =
the draft should mandate ALL
|media be looped-back, or "NO" if it should allow a hybrid approach of some=
 streams being looped while
|others are not.
|
|For reference, the draft is here:
|http://tools.ietf.org/html/draft-ietf-mmusic-media-loopback-24
|
|-hadriel
|
|_______________________________________________
|mmusic mailing list
|mmusic@ietf.org
|https://www.ietf.org/mailman/listinfo/mmusic

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From pkyzivat@alum.mit.edu  Wed Nov 21 06:29:13 2012
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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [MMUSIC] SDP Directorate review of draft-ietf-xrblock-rtcp-xr-burst-gap-discard-06
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Hi Qin,

What you propose is fine with me.

	Thanks,
	Paul

On 11/21/12 3:07 AM, Qin Wu wrote:
> Hi, Paul:
> Thank for your valuable comments, please see my reply inline below.
>
> Regards!
> -Qin
> ----- Original Message -----
> From: "Paul Kyzivat" <pkyzivat@alum.mit.edu>
> To: <mmusic@ietf.org>; <draft-ietf-xrblock-rtcp-xr-burst-gap-discard@tools.ietf.org>
> Sent: Tuesday, November 20, 2012 11:42 PM
> Subject: [MMUSIC] SDP Directorate review of draft-ietf-xrblock-rtcp-xr-burst-gap-discard-06
>
>
>> I am the assigned SDP directorate reviewer for
>> draft-ietf-xrblock-rtcp-xr-burst-gap-discard-06. For background on SDP
>> directorate, please see the FAQ at
>> <http://www.ietf.org/iesg/directorate/sdp.html>.
>>
>> Please wait for direction from your document shepherd or AD before
>> posting a new version of the draft.
>>
>> * Summary:
>>
>> The SDP syntax defined in this draft is trivial and fine.
>> It exercises a defined extension point in RFC3611. That RFC defines an
>> IANA registry for new values using this extension point. This document
>> nominally supplies the information called for when registering. However,
>> the semantics for processing the new syntax is underspecified, and needs
>> to be further elaborated.
>>
>> * New SDP information elements:
>>
>> Section 5 of this draft augments the SDP attribute "rtcp-xr" defined in
>> RFC3611, providing an additional value: "brst-gap-dscrd".
>>
>> * Technical Issues:
>>
>> RFC3611 establishes an IANA registry for new values, and requires that
>> the following information be supplied when registering a new value:
>>
>>     -  Contact information of the one doing the registration, including
>>        at least name, address, and email.
>>
>>     -  An Augmented Backus-Naur Form (ABNF) [2] definition of the
>>        parameter, in accordance with the "format-ext" definition of
>>        Section 5.1.
>>
>>     -  A description of what the parameter represents and how it shall be
>>        interpreted, both normally and in Offer/Answer.
>>
>> The first two of the above are properly supplied. But the offer/answer
>> behavior is insufficiently specified. ("When SDP is used in offer-answer
>> context, the SDP Offer/Answer usage defined in [RFC3611] applies.")
>>
>> RFC3611 states:
>>
>>     Each "rtcp-xr" parameter belongs to one of two categories.  The first
>>     category, the unilateral parameters, are for report blocks that
>>     simply report on the RTP stream and related metrics.  The second
>>     category, collaborative parameters, are for XR blocks that involve
>>     actions by more than one party in order to carry out their functions.
>>
>> It then goes on to give different offer/answer procedures for each
>> category of parameter. The reference in *this* draft ("the SDP
>> Offer/Answer usage defined in [RFC3611] applies") is insufficient
>> because there is no indication in this draft whether the new value is a
>> "unilateral" or "collaborative" parameter.
>>
>> The draft needs to be revised to address this issue. I *think* this new
>> parameter is a "unilateral" parameter. If so, it MAY be sufficient to
>> revise section 5.2 as follows:
>>
>> OLD:
>>     When SDP is used in offer-answer context, the SDP Offer/Answer usage
>>     defined in [RFC3611] applies.
>>
>> NEW:
>>     When SDP is used in offer-answer context, the SDP Offer/Answer usage
>>     defined in [RFC3611] for unilateral "rtcp-xr" attribute parameters
>>     applies.
>
> [Qin]: Thank you for digging deeper into SDP offer answer usage for burst gap discard.
> I fully agree with you that the new value we defined here is "unilateral" parameter.
> And your proposed change looks reasonable to me. Thanks.
>
>
>> It may be useful to provide an expanded explanation.
>
> [Qin]: Maybe we can append one more sentence to your proposed change to say:
> "
> For detailed usage in Offer/Answer for unilateral parameter, refer to section 5.2 of RFC3611.
> "
>
>> Regards,
>> Paul Kyzivat
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


From matthew@matthew.at  Wed Nov 21 07:39:48 2012
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Subject: Re: [MMUSIC] M-line philosophy (Re: Wisdom sought: Prioritization of codecs)
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On 11/19/2012 7:12 AM, Paul Kyzivat wrote:
>
> This just further illustrates the schizophrenic nature of the existing 
> SDP support that you are pointing out.

Explaining why the W3C has chosen to use strings containing SDP as the 
API for WEBRTC.

Matthew Kaufman

From partha@parthasarathi.co.in  Wed Nov 21 07:47:50 2012
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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Muthu Arul Mozhi Perumal \(mperumal\)'" <mperumal@cisco.com>, "'Hadriel Kaplan'" <HKaplan@acmepacket.com>, <mmusic@ietf.org>
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Cc: 'Gonzalo Camarillo' <gonzalo.camarillo@ericsson.com>, draft-ietf-mmusic-media-loopback@tools.ietf.org
Subject: Re: [MMUSIC] NEED WG CONSENSUS: media-loopback is all-or-nothing
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NO...it is possible that some of the device may prefer to loop back only one
of the stream (audio) and not the other streams (like Video).

Thanks
Partha

-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of
Muthu Arul Mozhi Perumal (mperumal)
Sent: Wednesday, November 21, 2012 10:01 AM
To: Hadriel Kaplan; mmusic@ietf.org (E-mail)
Cc: Gonzalo Camarillo; draft-ietf-mmusic-media-loopback@tools.ietf.org
Subject: Re: [MMUSIC] NEED WG CONSENSUS: media-loopback is all-or-nothing

|Please respond to this email by **Tuesday, December 4th** with a "YES" if
the 
|draft should mandate ALL media be looped-back, or "NO" if it should allow a

|hybrid approach of some streams being looped while others are not.

NO.

Muthu

|-----Original Message-----
|From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of
Hadriel Kaplan
|Sent: Tuesday, November 20, 2012 11:46 PM
|To: mmusic@ietf.org (E-mail)
|Cc: Gonzalo Camarillo; draft-ietf-mmusic-media-loopback@tools.ietf.org
|Subject: [MMUSIC] NEED WG CONSENSUS: media-loopback is all-or-nothing
|
|Howdy,
|during IESG review of the media-loopback draft, a discussion arose
regarding the language around
|indicating media-loopback support per media description (i.e., per m-line),
in sections 3.1 and 3.2.
|
|My interpretation/understanding was we were requiring ALL m-lines to
indicate media-loopback; in other
|words it's an all-or-nothing in both the offer+answer, and either every
media stream is looped back,
|or none are.  However the text currently doesn't make this clear, and it
could be argued either way,
|with pro's/con's either way too; so I'd like the WG's consensus on whether
it should in fact be an
|all-or-none approach, or not.
|
|Please respond to this email by **Tuesday, December 4th** with a "YES" if
the draft should mandate ALL
|media be looped-back, or "NO" if it should allow a hybrid approach of some
streams being looped while
|others are not.
|
|For reference, the draft is here:
|http://tools.ietf.org/html/draft-ietf-mmusic-media-loopback-24
|
|-hadriel
|
|_______________________________________________
|mmusic mailing list
|mmusic@ietf.org
|https://www.ietf.org/mailman/listinfo/mmusic
_______________________________________________
mmusic mailing list
mmusic@ietf.org
https://www.ietf.org/mailman/listinfo/mmusic


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From: Jonathan Lennox <jonathan@vidyo.com>
To: Bernard Aboba <bernard_aboba@hotmail.com>
Date: Wed, 21 Nov 2012 11:50:15 -0500
Thread-Topic: [MMUSIC] RFC 5576 is da answah
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Subject: Re: [MMUSIC] RFC 5576 is da answah
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On Nov 20, 2012, at 7:50 PM, Bernard Aboba wrote:

[BA] In the WebRTC context, IMHO there are a lot of unanswered questions in=
 RFC 5576.

Hi, Bernard --

Yes, RFC 5576 wasn't a complete solution for interactive scenarios; it left=
 some pieces open that needed additional standardization work.  Essentially=
, 5576 the declarative part, but not the negotiation part.

I followed it up with draft-lennox-mmusic-sdp-source-selection (still alive=
, though quiescent) and draft-lennox-mmusic-sdp-max-sources (expired, but m=
ore or less the same thing as draft-westerlund-mmusic-max-ssrc) which I thi=
nk should cover all the issues you raise.

At the time (the IETF Stockholm meeting in 2009) these drafts didn't get mu=
ch interest, because the benefits of source multiplexing weren't commonly u=
nderstood yet.  I'm glad that's now changed.


>From Section 5:

   Endpoints MUST NOT assume that only the sources mentioned in SDP will
   be present in an RTP session; additional sources, with previously
   unmentioned SSRC IDs, can be added at any time, and endpoints MUST be
   prepared to receive packets from these sources.  (How endpoints
   handle such packets is not specified here; they SHOULD be handled in
   the same manner as packets from additional sources would be handled
   had the endpoint not received any a=3Dssrc: attributes at all.)

[BA] While the advice above seems generally sensible, the meaning of "prepa=
red to receive" isn't very clear.  Implementations may differ in how they r=
espond if they don't receive any a=3Dssrc attributes.   For example, an imp=
lementation might only be able to deal with a single undeclared source or p=
erhaps not at all.  So being "prepared to receive" isn't enough to be able =
to deal with an RTP translator, for example.

Yes, you'd need a way to explain what you're prepared to receive; max-ssrc =
could handle this.

Generally, if you're speaking to a legacy SIP endpoint, the only safe thing=
 to assume in the absence of ssrc or max-ssrc attributes is one source per =
session; but since this actively contradicts the RTP spec, that couldn't ex=
actly be made normative.


>From Section 8:

   When used with the SDP Offer/Answer Model [RFC3264], SDP source-
   specific attributes describe only the sources that each party is
   willing to send (whether it is sending RTP data or RTCP report
   blocks).  No mechanism is provided by which an answer can accept or
   reject individual sources within a media stream; if the set of
   sources in a media stream is unacceptable, the answerer's only option
   is to reject the media stream or the entire multimedia session.

[BA] The advice above appears to only apply to an answerer that supports RF=
C 5576.  If it doesn't support RFC 5576, then it will presumably ignore the=
 a=3Dssrc lines, but may not reject the media stream (by setting the port t=
o zero) or the entire session.   In this case, it may not be clear to the O=
fferer whether in fact it can send all the declared ssrc's or not.  Contras=
t this with the CLUE approach in which receiver gets to indicate what media=
 captures it would like to receive.

The reason this is phrased as "no mechanism is provided" was to leave open =
the possibility of defining such a mechanism in the future, not to preclude=
 the possibility of there ever being such a mechanism.  My source-selection=
 draft would be one such mechanism; CLUE would be another.


>From Section 9:

   Source descriptions are specified in this document such that RTP
   endpoints that are compliant with the RTP specification [RFC3550]
   will be able to decode the media streams they describe whether or not
   they support explicit source descriptions.  However, some deployed
   RTP implementations may not actually support multiple media sources
   in a media stream.  Media senders MAY wish to restrict themselves to
   a single source at a time unless they have some means of concluding
   that the receivers of the media stream support source multiplexing.

[BA]  The obvious way for an answerer to indicate support for RFC 5576 woul=
d be to include a=3Dssrc lines in the answer.   If the Answer doesn't inclu=
de a=3Dssrc lines, should the Offerer conclude that only one source can be =
sent?   The above paragraph indicates this is optional.

It's perfectly reasonable to support receiving multiple sources, but not ha=
ve any sources you're interested in sending at the moment.  This problem is=
 what max-ssrc is for.

--
Jonathan Lennox
jonathan@vidyo.com<mailto:jonathan@vidyo.com>



--_000_265A48F7933044159BF9CAA29FE12BAEvidyocom_
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<html><head><base href=3D"x-msg://41/"></head><body style=3D"word-wrap: bre=
ak-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "=
><div>On Nov 20, 2012, at 7:50 PM, Bernard Aboba wrote:</div><div><div><br>=
</div><blockquote type=3D"cite"><span class=3D"Apple-style-span" style=3D"b=
order-collapse: separate; font-style: normal; font-variant: normal; font-we=
ight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text=
-align: -webkit-auto; text-indent: 0px; text-transform: none; white-space: =
normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0p=
x; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect=
: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><=
div class=3D"hmmessage"><div dir=3D"ltr"><font class=3D"Apple-style-span" f=
ace=3D"Calibri"><span class=3D"Apple-style-span" style=3D"font-size: 12pt;"=
>[BA] In the WebRTC context, IMHO there are a lot of unanswered questions i=
n RFC 5576.</span></font><span class=3D"Apple-converted-space" style=3D"fon=
t-family: Calibri; font-size: 12pt; ">&nbsp;</span><br></div></div></span><=
/blockquote><div><br></div><div><div>Hi, Bernard --</div><div><br></div><di=
v>Yes, RFC 5576 wasn't a complete solution for interactive scenarios; it le=
ft some pieces open that needed additional standardization work. &nbsp;Esse=
ntially, 5576 the declarative part, but not the negotiation part.</div><div=
><br></div><div>I followed it up with draft-lennox-mmusic-sdp-source-select=
ion (still alive, though quiescent) and&nbsp;draft-lennox-mmusic-sdp-max-so=
urces (expired, but more or less the same thing as&nbsp;draft-westerlund-mm=
usic-max-ssrc) which I think should cover all the issues you raise.</div><d=
iv><br></div><div>At the time (the IETF Stockholm meeting in 2009) these dr=
afts didn't get much interest, because the benefits of source multiplexing =
weren't commonly understood yet. &nbsp;I'm glad that's now changed.</div></=
div><div><br></div><br><blockquote type=3D"cite"><span class=3D"Apple-style=
-span" style=3D"border-collapse: separate; font-family: Helvetica; font-sty=
le: normal; font-variant: normal; font-weight: normal; letter-spacing: norm=
al; line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent:=
 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0=
px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing=
: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><div class=3D"hm=
message" style=3D"font-size: 12pt; font-family: Calibri; "><div dir=3D"ltr"=
>From Section 5:<br><br>&nbsp;&nbsp; Endpoints MUST NOT assume that only th=
e sources mentioned in SDP will<br>&nbsp;&nbsp; be present in an RTP sessio=
n; additional sources, with previously<br>&nbsp;&nbsp; unmentioned SSRC IDs=
, can be added at any time, and endpoints MUST be<br>&nbsp;&nbsp; prepared =
to receive packets from these sources.&nbsp; (How endpoints<br>&nbsp;&nbsp;=
 handle such packets is not specified here; they SHOULD be handled in<br>&n=
bsp;&nbsp; the same manner as packets from additional sources would be hand=
led<br>&nbsp;&nbsp; had the endpoint not received any a=3Dssrc: attributes =
at all.)<br><br>[BA] While the advice above seems generally sensible, the m=
eaning of "prepared to receive" isn't very clear.&nbsp; Implementations may=
 differ in how they respond if they don't receive any a=3Dssrc attributes.&=
nbsp;&nbsp; For example, an implementation might only be able to deal with =
a single undeclared source or perhaps not at all.&nbsp; So being "prepared =
to receive" isn't enough to be able to deal with an RTP translator, for exa=
mple.<span class=3D"Apple-converted-space">&nbsp;</span><br></div></div></s=
pan></blockquote><div><br></div><div>Yes, you'd need a way to explain what =
you're prepared to receive; max-ssrc could handle this.</div><div><br></div=
><div>Generally, if you're speaking to a legacy SIP endpoint, the only safe=
 thing to assume in the absence of ssrc or max-ssrc attributes is one sourc=
e per session; but since this actively contradicts the RTP spec, that could=
n't exactly be made normative.&nbsp;</div><div><br></div><br><blockquote ty=
pe=3D"cite"><span class=3D"Apple-style-span" style=3D"border-collapse: sepa=
rate; font-family: Helvetica; font-style: normal; font-variant: normal; fon=
t-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; =
text-align: -webkit-auto; text-indent: 0px; text-transform: none; white-spa=
ce: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing=
: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-ef=
fect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px;=
 font-size: medium; "><div class=3D"hmmessage" style=3D"font-size: 12pt; fo=
nt-family: Calibri; "><div dir=3D"ltr">From Section 8:<br><br>&nbsp;&nbsp; =
When used with the SDP Offer/Answer Model [RFC3264], SDP source-<br>&nbsp;&=
nbsp; specific attributes describe only the sources that each party is<br>&=
nbsp;&nbsp; willing to send (whether it is sending RTP data or RTCP report<=
br>&nbsp;&nbsp; blocks).&nbsp; No mechanism is provided by which an answer =
can accept or<br>&nbsp;&nbsp; reject individual sources within a media stre=
am; if the set of<br>&nbsp;&nbsp; sources in a media stream is unacceptable=
, the answerer's only option<br>&nbsp;&nbsp; is to reject the media stream =
or the entire multimedia session.<br><br>[BA] The advice above appears to o=
nly apply to an answerer that supports RFC 5576.&nbsp; If it doesn't suppor=
t RFC 5576, then it will presumably ignore the a=3Dssrc lines, but may not =
reject the media stream (by setting the port to zero) or the entire session=
.&nbsp;&nbsp; In this case, it may not be clear to the Offerer whether in f=
act it can send all the declared ssrc's or not.&nbsp; Contrast this with th=
e CLUE approach in which receiver gets to indicate what media captures it w=
ould like to receive.<span class=3D"Apple-converted-space">&nbsp;</span><br=
></div></div></span></blockquote><div><br></div><div>The reason this is phr=
ased as "no mechanism is provided" was to leave open the possibility of def=
ining such a mechanism in the future, not to preclude the possibility of th=
ere ever being such a mechanism. &nbsp;My source-selection draft would be o=
ne such mechanism; CLUE would be another.</div><div><br></div><br><blockquo=
te type=3D"cite"><span class=3D"Apple-style-span" style=3D"border-collapse:=
 separate; font-family: Helvetica; font-style: normal; font-variant: normal=
; font-weight: normal; letter-spacing: normal; line-height: normal; orphans=
: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: none; whit=
e-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-sp=
acing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-=
in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width:=
 0px; font-size: medium; "><div class=3D"hmmessage" style=3D"font-size: 12p=
t; font-family: Calibri; "><div dir=3D"ltr">From Section 9:<br><br>&nbsp;&n=
bsp; Source descriptions are specified in this document such that RTP<br>&n=
bsp;&nbsp; endpoints that are compliant with the RTP specification [RFC3550=
]<br>&nbsp;&nbsp; will be able to decode the media streams they describe wh=
ether or not<br>&nbsp;&nbsp; they support explicit source descriptions.&nbs=
p; However, some deployed<br>&nbsp;&nbsp; RTP implementations may not actua=
lly support multiple media sources<br>&nbsp;&nbsp; in a media stream.&nbsp;=
 Media senders MAY wish to restrict themselves to<br>&nbsp;&nbsp; a single =
source at a time unless they have some means of concluding<br>&nbsp;&nbsp; =
that the receivers of the media stream support source multiplexing.<br><br>=
[BA]&nbsp; The obvious way for an answerer to indicate support for RFC 5576=
 would be to include a=3Dssrc lines in the answer.&nbsp;&nbsp; If the Answe=
r doesn't include a=3Dssrc lines, should the Offerer conclude that only one=
 source can be sent?&nbsp;&nbsp; The above paragraph indicates this is opti=
onal.<span class=3D"Apple-converted-space">&nbsp;</span><br></div></div></s=
pan></blockquote><br></div><div>It's perfectly reasonable to support receiv=
ing multiple sources, but not have any sources you're interested in sending=
 at the moment. &nbsp;This problem is what max-ssrc is for.</div><br><div a=
pple-content-edited=3D"true">
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; color:=
 rgb(0, 0, 0); font-family: Helvetica; font-style: normal; font-variant: no=
rmal; font-weight: normal; letter-spacing: normal; line-height: normal; orp=
hans: 2; text-align: auto; text-indent: 0px; text-transform: none; white-sp=
ace: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacin=
g: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-e=
ffect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px=
; font-size: medium; "><span class=3D"Apple-style-span" style=3D"border-col=
lapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-style: n=
ormal; font-variant: normal; font-weight: normal; letter-spacing: normal; l=
ine-height: normal; orphans: 2; text-indent: 0px; text-transform: none; whi=
te-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-s=
pacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations=
-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width=
: 0px; font-size: medium; "><div style=3D"word-wrap: break-word; -webkit-nb=
sp-mode: space; -webkit-line-break: after-white-space; ">--</div><div style=
=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: af=
ter-white-space; ">Jonathan Lennox<br><a href=3D"mailto:jonathan@vidyo.com"=
>jonathan@vidyo.com</a><br><br></div></span></span>
</div>
<br></body></html>=

--_000_265A48F7933044159BF9CAA29FE12BAEvidyocom_--

From pkyzivat@alum.mit.edu  Wed Nov 21 09:13:49 2012
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Subject: Re: [MMUSIC] RFC 5576 is da answah
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On 11/21/12 11:50 AM, Jonathan Lennox wrote:

>> [BA] While the advice above seems generally sensible, the meaning of
>> "prepared to receive" isn't very clear.  Implementations may differ in
>> how they respond if they don't receive any a=ssrc attributes.   For
>> example, an implementation might only be able to deal with a single
>> undeclared source or perhaps not at all.  So being "prepared to
>> receive" isn't enough to be able to deal with an RTP translator, for
>> example.
>
> Yes, you'd need a way to explain what you're prepared to receive;
> max-ssrc could handle this.

It doesn't help me very much to know you are "prepared to receive" a 
bunch of media streams unless there is also a way to negotiate what they 
are for. You may be prepared to receive streams for purposes 
incompatible with the streams I am prepared to transmit.

Using a=ssrc it is possible to negotiate things about each stream. For 
those that aren't negotiated that way, ISTM that there must be some 
other mechanism for negotiating semantics. This could be SDP signaling 
that indicated "all otherwise unspecified streams are ...", or the 
semantics could be conveyed via some other protocol (e.g. CLUE) or by 
prior agreement.

	Thanks,
	Paul


From bernard_aboba@hotmail.com  Wed Nov 21 19:30:05 2012
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Subject: Re: [MMUSIC] RFC 5576 is da answah
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Paul Kyzivat said:=20

> It doesn't help me very much to know you are "prepared to receive" a=20
> bunch of media streams unless there is also a way to negotiate what they=
=20
> are for. You may be prepared to receive streams for purposes=20
> incompatible with the streams I am prepared to transmit.
>=20
> Using a=3Dssrc it is possible to negotiate things about each stream. For=
=20
> those that aren't negotiated that way=2C ISTM that there must be some=20
> other mechanism for negotiating semantics. This could be SDP signaling=20
> that indicated "all otherwise unspecified streams are ..."=2C or the=20
> semantics could be conveyed via some other protocol (e.g. CLUE) or by=20
> prior agreement.

[BA] If I read draft-lennox-mmusic-sdp-source-selection correctly=2C it is =
possible to describe aspects of what the answerer is "prepared to receive" =
but not details of what the Offerer is capable of sending.  For example=2C =
Section 4=2C includes the following offer/answer:
       Offer
   m=3Dvideo 49168 RTP/AVP 96=0A=
   a=3Drtpmap:96 H264/90000=0A=
   a=3Dssrc:12345 cname:user1@host1.example.com=0A=
   a=3Dssrc:67890 cname:user2@host2.example.com=0A=

   Answer=0A=
   m=3Dvideo 49170 RTP/AVP 96=0A=
   a=3Drtpmap:96 H264/90000=0A=
   a=3Dremote-ssrc:12345 recv:on=0A=
   a=3Dremote-ssrc:12345 imageattr:* [x=3D720=2Cy=3D576]=0A=
   a=3Dremote-ssrc:12345 framerate:15=0A=


While the above exchange might work for a very basic scenario=2C it doesn't=
 seem capable of handling more complex cases. In the above Offer/Answer exc=
hange=2C the Answerer doesn't know what resolutions or framerates are suppo=
rted by the Offerer.  For example=2C the Offerer might be capable of encodi=
ng 16:9 or 9:16 aspect ratios at 15=2C 30 or 64 frames/second:
16:9:  1280x720=2C 960x540=2C 848x480=2C  640x360=2C  480x270=2C 424x240=2C=
 320x180	  9:16: 720x1280=2C  540x960=2C 480x848=2C  360x640=2C 270x480=2C =
240x424=2C 180x320
Yet in the above exchange=2C the Answerer asks for 720 by 576=2C leaving th=
e Offer to guess what supported resolution is "closest".=20
Also=2C in the Offer=2C there is no indication of what the purpose of ssrc =
67890 is.  For example=2C this could be for slides=2C and by rejecting it=
=2C the Answerer will not be able to usefully participate in the conference=
.  So a content: attribute could come in handy. =20
IMHO=2C it became clear to me at IETF 85 that we are indeed straddling a st=
eadily widening canyon as Harald has said=2C though I see the two cliffs be=
ing CLUE on one side and RFC 5576 on the other. If RFC 5576 is to be "the a=
nswah"=2C then MMUSIC needs to fill in the details of how the SDP negotiati=
on works. Alternatively=2C if the negotiation is to occur outside SDP (as a=
dvocated in CLUE)=2C then we need an explicit statement of where the bounda=
ries lie.  The middle path (pursuing neither with conviction) doesn't seem =
viable to me.  		 	   		  =

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<html>
<head>
<style><!--
.hmmessage P
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{
font-size: 12pt=3B
font-family:Calibri
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<body class=3D'hmmessage'><div dir=3D'ltr'><font face=3D"Calibri=2C sans-se=
rif" size=3D"3">Paul Kyzivat said:&nbsp=3B</font><br><br><div style=3D"font=
-family: Calibri=2C sans-serif=3B font-size: 12pt=3B"><div id=3D"SkyDrivePl=
aceholder"></div>&gt=3B It doesn't help me very much to know you are "prepa=
red to receive" a <br>&gt=3B bunch of media streams unless there is also a =
way to negotiate what they <br>&gt=3B are for. You may be prepared to recei=
ve streams for purposes <br>&gt=3B incompatible with the streams I am prepa=
red to transmit.<br>&gt=3B <br>&gt=3B Using a=3Dssrc it is possible to nego=
tiate things about each stream. For <br>&gt=3B those that aren't negotiated=
 that way=2C ISTM that there must be some <br>&gt=3B other mechanism for ne=
gotiating semantics. This could be SDP signaling <br>&gt=3B that indicated =
"all otherwise unspecified streams are ..."=2C or the <br>&gt=3B semantics =
could be conveyed via some other protocol (e.g. CLUE) or by <br>&gt=3B prio=
r agreement.<br><br></div><div style=3D"font-family: Calibri=2C sans-serif=
=3B font-size: 12pt=3B">[BA]&nbsp=3B<span style=3D"font-size: 12pt=3B">If</=
span><span style=3D"font-size: 12pt=3B">&nbsp=3BI read&nbsp=3Bdraft-lennox-=
mmusic-sdp-source-selection correctly=2C it is possible to describe aspects=
 of what the answerer is "prepared to receive" but not details of what the =
Offerer is capable of sending. &nbsp=3BFor example=2C Section 4=2C includes=
 the following offer/answer:</span></div><div style=3D"font-family: Calibri=
=2C sans-serif=3B font-size: 12pt=3B"><br></div><div style=3D"font-family: =
Calibri=2C sans-serif=3B font-size: 12pt=3B">&nbsp=3B &nbsp=3B &nbsp=3B &nb=
sp=3BOffer</div><div style=3D"font-family: Calibri=2C sans-serif=3B font-si=
ze: 12pt=3B"><br></div><div style=3D"font-family: Calibri=2C sans-serif=3B =
font-size: 12pt=3B"><pre class=3D"newpage" style=3D"font-size: 1em=3B margi=
n-top: 0px=3B margin-bottom: 0px=3B page-break-before: always=3B">   m=3Dvi=
deo 49168 RTP/AVP 96=0A=
   a=3Drtpmap:96 H264/90000=0A=
   a=3Dssrc:12345 cname:user1@host1.example.com=0A=
   a=3Dssrc:67890 cname:user2@host2.example.com=0A=
<br></pre><pre class=3D"newpage" style=3D"font-size: 1em=3B margin-top: 0px=
=3B margin-bottom: 0px=3B page-break-before: always=3B">   Answer</pre><pre=
 class=3D"newpage" style=3D"font-size: 1em=3B margin-top: 0px=3B margin-bot=
tom: 0px=3B page-break-before: always=3B">=0A=
   m=3Dvideo 49170 RTP/AVP 96=0A=
   a=3Drtpmap:96 H264/90000=0A=
   a=3Dremote-ssrc:12345 recv:on=0A=
   a=3Dremote-ssrc:12345 imageattr:* [x=3D720=2Cy=3D576]=0A=
   a=3Dremote-ssrc:12345 framerate:15=0A=
</pre><pre class=3D"newpage" style=3D"font-size: 1em=3B margin-top: 0px=3B =
margin-bottom: 0px=3B page-break-before: always=3B"><br></pre><pre class=3D=
"newpage" style=3D"font-size: 1em=3B margin-top: 0px=3B margin-bottom: 0px=
=3B page-break-before: always=3B"><br></pre></div><div style=3D"font-family=
: Calibri=2C sans-serif=3B font-size: 12pt=3B">While the above exchange mig=
ht work for a very basic scenario=2C it doesn't seem capable of handling mo=
re complex cases. In the above Offer/Answer exchange=2C the Answerer doesn'=
t know what resolutions or framerates are supported by the Offerer. &nbsp=
=3BFor example=2C the Offerer might be capable of encoding 16:9 or 9:16 asp=
ect ratios at 15=2C 30 or 64 frames/second:</div><div style=3D"font-family:=
 Calibri=2C sans-serif=3B font-size: 12pt=3B"><br></div><div><div><font fac=
e=3D"Calibri=2C sans-serif">16:9: &nbsp=3B1280x720=2C</font><span style=3D"=
font-family: Calibri=2C sans-serif=3B">&nbsp=3B960x540=2C&nbsp=3B</span><sp=
an style=3D"font-family: Calibri=2C sans-serif=3B">848x480=2C</span><span s=
tyle=3D"font-family: Calibri=2C sans-serif=3B">&nbsp=3B 640x360=2C</span><s=
pan style=3D"font-family: Calibri=2C sans-serif=3B">&nbsp=3B 480x270=2C</sp=
an>&nbsp=3B<span style=3D"font-family: Calibri=2C sans-serif=3B">424x240=2C=
</span>&nbsp=3B<span style=3D"font-family: Calibri=2C sans-serif=3B">320x18=
0</span><span class=3D"Apple-tab-span" style=3D"font-family: Calibri=2C san=
s-serif=3B white-space: pre=3B">	</span></div><div><font face=3D"Calibri=2C=
 sans-serif">&nbsp=3B&nbsp=3B</font></div><div><span style=3D"font-family: =
Calibri=2C sans-serif=3B">9:16: 720x1280=2C</span><span style=3D"font-famil=
y: Calibri=2C sans-serif=3B">&nbsp=3B 540x960=2C</span><span style=3D"font-=
family: Calibri=2C sans-serif=3B">&nbsp=3B480x848=2C</span><span style=3D"f=
ont-family: Calibri=2C sans-serif=3B">&nbsp=3B 360x640=2C</span><span style=
=3D"font-family: Calibri=2C sans-serif=3B">&nbsp=3B270x480=2C</span><span s=
tyle=3D"font-family: Calibri=2C sans-serif=3B">&nbsp=3B240x424=2C</span><sp=
an style=3D"font-family: Calibri=2C sans-serif=3B">&nbsp=3B180x320</span></=
div></div><div><span style=3D"font-family: Calibri=2C sans-serif=3B"><br></=
span></div><div><span style=3D"font-family: Calibri=2C sans-serif=3B">Yet i=
n the above exchange=2C the Answerer asks for 720 by 576=2C leaving the Off=
er to guess what supported resolution is "closest".&nbsp=3B</span></div><di=
v><span style=3D"font-family: Calibri=2C sans-serif=3B"><br></span></div><d=
iv><span style=3D"font-family: Calibri=2C sans-serif=3B">Also=2C in the Off=
er=2C there is no indication of what the purpose of ssrc 67890 is. &nbsp=3B=
For example=2C this could be for slides=2C and by rejecting it=2C the Answe=
rer will not be able to usefully participate in the conference. &nbsp=3BSo =
a content: attribute could come in handy. &nbsp=3B</span></div><div><span s=
tyle=3D"font-family: Calibri=2C sans-serif=3B"><br></span></div><div><span =
style=3D"font-family: Calibri=2C sans-serif=3B font-size: 16px=3B">IMHO=2C =
it became clear to me at IETF 85 that we are indeed straddling a steadily w=
idening canyon as Harald has said=2C though I see the two cliffs being CLUE=
 on one side and RFC 5576 on the other. If RFC 5576 is to be "the answah"=
=2C then MMUSIC needs to fill in the details of how the SDP negotiation wor=
ks. Alternatively=2C if the negotiation is to occur outside SDP (as advocat=
ed in CLUE)=2C then we need an explicit statement of where the boundaries l=
ie. &nbsp=3BThe middle path (pursuing neither with conviction) doesn't seem=
 viable to me.&nbsp=3B</span></div> 		 	   		  </div></body>
</html>=

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Bernard Aboba <bernard_aboba@hotmail.com>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] RFC 5576 is da answah
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Subject: Re: [MMUSIC] RFC 5576 is da answah
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Hi,

>From Section 8:
>
>   When used with the SDP Offer/Answer Model [RFC3264], SDP source-
>   specific attributes describe only the sources that each party is
>   willing to send (whether it is sending RTP data or RTCP report
>   blocks).  No mechanism is provided by which an answer can accept or
>   reject individual sources within a media stream; if the set of
>   sources in a media stream is unacceptable, the answerer's only option
>   is to reject the media stream or the entire multimedia session.
>
> [BA] The advice above appears to only apply to an answerer that supports =
RFC 5576.  If it doesn't support RFC 5576, then it
> will presumably ignore the a=3Dssrc lines, but may not reject the media s=
tream (by setting the port to zero) or the entire session.
> In this case, it may not be clear to the Offerer whether in fact it can s=
end all the declared ssrc's or not.  Contrast this with the
> CLUE approach in which receiver gets to indicate what media captures it w=
ould like to receive.

In CLUE we have a requirement to be able to communicate with "non-CLUE" and=
 "single stream" devices, so also there we may have to deal with the case w=
here an endpoint doesn't support CLUE and/or RFC 5576.

Also keep in mind that, eventhough max-ssrc allows you to specify the numbe=
r of streams, it is currently not a mechanism to indicate WHICH streams are=
 accepted/rejected.

Regards,

Christer


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<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,&quot=
;sans-serif&quot;;color:#1F497D">&gt;</span><span style=3D"font-family:&quo=
t;Calibri&quot;,&quot;sans-serif&quot;">From Section 8:<br>
<span style=3D"color:#1F497D">&gt;</span><br>
<span style=3D"color:#1F497D">&gt;</span>&nbsp;&nbsp; When used with the SD=
P Offer/Answer Model [RFC3264], SDP source-<br>
<span style=3D"color:#1F497D">&gt;</span>&nbsp;&nbsp; specific attributes d=
escribe only the sources that each party is<br>
<span style=3D"color:#1F497D">&gt;</span>&nbsp;&nbsp; willing to send (whet=
her it is sending RTP data or RTCP report<br>
<span style=3D"color:#1F497D">&gt;</span>&nbsp;&nbsp; blocks).&nbsp; No mec=
hanism is provided by which an answer can accept or<br>
<span style=3D"color:#1F497D">&gt;</span>&nbsp;&nbsp; reject individual sou=
rces within a media stream; if the set of<br>
<span style=3D"color:#1F497D">&gt;</span>&nbsp;&nbsp; sources in a media st=
ream is unacceptable, the answerer's only option<br>
<span style=3D"color:#1F497D">&gt;</span>&nbsp;&nbsp; is to reject the medi=
a stream or the entire multimedia session.<br>
<span style=3D"color:#1F497D">&gt;</span><br>
<span style=3D"color:#1F497D">&gt; </span>[BA] The advice above appears to =
only apply to an answerer that supports RFC 5576.&nbsp; If it doesn't suppo=
rt RFC 5576, then it
<span style=3D"color:#1F497D"><o:p></o:p></span></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&gt;
</span><span style=3D"font-family:&quot;Calibri&quot;,&quot;sans-serif&quot=
;">will presumably ignore the a=3Dssrc lines, but may not reject the media =
stream (by setting the port to zero) or the entire session.&nbsp;&nbsp;
<span style=3D"color:#1F497D"><o:p></o:p></span></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&gt;
</span><span style=3D"font-family:&quot;Calibri&quot;,&quot;sans-serif&quot=
;">In this case, it may not be clear to the Offerer whether in fact it can =
send all the declared ssrc's or not.&nbsp; Contrast this with the
<span style=3D"color:#1F497D"><o:p></o:p></span></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&gt;
</span><span style=3D"font-family:&quot;Calibri&quot;,&quot;sans-serif&quot=
;">CLUE approach in which receiver gets to indicate what media captures it =
would like to receive.
<span style=3D"color:#1F497D"><o:p></o:p></span></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">In CLUE we have a require=
ment to be able to communicate with &#8220;non-CLUE&#8221; and &#8220;singl=
e stream&#8221; devices, so also there we may have to deal with the case wh=
ere an
 endpoint doesn&#8217;t support CLUE and/or RFC 5576.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Also keep in mind that, e=
venthough max-ssrc allows you to specify the number of streams, it is curre=
ntly not a mechanism to indicate WHICH streams are accepted/rejected.<o:p><=
/o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Regards,<o:p></o:p></span=
></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Christer<o:p></o:p></span=
></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
</div>
</div>
</body>
</html>

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From harald@alvestrand.no  Thu Nov 22 04:27:29 2012
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From: Harald Alvestrand <harald@alvestrand.no>
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To: Bernard Aboba <bernard_aboba@hotmail.com>
References: <50A4AC20.5030306@alvestrand.no> <50A4AE7F.4040403@alvestrand.no>, <50A51F04.5090502@alum.mit.edu> <50A54895.7030006@nostrum.com>, <50A9BFD4.3000601@alvestrand.no> <BLU002-W219B21BE1C13A009AE59C0B93550@phx.gbl>
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] M-line philosophy (Re: Wisdom sought: Prioritization of codecs)
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On 11/20/2012 08:12 PM, Bernard Aboba wrote:
> > The M-line is problematic because it binds together multiple things:
> >
> > - A set of parameters for media transmission *in all directions* 
> (bound to payload type)
>
> [BA] RFC 3264 Section 5.1 makes it clear that the parameters need not 
> apply in all directions (though sendrecv is the default):

agreed. What I find more problematic / irritating is that with 
parameters that apply to both directions of a sendrecv stream, there's 
no way to specify one set that applies A->B and another set that 
applies  B->A.
With sendonly / recvonly streams, this is not an issue; with parameters 
that apply only in one direction (such as crypto keys), it is also not 
an issue. But for some parameters, it is an issue.

>
>     The list of media formats for each media stream conveys two pieces of
>     information, namely the set of formats (codecs and any parameters
>     associated with the codec, in the case of RTP) that the offerer is
>     capable of sending and/or receiving (depending on the direction
>     attributes), and, in the case of RTP, the RTP payload type numbers
>     used to identify those formats.  If multiple formats are listed, it
>     means that the offerer is capable of making use of any of those
>     formats during the session.  In other words, the answerer MAY change
>     formats in the middle of the session, making use of any of the
>     formats listed, without sending a new offer.  For a sendonly stream,
>     the offer SHOULD indicate those formats the offerer is willing to
>     send for this stream.  For a recvonly stream, the offer SHOULD
>     indicate those formats the offerer is willing to receive for this
>     stream.  For a sendrecv stream, the offer SHOULD indicate those
>     codecs that the offerer is willing to send and receive with.
>


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    <div class="moz-cite-prefix">On 11/20/2012 08:12 PM, Bernard Aboba
      wrote:<br>
    </div>
    <blockquote cite="mid:BLU002-W219B21BE1C13A009AE59C0B93550@phx.gbl"
      type="cite">
      <style><!--
.hmmessage P
{
margin:0px;
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{
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--></style>
      <div dir="ltr">&gt; The M-line is problematic because it binds
        together multiple things:<br>
        <div>&gt; <br>
          &gt; - A set of parameters for media transmission *in all
          directions* (bound to payload type)<br>
          <br>
          [BA] RFC 3264 Section 5.1 makes it clear that the parameters
          need not apply in all directions (though sendrecv is the
          default): <br>
        </div>
      </div>
    </blockquote>
    <br>
    agreed. What I find more problematic / irritating is that with
    parameters that apply to both directions of a sendrecv stream,
    there's no way to specify one set that applies A-&gt;B and another
    set that applies&nbsp; B-&gt;A.<br>
    With sendonly / recvonly streams, this is not an issue; with
    parameters that apply only in one direction (such as crypto keys),
    it is also not an issue. But for some parameters, it is an issue.<br>
    <br>
    <blockquote cite="mid:BLU002-W219B21BE1C13A009AE59C0B93550@phx.gbl"
      type="cite">
      <div dir="ltr">
        <div><br>
          <pre class="newpage">   The list of media formats for each media stream conveys two pieces of
   information, namely the set of formats (codecs and any parameters
   associated with the codec, in the case of RTP) that the offerer is
   capable of sending and/or receiving (depending on the direction
   attributes), and, in the case of RTP, the RTP payload type numbers
   used to identify those formats.  If multiple formats are listed, it
   means that the offerer is capable of making use of any of those
   formats during the session.  In other words, the answerer MAY change
   formats in the middle of the session, making use of any of the
   formats listed, without sending a new offer.  For a sendonly stream,
   the offer SHOULD indicate those formats the offerer is willing to
   send for this stream.  For a recvonly stream, the offer SHOULD
   indicate those formats the offerer is willing to receive for this
   stream.  For a sendrecv stream, the offer SHOULD indicate those
   codecs that the offerer is willing to send and receive with.</pre>
          <br>
        </div>
      </div>
    </blockquote>
    <br>
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From christer.holmberg@ericsson.com  Thu Nov 22 04:35:31 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>, Bernard Aboba <bernard_aboba@hotmail.com>
Thread-Topic: [MMUSIC] M-line philosophy (Re: Wisdom sought: Prioritization of codecs)
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] M-line philosophy (Re: Wisdom sought: Prioritization of codecs)
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Hi,

>> The M-line is problematic because it binds together multiple things:
>>=20
>> - A set of parameters for media transmission *in all directions* (bound =
to payload type)
>
>[BA] RFC 3264 Section 5.1 makes it clear that the parameters need not appl=
y in all directions (though sendrecv is the default):=20
>
>agreed. What I find more problematic / irritating is that with parameters =
that apply to both directions of a sendrecv stream, there's no way to speci=
fy one set that applies A->B and another set that applies=A0 B->A.
>With sendonly / recvonly streams, this is not an issue; with parameters th=
at apply only in one direction (such as crypto keys), it is also not an iss=
ue. But for some parameters, it is an issue.

The root to these problems is that fact that, at one point many earth spins=
 ago, the community made a decision that with SIP/SDP you only need to indi=
cate your receiving capabilities...

Regards,

Christer




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Subject: Re: [MMUSIC] RFC 5576 is da answah
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On 11/22/2012 04:30 AM, Bernard Aboba wrote:
> Paul Kyzivat said:
>
> > It doesn't help me very much to know you are "prepared to receive" a
> > bunch of media streams unless there is also a way to negotiate what 
> they
> > are for. You may be prepared to receive streams for purposes
> > incompatible with the streams I am prepared to transmit.
> >
> > Using a=ssrc it is possible to negotiate things about each stream. For
> > those that aren't negotiated that way, ISTM that there must be some
> > other mechanism for negotiating semantics. This could be SDP signaling
> > that indicated "all otherwise unspecified streams are ...", or the
> > semantics could be conveyed via some other protocol (e.g. CLUE) or by
> > prior agreement.
>
> [BA] If I read draft-lennox-mmusic-sdp-source-selection correctly, it 
> is possible to describe aspects of what the answerer is "prepared to 
> receive" but not details of what the Offerer is capable of sending. 
>  For example, Section 4, includes the following offer/answer:
>
>      Offer
>
>     m=video 49168 RTP/AVP 96
>     a=rtpmap:96 H264/90000
>     a=ssrc:12345 cname:user1@host1.example.com
>     a=ssrc:67890 cname:user2@host2.example.com
>
>     Answer
>     m=video 49170 RTP/AVP 96
>     a=rtpmap:96 H264/90000
>     a=remote-ssrc:12345 recv:on
>     a=remote-ssrc:12345 imageattr:* [x=720,y=576]
>     a=remote-ssrc:12345 framerate:15
> While the above exchange might work for a very basic scenario, it 
> doesn't seem capable of handling more complex cases. In the above 
> Offer/Answer exchange, the Answerer doesn't know what resolutions or 
> framerates are supported by the Offerer.  For example, the Offerer 
> might be capable of encoding 16:9 or 9:16 aspect ratios at 15, 30 or 
> 64 frames/second:
>
> 16:9:  1280x720, 960x540, 848x480,  640x360,  480x270, 424x240, 320x180
> 9:16: 720x1280,  540x960, 480x848,  360x640, 270x480, 240x424, 180x320
>
> Yet in the above exchange, the Answerer asks for 720 by 576, leaving 
> the Offer to guess what supported resolution is "closest".
That's a misreading of RRC 6236. The answer asks for *exactly* 720 by 
576. The offerer has to either send in that resolution or renegotiate.
If a range was acceptable, the answerer should have indicated that 
([x=700:800,y=500:600]).
>
> Also, in the Offer, there is no indication of what the purpose of ssrc 
> 67890 is.  For example, this could be for slides, and by rejecting it, 
> the Answerer will not be able to usefully participate in the 
> conference.  So a content: attribute could come in handy.
>
> IMHO, it became clear to me at IETF 85 that we are indeed straddling a 
> steadily widening canyon as Harald has said, though I see the two 
> cliffs being CLUE on one side and RFC 5576 on the other.
Are you seeing the canyon as being in-SDP signalling vs out-of-SDP 
signalling?
If CLUE goes with multiple streams per m-line, and in-SDP signalling, it 
needs RFC 5576.
If out-of-band signalling needs to refer to SSRCs, we need unique 
identifiers (SRCNAME or MSID) that out-of-band signalling can use to 
refer to the flows. Those will likely use RFC 5576 too.

> If RFC 5576 is to be "the answah", then MMUSIC needs to fill in the 
> details of how the SDP negotiation works. Alternatively, if the 
> negotiation is to occur outside SDP (as advocated in CLUE), then we 
> need an explicit statement of where the boundaries lie.  The middle 
> path (pursuing neither with conviction) doesn't seem viable to me.
Our ability to converge on a boundary definition with conviction seems 
limited. So since we're presently implementing, it seems that our 
options are limited to what can be achieved without such convergence.
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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    <div class="moz-cite-prefix">On 11/22/2012 04:30 AM, Bernard Aboba
      wrote:<br>
    </div>
    <blockquote cite="mid:BLU002-W158A3D7EDFBFADEA151D8E935B0@phx.gbl"
      type="cite">
      <style><!--
.hmmessage P
{
margin:0px;
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}
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{
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font-family:Calibri
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--></style>
      <div dir="ltr"><font face="Calibri, sans-serif" size="3">Paul
          Kyzivat said:&nbsp;</font><br>
        <br>
        <div style="font-family: Calibri, sans-serif; font-size: 12pt;">&gt;
          It doesn't help me very much to know you are "prepared to
          receive" a <br>
          &gt; bunch of media streams unless there is also a way to
          negotiate what they <br>
          &gt; are for. You may be prepared to receive streams for
          purposes <br>
          &gt; incompatible with the streams I am prepared to transmit.<br>
          &gt; <br>
          &gt; Using a=ssrc it is possible to negotiate things about
          each stream. For <br>
          &gt; those that aren't negotiated that way, ISTM that there
          must be some <br>
          &gt; other mechanism for negotiating semantics. This could be
          SDP signaling <br>
          &gt; that indicated "all otherwise unspecified streams are
          ...", or the <br>
          &gt; semantics could be conveyed via some other protocol (e.g.
          CLUE) or by <br>
          &gt; prior agreement.<br>
          <br>
        </div>
        <div style="font-family: Calibri, sans-serif; font-size: 12pt;">[BA]&nbsp;<span
            style="font-size: 12pt;">If</span><span style="font-size:
            12pt;">&nbsp;I read&nbsp;draft-lennox-mmusic-sdp-source-selection
            correctly, it is possible to describe aspects of what the
            answerer is "prepared to receive" but not details of what
            the Offerer is capable of sending. &nbsp;For example, Section 4,
            includes the following offer/answer:</span></div>
        <div style="font-family: Calibri, sans-serif; font-size: 12pt;"><br>
        </div>
        <div style="font-family: Calibri, sans-serif; font-size: 12pt;">&nbsp;
          &nbsp; &nbsp; &nbsp;Offer</div>
        <div style="font-family: Calibri, sans-serif; font-size: 12pt;"><br>
        </div>
        <div style="font-family: Calibri, sans-serif; font-size: 12pt;">
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">   m=video 49168 RTP/AVP 96
   a=rtpmap:96 H264/90000
   a=ssrc:12345 <a class="moz-txt-link-abbreviated" href="mailto:cname:user1@host1.example.com">cname:user1@host1.example.com</a>
   a=ssrc:67890 <a class="moz-txt-link-abbreviated" href="mailto:cname:user2@host2.example.com">cname:user2@host2.example.com</a>

</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">   Answer</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">   m=video 49170 RTP/AVP 96
   a=rtpmap:96 H264/90000
   a=remote-ssrc:12345 recv:on
   a=remote-ssrc:12345 imageattr:* [x=720,y=576]
   a=remote-ssrc:12345 framerate:15
</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">
</pre>
          <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always;">
</pre>
        </div>
        <div style="font-family: Calibri, sans-serif; font-size: 12pt;">While
          the above exchange might work for a very basic scenario, it
          doesn't seem capable of handling more complex cases. In the
          above Offer/Answer exchange, the Answerer doesn't know what
          resolutions or framerates are supported by the Offerer. &nbsp;For
          example, the Offerer might be capable of encoding 16:9 or 9:16
          aspect ratios at 15, 30 or 64 frames/second:</div>
        <div style="font-family: Calibri, sans-serif; font-size: 12pt;"><br>
        </div>
        <div>
          <div><font face="Calibri, sans-serif">16:9: &nbsp;1280x720,</font><span
              style="font-family: Calibri, sans-serif;">&nbsp;960x540,&nbsp;</span><span
              style="font-family: Calibri, sans-serif;">848x480,</span><span
              style="font-family: Calibri, sans-serif;">&nbsp; 640x360,</span><span
              style="font-family: Calibri, sans-serif;">&nbsp; 480x270,</span>&nbsp;<span
              style="font-family: Calibri, sans-serif;">424x240,</span>&nbsp;<span
              style="font-family: Calibri, sans-serif;">320x180</span><span
              class="Apple-tab-span" style="font-family: Calibri,
              sans-serif; white-space: pre;"> </span></div>
          <div><font face="Calibri, sans-serif">&nbsp;&nbsp;</font></div>
          <div><span style="font-family: Calibri, sans-serif;">9:16:
              720x1280,</span><span style="font-family: Calibri,
              sans-serif;">&nbsp; 540x960,</span><span style="font-family:
              Calibri, sans-serif;">&nbsp;480x848,</span><span
              style="font-family: Calibri, sans-serif;">&nbsp; 360x640,</span><span
              style="font-family: Calibri, sans-serif;">&nbsp;270x480,</span><span
              style="font-family: Calibri, sans-serif;">&nbsp;240x424,</span><span
              style="font-family: Calibri, sans-serif;">&nbsp;180x320</span></div>
        </div>
        <div><span style="font-family: Calibri, sans-serif;"><br>
          </span></div>
        <div><span style="font-family: Calibri, sans-serif;">Yet in the
            above exchange, the Answerer asks for 720 by 576, leaving
            the Offer to guess what supported resolution is "closest". <br>
          </span></div>
      </div>
    </blockquote>
    That's a misreading of RRC 6236. The answer asks for *exactly* 720
    by 576. The offerer has to either send in that resolution or
    renegotiate.<br>
    If a range was acceptable, the answerer should have indicated that
    ([x=700:800,y=500:600]).<br>
    <blockquote cite="mid:BLU002-W158A3D7EDFBFADEA151D8E935B0@phx.gbl"
      type="cite">
      <div dir="ltr">
        <div><span style="font-family: Calibri, sans-serif;"><br>
          </span></div>
        <div><span style="font-family: Calibri, sans-serif;">Also, in
            the Offer, there is no indication of what the purpose of
            ssrc 67890 is. &nbsp;For example, this could be for slides, and
            by rejecting it, the Answerer will not be able to usefully
            participate in the conference. &nbsp;So a content: attribute
            could come in handy. &nbsp;</span></div>
        <div><span style="font-family: Calibri, sans-serif;"><br>
          </span></div>
        <div><span style="font-family: Calibri, sans-serif; font-size:
            16px;">IMHO, it became clear to me at IETF 85 that we are
            indeed straddling a steadily widening canyon as Harald has
            said, though I see the two cliffs being CLUE on one side and
            RFC 5576 on the other.</span></div>
      </div>
    </blockquote>
    Are you seeing the canyon as being in-SDP signalling vs out-of-SDP
    signalling?<br>
    If CLUE goes with multiple streams per m-line, and in-SDP
    signalling, it needs RFC 5576.<br>
    If out-of-band signalling needs to refer to SSRCs, we need unique
    identifiers (SRCNAME or MSID) that out-of-band signalling can use to
    refer to the flows. Those will likely use RFC 5576 too.<br>
    <br>
    <blockquote cite="mid:BLU002-W158A3D7EDFBFADEA151D8E935B0@phx.gbl"
      type="cite">
      <div dir="ltr">
        <div><span style="font-family: Calibri, sans-serif; font-size:
            16px;"> If RFC 5576 is to be "the answah", then MMUSIC needs
            to fill in the details of how the SDP negotiation works.
            Alternatively, if the negotiation is to occur outside SDP
            (as advocated in CLUE), then we need an explicit statement
            of where the boundaries lie. &nbsp;The middle path (pursuing
            neither with conviction) doesn't seem viable to me. <br>
          </span></div>
      </div>
    </blockquote>
    Our ability to converge on a boundary definition with conviction
    seems limited. So since we're presently implementing, it seems that
    our options are limited to what can be achieved without such
    convergence.<br>
    <blockquote cite="mid:BLU002-W158A3D7EDFBFADEA151D8E935B0@phx.gbl"
      type="cite">
      <div dir="ltr"> </div>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
mmusic mailing list
<a class="moz-txt-link-abbreviated" href="mailto:mmusic@ietf.org">mmusic@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a>
</pre>
    </blockquote>
    <br>
  </body>
</html>

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From stefan.lk.hakansson@ericsson.com  Thu Nov 22 05:39:18 2012
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Subject: Re: [MMUSIC] T.38 (Re: M-line philosophy (Re: Wisdom sought: Prioritization of codecs))
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On 11/20/2012 15:56, Adam Roach wrote:
>
>
>    +---------+   +------------+   +---------+
>    | Web App |---| SIP Server |---|         |
>    +---------+   +------------+   |         | +-------------+
>         |                         | PSTN GW |--| Fax Machine |
>    +---------+      media         |         | +-------------+
>    | Browser |--------------------|         |
>    +---------+                    +---------+

For cases when one client is not a proper fax machine, is not the usual 
thing that is done that the facsimile is converted to/from TIFF, JPEG, 
PDF or similar somewhere between that client and the fax machine? (And 
often sent as a mail attachment)

If so, all tools are already available (e.g. File API if you want to 
store/access on the local file system, XHR or WebSockets - or even the 
rtcweb data channel for transport) to build a client that can handle 
fax. (I am probably missing something)

Stefan


From albrecht.schwarz@alcatel-lucent.com  Thu Nov 22 06:13:35 2012
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From: "Schwarz, Albrecht (Albrecht)" <albrecht.schwarz@alcatel-lucent.com>
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Date: Thu, 22 Nov 2012 14:59:42 +0100
Thread-Topic: [MMUSIC] T.38 (Re: M-line philosophy (Re: Wisdom sought: Prioritization of codecs))
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Subject: Re: [MMUSIC] T.38 (Re: M-line philosophy (Re: Wisdom sought: Prioritization of codecs))
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Looks like that your are describing T.37 FoIP behaviour, whereas Adam did p=
oint on T.38 FoIP.

WebRTC should firstly agree
a) whether fax is part of this conversational realtime multimedia service a=
t all,
and if,
b) which transport mode via IP.

To a): support of F.185 (Internet fax) Yes/No
To b):=20
	b.1) T.37 FoIP store-and-forward already sufficient Yes/No
	b.2) T.38 FoIP realtime required Yes/No ((perhaps T.38 over RTP/UDP or UDP=
TL/UDP))
	b.3) or any alternative to T.38 FoIP realtime Yes/No

NOTE: T.38 would specify the behaviour of the=20
1) PSTN gateway (as "T.38 gateway", which is always a PSTN-to-IP gateway) a=
nd
2) Browser (as "T.38 IAD" (Internet aware fax device))
in below illustration.

-Albrecht

-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 Stefan Hakansson LK
Sent: Donnerstag, 22. November 2012 14:39
To: mmusic@ietf.org
Subject: Re: [MMUSIC] T.38 (Re: M-line philosophy (Re: Wisdom sought: Prior=
itization of codecs))

On 11/20/2012 15:56, Adam Roach wrote:
>
>
>    +---------+   +------------+   +---------+
>    | Web App |---| SIP Server |---|         |
>    +---------+   +------------+   |         | +-------------+
>         |                         | PSTN GW |--| Fax Machine |
>    +---------+      media         |         | +-------------+
>    | Browser |--------------------|         |
>    +---------+                    +---------+

For cases when one client is not a proper fax machine, is not the usual=20
thing that is done that the facsimile is converted to/from TIFF, JPEG,=20
PDF or similar somewhere between that client and the fax machine? (And=20
often sent as a mail attachment)

If so, all tools are already available (e.g. File API if you want to=20
store/access on the local file system, XHR or WebSockets - or even the=20
rtcweb data channel for transport) to build a client that can handle=20
fax. (I am probably missing something)

Stefan

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

From christer.holmberg@ericsson.com  Thu Nov 22 11:49:56 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Harald Alvestrand' <harald@alvestrand.no>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] RFC 5576 is da answah
Thread-Index: AQHNx4I7LDBSKk491E6gDy7Y6A3iQpf0cMeAgAAGk4CAAKwvgIAAmg2AgABnQNA=
Date: Thu, 22 Nov 2012 19:49:51 +0000
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References: <BLU002-W21780C8EDF047E1811F270D93540@phx.gbl>, <265A48F7-9330-4415-9BF9-CAA29FE12BAE@vidyo.com>, <50AD0BCB.9090508@alum.mit.edu> <BLU002-W158A3D7EDFBFADEA151D8E935B0@phx.gbl> <50AE1D75.1070100@alvestrand.no>
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Subject: Re: [MMUSIC] RFC 5576 is da answah
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Hi,

> Are you seeing the canyon as being in-SDP signalling vs out-of-SDP signal=
ling?
> If CLUE goes with multiple streams per m-line, and in-SDP signalling, it =
needs RFC 5576.
> If out-of-band signalling needs to refer to SSRCs, we need unique identif=
iers (SRCNAME or MSID) that out-of-band=20
> signalling can use to refer to the flows. Those will likely use RFC 5576 =
too.
>
> If RFC 5576 is to be "the answah", then MMUSIC needs to fill in the detai=
ls of how the SDP negotiation works.=20

Agree. The same issues apply for both CLUE and RTCWEB, so it needs to be de=
scribed on a general level.

> Alternatively, if the negotiation is to occur outside SDP (as advocated i=
n CLUE), then we need an explicit statement of where=20
> the boundaries lie.  The middle path (pursuing neither with conviction) d=
oesn't seem viable to me.=20

CLUE has not yet decided what goes in SDP, and what goes into the CLUE chan=
nel.

My suggestion has been that "CLUE application specific data" (e.g. spatial =
relationship information) goes into the CLUE channel, while "transport data=
" (especially data which might be needed by intermediaries etc) goes into S=
DP.=20

Now, there ARE some gray areas, where it can be discussed whether specific =
data is "CLUE app" or "transport". BUT, I would be concerned about CLUE put=
ting data in the CLUE channel as a workaround, simply because it doesn't "f=
it" into SDP - even if the data as such would belong SDP.

Regards,

Christer

=09



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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>,  "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] T.38 (Re: M-line philosophy (Re: Wisdom sought: Prioritization of codecs))
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Subject: Re: [MMUSIC] T.38 (Re: M-line philosophy (Re: Wisdom sought: Prioritization of codecs))
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Hi,=20

>>    +---------+   +------------+   +---------+
>>    | Web App |---| SIP Server |---|         |
>>    +---------+   +------------+   |         | +-------------+
>>         |                         | PSTN GW |--| Fax Machine |
>>    +---------+      media         |         | +-------------+
>>    | Browser |--------------------|         |
>>    +---------+                    +---------+
>
> For cases when one client is not a proper fax machine, is not the usual t=
hing that is done that the=20
> facsimile is converted to/from TIFF, JPEG, PDF or similar somewhere betwe=
en that client and the fax machine? (And often sent as a mail attachment)

I don't think you can make that assumption. Note that I am not talking spec=
ifically about browsers, but IP clients in general.

Now, there is ongoing work (draft-hanes-dispatch-fax-capability) in definin=
g a media feature tag, which clients can use to indicate their fax capabili=
ties. Maybe "picture" should be addded as an option...

Regards,

Christer=

From bernard_aboba@hotmail.com  Thu Nov 22 14:17:20 2012
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Subject: Re: [MMUSIC] RFC 5576 is da answah
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Christer said:

"My suggestion has been that "CLUE application specific data" (e.g. spatial=
 relationship information) goes into the CLUE channel=2C while "transport d=
ata" (especially data which might be needed by intermediaries etc) goes int=
o SDP. "

[BA]  That seems like a reasonable guideline to me.  Looking at draft-ietf-=
clue-framework Section 6.1.1=2C many of the attributes other than the "spat=
ial relationship" ones (such as Point of Capture=2C Line of Capture and Are=
a of Capture) do seem like they can be handled in SDP.  For example=2C  the=
 max-send-ssrc:  and max-recv-ssrc: parameters defined in draft-westerlund-=
mmusic-max-ssrc seem to communicate similar information to Max Capture Enco=
dings=2C and Encoding Group might be taken care of via the a=3Dssrc-group c=
onstruct in RFC 5576.   Height=2C Width and Framerate also can be handled i=
n SDP (though IMHO the negotiation mechanism in draft-lennox-mmusic-sdp-sou=
rce-selection needs work).=20

=0A=
    =0A=
    [Harald]  The answer asks for *exactly* 720=0A=
    by 576. The offerer has to either send in that resolution or=0A=
    renegotiate.
=0A=
    If a range was acceptable=2C the answerer should have indicated that=0A=
    ([x=3D700:800=2Cy=3D500:600]).

[BA] Yes=2C but it seems clumsy for the receiver to indicate its preference=
s without information on what the sender supports.  In draft-lennox-mmusic-=
sdp-source-selection=2C imageattr information only seems to be supported in=
 the answer=2C whereas in RFC 6236=2C the Offer may include both send and r=
eceive information as described in Section 6.1:
a=3Dimageattr:PT send attr-list recv attr-listIncluding both send and recei=
ve information in the Offer seems more likely to enable the negotiation to =
conclude in a single Offer/Answer exchange.
For example=2C the Offer can describe what can be sent by including an a=3D=
max-send-ssrc: line as well as information on the individual a=3Dssrc: line=
s (e.g.=2C  imageattr:=2C framerate:=2C content:).  In addition=2C the Offe=
r can communicate about what can be received via an a=3Dmax-recv-ssrc: line=
 (and perhaps a=3Dremote-ssrc: lines with some imageattr: and framerate: su=
ggestions).  Then the Answer can include a=3Dremote-ssrc lines correspondin=
g to  the sources described in the Offer=2C as well as a=3Dssrc: lines desc=
ribing what the Offer can send (which should be compatible with the receive=
 information in the Offer).  This may allow the negotiation to be concluded=
 in a single Offer/Answer exchange. =20

[Harald]  If CLUE goes with multiple streams per m-line=2C and in-SDP=0A=
    signalling=2C it needs RFC 5576. If out-of-band signalling needs to ref=
er to SSRCs=2C we need unique=0A=
    identifiers (SRCNAME or MSID) that out-of-band signalling can use to=0A=
    refer to the flows. Those will likely use RFC 5576 too.

[BA] Christer has suggested that CLUE out-of-band signalling should only be=
 required to communicate parameters (such as spatial relationships) that ar=
e unique to CLUE.   This seems reasonable to me=2C but today there is consi=
derable overlap in the parameters described in =0A=
draft-ietf-clue-framework and SDP documents such as RFC 5576=2C 6236=2C =0A=
draft-lennox-mmusic-sdp-source-selection and =0A=
draft-westerlund-mmusic-max-ssrc. =20
=0A=
   =20
=0A=
    =0A=
      =0A=
         If RFC 5576 is to be "the answah"=2C then MMUSIC needs=0A=
            to fill in the details of how the SDP negotiation works.=0A=
            Alternatively=2C if the negotiation is to occur outside SDP=0A=
            (as advocated in CLUE)=2C then we need an explicit statement=0A=
            of where the boundaries lie.  The middle path (pursuing=0A=
            neither with conviction) doesn't seem viable to me.=20

=0A=
          =0A=
      =0A=
    =0A=
    [Harald] Our ability to converge on a boundary definition with convicti=
on=0A=
    seems limited. So since we're presently implementing=2C it seems that=
=0A=
    our options are limited to what can be achieved without such=0A=
    convergence.

[BA] We can at least describe how the source negotiation is supposed to wor=
k in the absence of CLUE.  That exercise would help in understand the limit=
s of what can be accomplished in a single O/A exchange. =20
 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>Christer said:<br><br>"My sugges=
tion has been that "CLUE application specific data" (e.g. spatial relations=
hip information) goes into the CLUE channel=2C while "transport data" (espe=
cially data which might be needed by intermediaries etc) goes into SDP. "<b=
r><div><br>[BA]&nbsp=3B That seems like a reasonable guideline to me.&nbsp=
=3B Looking at draft-ietf-clue-framework Section 6.1.1=2C many of the attri=
butes other than the "spatial relationship" ones (such as Point of Capture=
=2C Line of Capture and Area of Capture) do seem like they can be handled i=
n SDP.&nbsp=3B For example=2C&nbsp=3B the max-send-ssrc:&nbsp=3B and max-re=
cv-ssrc: parameters defined in draft-westerlund-mmusic-max-ssrc seem to com=
municate similar information to Max Capture Encodings=2C and Encoding Group=
 might be taken care of via the a=3Dssrc-group construct in RFC 5576.&nbsp=
=3B&nbsp=3B Height=2C Width and Framerate also can be handled in SDP (thoug=
h IMHO the negotiation mechanism in draft-lennox-mmusic-sdp-source-selectio=
n needs work). <br><br>=0A=
    =0A=
    [Harald]&nbsp=3B The answer asks for *exactly* 720=0A=
    by 576. The offerer has to either send in that resolution or=0A=
    renegotiate.<br>=0A=
    If a range was acceptable=2C the answerer should have indicated that=0A=
    ([x=3D700:800=2Cy=3D500:600]).<br><br>[BA] Yes=2C but it seems clumsy f=
or the receiver to indicate its preferences without information on what the=
 sender supports.&nbsp=3B In draft-lennox-mmusic-sdp-source-selection=2C im=
ageattr information only seems to be supported in the answer=2C whereas in =
RFC 6236=2C the Offer may include both send and receive information as desc=
ribed in Section 6.1:<br><pre>a=3Dimageattr:PT send attr-list recv attr-lis=
t</pre>Including both send and receive information in the Offer seems more =
likely to enable the negotiation to conclude in a single Offer/Answer excha=
nge.<br>For example=2C the Offer can describe what can be sent by including=
 an a=3Dmax-send-ssrc: line as well as information on the individual a=3Dss=
rc: lines (e.g.=2C&nbsp=3B imageattr:=2C framerate:=2C content:).&nbsp=3B I=
n addition=2C the Offer can communicate about what can be received via an a=
=3Dmax-recv-ssrc: line (and perhaps a=3Dremote-ssrc: lines with some imagea=
ttr: and framerate: suggestions).&nbsp=3B Then the Answer can include a=3Dr=
emote-ssrc lines corresponding to&nbsp=3B the sources described in the Offe=
r=2C as well as a=3Dssrc: lines describing what the Offer can send (which s=
hould be compatible with the receive information in the Offer).&nbsp=3B Thi=
s may allow the negotiation to be concluded in a single Offer/Answer exchan=
ge.&nbsp=3B <br><br>[Harald]&nbsp=3B If CLUE goes with multiple streams per=
 m-line=2C and in-SDP=0A=
    signalling=2C it needs RFC 5576. If out-of-band signalling needs to ref=
er to SSRCs=2C we need unique=0A=
    identifiers (SRCNAME or MSID) that out-of-band signalling can use to=0A=
    refer to the flows. Those will likely use RFC 5576 too.<br><br>[BA] Chr=
ister has suggested that CLUE out-of-band signalling should only be require=
d to communicate parameters (such as spatial relationships) that are unique=
 to CLUE.&nbsp=3B&nbsp=3B This seems reasonable to me=2C but today there is=
 considerable overlap in the parameters described in =0A=
draft-ietf-clue-framework and SDP documents such as RFC 5576=2C 6236=2C =0A=
draft-lennox-mmusic-sdp-source-selection and =0A=
draft-westerlund-mmusic-max-ssrc.&nbsp=3B <br>=0A=
    <br>=0A=
    <blockquote cite=3D"" type=3D"cite">=0A=
      <div dir=3D"ltr">=0A=
        <div><span style=3D"font-family: Calibri=2C sans-serif=3B font-size=
:=0A=
            16px=3B"> If RFC 5576 is to be "the answah"=2C then MMUSIC need=
s=0A=
            to fill in the details of how the SDP negotiation works.=0A=
            Alternatively=2C if the negotiation is to occur outside SDP=0A=
            (as advocated in CLUE)=2C then we need an explicit statement=0A=
            of where the boundaries lie. &nbsp=3BThe middle path (pursuing=
=0A=
            neither with conviction) doesn't seem viable to me. <br><br>=0A=
          </span></div>=0A=
      </div>=0A=
    </blockquote>=0A=
    [Harald] Our ability to converge on a boundary definition with convicti=
on=0A=
    seems limited. So since we're presently implementing=2C it seems that=
=0A=
    our options are limited to what can be achieved without such=0A=
    convergence.<br><br>[BA] We can at least describe how the source negoti=
ation is supposed to work in the absence of CLUE.&nbsp=3B That exercise wou=
ld help in understand the limits of what can be accomplished in a single O/=
A exchange.&nbsp=3B <br></div> 		 	   		  </div></body>
</html>=

--_d7f653f9-915f-48fa-8050-547343466fb2_--

From stefan.lk.hakansson@ericsson.com  Fri Nov 23 03:08:40 2012
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Subject: Re: [MMUSIC] T.38 (Re: M-line philosophy (Re: Wisdom sought: Prioritization of codecs))
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Thanks for the clarifications, and I agree to what rtcweb/webrtc should
decide.

What I was implying in my original mail is that I think T.38 support
should not be added to browsers (at least not as part of the
rtcweb/webrtc effort), and there are at least two reasons:

* It is not in any of the use-cases
* You can support store-and-forward FoIP in browsers already

But this discussion should happen in the rtcweb/webrtc space.

Stefan

On 11/22/2012 02:59 PM, Schwarz, Albrecht (Albrecht) wrote:
> Looks like that your are describing T.37 FoIP behaviour, whereas Adam
> did point on T.38 FoIP.
>
> WebRTC should firstly agree
> a) whether fax is part of this conversational realtime multimedia
> service at all,
> and if,
> b) which transport mode via IP.
>
> To a): support of F.185 (Internet fax) Yes/No
> To b):
>      b.1) T.37 FoIP store-and-forward already sufficient Yes/No
>      b.2) T.38 FoIP realtime required Yes/No ((perhaps T.38 over RTP/UDP
> or UDPTL/UDP))
>      b.3) or any alternative to T.38 FoIP realtime Yes/No
>
> NOTE: T.38 would specify the behaviour of the
> 1) PSTN gateway (as "T.38 gateway", which is always a PSTN-to-IP
> gateway) and
> 2) Browser (as "T.38 IAD" (Internet aware fax device))
> in below illustration.
>
> -Albrecht
>
> -----Original Message-----
> From: mmusic-bounces at ietf.org [mailto:mmusic-bounces at ietf.org] On
> Behalf Of Stefan Hakansson LK
> Sent: Donnerstag, 22. November 2012 14:39
> To: mmusic at ietf.org
> Subject: Re: [MMUSIC] T.38 (Re: M-line philosophy (Re: Wisdom sought:
> Prioritization of codecs))
>
> On 11/20/2012 15:56, Adam Roach wrote:
>  >
>  >
>  >    +---------+   +------------+   +---------+
>  >    | Web App |---| SIP Server |---|         |
>  >    +---------+   +------------+   |         | +-------------+
>  >         |                         | PSTN GW |--| Fax Machine |
>  >    +---------+      media         |         | +-------------+
>  >    | Browser |--------------------|         |
>  >    +---------+                    +---------+
>
> For cases when one client is not a proper fax machine, is not the usual
> thing that is done that the facsimile is converted to/from TIFF, JPEG,
> PDF or similar somewhere between that client and the fax machine? (And
> often sent as a mail attachment)
>
> If so, all tools are already available (e.g. File API if you want to
> store/access on the local file system, XHR or WebSockets - or even the
> rtcweb data channel for transport) to build a client that can handle
> fax. (I am probably missing something)
>
> Stefan
>
> _______________________________________________
> mmusic mailing list
> mmusic at ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic




From albrecht.schwarz@alcatel-lucent.com  Fri Nov 23 03:11:41 2012
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From: "Schwarz, Albrecht (Albrecht)" <albrecht.schwarz@alcatel-lucent.com>
To: Stefan Hakansson LK <stefan.lk.hakansson@ericsson.com>, "mmusic@ietf.org" <mmusic@ietf.org>
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Thread-Topic: [MMUSIC] T.38 (Re: M-line philosophy (Re: Wisdom sought: Prioritization of codecs))
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Subject: Re: [MMUSIC] T.38 (Re: M-line philosophy (Re: Wisdom sought: Prioritization of codecs))
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Fully agree!

-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 Stefan Hakansson LK
Sent: Freitag, 23. November 2012 12:09
To: mmusic@ietf.org
Subject: Re: [MMUSIC] T.38 (Re: M-line philosophy (Re: Wisdom sought: Prior=
itization of codecs))

Thanks for the clarifications, and I agree to what rtcweb/webrtc should
decide.

What I was implying in my original mail is that I think T.38 support
should not be added to browsers (at least not as part of the
rtcweb/webrtc effort), and there are at least two reasons:

* It is not in any of the use-cases
* You can support store-and-forward FoIP in browsers already

But this discussion should happen in the rtcweb/webrtc space.

Stefan

On 11/22/2012 02:59 PM, Schwarz, Albrecht (Albrecht) wrote:
> Looks like that your are describing T.37 FoIP behaviour, whereas Adam
> did point on T.38 FoIP.
>
> WebRTC should firstly agree
> a) whether fax is part of this conversational realtime multimedia
> service at all,
> and if,
> b) which transport mode via IP.
>
> To a): support of F.185 (Internet fax) Yes/No
> To b):
>      b.1) T.37 FoIP store-and-forward already sufficient Yes/No
>      b.2) T.38 FoIP realtime required Yes/No ((perhaps T.38 over RTP/UDP
> or UDPTL/UDP))
>      b.3) or any alternative to T.38 FoIP realtime Yes/No
>
> NOTE: T.38 would specify the behaviour of the
> 1) PSTN gateway (as "T.38 gateway", which is always a PSTN-to-IP
> gateway) and
> 2) Browser (as "T.38 IAD" (Internet aware fax device))
> in below illustration.
>
> -Albrecht
>
> -----Original Message-----
> From: mmusic-bounces at ietf.org [mailto:mmusic-bounces at ietf.org] On
> Behalf Of Stefan Hakansson LK
> Sent: Donnerstag, 22. November 2012 14:39
> To: mmusic at ietf.org
> Subject: Re: [MMUSIC] T.38 (Re: M-line philosophy (Re: Wisdom sought:
> Prioritization of codecs))
>
> On 11/20/2012 15:56, Adam Roach wrote:
>  >
>  >
>  >    +---------+   +------------+   +---------+
>  >    | Web App |---| SIP Server |---|         |
>  >    +---------+   +------------+   |         | +-------------+
>  >         |                         | PSTN GW |--| Fax Machine |
>  >    +---------+      media         |         | +-------------+
>  >    | Browser |--------------------|         |
>  >    +---------+                    +---------+
>
> For cases when one client is not a proper fax machine, is not the usual
> thing that is done that the facsimile is converted to/from TIFF, JPEG,
> PDF or similar somewhere between that client and the fax machine? (And
> often sent as a mail attachment)
>
> If so, all tools are already available (e.g. File API if you want to
> store/access on the local file system, XHR or WebSockets - or even the
> rtcweb data channel for transport) to build a client that can handle
> fax. (I am probably missing something)
>
> Stefan
>
> _______________________________________________
> mmusic mailing list
> mmusic at ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic



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

From internet-drafts@ietf.org  Fri Nov 23 05:04:39 2012
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-rtsp-nat-evaluation-06.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Multiparty Multimedia Session Control Wor=
king Group of the IETF.

	Title           : The Evaluation of Different Network Address Translator (=
NAT) Traversal Techniques for Media Controlled by Real-time Streaming Proto=
col (RTSP)
	Author(s)       : Magnus Westerlund
                          Thomas Zeng
	Filename        : draft-ietf-mmusic-rtsp-nat-evaluation-06.txt
	Pages           : 41
	Date            : 2012-11-23

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


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

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

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


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


From magnus.westerlund@ericsson.com  Fri Nov 23 05:10:02 2012
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Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-rtsp-nat-evaluation-06.txt
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WG,

This version is the update to address all of Flemming's comments from
the WG last call. If you check the diff:

http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-rtsp-nat-evaluation-06

You will notice that this version has gone through a number of
significant changes. Including clearer section structure around
variants. Expanded comparison table and conclusions. Many, many
editorial fixes and quite a lot of corrections and clarifications.

Based on that I would recommend the WG chairs to make a second call on
these changes before requesting publication.

Cheers

Magnus


On 2012-11-23 14:04, internet-drafts@ietf.org wrote:
> 
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>  This draft is a work item of the Multiparty Multimedia Session Control Working Group of the IETF.
> 
> 	Title           : The Evaluation of Different Network Address Translator (NAT) Traversal Techniques for Media Controlled by Real-time Streaming Protocol (RTSP)
> 	Author(s)       : Magnus Westerlund
>                           Thomas Zeng
> 	Filename        : draft-ietf-mmusic-rtsp-nat-evaluation-06.txt
> 	Pages           : 41
> 	Date            : 2012-11-23
> 
> Abstract:
>    This document describes several Network Address Translator (NAT)
>    traversal techniques that were considered to be used for establishing
>    the RTP media flows controlled by the Real-time Streaming Protocol
>    (RTSP).  Each technique includes a description on how it would be
>    used, the security implications of using it and any other deployment
>    considerations it has.  There are also discussions on how NAT
>    traversal techniques relates to firewalls and how each technique can
>    be applied in different use cases.  These findings where used when
>    selecting the NAT traversal for RTSP 2.0.
> 
> 
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-mmusic-rtsp-nat-evaluation
> 
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-mmusic-rtsp-nat-evaluation-06
> 
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-rtsp-nat-evaluation-06
> 
> 
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
> 
> _______________________________________________
> I-D-Announce mailing list
> I-D-Announce@ietf.org
> https://www.ietf.org/mailman/listinfo/i-d-announce
> Internet-Draft directories: http://www.ietf.org/shadow.html
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> 
> 


-- 

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 partha@parthasarathi.co.in  Fri Nov 23 08:23:53 2012
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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Stefan Hakansson LK'" <stefan.lk.hakansson@ericsson.com>, <mmusic@ietf.org>
References: <50AF589F.50501@ericsson.com> <50AF5934.8010307@ericsson.com>
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Date: Fri, 23 Nov 2012 21:53:41 +0530
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Subject: Re: [MMUSIC] T.38 (Re: M-line philosophy (Re: Wisdom sought: Prioritization of codecs))
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Stefan,

I agree with you that usecase does not exists.

I think that T.38 shall be considered as part of PSTN interop as DTMF
usecase is already considered for PSTN interop. In fact, DTMF interop with
PSTN gateway shall be easily achieved using non-media (RTP) mechanism but it
is tough achieve real-time fax interop without media (T.38) support.

Thanks
Partha

-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of
Stefan Hakansson LK
Sent: Friday, November 23, 2012 4:39 PM
To: mmusic@ietf.org
Subject: Re: [MMUSIC] T.38 (Re: M-line philosophy (Re: Wisdom sought:
Prioritization of codecs))

Thanks for the clarifications, and I agree to what rtcweb/webrtc should
decide.

What I was implying in my original mail is that I think T.38 support
should not be added to browsers (at least not as part of the
rtcweb/webrtc effort), and there are at least two reasons:

* It is not in any of the use-cases
* You can support store-and-forward FoIP in browsers already

But this discussion should happen in the rtcweb/webrtc space.

Stefan

On 11/22/2012 02:59 PM, Schwarz, Albrecht (Albrecht) wrote:
> Looks like that your are describing T.37 FoIP behaviour, whereas Adam
> did point on T.38 FoIP.
>
> WebRTC should firstly agree
> a) whether fax is part of this conversational realtime multimedia
> service at all,
> and if,
> b) which transport mode via IP.
>
> To a): support of F.185 (Internet fax) Yes/No
> To b):
>      b.1) T.37 FoIP store-and-forward already sufficient Yes/No
>      b.2) T.38 FoIP realtime required Yes/No ((perhaps T.38 over RTP/UDP
> or UDPTL/UDP))
>      b.3) or any alternative to T.38 FoIP realtime Yes/No
>
> NOTE: T.38 would specify the behaviour of the
> 1) PSTN gateway (as "T.38 gateway", which is always a PSTN-to-IP
> gateway) and
> 2) Browser (as "T.38 IAD" (Internet aware fax device))
> in below illustration.
>
> -Albrecht
>
> -----Original Message-----
> From: mmusic-bounces at ietf.org [mailto:mmusic-bounces at ietf.org] On
> Behalf Of Stefan Hakansson LK
> Sent: Donnerstag, 22. November 2012 14:39
> To: mmusic at ietf.org
> Subject: Re: [MMUSIC] T.38 (Re: M-line philosophy (Re: Wisdom sought:
> Prioritization of codecs))
>
> On 11/20/2012 15:56, Adam Roach wrote:
>  >
>  >
>  >    +---------+   +------------+   +---------+
>  >    | Web App |---| SIP Server |---|         |
>  >    +---------+   +------------+   |         | +-------------+
>  >         |                         | PSTN GW |--| Fax Machine |
>  >    +---------+      media         |         | +-------------+
>  >    | Browser |--------------------|         |
>  >    +---------+                    +---------+
>
> For cases when one client is not a proper fax machine, is not the usual
> thing that is done that the facsimile is converted to/from TIFF, JPEG,
> PDF or similar somewhere between that client and the fax machine? (And
> often sent as a mail attachment)
>
> If so, all tools are already available (e.g. File API if you want to
> store/access on the local file system, XHR or WebSockets - or even the
> rtcweb data channel for transport) to build a client that can handle
> fax. (I am probably missing something)
>
> Stefan
>
> _______________________________________________
> mmusic mailing list
> mmusic at ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic



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


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Subject: Re: [MMUSIC] T.38 (Re: M-line philosophy (Re: Wisdom sought: Prioritization of codecs))
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Hi Partha,
On 2012-11-23 17:23, Parthasarathi R wrote:
> Stefan,
>
> I agree with you that usecase does not exists.
>
> I think that T.38 shall be considered as part of PSTN interop as DTMF
> usecase is already considered for PSTN interop. In fact, DTMF interop with
> PSTN gateway shall be easily achieved using non-media (RTP) mechanism but it
> is tough achieve real-time fax interop without media (T.38) support.


This is correct AFAIK. And personally I did not really want DTMF as part 
of rtcweb - it is just for legacy interop, and can be solved by other 
means as you point out. But the wg's view was that sending DTMF should 
be supported, and perhaps that makes sense - DTMF is quite often used, 
by a lot of people (I use it almost daily to navigate IVRs).

I don't think this goes for fax, I think fax these days is only used by 
certain (few) organizations and only for certain types of communication.

But if you think differently, please make a proposal on the rtcweb mail 
list!

Thanks,
Stefan

>
> Thanks
> Partha
>
> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of
> Stefan Hakansson LK
> Sent: Friday, November 23, 2012 4:39 PM
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] T.38 (Re: M-line philosophy (Re: Wisdom sought:
> Prioritization of codecs))
>
> Thanks for the clarifications, and I agree to what rtcweb/webrtc should
> decide.
>
> What I was implying in my original mail is that I think T.38 support
> should not be added to browsers (at least not as part of the
> rtcweb/webrtc effort), and there are at least two reasons:
>
> * It is not in any of the use-cases
> * You can support store-and-forward FoIP in browsers already
>
> But this discussion should happen in the rtcweb/webrtc space.
>
> Stefan
>
> On 11/22/2012 02:59 PM, Schwarz, Albrecht (Albrecht) wrote:
>> Looks like that your are describing T.37 FoIP behaviour, whereas Adam
>> did point on T.38 FoIP.
>>
>> WebRTC should firstly agree
>> a) whether fax is part of this conversational realtime multimedia
>> service at all,
>> and if,
>> b) which transport mode via IP.
>>
>> To a): support of F.185 (Internet fax) Yes/No
>> To b):
>>       b.1) T.37 FoIP store-and-forward already sufficient Yes/No
>>       b.2) T.38 FoIP realtime required Yes/No ((perhaps T.38 over RTP/UDP
>> or UDPTL/UDP))
>>       b.3) or any alternative to T.38 FoIP realtime Yes/No
>>
>> NOTE: T.38 would specify the behaviour of the
>> 1) PSTN gateway (as "T.38 gateway", which is always a PSTN-to-IP
>> gateway) and
>> 2) Browser (as "T.38 IAD" (Internet aware fax device))
>> in below illustration.
>>
>> -Albrecht
>>
>> -----Original Message-----
>> From: mmusic-bounces at ietf.org [mailto:mmusic-bounces at ietf.org] On
>> Behalf Of Stefan Hakansson LK
>> Sent: Donnerstag, 22. November 2012 14:39
>> To: mmusic at ietf.org
>> Subject: Re: [MMUSIC] T.38 (Re: M-line philosophy (Re: Wisdom sought:
>> Prioritization of codecs))
>>
>> On 11/20/2012 15:56, Adam Roach wrote:
>>   >
>>   >
>>   >    +---------+   +------------+   +---------+
>>   >    | Web App |---| SIP Server |---|         |
>>   >    +---------+   +------------+   |         | +-------------+
>>   >         |                         | PSTN GW |--| Fax Machine |
>>   >    +---------+      media         |         | +-------------+
>>   >    | Browser |--------------------|         |
>>   >    +---------+                    +---------+
>>
>> For cases when one client is not a proper fax machine, is not the usual
>> thing that is done that the facsimile is converted to/from TIFF, JPEG,
>> PDF or similar somewhere between that client and the fax machine? (And
>> often sent as a mail attachment)
>>
>> If so, all tools are already available (e.g. File API if you want to
>> store/access on the local file system, XHR or WebSockets - or even the
>> rtcweb data channel for transport) to build a client that can handle
>> fax. (I am probably missing something)
>>
>> Stefan
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic at ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


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Date: Sun, 25 Nov 2012 11:43:46 -0800
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Subject: Re: [MMUSIC] T.38 (Re: M-line philosophy (Re: Wisdom sought: Prioritization of codecs))
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Stefan said:

> I don't think this goes for fax=2C I think fax these days is only used by=
=20
> certain (few) organizations and only for certain types of communication.
>=20
> But if you think differently=2C please make a proposal on the rtcweb mail=
 list!

[BA]  Given the steep decline in landlines=2C POTS is on the road to becomi=
ng economically non-viable by the end of the decade=2C if not sooner.   Onc=
e POTS shuts down=2C the legal and regulatory requirements which prop up th=
e use of fax will need to be re-thought.  IMHO given the bleak future of fa=
x=2C supporting it in WebRTC is not worth the effort.=20

BTW=2C support for RFC 4733 (DTMF) is now required in draft-ietf-rtcweb-aud=
io-01 instead of  RFC 4734 (Fax=2C Modem and Text Telephony) which was cite=
d (mistakenly) in -00.  =20

Bernard "Spent several hours debugging T.38 last week" Aboba
 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>Stefan said:<br><br><div>&gt=3B =
I don't think this goes for fax=2C I think fax these days is only used by <=
br>&gt=3B certain (few) organizations and only for certain types of communi=
cation.<br>&gt=3B <br>&gt=3B But if you think differently=2C please make a =
proposal on the rtcweb mail list!<br><br>[BA]&nbsp=3B Given the steep decli=
ne in landlines=2C POTS is on the road to becoming economically non-viable =
by the end of the decade=2C if not sooner. &nbsp=3B Once POTS shuts down=2C=
 the legal and regulatory requirements which prop up the use of fax will ne=
ed to be re-thought.&nbsp=3B IMHO given the bleak future of fax=2C supporti=
ng it in WebRTC is not worth the effort. <br><br>BTW=2C support for RFC 473=
3 (DTMF) is now required in draft-ietf-rtcweb-audio-01 instead of&nbsp=3B R=
FC 4734 (Fax=2C Modem and Text Telephony) which was cited (mistakenly) in -=
00. &nbsp=3B <br><br>Bernard "Spent several hours debugging T.38 last week"=
 Aboba<br></div> 		 	   		  </div></body>
</html>=

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From internet-drafts@ietf.org  Sun Nov 25 23:40:04 2012
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-g723-g729-00.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Multiparty Multimedia Session Control Wor=
king Group of the IETF.

	Title           : Offer/Answer Considerations for G.723 Annex A and G.729 =
Annex B
	Author(s)       : Muthu Arul Mozhi Perumal
                          Parthasarathi Ravindran
	Filename        : draft-ietf-mmusic-sdp-g723-g729-00.txt
	Pages           : 8
	Date            : 2012-11-23

Abstract:
   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 and updates RFC4856.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-g723-g729

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-sdp-g723-g729-00


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


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Subject: Re: [MMUSIC] RFC 5576 is da answah
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On 11/22/2012 11:17 PM, Bernard Aboba wrote:
> Christer said:
>
> "My suggestion has been that "CLUE application specific data" (e.g. 
> spatial relationship information) goes into the CLUE channel, while 
> "transport data" (especially data which might be needed by 
> intermediaries etc) goes into SDP. "
>
> [BA]  That seems like a reasonable guideline to me.  Looking at 
> draft-ietf-clue-framework Section 6.1.1, many of the attributes other 
> than the "spatial relationship" ones (such as Point of Capture, Line 
> of Capture and Area of Capture) do seem like they can be handled in 
> SDP.  For example,  the max-send-ssrc:  and max-recv-ssrc: parameters 
> defined in draft-westerlund-mmusic-max-ssrc seem to communicate 
> similar information to Max Capture Encodings, and Encoding Group might 
> be taken care of via the a=ssrc-group construct in RFC 5576. Height, 
> Width and Framerate also can be handled in SDP (though IMHO the 
> negotiation mechanism in draft-lennox-mmusic-sdp-source-selection 
> needs work).
>
> [Harald]  The answer asks for *exactly* 720 by 576. The offerer has to 
> either send in that resolution or renegotiate.
> If a range was acceptable, the answerer should have indicated that 
> ([x=700:800,y=500:600]).
>
> [BA] Yes, but it seems clumsy for the receiver to indicate its 
> preferences without information on what the sender supports. In 
> draft-lennox-mmusic-sdp-source-selection, imageattr information only 
> seems to be supported in the answer, whereas in RFC 6236, the Offer 
> may include both send and receive information as described in Section 6.1:
> a=imageattr:PT send attr-list recv attr-list
> Including both send and receive information in the Offer seems more 
> likely to enable the negotiation to conclude in a single Offer/Answer 
> exchange.

So are you arguing that draft-lennox should specify "imageattr" with the 
"a=ssrc" construct as well as with the "a=remote-ssrc" construct? I 
could support that.

Sender's generic support can be indicated using 6236, but the sender 
might want to specify it per stream too.


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    <div class="moz-cite-prefix">On 11/22/2012 11:17 PM, Bernard Aboba
      wrote:<br>
    </div>
    <blockquote cite="mid:BLU002-W568799D8C49E68A19F7DE0935B0@phx.gbl"
      type="cite">
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      <div dir="ltr">Christer said:<br>
        <br>
        "My suggestion has been that "CLUE application specific data"
        (e.g. spatial relationship information) goes into the CLUE
        channel, while "transport data" (especially data which might be
        needed by intermediaries etc) goes into SDP. "<br>
        <div><br>
          [BA]&nbsp; That seems like a reasonable guideline to me.&nbsp; Looking
          at draft-ietf-clue-framework Section 6.1.1, many of the
          attributes other than the "spatial relationship" ones (such as
          Point of Capture, Line of Capture and Area of Capture) do seem
          like they can be handled in SDP.&nbsp; For example,&nbsp; the
          max-send-ssrc:&nbsp; and max-recv-ssrc: parameters defined in
          draft-westerlund-mmusic-max-ssrc seem to communicate similar
          information to Max Capture Encodings, and Encoding Group might
          be taken care of via the a=ssrc-group construct in RFC 5576.&nbsp;&nbsp;
          Height, Width and Framerate also can be handled in SDP (though
          IMHO the negotiation mechanism in
          draft-lennox-mmusic-sdp-source-selection needs work). <br>
          <br>
          [Harald]&nbsp; The answer asks for *exactly* 720 by 576. The
          offerer has to either send in that resolution or renegotiate.<br>
          If a range was acceptable, the answerer should have indicated
          that ([x=700:800,y=500:600]).<br>
          <br>
          [BA] Yes, but it seems clumsy for the receiver to indicate its
          preferences without information on what the sender supports.&nbsp;
          In draft-lennox-mmusic-sdp-source-selection, imageattr
          information only seems to be supported in the answer, whereas
          in RFC 6236, the Offer may include both send and receive
          information as described in Section 6.1:<br>
          <pre>a=imageattr:PT send attr-list recv attr-list</pre>
          Including both send and receive information in the Offer seems
          more likely to enable the negotiation to conclude in a single
          Offer/Answer exchange.<br>
        </div>
      </div>
    </blockquote>
    <br>
    So are you arguing that draft-lennox should specify "imageattr" with
    the "a=ssrc" construct as well as with the "a=remote-ssrc"
    construct? I could support that.<br>
    <br>
    Sender's generic support can be indicated using 6236, but the sender
    might want to specify it per stream too.<br>
    <br>
  </body>
</html>

--------------070504080009050803050103--

From bernard_aboba@hotmail.com  Mon Nov 26 00:02:52 2012
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Subject: Re: [MMUSIC] RFC 5576 is da answah
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Harald said:

=0A=
    "So are you arguing that draft-lennox should specify "imageattr" with=
=0A=
    the "a=3Dssrc" construct as well as with the "a=3Dremote-ssrc"=0A=
    construct? I could support that.
=0A=
   =20
=0A=
    Sender's generic support can be indicated using 6236=2C but the sender=
=0A=
    might want to specify it per stream too."

[BA] Yes.  There are situations in which per-stream information would be us=
eful (e.g. if the cameras do not have the same capabilities).  Including a=
=3Dssrc and a=3Dremote-ssrc in both the Offer and the Answer=2C along with =
parameters from RFC 6236=2C it seems to me that it would be possible to neg=
otiate per-stream parameters in a single O/A exchange in many situations. =
=20
 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>Harald said:<br><br><div>=0A=
    "So are you arguing that draft-lennox should specify "imageattr" with=
=0A=
    the "a=3Dssrc" construct as well as with the "a=3Dremote-ssrc"=0A=
    construct? I could support that.<br>=0A=
    <br>=0A=
    Sender's generic support can be indicated using 6236=2C but the sender=
=0A=
    might want to specify it per stream too."<br><br>[BA] Yes.&nbsp=3B Ther=
e are situations in which per-stream information would be useful (e.g. if t=
he cameras do not have the same capabilities).&nbsp=3B Including a=3Dssrc a=
nd a=3Dremote-ssrc in both the Offer and the Answer=2C along with parameter=
s from RFC 6236=2C it seems to me that it would be possible to negotiate pe=
r-stream parameters in a single O/A exchange in many situations.&nbsp=3B <b=
r></div> 		 	   		  </div></body>
</html>=

--_6f0d5894-eda6-4812-9f23-1250a284bed1_--

From keith.drage@alcatel-lucent.com  Mon Nov 26 01:56:07 2012
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Bernard Aboba <bernard_aboba@hotmail.com>, =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
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Thread-Topic: [MMUSIC] T.38 (Re: M-line philosophy (Re: Wisdom sought: Prioritization of codecs))
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Be careful to address this issue from a worldwide context rather than a Nor=
th America specific one.

There are still plenty of people raising fax issues from the Asian countrie=
s.

Keith

________________________________
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 Bernard Aboba
Sent: 25 November 2012 19:44
To: Stefan H=E5kansson LK
Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] T.38 (Re: M-line philosophy (Re: Wisdom sought: Prior=
itization of codecs))

Stefan said:
> I don't think this goes for fax, I think fax these days is only used by
> certain (few) organizations and only for certain types of communication.
>
> But if you think differently, please make a proposal on the rtcweb mail l=
ist!

[BA]  Given the steep decline in landlines, POTS is on the road to becoming=
 economically non-viable by the end of the decade, if not sooner.   Once PO=
TS shuts down, the legal and regulatory requirements which prop up the use =
of fax will need to be re-thought.  IMHO given the bleak future of fax, sup=
porting it in WebRTC is not worth the effort.

BTW, support for RFC 4733 (DTMF) is now required in draft-ietf-rtcweb-audio=
-01 instead of  RFC 4734 (Fax, Modem and Text Telephony) which was cited (m=
istakenly) in -00.

Bernard "Spent several hours debugging T.38 last week" Aboba

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<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Be careful to address this issue from =
a
worldwide context rather than a <st1:place w:st=3D"on">North America</st1:p=
lace>
specific one.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>There are still plenty of people raisi=
ng fax
issues from the Asian countries.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Keith<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

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4.0pt'>

<div>

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e=3D3
face=3D"Times New Roman"><span lang=3DEN-US style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</span>=
</font></b><font
size=3D2 face=3DTahoma><span lang=3DEN-US style=3D'font-size:10.0pt;font-fa=
mily:Tahoma'>
mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] <b><span
style=3D'font-weight:bold'>On Behalf Of </span></b>Bernard Aboba<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> 25 November 2012 19:44=
<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Stefan H=E5kansson LK<br=
>
<b><span style=3D'font-weight:bold'>Cc:</span></b> mmusic@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [MMUSIC] T.38 (=
Re:
M-line philosophy (Re: Wisdom sought: Prioritization of codecs))</span></fo=
nt><span
lang=3DEN-US><o:p></o:p></span></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

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<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D3 face=3DC=
alibri><span
style=3D'font-size:12.0pt;font-family:Calibri'>Stefan said:<o:p></o:p></spa=
n></font></p>

<div>

<p class=3DMsoNormal><font size=3D3 face=3DCalibri><span style=3D'font-size=
:12.0pt;
font-family:Calibri'>&gt; I don't think this goes for fax, I think fax thes=
e
days is only used by <br>
&gt; certain (few) organizations and only for certain types of communicatio=
n.<br>
&gt; <br>
&gt; But if you think differently, please make a proposal on the rtcweb mai=
l
list!<br>
<br>
[BA]&nbsp; Given the steep decline in landlines, POTS is on the road to
becoming economically non-viable by the end of the decade, if not sooner.
&nbsp; Once POTS shuts down, the legal and regulatory requirements which pr=
op
up the use of fax will need to be re-thought.&nbsp; IMHO given the bleak fu=
ture
of fax, supporting it in WebRTC is not worth the effort. <br>
<br>
BTW, support for RFC 4733 (DTMF) is now required in draft-ietf-rtcweb-audio=
-01
instead of&nbsp; RFC 4734 (Fax, Modem and Text Telephony) which was cited
(mistakenly) in -00. &nbsp; <br>
<br>
Bernard &quot;Spent several hours debugging T.38 last week&quot; Aboba<o:p>=
</o:p></span></font></p>

</div>

</div>

</div>

</div>

</body>

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From internet-drafts@ietf.org  Mon Nov 26 02:26:00 2012
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-14.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Multiparty Multimedia Session Control Wor=
king Group of the IETF.

	Title           : Session Description Protocol (SDP) Extension For Setting=
 Up Audio and Video Media Streams Over Circuit-Switched Bearers In The Publ=
ic Switched Telephone Network (PSTN)
	Author(s)       : Miguel A. Garcia-Martin
                          Simo Veikkolainen
	Filename        : draft-ietf-mmusic-sdp-cs-14.txt
	Pages           : 37
	Date            : 2012-11-26

Abstract:
   This memo describes use cases, requirements, and protocol extensions
   for using the Session Description Protocol (SDP) Offer/Answer model
   for establishing audio and video media streams over circuit-switched
   bearers in the Public Switched Telephone Network (PSTN).


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-14

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sdp-cs-14


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


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Hello,

We have submitted a new version of the sdp-cs draft which now includes chan=
ges requested by Flemming for publication of the document:

- Typos corrected
- Clarified that codecs used should be those can be run natively on top of =
a CS bearer
- Allow visual separators in E.164 numbers (refer to 'global-number' in RFC=
 3966)
- Removed the pre-Nov 2008 copyright statement
- Modified the example so that "a=3Dcs-correlation" is a media level attrib=
ute
- Moved RFC3108 to be an informative reference

Regards,
Simo


-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 ext internet-drafts@ietf.org
Sent: 26. marraskuuta 2012 12:26
To: i-d-announce@ietf.org
Cc: mmusic@ietf.org
Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-14.txt


A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Multiparty Multimedia Session Control Wor=
king Group of the IETF.

	Title           : Session Description Protocol (SDP) Extension For Setting=
 Up Audio and Video Media Streams Over Circuit-Switched Bearers In The Publ=
ic Switched Telephone Network (PSTN)
	Author(s)       : Miguel A. Garcia-Martin
                          Simo Veikkolainen
	Filename        : draft-ietf-mmusic-sdp-cs-14.txt
	Pages           : 37
	Date            : 2012-11-26

Abstract:
   This memo describes use cases, requirements, and protocol extensions
   for using the Session Description Protocol (SDP) Offer/Answer model
   for establishing audio and video media streams over circuit-switched
   bearers in the Public Switched Telephone Network (PSTN).


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-14

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sdp-cs-14


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

_______________________________________________
mmusic mailing list
mmusic@ietf.org
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From partha@parthasarathi.co.in  Mon Nov 26 09:12:08 2012
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From: "Parthasarathi R" <partha@parthasarathi.co.in>
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Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-g723-g729-00.txt
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Hi all,

Offer/Answer Considerations for G.723 Annex A and G.729 Annex B draft is
published with the change in the draft filename as
draft-ietf-mmusic-sdp-g723-g729-00.txt. As per IETF-84 discussion, there is
no open item in this revision of the draft. Could you please review and
provide your valuable experts comments for further progress.

Thanks
Partha & Muthu  

-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of
internet-drafts@ietf.org
Sent: Monday, November 26, 2012 1:10 PM
To: i-d-announce@ietf.org
Cc: mmusic@ietf.org
Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-g723-g729-00.txt


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

	Title           : Offer/Answer Considerations for G.723 Annex A and
G.729 Annex B
	Author(s)       : Muthu Arul Mozhi Perumal
                          Parthasarathi Ravindran
	Filename        : draft-ietf-mmusic-sdp-g723-g729-00.txt
	Pages           : 8
	Date            : 2012-11-23

Abstract:
   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 and updates RFC4856.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-g723-g729

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-sdp-g723-g729-00


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

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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>
Thread-Topic: [MMUSIC] NEED WG CONSENSUS: media-loopback is all-or-nothing
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Cc: "mmusic@ietf.org \(E-mail\)" <mmusic@ietf.org>, "draft-ietf-mmusic-media-loopback@tools.ietf.org" <draft-ietf-mmusic-media-loopback@tools.ietf.org>, Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>
Subject: Re: [MMUSIC] NEED WG CONSENSUS: media-loopback is all-or-nothing
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no


On Nov 20, 2012, at 11:16 , Hadriel Kaplan <HKaplan@acmepacket.com> wrote:

> Howdy,
> during IESG review of the media-loopback draft, a discussion arose regard=
ing the language around indicating media-loopback support per media descrip=
tion (i.e., per m-line), in sections 3.1 and 3.2.
>=20
> My interpretation/understanding was we were requiring ALL m-lines to indi=
cate media-loopback; in other words it's an all-or-nothing in both the offe=
r+answer, and either every media stream is looped back, or none are.  Howev=
er the text currently doesn't make this clear, and it could be argued eithe=
r way, with pro's/con's either way too; so I'd like the WG's consensus on w=
hether it should in fact be an all-or-none approach, or not.
>=20
> Please respond to this email by **Tuesday, December 4th** with a "YES" if=
 the draft should mandate ALL media be looped-back, or "NO" if it should al=
low a hybrid approach of some streams being looped while others are not.
>=20
> For reference, the draft is here:
> http://tools.ietf.org/html/draft-ietf-mmusic-media-loopback-24
>=20
> -hadriel
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From andrew.hutton@siemens-enterprise.com  Tue Nov 27 02:58:19 2012
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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>, "mmusic@ietf.org (E-mail)" <mmusic@ietf.org>
Thread-Topic: NEED WG CONSENSUS: media-loopback is all-or-nothing
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Cc: Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>, "draft-ietf-mmusic-media-loopback@tools.ietf.org" <draft-ietf-mmusic-media-loopback@tools.ietf.org>
Subject: Re: [MMUSIC] NEED WG CONSENSUS: media-loopback is all-or-nothing
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NO.=20

I see no reason to mandate that all media is looped back.

Regards
Andy

> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Hadriel Kaplan
> Sent: 20 November 2012 18:16
> To: mmusic@ietf.org (E-mail)
> Cc: Gonzalo Camarillo; draft-ietf-mmusic-media-loopback@tools.ietf.org
> Subject: [MMUSIC] NEED WG CONSENSUS: media-loopback is all-or-nothing
>=20
> Howdy,
> during IESG review of the media-loopback draft, a discussion arose
> regarding the language around indicating media-loopback support per
> media description (i.e., per m-line), in sections 3.1 and 3.2.
>=20
> My interpretation/understanding was we were requiring ALL m-lines to
> indicate media-loopback; in other words it's an all-or-nothing in both
> the offer+answer, and either every media stream is looped back, or none
> are.  However the text currently doesn't make this clear, and it could
> be argued either way, with pro's/con's either way too; so I'd like the
> WG's consensus on whether it should in fact be an all-or-none approach,
> or not.
>=20
> Please respond to this email by **Tuesday, December 4th** with a "YES"
> if the draft should mandate ALL media be looped-back, or "NO" if it
> should allow a hybrid approach of some streams being looped while
> others are not.
>=20
> For reference, the draft is here:
> http://tools.ietf.org/html/draft-ietf-mmusic-media-loopback-24
>=20
> -hadriel
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

From enrico.marocco@telecomitalia.it  Wed Nov 28 01:00:51 2012
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Subject: Re: [MMUSIC] NEED WG CONSENSUS: media-loopback is all-or-nothing
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No.

I see no compelling reason for introducing such a limitation, but a
bunch of cases where it breaks things badly. In interconnection
scenarios, for instance.

Enrico

On 11/20/12 7:16 PM, Hadriel Kaplan wrote:
> Howdy,
> during IESG review of the media-loopback draft, a discussion arose rega=
rding the language around indicating media-loopback support per media des=
cription (i.e., per m-line), in sections 3.1 and 3.2.
>=20
> My interpretation/understanding was we were requiring ALL m-lines to in=
dicate media-loopback; in other words it's an all-or-nothing in both the =
offer+answer, and either every media stream is looped back, or none are. =
 However the text currently doesn't make this clear, and it could be argu=
ed either way, with pro's/con's either way too; so I'd like the WG's cons=
ensus on whether it should in fact be an all-or-none approach, or not.
>=20
> Please respond to this email by **Tuesday, December 4th** with a "YES" =
if the draft should mandate ALL media be looped-back, or "NO" if it shoul=
d allow a hybrid approach of some streams being looped while others are n=
ot.
>=20
> For reference, the draft is here:
> http://tools.ietf.org/html/draft-ietf-mmusic-media-loopback-24
>=20
> -hadriel
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>=20



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From harald@alvestrand.no  Wed Nov 28 06:50:58 2012
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Date: Wed, 28 Nov 2012 15:50:52 +0100
From: Harald Alvestrand <harald@alvestrand.no>
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Subject: [MMUSIC] M-lines: Wikifying it
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Perhaps we can do some real work rather than just disagreeing on principles.

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

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

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

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

I make it 179 entries in these categories, combined.

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

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

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

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

I've created a spreadsheet here:

https://docs.google.com/spreadsheet/ccc?key=0Av-aF69sk5C6dDhPblpRV2pkbXVGNV9lUFI3cHlIRFE

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

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

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

                       Harald






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Content-Transfer-Encoding: 7bit

<html>
  <head>

    <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    Perhaps we can do some real work rather than just disagreeing on
    principles.<br>
    <br>
    Perhaps the most contentious issue now is "what should an M-line
    be".<br>
    As far as I can tell, M-lines and their parameters describe 3
    things:<br>
    <br>
    - Properties of the transport flow / RTP session (Transport, or T
    for short)<br>
    - Properties that make sense to apply to an individual media stream,
    and may need to be different for different media streams in the same
    combined RTP session (Flow, or F for short)<br>
    - Properties that make sense to apply to all streams of a given type
    (Media, or M for short)<br>
    <br>
    The properties that can be represented in M-lines are found in two
    IANA registries, both on<br>
    <meta http-equiv="content-type" content="text/html;
      charset=ISO-8859-1">
    <a
      href="http://www.iana.org/assignments/sdp-parameters/sdp-parameters.xml">http://www.iana.org/assignments/sdp-parameters/sdp-parameters.xml</a>:<br>
    <br>
    - att-field (both session and media level)<br>
    - att-field (media level only)<br>
    <br>
    I make it 179 entries in these categories, combined.<br>
    <br>
    I see the same kind of work being necessary no matter which way we
    go:<br>
    <br>
    - If we go with M-lines describing a whole RTP session (the MMT
    proposal), each property of the "F" class that we stil need needs to
    be copied onto a property in the "att-field (source level)"
    registry.<br>
    <br>
    - If we go with M-lines describing groups of media streams,
    degenerating to a single media stream in the case of using
    "a=inactive" for stream muting (the BUNDLE and TOGETHER proposals),
    each property of the "T" class needs to be described as "MUST have
    consistent values for all the combined M-lines", "MUST be taken only
    from the first M-line in the bundle argument", or "must get special
    treatment".<br>
    <br>
    - In both cases, the "M"-class lines don't need any special
    treatment. They just need to be present in all the M-lines where
    they're relevant.<br>
    <br>
    I've created a spreadsheet here:<br>
    <br>
<a class="moz-txt-link-freetext" href="https://docs.google.com/spreadsheet/ccc?key=0Av-aF69sk5C6dDhPblpRV2pkbXVGNV9lUFI3cHlIRFE">https://docs.google.com/spreadsheet/ccc?key=0Av-aF69sk5C6dDhPblpRV2pkbXVGNV9lUFI3cHlIRFE</a><br>
    <br>
    with 2 extra columns: "Class" (M, F or T) and "Importance" (1 for
    "necessary" and 0 for "will never matter").<br>
    <br>
    Anyone should be able to write it.<br>
    If a few of us do a quick job of markup on it, I think we should be
    able to have a more realistic assessment in a few days.<br>
    <br>
    I've started, but much of this is unknown to me. Can others help?<br>
    <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
    <br>
    <br>
    <br>
    <br>
    <br>
  </body>
</html>

--------------000407040007040107030804--

From jonathan@vidyo.com  Wed Nov 28 08:29:57 2012
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From: Jonathan Lennox <jonathan@vidyo.com>
To: Harald Alvestrand <harald@alvestrand.no>, mmusic <mmusic@ietf.org>
Date: Wed, 28 Nov 2012 11:29:47 -0500
Thread-Topic: [MMUSIC] M-lines: Wikifying it
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Subject: Re: [MMUSIC] M-lines: Wikifying it
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Hi, Harald -

I think this is a good starting point, though I think things are somewhat m=
ore complicated than you've described.

First of all, for the SHIM approaches, you need to separate Transport param=
eters (e.g. ICE candidates) from Session parameters (e.g., SRTP keying).  T=
his isn't relevant for Bundle/MMT, but as long as we're doing the work it m=
ight as well be done completely.

More significantly, I think, a number of parameters actually fall into mult=
iple categories depending on how they're used.  For example, rtcp-fb is Tra=
nsport (or, if you believe my previous argument, Session) when the rtcp-fb-=
pt field is "*", but it's Media when a specific payload type is given.

From: Harald Alvestrand [mailto:harald@alvestrand.no]
Sent: Wednesday, November 28, 2012 9:51 AM
To: mmusic
Subject: [MMUSIC] M-lines: Wikifying it

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

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

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

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

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

I make it 179 entries in these categories, combined.

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

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

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

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

I've created a spreadsheet here:

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

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

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

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

                      Harald





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</o:shapelayout></xml><![endif]--></head><body bgcolor=3Dwhite lang=3DEN-US=
 link=3Dblue vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal>=
<span style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1=
F497D'>Hi, Harald &#8211;<o:p></o:p></span></p><p class=3DMsoNormal><span s=
tyle=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>=
<o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span style=3D'font-size:1=
1.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>I think this is a g=
ood starting point, though I think things are somewhat more complicated tha=
n you&#8217;ve described.<o:p></o:p></span></p><p class=3DMsoNormal><span s=
tyle=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>=
<o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span style=3D'font-size:1=
1.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>First of all, for t=
he SHIM approaches, you need to separate Transport parameters (e.g. ICE can=
didates) from Session parameters (e.g., SRTP keying).&nbsp; This isn&#8217;=
t relevant for Bundle/MMT, but as long as we&#8217;re doing the work it mig=
ht as well be done completely.<o:p></o:p></span></p><p class=3DMsoNormal><s=
pan style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F4=
97D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span style=3D'font-s=
ize:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>More significa=
ntly, I think, a number of parameters actually fall into multiple categorie=
s depending on how they&#8217;re used.&nbsp; For example, rtcp-fb is Transp=
ort (or, if you believe my previous argument, Session) when the rtcp-fb-pt =
field is &#8220;*&#8221;, but it&#8217;s Media when a specific payload type=
 is given.<o:p></o:p></span></p><p class=3DMsoNormal><span style=3D'font-si=
ze:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p>&nbsp;</o:=
p></span></p><div><div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;=
padding:3.0pt 0in 0in 0in'><p class=3DMsoNormal><b><span style=3D'font-size=
:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext'>From:</span></b=
><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:wi=
ndowtext'> Harald Alvestrand [mailto:harald@alvestrand.no] <br><b>Sent:</b>=
 Wednesday, November 28, 2012 9:51 AM<br><b>To:</b> mmusic<br><b>Subject:</=
b> [MMUSIC] M-lines: Wikifying it<o:p></o:p></span></p></div></div><p class=
=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal style=3D'margin-bott=
om:12.0pt'>Perhaps we can do some real work rather than just disagreeing on=
 principles.<br><br>Perhaps the most contentious issue now is &quot;what sh=
ould an M-line be&quot;.<br>As far as I can tell, M-lines and their paramet=
ers describe 3 things:<br><br>- Properties of the transport flow / RTP sess=
ion (Transport, or T for short)<br>- Properties that make sense to apply to=
 an individual media stream, and may need to be different for different med=
ia streams in the same combined RTP session (Flow, or F for short)<br>- Pro=
perties that make sense to apply to all streams of a given type (Media, or =
M for short)<br><br>The properties that can be represented in M-lines are f=
ound in two IANA registries, both on<br><a href=3D"http://www.iana.org/assi=
gnments/sdp-parameters/sdp-parameters.xml">http://www.iana.org/assignments/=
sdp-parameters/sdp-parameters.xml</a>:<br><br>- att-field (both session and=
 media level)<br>- att-field (media level only)<br><br>I make it 179 entrie=
s in these categories, combined.<br><br>I see the same kind of work being n=
ecessary no matter which way we go:<br><br>- If we go with M-lines describi=
ng a whole RTP session (the MMT proposal), each property of the &quot;F&quo=
t; class that we stil need needs to be copied onto a property in the &quot;=
att-field (source level)&quot; registry.<br><br>- If we go with M-lines des=
cribing groups of media streams, degenerating to a single media stream in t=
he case of using &quot;a=3Dinactive&quot; for stream muting (the BUNDLE and=
 TOGETHER proposals), each property of the &quot;T&quot; class needs to be =
described as &quot;MUST have consistent values for all the combined M-lines=
&quot;, &quot;MUST be taken only from the first M-line in the bundle argume=
nt&quot;, or &quot;must get special treatment&quot;.<br><br>- In both cases=
, the &quot;M&quot;-class lines don't need any special treatment. They just=
 need to be present in all the M-lines where they're relevant.<br><br>I've =
created a spreadsheet here:<br><br><a href=3D"https://docs.google.com/sprea=
dsheet/ccc?key=3D0Av-aF69sk5C6dDhPblpRV2pkbXVGNV9lUFI3cHlIRFE">https://docs=
.google.com/spreadsheet/ccc?key=3D0Av-aF69sk5C6dDhPblpRV2pkbXVGNV9lUFI3cHlI=
RFE</a><br><br>with 2 extra columns: &quot;Class&quot; (M, F or T) and &quo=
t;Importance&quot; (1 for &quot;necessary&quot; and 0 for &quot;will never =
matter&quot;).<br><br>Anyone should be able to write it.<br>If a few of us =
do a quick job of markup on it, I think we should be able to have a more re=
alistic assessment in a few days.<br><br>I've started, but much of this is =
unknown to me. Can others help?<br><br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp; Harald<br><br><br><br><br><o:p></o:p></p></div></body></htm=
l>=

--_000_C3759687E4991243A1A0BD44EAC823034DFAC1FAA9BE235maillan_--

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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [MMUSIC] M-lines: Wikifying it
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Inline

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

Don't forget the media formats listed in the m-line itself. The meaning 
of these depends on the <proto> param. The focus of this discussion is 
on those <proto> values that correspond to various flavors of RTP. There 
are a bunch of <proto> values that don't correspond to RTP, and that 
assign different meanings to the format parameters.

AFAIK, of the currently defined proto values only the RTP-related ones 
have the notion of multiple flows within the session. But the sdp 
extensions for SCTP media are currently being defined. That certainly 
has the potential to be considered as multiple flows, and should be 
considered in this discussion.

For RTP transports, the formats are payload type numbers, and these are 
mapped to actual formats (codecs) via a=rtpmap. I see that you have 
proposed that rtpmap be per-media, not transport or flow. I guess this 
means that the same payload type number could mean one thing for all 
audio flows over the transport, and something different for all video 
flows over the same transport. But the payload numbers to be used 
(specified on the m-line) is negotiated in O/A. How would that work for 
audio vs. video? ISTM it would be preferred to negotiate payload numbers 
and types independently per flow.

Regarding the "importance" column in the table: I don't understand how 
you have assigned values. In what frame of reference is importance to be 
assessed?

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

(I'm not convinced we need to identify an "M" scope distinct from T and F.)

	Thanks,
	Paul

From harald@alvestrand.no  Wed Nov 28 12:25:01 2012
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Subject: Re: [MMUSIC] M-lines: Wikifying it
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On 11/28/2012 05:29 PM, Jonathan Lennox wrote:
>
> Hi, Harald --
>
> I think this is a good starting point, though I think things are 
> somewhat more complicated than you've described.
>
> First of all, for the SHIM approaches, you need to separate Transport 
> parameters (e.g. ICE candidates) from Session parameters (e.g., SRTP 
> keying).  This isn't relevant for Bundle/MMT, but as long as we're 
> doing the work it might as well be done completely.
>
Yes, if we do SHIM, they're different. Good point! Will you update the 
header for that?
(I tend to forget SHIM)
>
> More significantly, I think, a number of parameters actually fall into 
> multiple categories depending on how they're used.  For example, 
> rtcp-fb is Transport (or, if you believe my previous argument, 
> Session) when the rtcp-fb-pt field is "*", but it's Media when a 
> specific payload type is given.
>
That means we'll have to make 2 lines for it. I suspect it's not the 
only one.

> *From:*Harald Alvestrand [mailto:harald@alvestrand.no]
> *Sent:* Wednesday, November 28, 2012 9:51 AM
> *To:* mmusic
> *Subject:* [MMUSIC] M-lines: Wikifying it
>
> Perhaps we can do some real work rather than just disagreeing on 
> principles.
>
> Perhaps the most contentious issue now is "what should an M-line be".
> As far as I can tell, M-lines and their parameters describe 3 things:
>
> - Properties of the transport flow / RTP session (Transport, or T for 
> short)
> - Properties that make sense to apply to an individual media stream, 
> and may need to be different for different media streams in the same 
> combined RTP session (Flow, or F for short)
> - Properties that make sense to apply to all streams of a given type 
> (Media, or M for short)
>
> The properties that can be represented in M-lines are found in two 
> IANA registries, both on
> http://www.iana.org/assignments/sdp-parameters/sdp-parameters.xml:
>
> - att-field (both session and media level)
> - att-field (media level only)
>
> I make it 179 entries in these categories, combined.
>
> I see the same kind of work being necessary no matter which way we go:
>
> - If we go with M-lines describing a whole RTP session (the MMT 
> proposal), each property of the "F" class that we stil need needs to 
> be copied onto a property in the "att-field (source level)" registry.
>
> - If we go with M-lines describing groups of media streams, 
> degenerating to a single media stream in the case of using 
> "a=inactive" for stream muting (the BUNDLE and TOGETHER proposals), 
> each property of the "T" class needs to be described as "MUST have 
> consistent values for all the combined M-lines", "MUST be taken only 
> from the first M-line in the bundle argument", or "must get special 
> treatment".
>
> - In both cases, the "M"-class lines don't need any special treatment. 
> They just need to be present in all the M-lines where they're relevant.
>
> I've created a spreadsheet here:
>
> https://docs.google.com/spreadsheet/ccc?key=0Av-aF69sk5C6dDhPblpRV2pkbXVGNV9lUFI3cHlIRFE
>
> with 2 extra columns: "Class" (M, F or T) and "Importance" (1 for 
> "necessary" and 0 for "will never matter").
>
> Anyone should be able to write it.
> If a few of us do a quick job of markup on it, I think we should be 
> able to have a more realistic assessment in a few days.
>
> I've started, but much of this is unknown to me. Can others help?
>
>                       Harald
>
>
>
>


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    <div class="moz-cite-prefix">On 11/28/2012 05:29 PM, Jonathan Lennox
      wrote:<br>
    </div>
    <blockquote
      cite="mid:C3759687E4991243A1A0BD44EAC823034DFAC1FAA9@BE235.mail.lan"
      type="cite">
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      <div class="WordSection1">
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi,
            Harald &#8211;<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">I
            think this is a good starting point, though I think things
            are somewhat more complicated than you&#8217;ve described.<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">First
            of all, for the SHIM approaches, you need to separate
            Transport parameters (e.g. ICE candidates) from Session
            parameters (e.g., SRTP keying).&nbsp; This isn&#8217;t relevant for
            Bundle/MMT, but as long as we&#8217;re doing the work it might as
            well be done completely.</span></p>
      </div>
    </blockquote>
    Yes, if we do SHIM, they're different. Good point! Will you update
    the header for that?<br>
    (I tend to forget SHIM)<br>
    <blockquote
      cite="mid:C3759687E4991243A1A0BD44EAC823034DFAC1FAA9@BE235.mail.lan"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">More
            significantly, I think, a number of parameters actually fall
            into multiple categories depending on how they&#8217;re used.&nbsp; For
            example, rtcp-fb is Transport (or, if you believe my
            previous argument, Session) when the rtcp-fb-pt field is
            &#8220;*&#8221;, but it&#8217;s Media when a specific payload type is given.</span></p>
      </div>
    </blockquote>
    That means we'll have to make 2 lines for it. I suspect it's not the
    only one.<br>
    <br>
    <blockquote
      cite="mid:C3759687E4991243A1A0BD44EAC823034DFAC1FAA9@BE235.mail.lan"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <div>
          <div style="border:none;border-top:solid #B5C4DF
            1.0pt;padding:3.0pt 0in 0in 0in">
            <p class="MsoNormal"><b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">From:</span></b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">
                Harald Alvestrand [<a class="moz-txt-link-freetext" href="mailto:harald@alvestrand.no">mailto:harald@alvestrand.no</a>] <br>
                <b>Sent:</b> Wednesday, November 28, 2012 9:51 AM<br>
                <b>To:</b> mmusic<br>
                <b>Subject:</b> [MMUSIC] M-lines: Wikifying it<o:p></o:p></span></p>
          </div>
        </div>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        <p class="MsoNormal" style="margin-bottom:12.0pt">Perhaps we can
          do some real work rather than just disagreeing on principles.<br>
          <br>
          Perhaps the most contentious issue now is "what should an
          M-line be".<br>
          As far as I can tell, M-lines and their parameters describe 3
          things:<br>
          <br>
          - Properties of the transport flow / RTP session (Transport,
          or T for short)<br>
          - Properties that make sense to apply to an individual media
          stream, and may need to be different for different media
          streams in the same combined RTP session (Flow, or F for
          short)<br>
          - Properties that make sense to apply to all streams of a
          given type (Media, or M for short)<br>
          <br>
          The properties that can be represented in M-lines are found in
          two IANA registries, both on<br>
          <a moz-do-not-send="true"
            href="http://www.iana.org/assignments/sdp-parameters/sdp-parameters.xml">http://www.iana.org/assignments/sdp-parameters/sdp-parameters.xml</a>:<br>
          <br>
          - att-field (both session and media level)<br>
          - att-field (media level only)<br>
          <br>
          I make it 179 entries in these categories, combined.<br>
          <br>
          I see the same kind of work being necessary no matter which
          way we go:<br>
          <br>
          - If we go with M-lines describing a whole RTP session (the
          MMT proposal), each property of the "F" class that we stil
          need needs to be copied onto a property in the "att-field
          (source level)" registry.<br>
          <br>
          - If we go with M-lines describing groups of media streams,
          degenerating to a single media stream in the case of using
          "a=inactive" for stream muting (the BUNDLE and TOGETHER
          proposals), each property of the "T" class needs to be
          described as "MUST have consistent values for all the combined
          M-lines", "MUST be taken only from the first M-line in the
          bundle argument", or "must get special treatment".<br>
          <br>
          - In both cases, the "M"-class lines don't need any special
          treatment. They just need to be present in all the M-lines
          where they're relevant.<br>
          <br>
          I've created a spreadsheet here:<br>
          <br>
          <a moz-do-not-send="true"
href="https://docs.google.com/spreadsheet/ccc?key=0Av-aF69sk5C6dDhPblpRV2pkbXVGNV9lUFI3cHlIRFE">https://docs.google.com/spreadsheet/ccc?key=0Av-aF69sk5C6dDhPblpRV2pkbXVGNV9lUFI3cHlIRFE</a><br>
          <br>
          with 2 extra columns: "Class" (M, F or T) and "Importance" (1
          for "necessary" and 0 for "will never matter").<br>
          <br>
          Anyone should be able to write it.<br>
          If a few of us do a quick job of markup on it, I think we
          should be able to have a more realistic assessment in a few
          days.<br>
          <br>
          I've started, but much of this is unknown to me. Can others
          help?<br>
          <br>
          &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
          <br>
          <br>
          <br>
          <br>
          <o:p></o:p></p>
      </div>
    </blockquote>
    <br>
  </body>
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From: "Nikos Leontsinis" <nikosietf@gmail.com>
To: "'Qin Wu'" <bill.wu@huawei.com>, "'Hadriel Kaplan'" <HKaplan@acmepacket.com>
References: <A444ACE8-4EDD-48DC-86C8-7CCBB40173CE@acmepacket.com> <C5E08FE080ACFD4DAE31E4BDBF944EB11326C452@xmb-aln-x02.cisco.com> <A6D6A7467A9A47D48B7505A482D852F0@china.huawei.com>
In-Reply-To: <A6D6A7467A9A47D48B7505A482D852F0@china.huawei.com>
Date: Thu, 29 Nov 2012 16:09:05 +0200
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Cc: mmusic@ietf.org, draft-ietf-mmusic-media-loopback@tools.ietf.org, 'Gonzalo Camarillo' <gonzalo.camarillo@ericsson.com>
Subject: Re: [MMUSIC] NEED WG CONSENSUS: media-loopback is all-or-nothing
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No for scalability reasons

/nikos

-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of
Qin Wu
Sent: Thursday, November 29, 2012 2:54 AM
To: Hadriel Kaplan
Cc: mmusic@ietf.org; draft-ietf-mmusic-media-loopback@tools.ietf.org;
Gonzalo Camarillo
Subject: Re: [MMUSIC] NEED WG CONSENSUS: media-loopback is all-or-nothing

No,
I see it will be more flexible to allow some being looped back while some
not.

Regards!
-Qin
----- Original Message ----- 
From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "Hadriel Kaplan" <HKaplan@acmepacket.com>
Cc: <mmusic@ietf.org>; <draft-ietf-mmusic-media-loopback@tools.ietf.org>;
"Gonzalo Camarillo" <gonzalo.camarillo@ericsson.com>
Sent: Tuesday, November 27, 2012 11:43 AM
Subject: Re: [MMUSIC] NEED WG CONSENSUS: media-loopback is all-or-nothing


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